asterisk users - Dec 2013 - 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 at macro-dialexternal:57] GotoIf("IAX2/S-14468",
"1?dialnormal") in new stack
    -- Goto (macro-dialexternal,s,60)
    -- Executing [s at 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'
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20131212/aa7cdb31/attachment.html>

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-bounces at lists.digium.com [asterisk-users-bounces at
lists.digium.com] On Behalf Of Michelle Dupuis [mdupuis at 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 at macro-dialexternal:57] GotoIf("IAX2/S-14468",
"1?dialnormal") in new stack
    -- Goto (macro-dialexternal,s,60)
    -- Executing [s at 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'
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20131213/8b1ed9be/attachment.html>