Hi All, happy new you to you ! using a sangoma card and when dialing a mobile number which is having ringback tune / caller tune ; but a plain ring is heard to the user dialing that mobile through the trunk. I am using below syntax to dial out:- <action application="bridge" data="{monitor_early_media_ring_total=3}freetdm/wp1/a/${destination_number}"/> Any suggestions. Regards Sam
These was the output to fscli :- EXECUTE sofia/internal/7001@192.168.2.190bridge({monitor_early_media_ring_total=3}freetdm/wp1/a/9322273640) 2011-01-03 10:15:25.554494 [DEBUG] switch_ivr_originate.c:1954 variable string 0 = [monitor_early_media_ring_total=3] 2011-01-03 10:15:25.554494 [DEBUG] mod_freetdm.c:378 Set codec PCMA 20ms 2011-01-03 10:15:25.554494 [DEBUG] mod_freetdm.c:1361 Connect outbound channel FreeTDM/1:1/9322273640 2011-01-03 10:15:25.554494 [NOTICE] switch_channel.c:784 New Channel FreeTDM/1:1/9322273640 [42a43181-4124-426a-ab14-6cea045e5c22] 2011-01-03 10:15:25.554494 [DEBUG] mod_freetdm.c:1374 (FreeTDM/1:1/9322273640) State Change CS_NEW -> CS_INIT 2011-01-03 10:15:25.554494 [DEBUG] switch_core_session.c:1083 Send signal FreeTDM/1:1/9322273640 [BREAK] 2011-01-03 10:15:25.554494 [DEBUG] ftmod_sangoma_isdn.c:926 [s1c1][1:1] Changed state from DOWN to DIALING 2011-01-03 10:15:25.555440 [DEBUG] ftmod_sangoma_isdn.c:634 [s1c1][1:1] processing state change to DIALING 2011-01-03 10:15:25.555440 [INFO] ftmod_sangoma_isdn_stack_out.c:61 [s1c1][1:1] Outgoing call: Called No:[9322273640] Calling No:[7001] 2011-01-03 10:15:25.555440 [INFO] ftmod_sangoma_isdn_stack_out.c:74 [s1c1][1:1] Sending SETUP (suId:1 suInstId:67 spInstId:0 dchan:1 ces:0) 2011-01-03 10:15:25.565340 [DEBUG] switch_core_state_machine.c:320 (FreeTDM/1:1/9322273640) Running State Change CS_INIT 2011-01-03 10:15:25.565340 [DEBUG] switch_core_state_machine.c:356 (FreeTDM/1:1/9322273640) State INIT 2011-01-03 10:15:25.565340 [DEBUG] mod_freetdm.c:406 (FreeTDM/1:1/9322273640) State Change CS_INIT -> CS_ROUTING 2011-01-03 10:15:25.565340 [DEBUG] switch_core_session.c:1083 Send signal FreeTDM/1:1/9322273640 [BREAK] 2011-01-03 10:15:25.565340 [DEBUG] switch_core_state_machine.c:356 (FreeTDM/1:1/9322273640) State INIT going to sleep 2011-01-03 10:15:25.565340 [DEBUG] switch_core_state_machine.c:320 (FreeTDM/1:1/9322273640) Running State Change CS_ROUTING 2011-01-03 10:15:25.565340 [DEBUG] switch_channel.c:1615 (FreeTDM/1:1/9322273640) Callstate Change DOWN -> RINGING 2011-01-03 10:15:25.565340 [DEBUG] switch_core_state_machine.c:359 (FreeTDM/1:1/9322273640) State ROUTING 2011-01-03 10:15:25.565340 [DEBUG] mod_freetdm.c:431 FreeTDM/1:1/9322273640 CHANNEL ROUTING 2011-01-03 10:15:25.565340 [DEBUG] switch_ivr_originate.c:66 (FreeTDM/1:1/9322273640) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2011-01-03 10:15:25.565340 [DEBUG] switch_core_session.c:1083 Send signal FreeTDM/1:1/9322273640 [BREAK] 2011-01-03 10:15:25.565340 [DEBUG] switch_core_state_machine.c:359 (FreeTDM/1:1/9322273640) State ROUTING going to sleep 2011-01-03 10:15:25.565340 [DEBUG] switch_core_state_machine.c:320 (FreeTDM/1:1/9322273640) Running State Change CS_CONSUME_MEDIA 2011-01-03 10:15:25.566306 [DEBUG] switch_core_state_machine.c:378 (FreeTDM/1:1/9322273640) State CONSUME_MEDIA 2011-01-03 10:15:25.566306 [DEBUG] switch_core_state_machine.c:378 (FreeTDM/1:1/9322273640) State CONSUME_MEDIA going to sleep 2011-01-03 10:15:25.609441 [INFO] ftmod_sangoma_isdn_stack_rcv.c:172 [s1c1][1:1] Received PROCEED (suId:1 suInstId:67 spInstId:67 ces:0) 2011-01-03 10:15:25.609441 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:351 [s1c1][1:1] Processing PROCEED (suId:1 suInstId:67 spInstId:67 ces:0) 2011-01-03 10:15:25.609441 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:380 [s1c1][1:1] Changed state from DIALING to PROCEED 2011-01-03 10:15:25.609441 [DEBUG] ftmod_sangoma_isdn.c:634 [s1c1][1:1] processing state change to PROCEED 2011-01-03 10:15:25.610382 [DEBUG] mod_freetdm.c:2116 got clear channel sig [PROCEED] 2011-01-03 10:15:26.837602 [INFO] ftmod_sangoma_isdn_stack_rcv.c:172 [s1c1][1:1] Received ALERT (suId:1 suInstId:67 spInstId:67 ces:0) 2011-01-03 10:15:26.838544 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:351 [s1c1][1:1] Processing ALERT (suId:1 suInstId:67 spInstId:67 ces:0) 2011-01-03 10:15:26.838544 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:385 [s1c1][1:1] Changed state from PROCEED to RINGING 2011-01-03 10:15:26.838544 [DEBUG] ftmod_sangoma_isdn.c:634 [s1c1][1:1] processing state change to RINGING 2011-01-03 10:15:26.838544 [DEBUG] mod_freetdm.c:2116 got clear channel sig [RINGING] 2011-01-03 10:15:26.838544 [NOTICE] mod_freetdm.c:2184 Ring-Ready FreeTDM/1:1/9322273640! 2011-01-03 10:15:26.839557 [NOTICE] mod_sofia.c:2156 Ring-Ready sofia/internal/7001@192.168.2.190! 2011-01-03 10:15:26.839557 [DEBUG] switch_core_session.c:676 Send signal sofia/internal/7001@192.168.2.190 [BREAK] 2011-01-03 10:15:26.839557 [NOTICE] switch_ivr_originate.c:472 Ring Ready sofia/internal/7001@192.168.2.190! 2011-01-03 10:15:26.839557 [DEBUG] sofia.c:4606 Channel sofia/internal/ 7001@192.168.2.190 entering state [early][180] 2011-01-03 10:15:30.204270 [INFO] ftmod_sangoma_isdn_stack_rcv.c:172 [s1c1][1:1] Received PROGRESS (suId:1 suInstId:67 spInstId:67 ces:0) 2011-01-03 10:15:30.204270 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:351 [s1c1][1:1] Processing PROGRESS (suId:1 suInstId:67 spInstId:67 ces:0) Regds Sam On Mon, Jan 3, 2011 at 10:00 AM, Sam <u2nsam@gmail.com> wrote:
Hi All,
happy new you to you !
using a sangoma card and when dialing a mobile number which is having ringback tune / caller tune ; but a plain ring is heard to the user dialing that mobile through the trunk.
I am using below syntax to dial out:- <action application="bridge" data="{monitor_early_media_ring_total=3}freetdm/wp1/a/${destination_number}"/>
Any suggestions.
