[Vtigercrm-developers] Asterisk Good News

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[Vtigercrm-developers] Asterisk Good News

vtiger@camden.net

after starting from scratch the Asterisk Connector and PBX Manager are working consistently with one minor issue

click to call:

when you call out the person you call sees caller ID as them self

IE:

I click my cell phone number, my extension rings, I answer my extension and Asterisk calls my cell

my cell phone rings, but the callerID on my cell is saying I am getting a incoming call from my cell, when it should be the default outbound CID of my extension

the call does connect, just has the wrong callerID

on phones that support both Cname and CiD, the CID still comes up as the person you are calling see the CID as them selfs but

the Cname is set to the Display Name of the Extension which is technically correct but the whole callerID string should be set to the Outbound CID string of the extensions settings

most companies will have the display name as the person real name or just the extensions number where outbound CID will be "NAME"<NUMBER> of how the company wants outside callers to see the callerID info 



for those that are going to ask how we got it to work

you must be on rev 14179 or higher, you must also update PBX Manager

you need to have

[vtiger_outbound]
exten => _X.,1,Agi(agi://0.0.0.0/incoming.agi)

in extensions.conf and outbound context in PBX Manager also set to "vtiger_outbound"
actually as long as both match it will work

the outbound trunk is the name of your trunk in Asterisk to use for outbound calling, keep in mind Asterisk has a general settings and a outbound settings for each trunk both have trunk name and they must match

for inbound even though we tried to set default context to "vtiger_inbound" in asterisk we never got the AGI to fire from there and for us it broke other stuff, we put "exten => _X.,1,Agi(agi://0.0.0.0/incoming.agi)" into the from-internal context in extensions.conf as that is our default context and it works just fine

yes that is correct you will have 2 contexts in extensions.conf both with "exten => _X.,1,Agi(agi://0.0.0.0/incoming.agi)"




_______________________________________________
http://www.vtiger.com/
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Re: Asterisk Good News

Zebra Hosting
Thanks for sharing the good news!

Unfortunately I can’t get it to work. Guess a few more steps needed in your descriptions :-)

This part:
keep in mind Asterisk has a general settings and a outbound settings for each trunk both have trunk name and they must match
Left me puzzled….

The outbound trunk you mention, is a SIP trunk or something you setup for vTiger only? (I am on FreePBX so things might be a little different.

Thanks,

Bastiaan Houtkooper
Zebra Hosting



From: "[hidden email]" <[hidden email]>
Reply-To: <[hidden email]>, <[hidden email]>
Date: Tue, 19 Aug 2014 21:36:16 -0400
To: Rishab K <[hidden email]>, <[hidden email]>
Subject: [Vtigercrm-developers] Asterisk Good News


after starting from scratch the Asterisk Connector and PBX Manager are working consistently with one minor issue

click to call:

when you call out the person you call sees caller ID as them self

IE:

I click my cell phone number, my extension rings, I answer my extension and Asterisk calls my cell

my cell phone rings, but the callerID on my cell is saying I am getting a incoming call from my cell, when it should be the default outbound CID of my extension

the call does connect, just has the wrong callerID

on phones that support both Cname and CiD, the CID still comes up as the person you are calling see the CID as them selfs but

the Cname is set to the Display Name of the Extension which is technically correct but the whole callerID string should be set to the Outbound CID string of the extensions settings

most companies will have the display name as the person real name or just the extensions number where outbound CID will be "NAME"<NUMBER> of how the company wants outside callers to see the callerID info 



for those that are going to ask how we got it to work

you must be on rev 14179 or higher, you must also update PBX Manager

you need to have

[vtiger_outbound]
exten => _X.,1,Agi(agi://0.0.0.0/incoming.agi)

in extensions.conf and outbound context in PBX Manager also set to "vtiger_outbound"
actually as long as both match it will work

the outbound trunk is the name of your trunk in Asterisk to use for outbound calling, keep in mind Asterisk has a general settings and a outbound settings for each trunk both have trunk name and they must match

for inbound even though we tried to set default context to "vtiger_inbound" in asterisk we never got the AGI to fire from there and for us it broke other stuff, we put "exten => _X.,1,Agi(agi://0.0.0.0/incoming.agi)" into the from-internal context in extensions.conf as that is our default context and it works just fine

yes that is correct you will have 2 contexts in extensions.conf both with "exten => _X.,1,Agi(agi://0.0.0.0/incoming.agi)"



_______________________________________________ http://www.vtiger.com/

_______________________________________________
http://www.vtiger.com/
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Re: Asterisk Good News

vtiger@camden.net
In FreePBX

Connectivity --> Trunks  (select your trunk you want to use for vtiger outbound calling, you might only have 1 trunk to use, you might a few to pick from, dont matter if its SIP, DAHDi, IAX or what ever, if you have more than 1 trunk you need to pick one)

General Settings and Outgoing Settings Sections will both have "Trunk Name"

In Vtiger 

Settings --> CRM Settings -- > Integration --> PBXManager 

"Outbound Trunk"

all 3 must match


*************************************************************************************************************************

In Vtiger 

Settings --> CRM Settings -- > Integration --> PBXManager 

"Outbound Context" must match a context in extensions.conf in that context you must have exten => _X.,1,agi(agi://0.0.0.0/incoming.agi)

i used "vtiger_outbound" so I have "vtiger_outbound" in the "Outbound Context" field in PBX Manager and

[vtiger_outbound]
exten => _X.,1,agi(agi://0.0.0.0/incoming.agi)

in extensions.conf


you still need all the other setting that have been posted before

check all the settings in VtigerAsteriskConnector.properties

ServerIP = 127.0.0.1
serverport = {pick a unused port}

make sure in the PBX Manager "vtiger Asterisk App URL matches, should be 

http://127.0.0.1:{the port you picked}


AsteriskServerIP = 127.0.0.1
AsteriskServerPort = {your AMI or Proxy Port}
AsteriskUserName = {your AMI or Proxy User Name}
AsteriskPassword = {your secret to the above user}


make sure your AMI user (in manger.conf or refer to your proxy documentation) for above with


read = all,system,call,log,verbose,command,agent,user,config, originate
write = all,system,call,log,verbose,command,agent,user,config, originate

*** proxy should be used for large deployments, astmanproxy works well


make sure webapp.sh and agi.sh are actually running on your server and set to start on boot


the above will get click to call to work and your 99% setup for inbound



for inbound notification you need to put exten => _X.,1,agi(agi://0.0.0.0/incoming.agi) in a existing context in extensions.conf that will fire on inbound, this may not be same for everyone

we used [from-internal] which will make vtiger agi fire for any call goign though PBX on our system from outside or inside to inside

we could have used [from-trunk] that would make vtiger agi fire only on outside calls coming from the trunk

yours may be different, this would be more of question for your asterisk admin then vtiger as to how you should do this



 





From: "Zebra Hosting" <[hidden email]>
Sent: Wednesday, August 20, 2014 10:49 AM
To: "[hidden email]" <[hidden email]>
Subject: Re: [Vtigercrm-developers] Asterisk Good News


Thanks for sharing the good news!

Unfortunately I can't get it to work. Guess a few more steps needed in your descriptions :-)

This part:
keep in mind Asterisk has a general settings and a outbound settings for each trunk both have trunk name and they must match
Left me puzzled..

The outbound trunk you mention, is a SIP trunk or something you setup for vTiger only? (I am on FreePBX so things might be a little different.

Thanks,

Bastiaan Houtkooper
Zebra Hosting



From: "[hidden email]" <[hidden email]>
Reply-To: <[hidden email]>, <[hidden email]>
Date: Tue, 19 Aug 2014 21:36:16 -0400
To: Rishab K <[hidden email]>, <[hidden email]>
Subject: [Vtigercrm-developers] Asterisk Good News


after starting from scratch the Asterisk Connector and PBX Manager are working consistently with one minor issue

click to call:

when you call out the person you call sees caller ID as them self

IE:

I click my cell phone number, my extension rings, I answer my extension and Asterisk calls my cell

my cell phone rings, but the callerID on my cell is saying I am getting a incoming call from my cell, when it should be the default outbound CID of my extension

the call does connect, just has the wrong callerID

on phones that support both Cname and CiD, the CID still comes up as the person you are calling see the CID as them selfs but

the Cname is set to the Display Name of the Extension which is technically correct but the whole callerID string should be set to the Outbound CID string of the extensions settings

most companies will have the display name as the person real name or just the extensions number where outbound CID will be "NAME"<NUMBER> of how the company wants outside callers to see the callerID info 



for those that are going to ask how we got it to work

you must be on rev 14179 or higher, you must also update PBX Manager

you need to have

[vtiger_outbound]
exten => _X.,1,Agi(agi://0.0.0.0/incoming.agi)

in extensions.conf and outbound context in PBX Manager also set to "vtiger_outbound"
actually as long as both match it will work

the outbound trunk is the name of your trunk in Asterisk to use for outbound calling, keep in mind Asterisk has a general settings and a outbound settings for each trunk both have trunk name and they must match

for inbound even though we tried to set default context to "vtiger_inbound" in asterisk we never got the AGI to fire from there and for us it broke other stuff, we put "exten => _X.,1,Agi(agi://0.0.0.0/incoming.agi)" into the from-internal context in extensions.conf as that is our default context and it works just fine

yes that is correct you will have 2 contexts in extensions.conf both with "exten => _X.,1,Agi(agi://0.0.0.0/incoming.agi)"



_______________________________________________ http://www.vtiger.com/


_______________________________________________
http://www.vtiger.com/
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Re: Asterisk Good News

Zebra Hosting
Had some time to play around with the PBX setup:

Setup is a fresh AsteriskNow with one trunk (voipbuster) where I only
allow to call out to my own mobile phone for now.
In and out calling works fine. Internal too.

