Double dialing fails when one of the SIP peers doesn't exist

Asterisk Asterisk Support
1

Hi guys,

I am currently having a very weird issue. I am managing about 10 IP PBX around the world for my company and I use everywhere a macro which allows me to automatically dial IP phone & Soft Phone at the same time when someone tries to reach a user.

Here is an example of the configuration :

sip.conf :

[1111]
account for ip phone for user having number 1111

[softphone-1111]
account for softphone for user having number 1111

The dialplan just does this : exten => 1111,1,Dial(SIP/1111&SIP/softphone-1111|20|rt).

It is working everywhere but not on one of the asterisk. If sip.conf contains the SIP account & the softphone SIP account, it work even if one of them is not registered. If only the SIP account is setup, the IP phone will ring, but when you answer, nothing happens. This only happens when there is no softphone sip account associated to the “extension”.

Note that on 9 of the Asterisk, if the softphone is not setup in sip.conf, it still goes through correctly. all sip.conf / extension.conf general parts are the same, and all softphone accounts & sip accounts are also exactly the same.

The call looks going through, but I just see “Called 1111” and no Ringing 1111 (even if IP phone rings). When hanging up the call, it tells me it could not find softphone-1111 as error etc.

Any idea on why this could happen ?

Thanks in advance !

2

It probably may help some of you, or maybe find out the real solution but I found a work around :

for all extension, I created a void softphone account with only :
[softphone-1111]
type=friend

[softphone-1112]
type=friend

This done, the automatic double dial works :

-- Executing [7103@international:2] Goto("SIP/7050-b7cd46c0", "international-cli|7103|1") in new stack -- Goto (international-cli,7103,1) -- Executing [7103@international-cli:1] Dial("SIP/7050-b7cd46c0", "SIP/7103&SIP/softphone-7103|20|rt") in new stack -- Called 7103 [Nov 25 09:45:17] WARNING[7013]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) -- SIP/7103-09c9d620 is ringing

I can now see the Ringing state and the voice goes through correctly.