Regards Sam
Can you clarify what the actual issue is? -MC On Sun, Jan 2, 2011 at 8:30 PM, Sam <u2nsam@gmail.com> wrote:
Hi All,
happy new you to you !
using a sangoma card and when dialing a mobile number which is having ringback tune / caller tune ; but a plain ring is heard to the user dialing that mobile through the trunk.
I am using below syntax to dial out:- <action application="bridge" data="{monitor_early_media_ring_total=3}freetdm/wp1/a/${destination_number}"/>
Any suggestions.
Regards Sam
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Some mobile operators give VAS as ringback tone / caller tune to be set as mobile ringback, now when dialing out such mobile users who has ringback enabled i just get plain ring and no proceeding with 183 for the media to listen, how to enable such so that some one dialing out will hear ringback generated by the mobile operators. call flow is FS --> Sangoma --> mobile (ringback enabled by mobile operator ) Regards Sam On Tue, Jan 4, 2011 at 12:02 AM, Michael Collins <msc@freeswitch.org> wrote:
Can you clarify what the actual issue is? -MC
On Sun, Jan 2, 2011 at 8:30 PM, Sam <u2nsam@gmail.com> wrote:
Hi All,
happy new you to you !
using a sangoma card and when dialing a mobile number which is having ringback tune / caller tune ; but a plain ring is heard to the user dialing that mobile through the trunk.
I am using below syntax to dial out:- <action application="bridge" data="{monitor_early_media_ring_total=3}freetdm/wp1/a/${destination_number}"/>
Any suggestions.
Regards Sam
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You don't have any control over whether they send you audio, no? Either they send it or they don't. Unless there's something else that I'm missing... -MC On Mon, Jan 3, 2011 at 7:03 PM, Sam <u2nsam@gmail.com> wrote:
Some mobile operators give VAS as ringback tone / caller tune to be set as mobile ringback, now when dialing out such mobile users who has ringback enabled i just get plain ring and no proceeding with 183 for the media to listen, how to enable such so that some one dialing out will hear ringback generated by the mobile operators.
call flow is FS --> Sangoma --> mobile (ringback enabled by mobile operator )
Regards Sam
On Tue, Jan 4, 2011 at 12:02 AM, Michael Collins <msc@freeswitch.org>wrote:
Can you clarify what the actual issue is? -MC
On Sun, Jan 2, 2011 at 8:30 PM, Sam <u2nsam@gmail.com> wrote:
Hi All,
happy new you to you !
using a sangoma card and when dialing a mobile number which is having ringback tune / caller tune ; but a plain ring is heard to the user dialing that mobile through the trunk.
I am using below syntax to dial out:- <action application="bridge" data="{monitor_early_media_ring_total=3}freetdm/wp1/a/${destination_number}"/>
Any suggestions.
Regards Sam
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But when i do it from asterisk, it gives back the audio but not when i do it from freeswitch with sangoma. is there any method to listen to the audio send from the operator with combination with sangoma? Regds Sam On Wed, Jan 5, 2011 at 12:00 AM, Michael Collins <msc@freeswitch.org> wrote:
You don't have any control over whether they send you audio, no? Either they send it or they don't. Unless there's something else that I'm missing... -MC
On Mon, Jan 3, 2011 at 7:03 PM, Sam <u2nsam@gmail.com> wrote:
Some mobile operators give VAS as ringback tone / caller tune to be set as mobile ringback, now when dialing out such mobile users who has ringback enabled i just get plain ring and no proceeding with 183 for the media to listen, how to enable such so that some one dialing out will hear ringback generated by the mobile operators.
call flow is FS --> Sangoma --> mobile (ringback enabled by mobile operator )
Regards Sam
On Tue, Jan 4, 2011 at 12:02 AM, Michael Collins <msc@freeswitch.org>wrote:
Can you clarify what the actual issue is? -MC
On Sun, Jan 2, 2011 at 8:30 PM, Sam <u2nsam@gmail.com> wrote:
Hi All,
happy new you to you !
using a sangoma card and when dialing a mobile number which is having ringback tune / caller tune ; but a plain ring is heard to the user dialing that mobile through the trunk.