Allow SIP guests and Allow Anonymus inbound SIP calls are both off for
security. (Unbelievable amount of idiots trying to hack in there)
0.0.0.0/0.0.0.0 is denied but 127.0.0.1, office and vTiger IP are allowed.

setup a manager called vtiger with the same permit/deny and basically all
rights. Did not work so went back to admin user/manager for now.

In the ³from-internal context I have exten =>
_X.,1,agi(agi://127.0.0.1/incoming.agi) but it stops all internal calls...

Any suggestions?


_______________________________________________
http://www.vtiger.com/

Screen Shot 2014-08-21 at 11.42.03.png (48K) Download Attachment
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Re: Asterisk Good News

vtiger@camden.net
try [from-trunk] or any other context that only fires on inbound call from outside, should take care of that issue
 
 
though this is not ready for production, after a week of testing its unstable
 
worked fine for a few days, then:
 
it got stuck in endless loop repetitively notifying vtiger users of a inbound call
 
over the weekend it did something in which all extensions that where in vtiger no longer ring though call queues, even after removing them from vtiger, all the extensions that where not in vtiger still work
 
now we are dealing trying to deal with (on click to call)
 
the webapp.sh data stream reports
 
ug 26, 2014 5:43:29 PM com.vtiger.apps.asterisk.webapp.helpers.b a
INFO: Sending call originate request to asterisk
Aug 26, 2014 5:43:29 PM com.vtiger.apps.asterisk.webapp.helpers.b a
INFO: Context:[vtiger_outbound]
Aug 26, 2014 5:43:29 PM com.vtiger.apps.asterisk.webapp.helpers.b a
INFO: From: 101 -> To: 18032433300
Aug 26, 2014 5:43:29 PM com.vtiger.apps.asterisk.webapp.helpers.b a
INFO: Outgoing Call Response: Error
Aug 26, 2014 5:43:29 PM com.vtiger.apps.asterisk.webapp.helpers.b a
INFO: Asterisk Response: Extension does not exist.
 
 
 
no data seen in AMI, CLI, or agi.sh, and it never has wrote any file to its log directory
 
101 is a good extension, we can send an origination command directly to AMI and it works just fine
 
 
our primary business is hosting PBX, some of our clients have requested a hosted CRM to go along with it, to replace products like ACT that work just fine with click to call and notification via TAPI drivers
 
we have some big concerns about this closed source java based AGI approach, its not needed for anything vtiger is really trying to do, at least as advertised, and it seems to want to take over and be a mini PBX inside the PBX, its making its own CDR records, forcing all calls to be recorded, and we are not sure what else but its apparently capable of changing how extensions work in call queues  
 
it would be nice if vtiger developers just gave in and used the AMI for origination and notification (they are doing that anyway in there AGI) and used SQL to tie vtiger records to Asterisk CDR records and call recordings if any even exist  
 
if this dont get fixed and these issues get addressed soon, we may look into trying to fool vtiger into firing our own php AGI for click to call, and just use the our own windows system tray call notification, or start looking into other CRM solutions altogether
 
 
 
 
 

From: "Zebra Hosting" <[hidden email]>
Sent: Tuesday, August 26, 2014 12:36 PM
To: "[hidden email]" <[hidden email]>
Subject: Re: [Vtigercrm-developers] Asterisk Good News
 
Had some time to play around with the PBX setup:

Setup is a fresh AsteriskNow with one trunk (voipbuster) where I only
allow to call out to my own mobile phone for now.
In and out calling works fine. Internal too.

Allow SIP guests and Allow Anonymus inbound SIP calls are both off for
security. (Unbelievable amount of idiots trying to hack in there)
0.0.0.0/0.0.0.0 is denied but 127.0.0.1, office and vTiger IP are allowed.

setup a manager called vtiger with the same permit/deny and basically all
rights. Did not work so went back to admin user/manager for now.

In the ³from-internal context I have exten =>
_X.,1,agi(agi://127.0.0.1/incoming.agi) but it stops all internal calls...

Any suggestions?

_______________________________________________
http://www.vtiger.com/

_______________________________________________
http://www.vtiger.com/

Attachment 1 (1000 bytes) Download Attachment
Screen Shot 2014-08-21 at 11.42.03.png (48K) Download Attachment
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Re: Asterisk Good News

Prasad-14
We don't have in-house expertise working with AMI.

VtigerAsteriskApp is a web-app that connects AGI and Vtiger CRM - 
does no magic  other than proxying the requests to AGI using the library.
We have no short-term plans to make it open-souce than making 
the distribution  will be made freely downloadable.

We would follow up on the issue reported soon.

Regards,
Prasad 

Connect with us on: Twitter I Facebook I Blog I Wiki Forums I Website


On Wed, Aug 27, 2014 at 4:43 AM, [hidden email] <[hidden email]> wrote:
try [from-trunk] or any other context that only fires on inbound call from outside, should take care of that issue
 
 
though this is not ready for production, after a week of testing its unstable
 
worked fine for a few days, then:
 
it got stuck in endless loop repetitively notifying vtiger users of a inbound call
 
over the weekend it did something in which all extensions that where in vtiger no longer ring though call queues, even after removing them from vtiger, all the extensions that where not in vtiger still work
 
now we are dealing trying to deal with (on click to call)
 
the webapp.sh data stream reports
 
ug 26, 2014 5:43:29 PM com.vtiger.apps.asterisk.webapp.helpers.b a
INFO: Sending call originate request to asterisk
Aug 26, 2014 5:43:29 PM com.vtiger.apps.asterisk.webapp.helpers.b a
INFO: Context:[vtiger_outbound]
Aug 26, 2014 5:43:29 PM com.vtiger.apps.asterisk.webapp.helpers.b a
INFO: From: 101 -> To: 18032433300
Aug 26, 2014 5:43:29 PM com.vtiger.apps.asterisk.webapp.helpers.b a
INFO: Outgoing Call Response: Error
Aug 26, 2014 5:43:29 PM com.vtiger.apps.asterisk.webapp.helpers.b a
INFO: Asterisk Response: Extension does not exist.
 
 
 
no data seen in AMI, CLI, or agi.sh, and it never has wrote any file to its log directory
 
101 is a good extension, we can send an origination command directly to AMI and it works just fine
 
 
our primary business is hosting PBX, some of our clients have requested a hosted CRM to go along with it, to replace products like ACT that work just fine with click to call and notification via TAPI drivers
 
we have some big concerns about this closed source java based AGI approach, its not needed for anything vtiger is really trying to do, at least as advertised, and it seems to want to take over and be a mini PBX inside the PBX, its making its own CDR records, forcing all calls to be recorded, and we are not sure what else but its apparently capable of changing how extensions work in call queues  
 
it would be nice if vtiger developers just gave in and used the AMI for origination and notification (they are doing that anyway in there AGI) and used SQL to tie vtiger records to Asterisk CDR records and call recordings if any even exist  
 
if this dont get fixed and these issues get addressed soon, we may look into trying to fool vtiger into firing our own php AGI for click to call, and just use the our own windows system tray call notification, or start looking into other CRM solutions altogether
 
 
 
 
 

From: "Zebra Hosting" <[hidden email]>
Sent: Tuesday, August 26, 2014 12:36 PM

To: "[hidden email]" <[hidden email]>
Subject: Re: [Vtigercrm-developers] Asterisk Good News
 
Had some time to play around with the PBX setup:

Setup is a fresh AsteriskNow with one trunk (voipbuster) where I only
allow to call out to my own mobile phone for now.
In and out calling works fine. Internal too.

Allow SIP guests and Allow Anonymus inbound SIP calls are both off for
security. (Unbelievable amount of idiots trying to hack in there)
0.0.0.0/0.0.0.0 is denied but 127.0.0.1, office and vTiger IP are allowed.

setup a manager called vtiger with the same permit/deny and basically all
rights. Did not work so went back to admin user/manager for now.

In the ³from-internal context I have exten =>
_X.,1,agi(agi://127.0.0.1/incoming.agi) but it stops all internal calls...

Any suggestions?

_______________________________________________
http://www.vtiger.com/

_______________________________________________
http://www.vtiger.com/


_______________________________________________
http://www.vtiger.com/
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Re: Asterisk Good News

vtiger@camden.net
closed source / open source no matter as long as it works and does no harm, but as it stands now I am surly not the the only asterisk admin that is going to feel this way about your product 
 
and you already are using AMI
 
from your webapp.sh data stream
 
65423 [Asterisk-Java ManagerConnection-0-Reader-0] INFO org.asteriskjava.manager.internal.ManagerConnectionImpl - Connected via Asterisk Call Manager/1.3
 
thats a AMI connection, made with the the AMI user credentials in vtiger pbx manager and manger.conf
 
 
 
 
 

From: "Prasad" <[hidden email]>
Sent: Tuesday, August 26, 2014 8:03 PM
To: "[hidden email]" <[hidden email]>, "[hidden email]" <[hidden email]>
Subject: Re: [Vtigercrm-developers] Asterisk Good News
 
We don't have in-house expertise working with AMI.
 
VtigerAsteriskApp is a web-app that connects AGI and Vtiger CRM - 
does no magic  other than proxying the requests to AGI using the library.
We have no short-term plans to make it open-souce than making 
the distribution  will be made freely downloadable.
 
We would follow up on the issue reported soon.
 
