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<div dir="ltr"><font color="#000000" size="2" face="Tahoma">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.</font></div>
<div dir="ltr"><font size="2" face="tahoma"></font> </div>
<div dir="ltr"><font size="2" face="tahoma">While the channels are up, I did an core show channel xxx and found Blocking in:</font></div>
<div dir="ltr">ast_waitfor_nandfds</div>
<div dir="ltr"><font face="times new roman"></font> </div>
<div dir="ltr"><font face="times new roman">Is this a bug? Or something I can fix through config?</font></div>
<div dir="ltr"><font face="times new roman"></font> </div>
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<font size="2" face="Tahoma"><b>From:</b> asterisk-users-bounces@lists.digium.com [asterisk-users-bounces@lists.digium.com] On Behalf Of Michelle Dupuis [mdupuis@ocg.ca]<br>
<b>Sent:</b> Thursday, December 12, 2013 5:08 PM<br>
<b>To:</b> Asterisk Users List<br>
<b>Subject:</b> [asterisk-users] IAX2 bridge failing<br>
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<div dir="ltr"><font color="#000000" size="2" face="Tahoma">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.</font></div>
<div dir="ltr"><font size="2" face="tahoma"></font> </div>
<div dir="ltr"><font size="2" face="tahoma">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.
</font></div>
<div dir="ltr"><font size="2" face="tahoma"></font> </div>
<div dir="ltr"><font size="2" face="tahoma">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)</font></div>
<div dir="ltr"><font size="2" face="tahoma"></font> </div>
<div dir="ltr"><font size="2" face="tahoma">Can someone point me to a solution? </font>
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<div dir="ltr"><font size="2" face="tahoma"></font> </div>
<div dir="ltr"><font size="2" face="tahoma">Thanks!</font></div>
<div dir="ltr"><font size="2" face="tahoma"></font> </div>
<div dir="ltr"><font size="2" face="tahoma"></font> </div>
<div dir="ltr"><font size="2" face="tahoma">(IP's and ISP and phone number disguised)</font></div>
<div dir="ltr"><font size="2" face="tahoma"></font> </div>
<div dir="ltr">- Executing [s@macro-dialexternal:57] GotoIf("IAX2/S-14468", "1?dialnormal") in new stack<br>
-- Goto (macro-dialexternal,s,60)<br>
-- Executing [s@macro-dialexternal:60] Dial("IAX2/S-14468", "IAX2/ISP123/1234567890|60|W") in new stack<br>
-- Called ISP123/1234567890<br>
-- Call accepted by 201.191.37.138 (format ulaw)<br>
-- Format for call is ulaw<br>
-- IAX2/ISP123-2261 answered IAX2/S-14468<br>
-- Channel 'IAX2/S-14468' ready to transfer<br>
-- Channel 'IAX2/ISP123-2261' ready to transfer<br>
-- Hungup 'IAX2/ISP123-2261'<br>
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