Some more details...I noticed that the call is bridged, and audio goes one way.
However, the dial command still times out after 35 seconds (approx), and
exists non-zero.
While the channels are up, I did an core show channel xxx and found Blocking in:
ast_waitfor_nandfds
Is this a bug? Or something I can fix through config?
________________________________
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis
[mdup...@ocg.ca]
Sent: Thursday, December 12, 2013 5:08 PM
To: Asterisk Users List
Subject: [asterisk-users] IAX2 bridge failing
I am trying to connect an IAX ATA to an Asterisk 1.4.21.2 system. The Asterisk
system has been stable for years, and has no trouble bridge SIP phone sets to
IAX trunks.
When I initiate a call from the IAX ATA, something goes wrong. One rare
occasion it works fine, but usually there is no audio passed. I have a snippet
of the console below. Notice no bridging message...not sure if that's a clue?
The dialplan seems to execute properly, and I can watch the destination system
which answers the call and starts playing media (monkeys) which I don't hear.
Any ideas on what is going on? Since this is IAX in and IAX out, NAT should
not be an issue (even through there is NAT on both sides). Since media moves
on the same UDP port as call setup, also proves should not be a network problem
(I think)
Can someone point me to a solution?
Thanks!
(IP's and ISP and phone number disguised)
- Executing [s@macro-dialexternal:57] GotoIf("IAX2/S-14468", "1?dialnormal") in
new stack
-- Goto (macro-dialexternal,s,60)
-- Executing [s@macro-dialexternal:60] Dial("IAX2/S-14468",
"IAX2/ISP123/1234567890|60|W") in new stack
-- Called ISP123/1234567890
-- Call accepted by 201.191.37.138 (format ulaw)
-- Format for call is ulaw
-- IAX2/ISP123-2261 answered IAX2/S-14468
-- Channel 'IAX2/S-14468' ready to transfer
-- Channel 'IAX2/ISP123-2261' ready to transfer
-- Hungup 'IAX2/ISP123-2261'
--
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