Regards,
Prasad 
 
Connect with us on: Twitter I Facebook I Blog I Wiki Forums I Website
 
On Wed, Aug 27, 2014 at 4:43 AM, [hidden email] <[hidden email]> wrote:
try [from-trunk] or any other context that only fires on inbound call from outside, should take care of that issue
 
 
though this is not ready for production, after a week of testing its unstable
 
worked fine for a few days, then:
 
it got stuck in endless loop repetitively notifying vtiger users of a inbound call
 
over the weekend it did something in which all extensions that where in vtiger no longer ring though call queues, even after removing them from vtiger, all the extensions that where not in vtiger still work
 
now we are dealing trying to deal with (on click to call)
 
the webapp.sh data stream reports
 
ug 26, 2014 5:43:29 PM com.vtiger.apps.asterisk.webapp.helpers.b a
INFO: Sending call originate request to asterisk
Aug 26, 2014 5:43:29 PM com.vtiger.apps.asterisk.webapp.helpers.b a
INFO: Context:[vtiger_outbound]
Aug 26, 2014 5:43:29 PM com.vtiger.apps.asterisk.webapp.helpers.b a
INFO: From: 101 -> To: 18032433300
Aug 26, 2014 5:43:29 PM com.vtiger.apps.asterisk.webapp.helpers.b a
INFO: Outgoing Call Response: Error
Aug 26, 2014 5:43:29 PM com.vtiger.apps.asterisk.webapp.helpers.b a
INFO: Asterisk Response: Extension does not exist.
 
 
 
no data seen in AMI, CLI, or agi.sh, and it never has wrote any file to its log directory
 
101 is a good extension, we can send an origination command directly to AMI and it works just fine
 
 
our primary business is hosting PBX, some of our clients have requested a hosted CRM to go along with it, to replace products like ACT that work just fine with click to call and notification via TAPI drivers
 
we have some big concerns about this closed source java based AGI approach, its not needed for anything vtiger is really trying to do, at least as advertised, and it seems to want to take over and be a mini PBX inside the PBX, its making its own CDR records, forcing all calls to be recorded, and we are not sure what else but its apparently capable of changing how extensions work in call queues  
 
it would be nice if vtiger developers just gave in and used the AMI for origination and notification (they are doing that anyway in there AGI) and used SQL to tie vtiger records to Asterisk CDR records and call recordings if any even exist  
 
if this dont get fixed and these issues get addressed soon, we may look into trying to fool vtiger into firing our own php AGI for click to call, and just use the our own windows system tray call notification, or start looking into other CRM solutions altogether
 
 
 
 
 

From: "Zebra Hosting" <[hidden email]>
Sent: Tuesday, August 26, 2014 12:36 PM

To: "[hidden email]" <[hidden email]>
Subject: Re: [Vtigercrm-developers] Asterisk Good News
 
Had some time to play around with the PBX setup:

Setup is a fresh AsteriskNow with one trunk (voipbuster) where I only
allow to call out to my own mobile phone for now.
In and out calling works fine. Internal too.

Allow SIP guests and Allow Anonymus inbound SIP calls are both off for
security. (Unbelievable amount of idiots trying to hack in there)
0.0.0.0/0.0.0.0 is denied but 127.0.0.1, office and vTiger IP are allowed.

setup a manager called vtiger with the same permit/deny and basically all
rights. Did not work so went back to admin user/manager for now.

In the ³from-internal context I have exten =>
_X.,1,agi(agi://127.0.0.1/incoming.agi) but it stops all internal calls...

Any suggestions?

 
_______________________________________________
http://www.vtiger.com/

_______________________________________________
http://www.vtiger.com/

_______________________________________________
http://www.vtiger.com/
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Re: Asterisk Good News

Prasad-14
We will further follow up off the list and update the thread here again.

Connect with us on: Twitter I Facebook I Blog I Wiki Forums I Website


On Wed, Aug 27, 2014 at 7:05 AM, [hidden email] <[hidden email]> wrote:
closed source / open source no matter as long as it works and does no harm, but as it stands now I am surly not the the only asterisk admin that is going to feel this way about your product 
 
and you already are using AMI
 
from your webapp.sh data stream
 
65423 [Asterisk-Java ManagerConnection-0-Reader-0] INFO org.asteriskjava.manager.internal.ManagerConnectionImpl - Connected via Asterisk Call Manager/1.3
 
thats a AMI connection, made with the the AMI user credentials in vtiger pbx manager and manger.conf
 
 
 
 
 

From: "Prasad" <[hidden email]>
Sent: Tuesday, August 26, 2014 8:03 PM
To: "[hidden email]" <[hidden email]>, "[hidden email]" <[hidden email]>

Subject: Re: [Vtigercrm-developers] Asterisk Good News
 
We don't have in-house expertise working with AMI.
 
VtigerAsteriskApp is a web-app that connects AGI and Vtiger CRM - 
does no magic  other than proxying the requests to AGI using the library.
We have no short-term plans to make it open-souce than making 
the distribution  will be made freely downloadable.
 
We would follow up on the issue reported soon.
 
Regards,
Prasad 
 
Connect with us on: Twitter I Facebook I Blog I Wiki Forums I Website
 
On Wed, Aug 27, 2014 at 4:43 AM, [hidden email] <[hidden email]> wrote:
try [from-trunk] or any other context that only fires on inbound call from outside, should take care of that issue
 
 
though this is not ready for production, after a week of testing its unstable
 
worked fine for a few days, then:
 
it got stuck in endless loop repetitively notifying vtiger users of a inbound call
 
over the weekend it did something in which all extensions that where in vtiger no longer ring though call queues, even after removing them from vtiger, all the extensions that where not in vtiger still work
 
now we are dealing trying to deal with (on click to call)
 
the webapp.sh data stream reports
 
ug 26, 2014 5:43:29 PM com.vtiger.apps.asterisk.webapp.helpers.b a
INFO: Sending call originate request to asterisk
Aug 26, 2014 5:43:29 PM com.vtiger.apps.asterisk.webapp.helpers.b a
INFO: Context:[vtiger_outbound]
Aug 26, 2014 5:43:29 PM com.vtiger.apps.asterisk.webapp.helpers.b a
INFO: From: 101 -> To: 18032433300
Aug 26, 2014 5:43:29 PM com.vtiger.apps.asterisk.webapp.helpers.b a
INFO: Outgoing Call Response: Error
Aug 26, 2014 5:43:29 PM com.vtiger.apps.asterisk.webapp.helpers.b a
INFO: Asterisk Response: Extension does not exist.
 
 
 
no data seen in AMI, CLI, or agi.sh, and it never has wrote any file to its log directory
 
101 is a good extension, we can send an origination command directly to AMI and it works just fine
 
 
our primary business is hosting PBX, some of our clients have requested a hosted CRM to go along with it, to replace products like ACT that work just fine with click to call and notification via TAPI drivers
 
we have some big concerns about this closed source java based AGI approach, its not needed for anything vtiger is really trying to do, at least as advertised, and it seems to want to take over and be a mini PBX inside the PBX, its making its own CDR records, forcing all calls to be recorded, and we are not sure what else but its apparently capable of changing how extensions work in call queues  
 
it would be nice if vtiger developers just gave in and used the AMI for origination and notification (they are doing that anyway in there AGI) and used SQL to tie vtiger records to Asterisk CDR records and call recordings if any even exist  
 
if this dont get fixed and these issues get addressed soon, we may look into trying to fool vtiger into firing our own php AGI for click to call, and just use the our own windows system tray call notification, or start looking into other CRM solutions altogether
 
 
 
 
 

From: "Zebra Hosting" <[hidden email]>
Sent: Tuesday, August 26, 2014 12:36 PM

To: "[hidden email]" <[hidden email]>
Subject: Re: [Vtigercrm-developers] Asterisk Good News
 
Had some time to play around with the PBX setup:

Setup is a fresh AsteriskNow with one trunk (voipbuster) where I only
allow to call out to my own mobile phone for now.
In and out calling works fine. Internal too.

Allow SIP guests and Allow Anonymus inbound SIP calls are both off for
security. (Unbelievable amount of idiots trying to hack in there)
0.0.0.0/0.0.0.0 is denied but 127.0.0.1, office and vTiger IP are allowed.

setup a manager called vtiger with the same permit/deny and basically all
rights. Did not work so went back to admin user/manager for now.

In the ³from-internal context I have exten =>
_X.,1,agi(agi://127.0.0.1/incoming.agi) but it stops all internal calls...

Any suggestions?

 
_______________________________________________
http://www.vtiger.com/

_______________________________________________
http://www.vtiger.com/


_______________________________________________
http://www.vtiger.com/
Reply | Threaded
Open this post in threaded view
|

Re: Asterisk Good News

Zebra Hosting
Would be good to have a solid working system.
In general clients react very positive about the PBX connection option and feel it is important for their customer relation to see who is calling and to record the call data.
I have a basic understanding of Asterisk and struggle for days to get things working. 

Good if there is a bit more attention to this module and how to get things working. I believe someone was working on a more standard SIP solution?

Bastiaan Houtkooper
Zebra Hosting




On 27 aug. 2014, at 03:42, Prasad <[hidden email]> wrote:

We will further follow up off the list and update the thread here again.

Connect with us on: Twitter I Facebook I Blog I Wiki Forums I Website


On Wed, Aug 27, 2014 at 7:05 AM, [hidden email] <[hidden email]> wrote:
closed source / open source no matter as long as it works and does no harm, but as it stands now I am surly not the the only asterisk admin that is going to feel this way about your product 
 
and you already are using AMI
 
from your webapp.sh data stream
 
65423 [Asterisk-Java ManagerConnection-0-Reader-0] INFO org.asteriskjava.manager.internal.ManagerConnectionImpl - Connected via Asterisk Call Manager/1.3
 
thats a AMI connection, made with the the AMI user credentials in vtiger pbx manager and manger.conf
 
 
 
 
 

From: "Prasad" <[hidden email]>
Sent: Tuesday, August 26, 2014 8:03 PM
To: "[hidden email]" <[hidden email]>, "[hidden email]" <[hidden email]>

Subject: Re: [Vtigercrm-developers] Asterisk Good News
 
We don't have in-house expertise working with AMI.
 