I am using below syntax to dial out:- <action application="bridge" data="{monitor_early_media_ring_total=3}freetdm/wp1/a/${destination_number}"/>
Any suggestions.
Regards Sam
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What kind of phone service is this? If it's ISDN then you can compare d-channel traces between the Asterisk and FreeSWITCH instances and see if there are any differences. Otherwise I'll have to defer to Sangoma on this one. -MC On Tue, Jan 4, 2011 at 8:14 PM, Sam <u2nsam@gmail.com> wrote:
But when i do it from asterisk, it gives back the audio but not when i do it from freeswitch with sangoma. is there any method to listen to the audio send from the operator with combination with sangoma?
Regds Sam
On Wed, Jan 5, 2011 at 12:00 AM, Michael Collins <msc@freeswitch.org>wrote:
You don't have any control over whether they send you audio, no? Either they send it or they don't. Unless there's something else that I'm missing... -MC
On Mon, Jan 3, 2011 at 7:03 PM, Sam <u2nsam@gmail.com> wrote:
Some mobile operators give VAS as ringback tone / caller tune to be set as mobile ringback, now when dialing out such mobile users who has ringback enabled i just get plain ring and no proceeding with 183 for the media to listen, how to enable such so that some one dialing out will hear ringback generated by the mobile operators.
call flow is FS --> Sangoma --> mobile (ringback enabled by mobile operator )
Regards Sam
On Tue, Jan 4, 2011 at 12:02 AM, Michael Collins <msc@freeswitch.org>wrote:
Can you clarify what the actual issue is? -MC
On Sun, Jan 2, 2011 at 8:30 PM, Sam <u2nsam@gmail.com> wrote:
Hi All,
happy new you to you !
using a sangoma card and when dialing a mobile number which is having ringback tune / caller tune ; but a plain ring is heard to the user dialing that mobile through the trunk.
I am using below syntax to dial out:- <action application="bridge" data="{monitor_early_media_ring_total=3}freetdm/wp1/a/${destination_number}"/>
Any suggestions.
Regards Sam
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I have attached the isdn pcap to this mail of FS --> Sangoma --> pri Regards Sam On Wed, Jan 5, 2011 at 10:36 AM, Michael Collins <msc@freeswitch.org> wrote:
What kind of phone service is this? If it's ISDN then you can compare d-channel traces between the Asterisk and FreeSWITCH instances and see if there are any differences. Otherwise I'll have to defer to Sangoma on this one.
-MC
On Tue, Jan 4, 2011 at 8:14 PM, Sam <u2nsam@gmail.com> wrote:
But when i do it from asterisk, it gives back the audio but not when i do it from freeswitch with sangoma. is there any method to listen to the audio send from the operator with combination with sangoma?
Regds Sam
On Wed, Jan 5, 2011 at 12:00 AM, Michael Collins <msc@freeswitch.org>wrote:
You don't have any control over whether they send you audio, no? Either they send it or they don't. Unless there's something else that I'm missing... -MC
On Mon, Jan 3, 2011 at 7:03 PM, Sam <u2nsam@gmail.com> wrote:
Some mobile operators give VAS as ringback tone / caller tune to be set as mobile ringback, now when dialing out such mobile users who has ringback enabled i just get plain ring and no proceeding with 183 for the media to listen, how to enable such so that some one dialing out will hear ringback generated by the mobile operators.
call flow is FS --> Sangoma --> mobile (ringback enabled by mobile operator )
Regards Sam
On Tue, Jan 4, 2011 at 12:02 AM, Michael Collins <msc@freeswitch.org>wrote:
Can you clarify what the actual issue is? -MC
On Sun, Jan 2, 2011 at 8:30 PM, Sam <u2nsam@gmail.com> wrote:
Hi All,
happy new you to you !
using a sangoma card and when dialing a mobile number which is having ringback tune / caller tune ; but a plain ring is heard to the user dialing that mobile through the trunk.
