I recently upgraded to asterisk 1.4.2 because of the following problem, but was forced to go back to asterisk 1.2.13 because MOH wasn’t working when used in the dial command.
The problem on asterisk 1.2.13 is this:
I have dial() in extensions.conf to dial out to my home landline and mobile from my server:
exten => _X.,4,Dial(SIP/enginout/0882425603&SIP/enginout/0423289232,45,m(default))
When I dial the number for the incoming voip trunk from an external telephone it answers fine then initiates the command above, both my landline and my mobile ring, but when I answer either of them both the calling party and myself are disconnected and the following error is displayed:
– Attempting native bridge of SIP/184.108.40.206-08194000 and SIP/enginout-086d6000
Apr 18 15:54:21 NOTICE: chan_sip.c:9810 handle_response_invite: Failed to authenticate on INVITE to ‘sip:email@example.com;tag=as6557605d’
This problem isn’t present in 1.4.2 but because moh doesn’t work in the dial command it’s useless.
The configurations for sip.conf, musiconhold.conf and extensions.conf are the same for both versions. One has one problem the other has another problem so I’ve posted both issues on the forum in seperate posts and hope I get a solution for at least one of them.
Tried googling both the issues extensively and got nothing for either.