VtigerAsteriskApp is a web-app that connects AGI and Vtiger CRM - 
does no magic  other than proxying the requests to AGI using the library.
We have no short-term plans to make it open-souce than making 
the distribution  will be made freely downloadable.
 
We would follow up on the issue reported soon.
 
Regards,
Prasad 
 
Connect with us on: Twitter I Facebook I Blog I Wiki Forums I Website
 
On Wed, Aug 27, 2014 at 4:43 AM, [hidden email] <[hidden email]> wrote:
try [from-trunk] or any other context that only fires on inbound call from outside, should take care of that issue
 
 
though this is not ready for production, after a week of testing its unstable
 
worked fine for a few days, then:
 
it got stuck in endless loop repetitively notifying vtiger users of a inbound call
 
over the weekend it did something in which all extensions that where in vtiger no longer ring though call queues, even after removing them from vtiger, all the extensions that where not in vtiger still work
 
now we are dealing trying to deal with (on click to call)
 
the webapp.sh data stream reports
 
ug 26, 2014 5:43:29 PM com.vtiger.apps.asterisk.webapp.helpers.b a
INFO: Sending call originate request to asterisk
Aug 26, 2014 5:43:29 PM com.vtiger.apps.asterisk.webapp.helpers.b a
INFO: Context:[vtiger_outbound]
Aug 26, 2014 5:43:29 PM com.vtiger.apps.asterisk.webapp.helpers.b a
INFO: From: 101 -> To: 18032433300
Aug 26, 2014 5:43:29 PM com.vtiger.apps.asterisk.webapp.helpers.b a
INFO: Outgoing Call Response: Error
Aug 26, 2014 5:43:29 PM com.vtiger.apps.asterisk.webapp.helpers.b a
INFO: Asterisk Response: Extension does not exist.
 
 
 
no data seen in AMI, CLI, or agi.sh, and it never has wrote any file to its log directory
 
101 is a good extension, we can send an origination command directly to AMI and it works just fine
 
 
our primary business is hosting PBX, some of our clients have requested a hosted CRM to go along with it, to replace products like ACT that work just fine with click to call and notification via TAPI drivers
 
we have some big concerns about this closed source java based AGI approach, its not needed for anything vtiger is really trying to do, at least as advertised, and it seems to want to take over and be a mini PBX inside the PBX, its making its own CDR records, forcing all calls to be recorded, and we are not sure what else but its apparently capable of changing how extensions work in call queues  
 
it would be nice if vtiger developers just gave in and used the AMI for origination and notification (they are doing that anyway in there AGI) and used SQL to tie vtiger records to Asterisk CDR records and call recordings if any even exist  
 
if this dont get fixed and these issues get addressed soon, we may look into trying to fool vtiger into firing our own php AGI for click to call, and just use the our own windows system tray call notification, or start looking into other CRM solutions altogether
 
 
 
 
 

From: "Zebra Hosting" <[hidden email]>
Sent: Tuesday, August 26, 2014 12:36 PM

To: "[hidden email]" <[hidden email]>
Subject: Re: [Vtigercrm-developers] Asterisk Good News
 
Had some time to play around with the PBX setup:

Setup is a fresh AsteriskNow with one trunk (voipbuster) where I only
allow to call out to my own mobile phone for now.
In and out calling works fine. Internal too.

Allow SIP guests and Allow Anonymus inbound SIP calls are both off for
security. (Unbelievable amount of idiots trying to hack in there)
0.0.0.0/0.0.0.0 is denied but 127.0.0.1, office and vTiger IP are allowed.

setup a manager called vtiger with the same permit/deny and basically all
rights. Did not work so went back to admin user/manager for now.

In the ³from-internal context I have exten =>
_X.,1,agi(agi://127.0.0.1/incoming.agi) but it stops all internal calls...

Any suggestions?

 
_______________________________________________
http://www.vtiger.com/

_______________________________________________
http://www.vtiger.com/

_______________________________________________
http://www.vtiger.com/


_______________________________________________
http://www.vtiger.com/
Reply | Threaded
Open this post in threaded view
|

Re: Asterisk Good News

vtiger@camden.net
In reply to this post by Zebra Hosting
yes I am aware of what it was saying... but 101 was most certainly a valid extension, at the same time this was going on we could send our own origination command to AMI for 101 to call out and it worked just fine, that error is not produced by asterisk at all, its not a valid asterisk error return, you are mis-interrupting some other error and reporting it back as a asterisk error 
 
I think it was some error inside your java, at the time this was happening we only saw data stream on webapp.sh on the click to call attempts, we where also looking at data streams on your agi.sh saw nothing, the cli with agi and verbose 10 notifications enabled saw nothing, and we also looking at the AMI stream, nothing there as well, so we dont even think your connector was even talking to asterisk at all
 
we fixed the issue last night, stopped the webapp.sh and agi.sh files and prevented them from start @ boot, rebooted, cleared settings in vtiger PBX manager or rather put in a bunch of junk, saved, rebooted, put settings back into PBX manager, rebooted then turned the .sh files back on, not sure what steps fixed but simple stop and start of .sh files and/or system reboots did not fix
 
 
we still have issue on on test server where it seems vtiger altered something in asterisk where all the extensions we had in vtiger no longer will ring though a call queue, we have no data on the event we where not actively monitoring data streams, and your asterisk connector has never written 1 log to its own log directory, everything was working Friday, and Monday we checked on it again and found that vtiger was stuck in endless loop of inbound notification and those extensions would no longer sent a sip invite though call queue, even after they where removed from vtiger and a reboot, and the asterisk connector was completely shut down, keep in mind these are test servers dedicated to getting vtiger to work, they are lamp + asterisk + freepbx + vtiger only we have not moved into phase of testing with other packages so the problems we are having can only be vtiger, this is what concerns us about closed source java+agi, beyond we can not see code to try to help you we now have situation where it seems your connector and AGI have altered asterisk, we dont know whats it doing.
 
the whole reason we advocate for just use the AMI is its simple and works and your not really doing anything that requires the interactivity that AGI allows, AMI can be used to originate call as well as be monitored for inbound, this is how tapi drivers work that plug into on site CRMs like ACT, SAGE CRM, Lexis Nexus, ect, its also how sugar / sweet crm are handling it
 
I think the tiring the call records to data in vtiger is a great idea, but why are you generating your own CDR log's via AGI, and this is where I think you guys are hung up on using AGI, but asterisk does have it own MySQL tables and CDR reccords, why not during the isntall of vtiger ask for the asterisk database name and credentials and query them for the info, and vtiger should not be forcing all calls to record, asterisk gives us ways to record and not record calls, if you want to link to recordings fine but again you should be looking to see if a call was recorded, beyond a performance issue and potential duplication of whats already being done in the Asterisk, this ventures into a legal issue, a lot of places all parties must be notified and consent to call recording, with vtiger it forces the recording
 
the general rule of thumb here is if you need user input like your making a IVR and logging what was entered then you need AGI, if your just originating calls and monitoring for inbound use AMI, in either case SQL should be used to get call record data
 
AGI has it problems that you are unnecessarily dealing with, they are common for AGI, what if my agi script finishes before the call ends, what if the call ends before my agi script finishes and a whole host of other issues that you simply dont need to deal with for what your trying to do   
 
 
 
 
 
 
 
 
 
 

From: "Akshath T.A" <[hidden email]>
Sent: Wednesday, August 27, 2014 3:19 AM
To: [hidden email]
Subject: Re: [Vtigercrm-developers] Asterisk Good News
 
Hi,
 
From the logs, it clearly states that the extension - 101 was not found by your Asterisk server when AMI originate request sent from Vtiger Connector.
 
In order to nail down the issue, can we have a online meeting today or tomorrow so that it would help us know what really is causing the problem. 
 
I will be available from 9:00 AM to 8:00 PM IST at my office. Please tell us your convenient time within this so that I can send you meeting link.
 
Thank you.
 
 
regards,
Akshath
  Vtiger

_______________________________________________
http://www.vtiger.com/
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Re: Asterisk Good News

vtiger@camden.net
In reply to this post by Zebra Hosting
SIP would not be the best idea, yes its even easier to deal with than AMI but asterisk can only handle 1 sip registration per extension so this will lead to the need to have 2 extensions, one for your phone and one for vtiger you would then need to daisy chain them in follow me
 
same reason SIPTapi from IPcom never took off, yes you can get the free version and click to call outbound works with only one extension but if you want inbound you have to pay and will find out after you need 2 extensions  
 
 
 

From: "Zebra Hosting" <[hidden email]>
Sent: Wednesday, August 27, 2014 2:18 AM
To: "[hidden email]" <[hidden email]>
Subject: Re: [Vtigercrm-developers] Asterisk Good News
 
Would be good to have a solid working system.
In general clients react very positive about the PBX connection option and feel it is important for their customer relation to see who is calling and to record the call data.
I have a basic understanding of Asterisk and struggle for days to get things working. 
 
Good if there is a bit more attention to this module and how to get things working. I believe someone was working on a more standard SIP solution?
 
Bastiaan Houtkooper
Zebra Hosting
 
 
 
 
On 27 aug. 2014, at 03:42, Prasad <[hidden email]> wrote:
 
We will further follow up off the list and update the thread here again.
 
Connect with us on: Twitter I Facebook I Blog I Wiki Forums I Website
 
On Wed, Aug 27, 2014 at 7:05 AM, [hidden email] <[hidden email]> wrote:
closed source / open source no matter as long as it works and does no harm, but as it stands now I am surly not the the only asterisk admin that is going to feel this way about your product 
 
and you already are using AMI
 
from your webapp.sh data stream
 
65423 [Asterisk-Java ManagerConnection-0-Reader-0] INFO org.asteriskjava.manager.internal.ManagerConnectionImpl - Connected via Asterisk Call Manager/1.3
 
thats a AMI connection, made with the the AMI user credentials in vtiger pbx manager and manger.conf
 
 
 
 
 

From: "Prasad" <[hidden email]>
Sent: Tuesday, August 26, 2014 8:03 PM
To: "[hidden email]" <[hidden email]>, "[hidden email]" <[hidden email]>

Subject: Re: [Vtigercrm-developers] Asterisk Good News
 
We don't have in-house expertise working with AMI.
 
