Hi,<br><br>I have trouble getting MOH to work after an upgrade from asterisk 1.4 to 1.6.1.4. The call goes on hold, MOH is started, and then stops right away.<br><br>Here are the files both of type .raw:<br><br>Tsunami*CLI> moh show files<br>
Class: default<br> File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-2<br> File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-1<br><br>These files were generated by SoX:<br>Channels : 1<br>
Sample Rate : 8000<br>Precision : 16-bit<br>Sample Encoding: 16-bit Signed Integer PCM<br>Endian Type : little<br>Reverse Nibbles: no<br>Reverse Bits : no<br>Comment : 'Processed by SoX'<br><br>
This prints in the asterisk console when you attempt to put someone in hold:<br><br> -- Started music on hold, class 'default', on SIP/link2voip-sw1-02477668<br> -- Stopped music on hold on SIP/link2voip-sw1-02477668<br>
<br>No errors are printed, however the other side just hears silence.<br><br>Here is the full debug output (asterisk -rvvvvv):<br><br> == Using SIP RTP CoS mark 5<br> -- Executing [xxxxxxx@phones:1] Goto("SIP/ATA-xxxxxxxxxx-L1-024b6d88", "1xxxxxxxxxx,1") in new stack<br>
-- Goto (phones,1xxxxxxxxxx,1)<br> -- Executing [1xxxxxxxxxx@phones:1] MSet("SIP/ATA-xxxxxxxxxx-L1-024b6d88", "oldcidnum=0") in new stack<br> -- Executing [1xxxxxxxxxx@phones:2] MSet("SIP/ATA-xxxxxxxxxx-L1-024b6d88", "CALLERID(name)=""") in new stack<br>
-- Executing [1xxxxxxxxxx@phones:3] MSet("SIP/ATA-xxxxxxxxxx-L1-024b6d88", "CALLERID(num)=xxxxxxxxxx") in new stack<br> -- Executing [1xxxxxxxxxx@phones:4] Monitor("SIP/ATA-xxxxxxxxxx-L1-024b6d88", "wav,/tmp/out 0 2009-09-17 03h 04m 51s CST xxxxxxxxxx,m") in new stack<br>
-- Executing [1xxxxxxxxxx@phones:5] Gosub("SIP/ATA-xxxxxxxxxx-L1-024b6d88", "ExternalDial,s,1(1xxxxxxxxxx)") in new stack<br> -- Executing [s@ExternalDial:1] MSet("SIP/ATA-xxxxxxxxxx-L1-024b6d88", "LOCAL(num)=1xxxxxxxxxx") in new stack<br>
-- Executing [s@ExternalDial:2] MSet("SIP/ATA-xxxxxxxxxx-L1-024b6d88", "~~EXTEN~~=s") in new stack<br> -- Executing [s@ExternalDial:3] Dial("SIP/ATA-xxxxxxxxxx-L1-024b6d88", "SIP/1xxxxxxxxxx@link2voip-sw1,120") in new stack<br>
== Using SIP RTP CoS mark 5<br> -- Called 1xxxxxxxxxx@link2voip-sw1<br> -- SIP/link2voip-sw1-02477668 is making progress passing it to SIP/ATA-xxxxxxxxxx-L1-024b6d88<br> -- SIP/link2voip-sw1-02477668 answered SIP/ATA-xxxxxxxxxx-L1-024b6d88<br>
-- Started music on hold, class 'default', on SIP/link2voip-sw1-02477668<br> -- Stopped music on hold on SIP/link2voip-sw1-02477668<br> > doing dnsmgr_lookup for '<a href="http://sip.ca2.link2voip.com">sip.ca2.link2voip.com</a>'<br>
> doing dnsmgr_lookup for '<a href="http://sip.ca1.link2voip.com">sip.ca1.link2voip.com</a>'<br> == Spawn extension (ExternalDial, s, 3) exited non-zero on 'SIP/ATA-xxxxxxxxxx-L1-024b6d88'<br><br>
Any thoughts or ideas? If there were an error I could work on solving that, but there is none... Thanks.<br><br>