Voice not working

I have a fun little situation that I’m sure someone, somewhere has faced in the past. I have an asterisk server set up with SIP extensions only. When I attempt to dial another sip phone directly, it rings and connects just fine, but neither of us can hear the other person speak. When we both call in to the same MeetMe conference, we can hear each other just fine.

Here is the SIP call that I have used in my dialplan:
exten => _3XXX,1,Dial(SIP/${EXTEN},10,m)
exten => _3XXX,2,Macro(voicemail,${EXTEN})
exten => _3XXX,102,Macro(voicemail,${EXTEN})
exten => _3XXX,103,Hangup()

As I stated, it calls just fine and both people can see the status of the call. But when it connects, there is no sound. I must also mention that with the conference, I DO have music on hold working for that one as well, so it’s not like all audio is being suppressed.

Has anyone seen this before and if so, do you remember how to fix it?

Thanks in advance.

canreinvite=no to sip.conf?

Thank you so much! I can’t believe I forgot that line. I’ve set these up a hundred times, but somehow managed to forget one simple line that makes or breaks the system.
After putting that back in, it works like a charm.


same for me. But I dont hear the music on hold also:
– Executing Answer(“SIP/32-00698890”, “”) in new stack
– Executing MusicOnHold(“SIP/32-00698890”, “”) in new stack
– Started music on hold, class ‘default’, on channel 'SIP/32-00698890’
Everything is silence, except “The person you are calling is unavailable …”, but maybe this voice comes directly from my x-lite and not from asterisk.

canreinvite=no is set in sip.conf for [32]

Asterisk 1.2.11 built by root @ webserv01 on a x86_64 running Linux on 2006-08-29 12:23:34 UTC
Linux webserv01 2.6.8-24-smp #1 SMP Wed Oct 6 09:16:23 UTC 2004 x86_64 x86_64 x86_64 GNU/Linux

write a line [color=blue]musiconhold=default [/color]in your zapata.conf

it worked for me & few otherz, so i hope it may work for you tOo