VtigerAsteriskApp is a web-app that connects AGI and Vtiger CRM - 
does no magic  other than proxying the requests to AGI using the library.
We have no short-term plans to make it open-souce than making 
the distribution  will be made freely downloadable.
 
We would follow up on the issue reported soon.
 
Regards,
Prasad 
 
Connect with us on: Twitter I Facebook I Blog I Wiki Forums I Website
 
On Wed, Aug 27, 2014 at 4:43 AM, [hidden email] <[hidden email]> wrote:
try [from-trunk] or any other context that only fires on inbound call from outside, should take care of that issue
 
 
though this is not ready for production, after a week of testing its unstable
 
worked fine for a few days, then:
 
it got stuck in endless loop repetitively notifying vtiger users of a inbound call
 
over the weekend it did something in which all extensions that where in vtiger no longer ring though call queues, even after removing them from vtiger, all the extensions that where not in vtiger still work
 
now we are dealing trying to deal with (on click to call)
 
the webapp.sh data stream reports
 
ug 26, 2014 5:43:29 PM com.vtiger.apps.asterisk.webapp.helpers.b a
INFO: Sending call originate request to asterisk
Aug 26, 2014 5:43:29 PM com.vtiger.apps.asterisk.webapp.helpers.b a
INFO: Context:[vtiger_outbound]
Aug 26, 2014 5:43:29 PM com.vtiger.apps.asterisk.webapp.helpers.b a
INFO: From: 101 -> To: 18032433300
Aug 26, 2014 5:43:29 PM com.vtiger.apps.asterisk.webapp.helpers.b a
INFO: Outgoing Call Response: Error
Aug 26, 2014 5:43:29 PM com.vtiger.apps.asterisk.webapp.helpers.b a
INFO: Asterisk Response: Extension does not exist.
 
 
 
no data seen in AMI, CLI, or agi.sh, and it never has wrote any file to its log directory
 
101 is a good extension, we can send an origination command directly to AMI and it works just fine
 
 
our primary business is hosting PBX, some of our clients have requested a hosted CRM to go along with it, to replace products like ACT that work just fine with click to call and notification via TAPI drivers
 
we have some big concerns about this closed source java based AGI approach, its not needed for anything vtiger is really trying to do, at least as advertised, and it seems to want to take over and be a mini PBX inside the PBX, its making its own CDR records, forcing all calls to be recorded, and we are not sure what else but its apparently capable of changing how extensions work in call queues  
 
it would be nice if vtiger developers just gave in and used the AMI for origination and notification (they are doing that anyway in there AGI) and used SQL to tie vtiger records to Asterisk CDR records and call recordings if any even exist  
 
if this dont get fixed and these issues get addressed soon, we may look into trying to fool vtiger into firing our own php AGI for click to call, and just use the our own windows system tray call notification, or start looking into other CRM solutions altogether
 
 
 
 
 

From: "Zebra Hosting" <[hidden email]>
Sent: Tuesday, August 26, 2014 12:36 PM

To: "[hidden email]" <[hidden email]>
Subject: Re: [Vtigercrm-developers] Asterisk Good News
 
Had some time to play around with the PBX setup:

Setup is a fresh AsteriskNow with one trunk (voipbuster) where I only
allow to call out to my own mobile phone for now.
In and out calling works fine. Internal too.

Allow SIP guests and Allow Anonymus inbound SIP calls are both off for
security. (Unbelievable amount of idiots trying to hack in there)
0.0.0.0/0.0.0.0 is denied but 127.0.0.1, office and vTiger IP are allowed.

setup a manager called vtiger with the same permit/deny and basically all
rights. Did not work so went back to admin user/manager for now.

In the ³from-internal context I have exten =>
_X.,1,agi(agi://127.0.0.1/incoming.agi) but it stops all internal calls...

Any suggestions?

 
_______________________________________________
http://www.vtiger.com/

_______________________________________________
http://www.vtiger.com/
_______________________________________________
http://www.vtiger.com/

_______________________________________________
http://www.vtiger.com/

Attachment 1 (22K) Download Attachment
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Re: Asterisk Good News

Alan Bell-6
I did write most of a SIP module, I am planning to finish it off and perhaps put it in the marketplace once that is all up and running, it certainly doesn't require multiple extensions to do click to call and inbound, it uses a SIP proxy, basically a benevolent man in the middle attack on the sip signalling :) this proxy can see the calls in both directions being set up, the handsets think that the proxy is the PBX, the PBX thinks the proxy is the handsets. The proxy could be running on the same machine as the PBX, or the same machine as vtiger, or somewhere else altogether.

Alan.

On 27/08/14 15:16, [hidden email] wrote:
SIP would not be the best idea, yes its even easier to deal with than AMI but asterisk can only handle 1 sip registration per extension so this will lead to the need to have 2 extensions, one for your phone and one for vtiger you would then need to daisy chain them in follow me
 
same reason SIPTapi from IPcom never took off, yes you can get the free version and click to call outbound works with only one extension but if you want inbound you have to pay and will find out after you need 2 extensions  
 
 
 



_______________________________________________
http://www.vtiger.com/


-- 
Libertus Solutions
http://libertus.co.uk

_______________________________________________
http://www.vtiger.com/
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|

Re: Asterisk Good News

Akshath T.A
In reply to this post by vtiger@camden.net
Are you free for meeting now? I will send you the meeting link.

​​



On Wed, Aug 27, 2014 at 7:39 PM, [hidden email] <[hidden email]> wrote:
yes I am aware of what it was saying... but 101 was most certainly a valid extension, at the same time this was going on we could send our own origination command to AMI for 101 to call out and it worked just fine, that error is not produced by asterisk at all, its not a valid asterisk error return, you are mis-interrupting some other error and reporting it back as a asterisk error 
 
I think it was some error inside your java, at the time this was happening we only saw data stream on webapp.sh on the click to call attempts, we where also looking at data streams on your agi.sh saw nothing, the cli with agi and verbose 10 notifications enabled saw nothing, and we also looking at the AMI stream, nothing there as well, so we dont even think your connector was even talking to asterisk at all
 
we fixed the issue last night, stopped the webapp.sh and agi.sh files and prevented them from start @ boot, rebooted, cleared settings in vtiger PBX manager or rather put in a bunch of junk, saved, rebooted, put settings back into PBX manager, rebooted then turned the .sh files back on, not sure what steps fixed but simple stop and start of .sh files and/or system reboots did not fix
 
 
we still have issue on on test server where it seems vtiger altered something in asterisk where all the extensions we had in vtiger no longer will ring though a call queue, we have no data on the event we where not actively monitoring data streams, and your asterisk connector has never written 1 log to its own log directory, everything was working Friday, and Monday we checked on it again and found that vtiger was stuck in endless loop of inbound notification and those extensions would no longer sent a sip invite though call queue, even after they where removed from vtiger and a reboot, and the asterisk connector was completely shut down, keep in mind these are test servers dedicated to getting vtiger to work, they are lamp + asterisk + freepbx + vtiger only we have not moved into phase of testing with other packages so the problems we are having can only be vtiger, this is what concerns us about closed source java+agi, beyond we can not see code to try to help you we now have situation where it seems your connector and AGI have altered asterisk, we dont know whats it doing.
 
the whole reason we advocate for just use the AMI is its simple and works and your not really doing anything that requires the interactivity that AGI allows, AMI can be used to originate call as well as be monitored for inbound, this is how tapi drivers work that plug into on site CRMs like ACT, SAGE CRM, Lexis Nexus, ect, its also how sugar / sweet crm are handling it
 
I think the tiring the call records to data in vtiger is a great idea, but why are you generating your own CDR log's via AGI, and this is where I think you guys are hung up on using AGI, but asterisk does have it own MySQL tables and CDR reccords, why not during the isntall of vtiger ask for the asterisk database name and credentials and query them for the info, and vtiger should not be forcing all calls to record, asterisk gives us ways to record and not record calls, if you want to link to recordings fine but again you should be looking to see if a call was recorded, beyond a performance issue and potential duplication of whats already being done in the Asterisk, this ventures into a legal issue, a lot of places all parties must be notified and consent to call recording, with vtiger it forces the recording
 
the general rule of thumb here is if you need user input like your making a IVR and logging what was entered then you need AGI, if your just originating calls and monitoring for inbound use AMI, in either case SQL should be used to get call record data
 
AGI has it problems that you are unnecessarily dealing with, they are common for AGI, what if my agi script finishes before the call ends, what if the call ends before my agi script finishes and a whole host of other issues that you simply dont need to deal with for what your trying to do   
 
 
 
 
 
 
 
 
 
 

From: "Akshath T.A" <[hidden email]>
Sent: Wednesday, August 27, 2014 3:19 AM
To: [hidden email]
Subject: Re: [Vtigercrm-developers] Asterisk Good News
 
Hi,
 
From the logs, it clearly states that the extension - 101 was not found by your Asterisk server when AMI originate request sent from Vtiger Connector.
 
In order to nail down the issue, can we have a online meeting today or tomorrow so that it would help us know what really is causing the problem. 
 
I will be available from 9:00 AM to 8:00 PM IST at my office. Please tell us your convenient time within this so that I can send you meeting link.
 
