No Audio,Either Direction-Answered by External Queue Member

I have Asterisk 1.6.2 installed on Ubuntu 10.4, I have queues working fine with 3 IP phones on the same LAN as the server. PSTN connectivity is via 3 SIP trunks from a local provider. The provider is sending me 20 DID phone numbers (they are delivering 4-digits).
My Asterisk box sits behind a Cisco router, but I do not believe I have any NAT issues, for the following reason:
I can place and receive normal calls (not using the queues) to and from the PSTN from my IP phones with no issue.
I can place a call from my cell phone to the “main” DID by dialing the full 10-digit phone number, which sends the call into the queue. The queue is set to “Ring All” and all three phones ring (as long as they are logged into the queue). I can answer the inbound call and all works fine.
I have also successfully used Softphones “off net” as queue members and they work fine.

Now I add a 4th queue member which is an analog phone line which is completely unrelated to Asterisk. Here are the lines from my extentions.config where I set up the add and remove of queue members:
exten => 6000,1,AddQueueMember(practice1q,SIP/${CALLERID(num)})
exten => 6100,1,RemoveQueueMember(practice1q,SIP/${CALLERID(num)})
exten => 6200,1,AddQueueMember(practice1q,SIP/trunk_1/91406327XXXX)
exten => 6300,1,RemoveQueueMember(practice1q,SIP/trunk_1/91406327XXXX)

The queue member added by dialing 6200 and removed by dialing 6300, is the external queue member. When I dial “6200” Asterisk shows the member being added:
[Feb 13 16:16:53] NOTICE[11472]: app_queue.c:4930 aqm_exec: Added interface ‘SIP/trunk_1/91406327XXXX’ to queue 'practice1q’
When I place the call to the main DID, the 3 phone on the LAN AND the analog phone (406-327-XXXX) all ring as expected. I can answer the analog phone and all the other quit ringing but I have no audio in either direction.
Is anyone aware of specific setting that might cause this behavior? Perhaps some sort of conflict with inbound calls traversing two of my three SIP trunks (which by the way are all delivered as a single IP and port)?
Thanks in advance for any comments

OK I think I have refined my issue. I now believe that what I am seeing is related to PSTN call legs being “joined” by Asterisk but when Asterisk drops out of the loop the Audio stream “gets lost”.

I can duplicate similar behavior by simply placing a call from the PSTN to an Asterisk extension via an inbound SIP trunk. If I “Conference” a different phone out on the PSTN, all three phones work as expected (full audio both directions between all phones). However, If I place an inbound call to an Asterisk extension and then subsequently forward that call to a different PSTN phone, the phone rings, I can answer, BUT there is no audio in either direction.

Any idea where I sould begin to look?

I have captured a SIP debug of just such a session, but I do not know of an easy way to remove all of the phone numbers and IP references that I would be uncomfortable posting in a public forum.


canreinvite, or its replacement.

A conference inhibits external bridging.

Thanks for the tip David. I made the changes “can reinvite = no” to “can reinvite = yes” in both my sip.config and my users.config (just for good measure). I then did a reload at the CLI .

Now when I place a call from the PSTN into an Asterisk Extension, it rings, I answer, and then the call is immediately dropped?

I am seeing a “481 call leg/transaction does not exist”

Any idea what I’ve done wrong?


I was assuming you might have to change it to “no”, from “yes”!

OK I’ve found some more information but I am stuck on the terminology or how to control it!

I read that somewhere in Asterisk there is the ability to do the following:

Caller “A” places a call to one of my Asterisk Extensions.- let’s call it caller "B"
Caller “B” now transfers the call to caller “C” which is outside my Asterisk System
Asterisk then attempts to connect caller "A’ and caller “C” directly by setting up the audio stream directly between endpoint “A” and enpoint “C”.

I do not remember what the process is called or how to control it.

My SIP carrier does not support it.

Any ideas??



That is a reinvite (or something like directmedia, in 1.8).


You are the man! Thankyou very much for the assistance.

“directmedia=no” in my sip.conf resolved the issue.

Thanks once more,