Hi, I have been using Sangoma A102 card with wanpipe-3.5.28 and FS 1.2.8. At a couple of different customer sites, common issue reported is, when a call is made from external world (mobile/fixed line), although the destination extension rings, no ringback is heard at the caller end. Extension dialplan is to bridge to the desired extension, something like: <action application="bridge" data="user/3001" /> On capturing the isdn traces, I see that packet exchanges involve SETUP, CALL PROCEEDING, and ALERTING (with a progress indicator saying "Call is not end-to-end ISDN - progress information available in-band") After experimenting a bit, what I find is, ringback is heard if I were to modify my dialplan as follows: <action application="ring_ready" /> <action application="set" data="ringback=$${uk-ring}"/> <action application="bridge" data="user/3001" /> This time, ISDN traces have the following: SETUP, CALL PROCEEDING, ALERTING (with a progress indicator saying "Call is not end-to-end ISDN - progress information available in-band"), and PROGRESS (with a progress indicator saying "In-band information or an appropriate pattern is now available") My question is, while working with FreeTDM, do I need to always set ringback explicitly? Could somebody please help me understand this correctly? Thanks!
On Sun, Dec 1, 2013 at 6:29 PM, Siri MM <sirimmfs@gmail.com> wrote:
Hi,
I have been using Sangoma A102 card with wanpipe-3.5.28 and FS 1.2.8.
At a couple of different customer sites, common issue reported is, when a call is made from external world (mobile/fixed line), although the destination extension rings, no ringback is heard at the caller end.
Extension dialplan is to bridge to the desired extension, something like: <action application="bridge" data="user/3001" />
On capturing the isdn traces, I see that packet exchanges involve SETUP, CALL PROCEEDING, and ALERTING (with a progress indicator saying "Call is not end-to-end ISDN - progress information available in-band")
After experimenting a bit, what I find is, ringback is heard if I were to modify my dialplan as follows: <action application="ring_ready" /> <action application="set" data="ringback=$${uk-ring}"/> <action application="bridge" data="user/3001" />
This time, ISDN traces have the following: SETUP, CALL PROCEEDING, ALERTING (with a progress indicator saying "Call is not end-to-end ISDN - progress information available in-band"), and PROGRESS (with a progress indicator saying "In-band information or an appropriate pattern is now available")
My question is, while working with FreeTDM, do I need to always set ringback explicitly? Could somebody please help me understand this correctly?
Not always. It depends very much on your situation. In many cases the telco plays the media regardless of what you send. In your case it seems you want to bridge the call to a given SIP endpoint. My guess is that SIP endpoint is never sending 183 or even if it sends 183 is not sending media. In such cases, if you want to provide media, yes you have to execute ring_ready and let FreeSWITCH play the media (ie, through the ringback variable, using playback, etc) Moy *Moises Silva**Manager, Software Engineering* msilva@sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 <http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email+signatures> Products<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | Solutions<http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | Events<http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | Contact<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | Wiki<http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | Facebook<http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | Twitter<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>`| | YouTube<http://www.youtube.com/sangomatechnologies?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
Hi Moises, Thanks for the reply. I am using a SNOM 320 phone as SIP end point. On being invited, it sends back a SIP/2.0 180 Ringing. How does FreeSWITCH/FreeTDM behave in this case - would it send media as part of the Progress message (in band)? Is there a way for me to check if media is being sent out to the caller? On Mon, Dec 2, 2013 at 5:12 PM, Moises Silva <moises.silva@gmail.com> wrote:
On Sun, Dec 1, 2013 at 6:29 PM, Siri MM <sirimmfs@gmail.com> wrote:
Hi,
I have been using Sangoma A102 card with wanpipe-3.5.28 and FS 1.2.8.
At a couple of different customer sites, common issue reported is, when a call is made from external world (mobile/fixed line), although the destination extension rings, no ringback is heard at the caller end.