Thank you.
 
 
regards,
Akshath
  Vtiger


_______________________________________________
http://www.vtiger.com/
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Re: Asterisk Good News

vtiger@camden.net
In reply to this post by Alan Bell-6
very interesting using a SIP proxy ;-), would love to check it out when you get it finished
 
how will it handle inbound notification? a in browser pop like vtiger does now? 
 
when you say marketplace I assume extension store? will this support complete 3rd party apps like the sip proxy server your building, from everything I read it seemed more geared to modules that install directly into vtiger
 
if its going to support 3rd party applications we are working on a windows system trey call pop for vtiger so your browser can be minimized and you are still altered to the call with links to relevant records,  in alpha testing now but we would be interested in this extension store     
 
 
 

From: "Alan Bell" <[hidden email]>
Sent: Wednesday, August 27, 2014 10:25 AM
To: [hidden email]
Subject: Re: [Vtigercrm-developers] Asterisk Good News
 
I did write most of a SIP module, I am planning to finish it off and perhaps put it in the marketplace once that is all up and running, it certainly doesn't require multiple extensions to do click to call and inbound, it uses a SIP proxy, basically a benevolent man in the middle attack on the sip signalling :) this proxy can see the calls in both directions being set up, the handsets think that the proxy is the PBX, the PBX thinks the proxy is the handsets. The proxy could be running on the same machine as the PBX, or the same machine as vtiger, or somewhere else altogether.

Alan.

On 27/08/14 15:16, [hidden email] wrote:
SIP would not be the best idea, yes its even easier to deal with than AMI but asterisk can only handle 1 sip registration per extension so this will lead to the need to have 2 extensions, one for your phone and one for vtiger you would then need to daisy chain them in follow me
 
same reason SIPTapi from IPcom never took off, yes you can get the free version and click to call outbound works with only one extension but if you want inbound you have to pay and will find out after you need 2 extensions  
 
 
 

 
 
 
_______________________________________________
http://www.vtiger.com/
 
--
Libertus Solutions
http://libertus.co.uk

_______________________________________________
http://www.vtiger.com/
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Re: Asterisk Good News

Akshath T.A
In reply to this post by vtiger@camden.net
VtigerAsteriskApp does not modify any configuration of Asterisk directly. We have recommended
changes to be made in Asterisk configuration to get the AGI request back to the App.
Please make sure to track the changes in case you need to revert back.

AGI vs AMI

AGI events are listened for incoming calls as AMI event parsing had to deal with bit-of overhead 
(as we did in 5.4.0 Asterisk Integration). To origniate outgoing call AMI action is used 
(borrowed from our earlier implementation).

Working with AGI events is lot-easier than processing AMI ourselves. The hard-work
is taken care by the asteriskjava library.

Polling Asterisk DB from CRM server was not a model considered as both the applications can be setup on different machines. 
Also php script used for regular polling in short-intervals (specially during active call handling) demands
server resources and impatcts CRM performance - we had encountered this in our earlier implementation.

The approach of implementation we have taken may not be the only way to get the integration.
We are hopeful to see more such coming from community and take away the learning from the same.


regards,
AKshath





On Wed, Aug 27, 2014 at 7:39 PM, [hidden email] <[hidden email]> wrote:
yes I am aware of what it was saying... but 101 was most certainly a valid extension, at the same time this was going on we could send our own origination command to AMI for 101 to call out and it worked just fine, that error is not produced by asterisk at all, its not a valid asterisk error return, you are mis-interrupting some other error and reporting it back as a asterisk error 
 
I think it was some error inside your java, at the time this was happening we only saw data stream on webapp.sh on the click to call attempts, we where also looking at data streams on your agi.sh saw nothing, the cli with agi and verbose 10 notifications enabled saw nothing, and we also looking at the AMI stream, nothing there as well, so we dont even think your connector was even talking to asterisk at all
 
we fixed the issue last night, stopped the webapp.sh and agi.sh files and prevented them from start @ boot, rebooted, cleared settings in vtiger PBX manager or rather put in a bunch of junk, saved, rebooted, put settings back into PBX manager, rebooted then turned the .sh files back on, not sure what steps fixed but simple stop and start of .sh files and/or system reboots did not fix
 
 
we still have issue on on test server where it seems vtiger altered something in asterisk where all the extensions we had in vtiger no longer will ring though a call queue, we have no data on the event we where not actively monitoring data streams, and your asterisk connector has never written 1 log to its own log directory, everything was working Friday, and Monday we checked on it again and found that vtiger was stuck in endless loop of inbound notification and those extensions would no longer sent a sip invite though call queue, even after they where removed from vtiger and a reboot, and the asterisk connector was completely shut down, keep in mind these are test servers dedicated to getting vtiger to work, they are lamp + asterisk + freepbx + vtiger only we have not moved into phase of testing with other packages so the problems we are having can only be vtiger, this is what concerns us about closed source java+agi, beyond we can not see code to try to help you we now have situation where it seems your connector and AGI have altered asterisk, we dont know whats it doing.
 
the whole reason we advocate for just use the AMI is its simple and works and your not really doing anything that requires the interactivity that AGI allows, AMI can be used to originate call as well as be monitored for inbound, this is how tapi drivers work that plug into on site CRMs like ACT, SAGE CRM, Lexis Nexus, ect, its also how sugar / sweet crm are handling it
 
I think the tiring the call records to data in vtiger is a great idea, but why are you generating your own CDR log's via AGI, and this is where I think you guys are hung up on using AGI, but asterisk does have it own MySQL tables and CDR reccords, why not during the isntall of vtiger ask for the asterisk database name and credentials and query them for the info, and vtiger should not be forcing all calls to record, asterisk gives us ways to record and not record calls, if you want to link to recordings fine but again you should be looking to see if a call was recorded, beyond a performance issue and potential duplication of whats already being done in the Asterisk, this ventures into a legal issue, a lot of places all parties must be notified and consent to call recording, with vtiger it forces the recording
 
the general rule of thumb here is if you need user input like your making a IVR and logging what was entered then you need AGI, if your just originating calls and monitoring for inbound use AMI, in either case SQL should be used to get call record data
 
AGI has it problems that you are unnecessarily dealing with, they are common for AGI, what if my agi script finishes before the call ends, what if the call ends before my agi script finishes and a whole host of other issues that you simply dont need to deal with for what your trying to do   
 
 
 
 
 
 
 
 
 
 

From: "Akshath T.A" <[hidden email]>
Sent: Wednesday, August 27, 2014 3:19 AM
To: [hidden email]
Subject: Re: [Vtigercrm-developers] Asterisk Good News
 
Hi,
 
From the logs, it clearly states that the extension - 101 was not found by your Asterisk server when AMI originate request sent from Vtiger Connector.
 
In order to nail down the issue, can we have a online meeting today or tomorrow so that it would help us know what really is causing the problem. 
 
I will be available from 9:00 AM to 8:00 PM IST at my office. Please tell us your convenient time within this so that I can send you meeting link.
 
Thank you.
 
 
regards,
Akshath
  Vtiger


_______________________________________________
http://www.vtiger.com/
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Re: Asterisk Good News

Prasad-14
In reply to this post by Akshath T.A
I noted a scenario that can get the popup stuck on UI due to unexpected / no-response - 
hopefully when we meet this can be reviewed: http://trac.vtiger.com/cgi-bin/trac.cgi/ticket/8195

Regards,
Prasad

Connect with us on: Twitter I Facebook I Blog I Wiki Forums I Website


On Wed, Aug 27, 2014 at 8:10 PM, Akshath T.A <[hidden email]> wrote:
Are you free for meeting now? I will send you the meeting link.

​​



On Wed, Aug 27, 2014 at 7:39 PM, [hidden email] <[hidden email]> wrote:
yes I am aware of what it was saying... but 101 was most certainly a valid extension, at the same time this was going on we could send our own origination command to AMI for 101 to call out and it worked just fine, that error is not produced by asterisk at all, its not a valid asterisk error return, you are mis-interrupting some other error and reporting it back as a asterisk error 
 
I think it was some error inside your java, at the time this was happening we only saw data stream on webapp.sh on the click to call attempts, we where also looking at data streams on your agi.sh saw nothing, the cli with agi and verbose 10 notifications enabled saw nothing, and we also looking at the AMI stream, nothing there as well, so we dont even think your connector was even talking to asterisk at all
 
we fixed the issue last night, stopped the webapp.sh and agi.sh files and prevented them from start @ boot, rebooted, cleared settings in vtiger PBX manager or rather put in a bunch of junk, saved, rebooted, put settings back into PBX manager, rebooted then turned the .sh files back on, not sure what steps fixed but simple stop and start of .sh files and/or system reboots did not fix
 
 
we still have issue on on test server where it seems vtiger altered something in asterisk where all the extensions we had in vtiger no longer will ring though a call queue, we have no data on the event we where not actively monitoring data streams, and your asterisk connector has never written 1 log to its own log directory, everything was working Friday, and Monday we checked on it again and found that vtiger was stuck in endless loop of inbound notification and those extensions would no longer sent a sip invite though call queue, even after they where removed from vtiger and a reboot, and the asterisk connector was completely shut down, keep in mind these are test servers dedicated to getting vtiger to work, they are lamp + asterisk + freepbx + vtiger only we have not moved into phase of testing with other packages so the problems we are having can only be vtiger, this is what concerns us about closed source java+agi, beyond we can not see code to try to help you we now have situation where it seems your connector and AGI have altered asterisk, we dont know whats it doing.
 
the whole reason we advocate for just use the AMI is its simple and works and your not really doing anything that requires the interactivity that AGI allows, AMI can be used to originate call as well as be monitored for inbound, this is how tapi drivers work that plug into on site CRMs like ACT, SAGE CRM, Lexis Nexus, ect, its also how sugar / sweet crm are handling it
 
I think the tiring the call records to data in vtiger is a great idea, but why are you generating your own CDR log's via AGI, and this is where I think you guys are hung up on using AGI, but asterisk does have it own MySQL tables and CDR reccords, why not during the isntall of vtiger ask for the asterisk database name and credentials and query them for the info, and vtiger should not be forcing all calls to record, asterisk gives us ways to record and not record calls, if you want to link to recordings fine but again you should be looking to see if a call was recorded, beyond a performance issue and potential duplication of whats already being done in the Asterisk, this ventures into a legal issue, a lot of places all parties must be notified and consent to call recording, with vtiger it forces the recording
 
the general rule of thumb here is if you need user input like your making a IVR and logging what was entered then you need AGI, if your just originating calls and monitoring for inbound use AMI, in either case SQL should be used to get call record data
 
AGI has it problems that you are unnecessarily dealing with, they are common for AGI, what if my agi script finishes before the call ends, what if the call ends before my agi script finishes and a whole host of other issues that you simply dont need to deal with for what your trying to do   
 
 
 
 
 
 
 
 
 
 

From: "Akshath T.A" <[hidden email]>
Sent: Wednesday, August 27, 2014 3:19 AM
To: [hidden email]
Subject: Re: [Vtigercrm-developers] Asterisk Good News
 
Hi,
 
From the logs, it clearly states that the extension - 101 was not found by your Asterisk server when AMI originate request sent from Vtiger Connector.
 
