[Vtigercrm-developers] Vtiger Astersik Connector

[Alex Martin]

You need to use the exact name of your outgoing trunk, who are you using as a provider? Make a test call from one of the phones and look at the same info in the Asterisk CLI, it will show you your outgoing trunk name to use.



-----Original Message-----
From: vtigercrm-developers-bounces at lists.vtigercrm.com [mailto:vtigercrm-developers-bounces at lists.vtigercrm.com] On Behalf Of mahoutchian
Sent: Sunday, October 18, 2015 1:21 PM
To: vtigercrm-developers at lists.vtigercrm.com
Subject: Re: [Vtigercrm-developers] Vtiger Astersik Connector

Hi Dear Vladimir Karažija,
Tanx Tanx Tanx Tanx Tanx
My System is :
PBX : Elastix x86 32bit ( yum update : asterisk 11.17 and freepbx 2.11 ) Vtiger : Centos 7 x 64 , Vtiger 6.4 Asterisk Connector : 1.4 incoming call in vtiger : ok outgoing click to call for local extentions : ok

now i changed systen to this :
VTiger :

PBXManager

Asterisk Server Details
*Vtiger Asterisk App URL	http://192.168.33.10:8380
*Outbound Context	vtiger-outbound
*Outbound Trunk	DAHDI Channel g0
*Vtiger Secret Key	2463765574556bc57b74661

My Extentions_custom.conf ( part o extentions.conf ) is :

; This file contains the contexts the agents login for the module call center.
; and contains the context conferences for module conferences of elastix 1.0.
 
[from-internal-custom]				 
include => agentlogin
include => conferences
include => calendar-event
include => weather-wakeup

;include => vtiger-outbound ; ????????? i don't know ? need to with o without comment !!!!

exten => h,1,Hangup()


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


[from-pstn-custom]
exten => _X.,1,Agi(agi://0.0.0.0/incoming.agi)

but when clik on a number in vtiger , my phone is ringing with caller id , after pickup, phone say busy tone !!!!

My Asterisk LOG After Click to dial 

       > Channel SIP/2186-00000000 was answered
    -- Executing [2186 at vtiger-outbound:1] AGI("SIP/2186-00000000",
"agi://0.0.0.0/incoming.agi") in new stack
       > 0x9d2ecc8 -- Probation passed - setting RTP source address to
192.168.39.1:8000
    -- AGI Script Executing Application: (Dial) Options:
(SIP/09123081938 at DAHDI Channel g0, 60)
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
[2015-10-18 20:53:35] ERROR[4951][C-00000000]: netsock2.c:271
ast_sockaddr_resolve: getaddrinfo("DAHDI Channel g0", "(null)", ...): Name or service not known
[2015-10-18 20:53:35] WARNING[4951][C-00000000]: chan_sip.c:6058
create_addr: No such host: DAHDI Channel g0
[2015-10-18 20:53:35] WARNING[4951][C-00000000]: app_dial.c:2437
dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- <SIP/2186-00000000>AGI Script agi://0.0.0.0/incoming.agi completed, returning 0
    -- Auto fallthrough, channel 'SIP/2186-00000000' status is 'CHANUNAVAIL'
       > Saved useragent "Rainbow2 V1.5.0" for peer 2114 BranchVoip*CLI> 

i think problem is :

AGI Script Executing Application: (Dial) Options: (SIP/09123081938 at DAHDI Channel g0, 60)

 ERROR[4951][C-00000000]: netsock2.c:271 ast_sockaddr_resolve:
getaddrinfo("DAHDI Channel g0", "(null)", ...): Name or service not known





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