Alejandro Recarey
2011-Sep-28 01:44 UTC
[asterisk-users] Receiving musinc on hold instead of ring
Hi all and thanks for reading.
I am having a very strange issue. When dialing out with a certain
carrier, asterisk 1.6.20 will play music on hold instead of a ring
tone, although this behaviour is NOT what I want.
Example dialplan execution:
-- Executing [0021266xxx at test:13] Progress("SIP/100-00001e04",
"") in new stack
-- Executing [0021266xxx at test:14]
Dial("SIP/100-00001e04","SIP/21266xxx at x.x.x.x") in new
stack
-- Called 21266xxx at x.x.x.x
-- Call on SIP/x.x.x.x-00001e05 placed on hold
-- Started music on hold, class 'default', on SIP/100-00001e04
-- SIP/x.x.x.x-00001e05 is making progress passing it to SIP/100-00001e04
Now, a SIP packet capture shows no trace of the call being put on hold!
Sample wireshark capture for the same call:
x.x.x.x -> y.y.y.y SIP/SDP Request: INVITE sip:21266xxx at x.x.x.x, with
session description
y.y.y.y -> x.x.x.x SIP Status: 100 Giving a try
y.y.y.y -> x.x.x.x SIP/SDP Status: 180 Ringing, with session description
And I get the music on hold instead of the ringtone. I have tried
placing Progress() in front of Dial() but to no avail. I do not want
to use the "r" option in Dial() because then I lose the destination
ringtone in early media which is important to my customers.
Anybody had a similar issue? Any idea of what parameters I can try to
tweak, as I am stumped...
Thanks!
Alex
Sam Govind
2011-Sep-28 05:38 UTC
[asterisk-users] Receiving musinc on hold instead of ring
Very strange indeed! post the dialplan lines as well. Seems like a very normal Dial command execution. Also complete SIP packets for this particular behaviour can show some insider. Which version of Asterisk you are using? On Wed, Sep 28, 2011 at 6:44 AM, Alejandro Recarey <alexrecarey at gmail.com>wrote:> Hi all and thanks for reading. > > I am having a very strange issue. When dialing out with a certain > carrier, asterisk 1.6.20 will play music on hold instead of a ring > tone, although this behaviour is NOT what I want. > > Example dialplan execution: > > -- Executing [0021266xxx at test:13] Progress("SIP/100-00001e04", "") in new > stack > -- Executing [0021266xxx at test:14] > Dial("SIP/100-00001e04","SIP/21266xxx at x.x.x.x") in new stack > -- Called 21266xxx at x.x.x.x > -- Call on SIP/x.x.x.x-00001e05 placed on hold > -- Started music on hold, class 'default', on SIP/100-00001e04 > -- SIP/x.x.x.x-00001e05 is making progress passing it to SIP/100-00001e04 > > Now, a SIP packet capture shows no trace of the call being put on hold! > > Sample wireshark capture for the same call: > > x.x.x.x -> y.y.y.y SIP/SDP Request: INVITE sip:21266xxx at x.x.x.x, with > session description > y.y.y.y -> x.x.x.x SIP Status: 100 Giving a try > y.y.y.y -> x.x.x.x SIP/SDP Status: 180 Ringing, with session description > > And I get the music on hold instead of the ringtone. I have tried > placing Progress() in front of Dial() but to no avail. I do not want > to use the "r" option in Dial() because then I lose the destination > ringtone in early media which is important to my customers. > > Anybody had a similar issue? Any idea of what parameters I can try to > tweak, as I am stumped... > > Thanks! > > Alex > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110928/2e61799a/attachment.htm>
Tarek Sawah
2011-Sep-28 13:39 UTC
[asterisk-users] Receiving musinc on hold instead of ring
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. an sip debug log for a similar call will be more helpful? Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993> From: alexrecarey at gmail.com > Date: Wed, 28 Sep 2011 03:44:35 +0200 > To: asterisk-users at lists.digium.com > Subject: [asterisk-users] Receiving musinc on hold instead of ring > > Hi all and thanks for reading. > > I am having a very strange issue. When dialing out with a certain > carrier, asterisk 1.6.20 will play music on hold instead of a ring > tone, although this behaviour is NOT what I want. > > Example dialplan execution: > > -- Executing [0021266xxx at test:13] Progress("SIP/100-00001e04", "") in new stack > -- Executing [0021266xxx at test:14] > Dial("SIP/100-00001e04","SIP/21266xxx at x.x.x.x") in new stack > -- Called 21266xxx at x.x.x.x > -- Call on SIP/x.x.x.x-00001e05 placed on hold > -- Started music on hold, class 'default', on SIP/100-00001e04 > -- SIP/x.x.x.x-00001e05 is making progress passing it to SIP/100-00001e04 > > Now, a SIP packet capture shows no trace of the call being put on hold! > > Sample wireshark capture for the same call: > > x.x.x.x -> y.y.y.y SIP/SDP Request: INVITE sip:21266xxx at x.x.x.x, with > session description > y.y.y.y -> x.x.x.x SIP Status: 100 Giving a try > y.y.y.y -> x.x.x.x SIP/SDP Status: 180 Ringing, with session description > > And I get the music on hold instead of the ringtone. I have tried > placing Progress() in front of Dial() but to no avail. I do not want > to use the "r" option in Dial() because then I lose the destination > ringtone in early media which is important to my customers. > > Anybody had a similar issue? Any idea of what parameters I can try to > tweak, as I am stumped... > > Thanks! > > Alex > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110928/1dc681be/attachment.htm>