In order to nail down the issue, can we have a online meeting today or tomorrow so that it would help us know what really is causing the problem. 
 
I will be available from 9:00 AM to 8:00 PM IST at my office. Please tell us your convenient time within this so that I can send you meeting link.
 
Thank you.
 
 
regards,
Akshath
  Vtiger


_______________________________________________
http://www.vtiger.com/


_______________________________________________
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Re: Asterisk Good News

Alan Bell-6
In reply to this post by vtiger@camden.net
On 27/08/14 15:43, [hidden email] wrote:
very interesting using a SIP proxy ;-), would love to check it out when you get it finished
 
how will it handle inbound notification? a in browser pop like vtiger does now?
yeah, but probably based on web sockets rather than polling, if I can do that cleanly in a module without ridiculous dependencies.
 
when you say marketplace I assume extension store?
yup, it would be an open source solution, but packed up in the extension store for easy installation, possibly with a price tag on it a bit like the geotools module.
will this support complete 3rd party apps like the sip proxy server your building, from everything I read it seemed more geared to modules that install directly into vtiger
well, yeah, that is an interesting question, and one reason why I parked the project for a bit. It might be that I run an opensips server as a service in the cloud for users of the module, and give instructions on how to run your own opensips proxy for people who don't want to have their telephone metadata going outside the organisation (other than to the NSA of course). One of the things I would have to figure out for a cloud based sip proxy is getting it to work and be maintainable, but store no logs, I don't want that data!

 
if its going to support 3rd party applications we are working on a windows system trey call pop for vtiger so your browser can be minimized and you are still altered to the call with links to relevant records,  in alpha testing now but we would be interested in this extension store    

 
 
 

From: "Alan Bell" [hidden email]
Sent: Wednesday, August 27, 2014 10:25 AM
To: [hidden email]
Subject: Re: [Vtigercrm-developers] Asterisk Good News
 
I did write most of a SIP module, I am planning to finish it off and perhaps put it in the marketplace once that is all up and running, it certainly doesn't require multiple extensions to do click to call and inbound, it uses a SIP proxy, basically a benevolent man in the middle attack on the sip signalling :) this proxy can see the calls in both directions being set up, the handsets think that the proxy is the PBX, the PBX thinks the proxy is the handsets. The proxy could be running on the same machine as the PBX, or the same machine as vtiger, or somewhere else altogether.

Alan.

On 27/08/14 15:16, [hidden email] wrote:
SIP would not be the best idea, yes its even easier to deal with than AMI but asterisk can only handle 1 sip registration per extension so this will lead to the need to have 2 extensions, one for your phone and one for vtiger you would then need to daisy chain them in follow me
 
same reason SIPTapi from IPcom never took off, yes you can get the free version and click to call outbound works with only one extension but if you want inbound you have to pay and will find out after you need 2 extensions  
 
 
 

 
 
 
_______________________________________________
http://www.vtiger.com/
 
-- 
Libertus Solutions
http://libertus.co.uk


_______________________________________________
http://www.vtiger.com/


-- 
Libertus Solutions
http://libertus.co.uk

_______________________________________________
http://www.vtiger.com/
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Re: Asterisk Good News

vtiger@camden.net
In reply to this post by Akshath T.A
yes the overhead for AMI listen can be issue
 
normally dealt with by giving the AMI user Origination and Read Call only rights, that cuts down on 99% of what you need to parse
 
in larger deployments where even the overhead of incoming calls to other extensions is burdensome, AMI proxy where each user has a proxy user that only listens to AMI data relevant to there own channel(s) 
 
 
I would think you would only pull SQL data when user in vtiger clicks on PBXmanager (the user version where they can see the call logs, not the one for admin to change settings), and display the data rather than pulling it all the time and creating your own tables, if remote access to SQL is the concern its easy to write a php XML API, thats how our windows based call notification works for vtiger so we did not have to open SQL to outside world, when call comes in we take incomming phone number send it to our API, the API does all the SQL queries, then returns XML of the results, we did this with php, all that would be required is for you guys to say hey put this php file on your asterisk server and modify the database name, user and secret to match
 
  
 
 
 

From: "Akshath T.A" <[hidden email]>
Sent: Wednesday, August 27, 2014 11:09 AM
To: [hidden email]
Cc: [hidden email]
Subject: Re: [Vtigercrm-developers] Asterisk Good News
 
VtigerAsteriskApp does not modify any configuration of Asterisk directly. We have recommended
changes to be made in Asterisk configuration to get the AGI request back to the App.
Please make sure to track the changes in case you need to revert back.

AGI vs AMI

AGI events are listened for incoming calls as AMI event parsing had to deal with bit-of overhead 
(as we did in 5.4.0 Asterisk Integration). To origniate outgoing call AMI action is used 
(borrowed from our earlier implementation).

Working with AGI events is lot-easier than processing AMI ourselves. The hard-work
is taken care by the asteriskjava library.
 
Polling Asterisk DB from CRM server was not a model considered as both the applications can be setup on different machines. 
Also php script used for regular polling in short-intervals (specially during active call handling) demands
server resources and impatcts CRM performance - we had encountered this in our earlier implementation.
 
The approach of implementation we have taken may not be the only way to get the integration.
We are hopeful to see more such coming from community and take away the learning from the same.
 
 
regards,
AKshath
 
 
 
 
On Wed, Aug 27, 2014 at 7:39 PM, [hidden email] <[hidden email]> wrote:
yes I am aware of what it was saying... but 101 was most certainly a valid extension, at the same time this was going on we could send our own origination command to AMI for 101 to call out and it worked just fine, that error is not produced by asterisk at all, its not a valid asterisk error return, you are mis-interrupting some other error and reporting it back as a asterisk error 
 
I think it was some error inside your java, at the time this was happening we only saw data stream on webapp.sh on the click to call attempts, we where also looking at data streams on your agi.sh saw nothing, the cli with agi and verbose 10 notifications enabled saw nothing, and we also looking at the AMI stream, nothing there as well, so we dont even think your connector was even talking to asterisk at all
 
we fixed the issue last night, stopped the webapp.sh and agi.sh files and prevented them from start @ boot, rebooted, cleared settings in vtiger PBX manager or rather put in a bunch of junk, saved, rebooted, put settings back into PBX manager, rebooted then turned the .sh files back on, not sure what steps fixed but simple stop and start of .sh files and/or system reboots did not fix
 
 
we still have issue on on test server where it seems vtiger altered something in asterisk where all the extensions we had in vtiger no longer will ring though a call queue, we have no data on the event we where not actively monitoring data streams, and your asterisk connector has never written 1 log to its own log directory, everything was working Friday, and Monday we checked on it again and found that vtiger was stuck in endless loop of inbound notification and those extensions would no longer sent a sip invite though call queue, even after they where removed from vtiger and a reboot, and the asterisk connector was completely shut down, keep in mind these are test servers dedicated to getting vtiger to work, they are lamp + asterisk + freepbx + vtiger only we have not moved into phase of testing with other packages so the problems we are having can only be vtiger, this is what concerns us about closed source java+agi, beyond we can not see code to try to help you we now have situation where it seems your connector and AGI have altered asterisk, we dont know whats it doing.
 
the whole reason we advocate for just use the AMI is its simple and works and your not really doing anything that requires the interactivity that AGI allows, AMI can be used to originate call as well as be monitored for inbound, this is how tapi drivers work that plug into on site CRMs like ACT, SAGE CRM, Lexis Nexus, ect, its also how sugar / sweet crm are handling it
 
I think the tiring the call records to data in vtiger is a great idea, but why are you generating your own CDR log's via AGI, and this is where I think you guys are hung up on using AGI, but asterisk does have it own MySQL tables and CDR reccords, why not during the isntall of vtiger ask for the asterisk database name and credentials and query them for the info, and vtiger should not be forcing all calls to record, asterisk gives us ways to record and not record calls, if you want to link to recordings fine but again you should be looking to see if a call was recorded, beyond a performance issue and potential duplication of whats already being done in the Asterisk, this ventures into a legal issue, a lot of places all parties must be notified and consent to call recording, with vtiger it forces the recording
 
the general rule of thumb here is if you need user input like your making a IVR and logging what was entered then you need AGI, if your just originating calls and monitoring for inbound use AMI, in either case SQL should be used to get call record data
 
AGI has it problems that you are unnecessarily dealing with, they are common for AGI, what if my agi script finishes before the call ends, what if the call ends before my agi script finishes and a whole host of other issues that you simply dont need to deal with for what your trying to do   
 
 
 
 
 
 
 
 
 
 

From: "Akshath T.A" <[hidden email]>
Sent: Wednesday, August 27, 2014 3:19 AM
To: [hidden email]
Subject: Re: [Vtigercrm-developers] Asterisk Good News
 
Hi,
 
From the logs, it clearly states that the extension - 101 was not found by your Asterisk server when AMI originate request sent from Vtiger Connector.
 