Extension dialplan is to bridge to the desired extension, something like: <action application="bridge" data="user/3001" />
On capturing the isdn traces, I see that packet exchanges involve SETUP, CALL PROCEEDING, and ALERTING (with a progress indicator saying "Call is not end-to-end ISDN - progress information available in-band")
After experimenting a bit, what I find is, ringback is heard if I were to modify my dialplan as follows: <action application="ring_ready" /> <action application="set" data="ringback=$${uk-ring}"/> <action application="bridge" data="user/3001" />
This time, ISDN traces have the following: SETUP, CALL PROCEEDING, ALERTING (with a progress indicator saying "Call is not end-to-end ISDN - progress information available in-band"), and PROGRESS (with a progress indicator saying "In-band information or an appropriate pattern is now available")
My question is, while working with FreeTDM, do I need to always set ringback explicitly? Could somebody please help me understand this correctly?
Not always. It depends very much on your situation. In many cases the telco plays the media regardless of what you send. In your case it seems you want to bridge the call to a given SIP endpoint. My guess is that SIP endpoint is never sending 183 or even if it sends 183 is not sending media. In such cases, if you want to provide media, yes you have to execute ring_ready and let FreeSWITCH play the media (ie, through the ringback variable, using playback, etc)
Moy
*Moises Silva **Manager, Software Engineering*
msilva@sangoma.com
Sangoma Technologies
100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada
t. +1 800 388 2475 (N. America)
t. +1 905 474 1990 x128
f. +1 905 474 9223
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On Mon, Dec 2, 2013 at 1:38 AM, Siri MM <sirimmfs@gmail.com> wrote:
Hi Moises,
Thanks for the reply. I am using a SNOM 320 phone as SIP end point. On being invited, it sends back a SIP/2.0 180 Ringing. How does FreeSWITCH/FreeTDM behave in this case - would it send media as part of the Progress message (in band)? Is there a way for me to check if media is being sent out to the caller?
I think strictly speaking a 180 should not trigger any early media (ring tone). The SIP response code for early media is 183 (although I've heard about 180 sometimes containing SDP). If ring_ready solves your problem, what problem are you trying to solve now? I am not sure 180 should be triggering any media bridging code. *Moises Silva**Manager, Software Engineering* msilva@sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 <http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email+signatures> Products<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | Solutions<http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | Events<http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | Contact<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | Wiki<http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | Facebook<http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | Twitter<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>`| | YouTube<http://www.youtube.com/sangomatechnologies?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
Hi Moises, I am not trying to solve any further problems, but am interested in understanding this better. Am I right in saying that SNOM(or any other SIP end point for that matter) should ideally be sending over 183 for the ringback to be triggered from FreeSWITCH/FreeTDM, and if that's not the case, ring_ready should be used to simulate the same? On Mon, Dec 2, 2013 at 5:44 PM, Moises Silva <moises.silva@gmail.com> wrote:
On Mon, Dec 2, 2013 at 1:38 AM, Siri MM <sirimmfs@gmail.com> wrote:
Hi Moises,
Thanks for the reply. I am using a SNOM 320 phone as SIP end point. On being invited, it sends back a SIP/2.0 180 Ringing. How does FreeSWITCH/FreeTDM behave in this case - would it send media as part of the Progress message (in band)? Is there a way for me to check if media is being sent out to the caller?
I think strictly speaking a 180 should not trigger any early media (ring tone). The SIP response code for early media is 183 (although I've heard about 180 sometimes containing SDP).
If ring_ready solves your problem, what problem are you trying to solve now? I am not sure 180 should be triggering any media bridging code.
*Moises Silva**Manager, Software Engineering*
msilva@sangoma.com
Sangoma Technologies
100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada
t. +1 800 388 2475 (N. America)
t. +1 905 474 1990 x128
f. +1 905 474 9223
<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email+signatures>
Products<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | Solutions<http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | Events<http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | Contact<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | Wiki<http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | Facebook<http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | Twitter<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>`| | YouTube<http://www.youtube.com/sangomatechnologies?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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participants (2)
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Moises Silva
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Siri MM