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this is related to your carrier's SIP messages as they are sending a sendonly attribute instead of sendrecv (taking a wild guess here) your asterisk will act as if the call was placed on hold thus the MOH butts in. <br>an sip debug log for a similar call will be more helpful?<br><br>Tarek Sawah<br><br>Information Technology Adviser<br><br>Integrated Digital Systems<br><br>CCNP, MCSE, RHCE, TELECOM<br><br>USA: +1 386 492 9993<br><br><br><br><div>> From: alexrecarey@gmail.com<br>> Date: Wed, 28 Sep 2011 03:44:35 +0200<br>> To: asterisk-users@lists.digium.com<br>> Subject: [asterisk-users] Receiving musinc on hold instead of ring<br>> <br>> Hi all and thanks for reading.<br>> <br>> I am having a very strange issue. When dialing out with a certain<br>> carrier, asterisk 1.6.20 will play music on hold instead of a ring<br>> tone, although this behaviour is NOT what I want.<br>> <br>> Example dialplan execution:<br>> <br>> -- Executing [0021266xxx@test:13] Progress("SIP/100-00001e04", "") in new stack<br>> -- Executing [0021266xxx@test:14]<br>> Dial("SIP/100-00001e04","SIP/21266xxx@x.x.x.x") in new stack<br>> -- Called 21266xxx@x.x.x.x<br>> -- Call on SIP/x.x.x.x-00001e05 placed on hold<br>> -- Started music on hold, class 'default', on SIP/100-00001e04<br>> -- SIP/x.x.x.x-00001e05 is making progress passing it to SIP/100-00001e04<br>> <br>> Now, a SIP packet capture shows no trace of the call being put on hold!<br>> <br>> Sample wireshark capture for the same call:<br>> <br>> x.x.x.x -> y.y.y.y SIP/SDP Request: INVITE sip:21266xxx@x.x.x.x, with<br>> session description<br>> y.y.y.y -> x.x.x.x SIP Status: 100 Giving a try<br>> y.y.y.y -> x.x.x.x SIP/SDP Status: 180 Ringing, with session description<br>> <br>> And I get the music on hold instead of the ringtone. I have tried<br>> placing Progress() in front of Dial() but to no avail. I do not want<br>> to use the "r" option in Dial() because then I lose the destination<br>> ringtone in early media which is important to my customers.<br>> <br>> Anybody had a similar issue? Any idea of what parameters I can try to<br>> tweak, as I am stumped...<br>> <br>> Thanks!<br>> <br>> Alex<br>> <br>> --<br>> _____________________________________________________________________<br>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>> http://www.asterisk.org/hello<br>> <br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-users<br></div> </div></body>
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