In order to nail down the issue, can we have a online meeting today or tomorrow so that it would help us know what really is causing the problem. 
 
I will be available from 9:00 AM to 8:00 PM IST at my office. Please tell us your convenient time within this so that I can send you meeting link.
 
Thank you.
 
 
regards,
Akshath
  Vtiger

_______________________________________________
http://www.vtiger.com/
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Re: Asterisk Good News

vtiger@camden.net
In reply to this post by Prasad-14
this might fix all issues, we have noticed a collation of problems to the endless loop of inbound notification
 
but we have no way of knowing the root cause of this endless loop, all we can see is data streams on the ports your webapp.sh and agi.sh + the 2 streams on cli and AMI, and that only if we have stream open in terminal we dont listen all the time and log, takes far to much resource
 
what are we trying to accomplish in this meeting, we have a couple test servers are you wanting access? live testing what? also have a few guys that have more knowledge in certain areas so I might need to check there schedules
 
 
 
 
 
 
 
 

From: "Prasad" <[hidden email]>
Sent: Wednesday, August 27, 2014 11:32 AM
To: "[hidden email]" <[hidden email]>
Cc: "[hidden email]" <[hidden email]>
Subject: Re: [Vtigercrm-developers] Asterisk Good News
 
I noted a scenario that can get the popup stuck on UI due to unexpected / no-response - 
hopefully when we meet this can be reviewed: http://trac.vtiger.com/cgi-bin/trac.cgi/ticket/8195
 
Regards,
Prasad
 
Connect with us on: Twitter I Facebook I Blog I Wiki Forums I Website
 
On Wed, Aug 27, 2014 at 8:10 PM, Akshath T.A <[hidden email]> wrote:
Are you free for meeting now? I will send you the meeting link.
 
??
 
 
On Wed, Aug 27, 2014 at 7:39 PM, [hidden email] <[hidden email]> wrote:
yes I am aware of what it was saying... but 101 was most certainly a valid extension, at the same time this was going on we could send our own origination command to AMI for 101 to call out and it worked just fine, that error is not produced by asterisk at all, its not a valid asterisk error return, you are mis-interrupting some other error and reporting it back as a asterisk error 
 
I think it was some error inside your java, at the time this was happening we only saw data stream on webapp.sh on the click to call attempts, we where also looking at data streams on your agi.sh saw nothing, the cli with agi and verbose 10 notifications enabled saw nothing, and we also looking at the AMI stream, nothing there as well, so we dont even think your connector was even talking to asterisk at all
 
we fixed the issue last night, stopped the webapp.sh and agi.sh files and prevented them from start @ boot, rebooted, cleared settings in vtiger PBX manager or rather put in a bunch of junk, saved, rebooted, put settings back into PBX manager, rebooted then turned the .sh files back on, not sure what steps fixed but simple stop and start of .sh files and/or system reboots did not fix
 
 
we still have issue on on test server where it seems vtiger altered something in asterisk where all the extensions we had in vtiger no longer will ring though a call queue, we have no data on the event we where not actively monitoring data streams, and your asterisk connector has never written 1 log to its own log directory, everything was working Friday, and Monday we checked on it again and found that vtiger was stuck in endless loop of inbound notification and those extensions would no longer sent a sip invite though call queue, even after they where removed from vtiger and a reboot, and the asterisk connector was completely shut down, keep in mind these are test servers dedicated to getting vtiger to work, they are lamp + asterisk + freepbx + vtiger only we have not moved into phase of testing with other packages so the problems we are having can only be vtiger, this is what concerns us about closed source java+agi, beyond we can not see code to try to help you we now have situation where it seems your connector and AGI have altered asterisk, we dont know whats it doing.
 
the whole reason we advocate for just use the AMI is its simple and works and your not really doing anything that requires the interactivity that AGI allows, AMI can be used to originate call as well as be monitored for inbound, this is how tapi drivers work that plug into on site CRMs like ACT, SAGE CRM, Lexis Nexus, ect, its also how sugar / sweet crm are handling it
 
I think the tiring the call records to data in vtiger is a great idea, but why are you generating your own CDR log's via AGI, and this is where I think you guys are hung up on using AGI, but asterisk does have it own MySQL tables and CDR reccords, why not during the isntall of vtiger ask for the asterisk database name and credentials and query them for the info, and vtiger should not be forcing all calls to record, asterisk gives us ways to record and not record calls, if you want to link to recordings fine but again you should be looking to see if a call was recorded, beyond a performance issue and potential duplication of whats already being done in the Asterisk, this ventures into a legal issue, a lot of places all parties must be notified and consent to call recording, with vtiger it forces the recording
 
the general rule of thumb here is if you need user input like your making a IVR and logging what was entered then you need AGI, if your just originating calls and monitoring for inbound use AMI, in either case SQL should be used to get call record data
 
AGI has it problems that you are unnecessarily dealing with, they are common for AGI, what if my agi script finishes before the call ends, what if the call ends before my agi script finishes and a whole host of other issues that you simply dont need to deal with for what your trying to do   
 
 
 
 
 
 
 
 
 
 

From: "Akshath T.A" <[hidden email]>
Sent: Wednesday, August 27, 2014 3:19 AM
To: [hidden email]
Subject: Re: [Vtigercrm-developers] Asterisk Good News
 
Hi,
 
From the logs, it clearly states that the extension - 101 was not found by your Asterisk server when AMI originate request sent from Vtiger Connector.
 
In order to nail down the issue, can we have a online meeting today or tomorrow so that it would help us know what really is causing the problem. 
 
I will be available from 9:00 AM to 8:00 PM IST at my office. Please tell us your convenient time within this so that I can send you meeting link.
 
Thank you.
 
 
regards,
Akshath
  Vtiger

_______________________________________________
http://www.vtiger.com/

_______________________________________________
http://www.vtiger.com/
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Re: Asterisk Good News

vtiger@camden.net
In reply to this post by Alan Bell-6
self hosted opensips would be our preference and here is why
 
currently we have 2 internet connections to deal with, our connection to the ILECS that we pay top dollar for the QoS and SLA on SIP/PSTN traffic, then the internet connection to our EU, that EU connection is 99% of our support issues, though we have ways of dealing with that to some extent in EU;s QoS with use of ingress and egress QoS capable routers
 
a cloud based sip proxy would add a 3rd connection that we have no SLA and no real way of controlling QoS, no matter what it would add latency and jitter 
 
so for us its pretty important to keep our switches, PBXs and any proxy's all in the same network we control, this is a quality issue more than a privacy issue, though privacy is a legitimate concern
 
 
 
 
 

From: "Alan Bell" <[hidden email]>
Sent: Wednesday, August 27, 2014 12:02 PM
To: [hidden email]
Subject: Re: [Vtigercrm-developers] Asterisk Good News
 
On 27/08/14 15:43, [hidden email] wrote:
very interesting using a SIP proxy ;-), would love to check it out when you get it finished
 
how will it handle inbound notification? a in browser pop like vtiger does now?
yeah, but probably based on web sockets rather than polling, if I can do that cleanly in a module without ridiculous dependencies.
 
when you say marketplace I assume extension store?
yup, it would be an open source solution, but packed up in the extension store for easy installation, possibly with a price tag on it a bit like the geotools module.
will this support complete 3rd party apps like the sip proxy server your building, from everything I read it seemed more geared to modules that install directly into vtiger
well, yeah, that is an interesting question, and one reason why I parked the project for a bit. It might be that I run an opensips server as a service in the cloud for users of the module, and give instructions on how to run your own opensips proxy for people who don't want to have their telephone metadata going outside the organisation (other than to the NSA of course). One of the things I would have to figure out for a cloud based sip proxy is getting it to work and be maintainable, but store no logs, I don't want that data!
 
 
if its going to support 3rd party applications we are working on a windows system trey call pop for vtiger so your browser can be minimized and you are still altered to the call with links to relevant records,  in alpha testing now but we would be interested in this extension store    
 
 
 
 

From: "Alan Bell" [hidden email]
Sent: Wednesday, August 27, 2014 10:25 AM
To: [hidden email]
Subject: Re: [Vtigercrm-developers] Asterisk Good News
 
I did write most of a SIP module, I am planning to finish it off and perhaps put it in the marketplace once that is all up and running, it certainly doesn't require multiple extensions to do click to call and inbound, it uses a SIP proxy, basically a benevolent man in the middle attack on the sip signalling :) this proxy can see the calls in both directions being set up, the handsets think that the proxy is the PBX, the PBX thinks the proxy is the handsets. The proxy could be running on the same machine as the PBX, or the same machine as vtiger, or somewhere else altogether.

Alan.

On 27/08/14 15:16, [hidden email] wrote:
SIP would not be the best idea, yes its even easier to deal with than AMI but asterisk can only handle 1 sip registration per extension so this will lead to the need to have 2 extensions, one for your phone and one for vtiger you would then need to daisy chain them in follow me
 
same reason SIPTapi from IPcom never took off, yes you can get the free version and click to call outbound works with only one extension but if you want inbound you have to pay and will find out after you need 2 extensions  
 
 
 

 
 
 
_______________________________________________
http://www.vtiger.com/
 
--
Libertus Solutions
http://libertus.co.uk
 
 
 
_______________________________________________
http://www.vtiger.com/
 
--
Libertus Solutions
http://libertus.co.uk

_______________________________________________
http://www.vtiger.com/
12