Joseph Rothstein
2005-Oct-28 02:56 UTC
[Asterisk-Users] IAX voice problem, no voice at all
I have a strange problem with IAX. We have an IAX connection between two boxes which works fine. We can call from the two boxes, in and out using SIP phones, hard and soft without issue. We are testing some IAX softphones, and have come across a problem with the voice. Calls on the same box are fine, anything I call on the box where the IAX clients are registered is perfect. The problems occur when we try and call any extensions on the other box. Signalling seems to work OK, the phones ring, but there is no voice at all. Trace from Asterisk below (box where the IAX client is registered): -- Accepting AUTHENTICATED call from 195.27.242.103, requested format 1024, actual format = 4 -- Executing Dial("IAX2/test2@test2/4", "IAX2/RemoteServ/212||r") in new stack -- Called RemoteServ/212 -- Call accepted by 172.16.10.2 (format ulaw) -- Format for call is ulaw -- IAX2/RemoteServ/7 is ringing -- IAX2/RemoteServ/7 is ringing -- IAX2/RemoteServ/7 answered IAX2/test2@test2/4 -- Attempting native bridge of IAX2/test2@test2/4 and IAX2/RemoteServ/7 -- Channel 'IAX2/RemoteServ/7' ready to transfer -- Channel 'IAX2/test2@test2/4' ready to transfer -- Releasing IAX2/test2@test2/4 and IAX2/RemoteServ/7 -- Hungup 'IAX2/RemoteServ/7' == Spawn extension (iaxtest, 7212, 1) exited non-zero on 'IAX2/test2@test2/4' -- Hungup 'IAX2/test2@test2/4' This is what we see form the other side: -- Accepting unauthenticated call from 172.16.10.5, requested format = 4, actual format = 4 -- Executing Dial("IAX2/ClientServ@ClientServ/16386", "SIP/phone12|20|tr") in new stack -- Called phone12 -- SIP/phone12-fe21 is ringing -- SIP/phone12-fe21 answered IAX2/ClientServ@ClientServ/16386 == Spawn extension (default, 212, 1) exited non-zero on 'IAX2/ClientServ@ClientServ/16386' -- Hungup 'IAX2/ClientServ@ClientServ/16386' The only thing for me that looks strange is the numbers here. Why does on machine have only single digit numbers? Everything seems to progress as it should, there is just no voice. The IAX firmware is the same on both sides: asterisk_test*CLI> iax2 show firmware Device Version Size iaxy 23 39360 Both boxes are running 1.0.9. As I mentioned, the AIX connection between the two boxes runs fine for all the SIP traffic we pass over it. Any ideas with this would really help. Regards, Joe Joseph Rothstein, CCIE Senior Network Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051028/c231cf69/attachment.htm