Re: [asterisk-users] Receiving musinc on hold instead of ring
Hi Tarek, Yes, after running some more detailed packet captures, it seems that the SDP sent has the sendonly media attribute. I do not know if it is the Sonus switch, but the problem is identical to yours. Unfortunately setting canreinvite=yes for that peer does not solve the problem. I am guessing this is because the other leg of the call has canreinvite=no. This is necessary for correct billing. Should I submit it as an asterisk bug? Is there something else I can try to fix this interconnection? Thanks for your help! Alex On Wed, Sep 28, 2011 at 7:34 PM, Tarek Sawah wrote: > i have faced this problem with one of the major VoIP whole providers in > India .. they have a new platform with Sonus switches.. which does not > support sendrecv media attribute .. however a work around that may work for > you .. is enabling re-invite on their peer. > let me know if this works out for you. > > > Tarek Sawah > > Information Technology Adviser > > Integrated Digital Systems > > CCNP, MCSE, RHCE, TELECOM > > USA: +1 386 492 9993 > > > >> From: alexreca...@gmail.com >> Date: Wed, 28 Sep 2011 18:59:39 +0200 >> To: asterisk-users@lists.digium.com >> Subject: Re: [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? >> >> Thanks for the answer Tarek! I will try to obtain a full SIP trace >> tonight. If the problem is indeed that the carrier is sending the >> sendonly attribute in the SDP instead of sendrecv, what can I do? Is >> there anything I can configure on my side? >> >> Thanks again, >> >> 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 > > -- > _ > -- 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 > -- _ -- 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
Re: [asterisk-users] Receiving musinc on hold instead of ring
i have faced this problem with one of the major VoIP whole providers in India .. they have a new platform with Sonus switches.. which does not support sendrecv media attribute .. however a work around that may work for you .. is enabling re-invite on their peer. let me know if this works out for you. Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 > From: alexreca...@gmail.com > Date: Wed, 28 Sep 2011 18:59:39 +0200 > To: asterisk-users@lists.digium.com > Subject: Re: [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? > > Thanks for the answer Tarek! I will try to obtain a full SIP trace > tonight. If the problem is indeed that the carrier is sending the > sendonly attribute in the SDP instead of sendrecv, what can I do? Is > there anything I can configure on my side? > > Thanks again, > > 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 -- _ -- 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
Re: [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? Thanks for the answer Tarek! I will try to obtain a full SIP trace tonight. If the problem is indeed that the carrier is sending the sendonly attribute in the SDP instead of sendrecv, what can I do? Is there anything I can configure on my side? Thanks again, 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
Re: [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: alexreca...@gmail.com
> Date: Wed, 28 Sep 2011 03:44:35 +0200
> To: asterisk-users@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@test:13] Progress("SIP/100-1e04", "") in new
> stack
> -- Executing [0021266xxx@test:14]
> Dial("SIP/100-1e04","SIP/21266xxx@x.x.x.x") in new stack
> -- Called 21266xxx@x.x.x.x
> -- Call on SIP/x.x.x.x-1e05 placed on hold
> -- Started music on hold, class 'default', on SIP/100-1e04
> -- SIP/x.x.x.x-1e05 is making progress passing it to SIP/100-1e04
>
> 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@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
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Re: [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 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@test:13] Progress("SIP/100-1e04", "") in new
> stack
> -- Executing [0021266xxx@test:14]
> Dial("SIP/100-1e04","SIP/21266xxx@x.x.x.x") in new stack
> -- Called 21266xxx@x.x.x.x
> -- Call on SIP/x.x.x.x-1e05 placed on hold
> -- Started music on hold, class 'default', on SIP/100-1e04
> -- SIP/x.x.x.x-1e05 is making progress passing it to SIP/100-1e04
>
> 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@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
>
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_
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asterisk-users mailing list
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[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@test:13] Progress("SIP/100-1e04", "") in new stack
-- Executing [0021266xxx@test:14]
Dial("SIP/100-1e04","SIP/21266xxx@x.x.x.x") in new stack
-- Called 21266xxx@x.x.x.x
-- Call on SIP/x.x.x.x-1e05 placed on hold
-- Started music on hold, class 'default', on SIP/100-1e04
-- SIP/x.x.x.x-1e05 is making progress passing it to SIP/100-1e04
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@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
--
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asterisk-users mailing list
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