I am using below syntax to dial out:- <action application="bridge" data="{monitor_early_media_ring_total=3}freetdm/wp1/a/${destination_number}"/>
Any suggestions.
Regards Sam
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On Wed, Jan 5, 2011 at 12:21 AM, Sam <u2nsam@gmail.com> wrote:
I have attached the isdn pcap to this mail of FS --> Sangoma --> pri
Thanks Sam, The problem is we're not moving to PROGRESS_MEDIA state. I confirmed that the ISDN pcap has the progress indicator in the PROGRESS message, therefore we should be moving to PROGRESS_MEDIA. Are you working with latest libsng_isdn and freetdm? Latest version for sure should be moving to PROGRESS_MEDIA. Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. moy@sangoma.com
Hi, We are using libsng_isdn-7.0.0.x86_64 and latest freetdm, also there is 1 more problem you could see on the pcap, that we are sending callerid but its not presented to the callee instead of that the callee gets the presentation as the pilot number of pri which is 67287000. Regds Sam On Wed, Jan 5, 2011 at 7:54 PM, Moises Silva <moises.silva@gmail.com> wrote:
On Wed, Jan 5, 2011 at 12:21 AM, Sam <u2nsam@gmail.com> wrote:
I have attached the isdn pcap to this mail of FS --> Sangoma --> pri
Thanks Sam,
The problem is we're not moving to PROGRESS_MEDIA state. I confirmed that the ISDN pcap has the progress indicator in the PROGRESS message, therefore we should be moving to PROGRESS_MEDIA.
Are you working with latest libsng_isdn and freetdm?
Latest version for sure should be moving to PROGRESS_MEDIA.
Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. moy@sangoma.com
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On Wed, Jan 5, 2011 at 11:12 AM, Sam <u2nsam@gmail.com> wrote:
Hi,
We are using libsng_isdn-7.0.0.x86_64 and latest freetdm, also there is 1 more problem you could see on the pcap, that we are sending callerid but its not presented to the callee instead of that the callee gets the presentation as the pilot number of pri which is 67287000.
The log you posted has line numbers that do not match latest FreeTDM. Please update, there is a chance this was fixed in any of the multiple changes done in the past 2 weeks that were just merged yesterday into FreeSWITCH repository. As for the caller id, you mean the number 7001? and the callee (for which I don't have a trace here) receives 67287000? That seems to be your telco not liking our caller id (or where we send it) and putting the default (your PRI line number). May be your telco wants the caller id coming in a facility message? If you have a working pcap (if working from Asterisk, take the pcap when dialing from Asterisk) to see how the caller id is being sent, that would be useful. Please open a jira ticket to keep working on your issue. ( http://jira.freeswitch.org/) In the jira ticket add a debug log and matching pcap (by matching I mean the debug log and the pcap must be for the same call), and do not cut the debug log until the call is completely done. Thanks! Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. moy@sangoma.com
Tried installing the latest git-4272dcb 2011-01-05 20-12-19 -0600 it giving the sangoma codec error while instaling. Regards Sam On Wed, Jan 5, 2011 at 11:18 PM, Moises Silva <moises.silva@gmail.com>wrote:
On Wed, Jan 5, 2011 at 11:12 AM, Sam <u2nsam@gmail.com> wrote:
Hi,
We are using libsng_isdn-7.0.0.x86_64 and latest freetdm, also there is 1 more problem you could see on the pcap, that we are sending callerid but its not presented to the callee instead of that the callee gets the presentation as the pilot number of pri which is 67287000.
The log you posted has line numbers that do not match latest FreeTDM. Please update, there is a chance this was fixed in any of the multiple changes done in the past 2 weeks that were just merged yesterday into FreeSWITCH repository.
As for the caller id, you mean the number 7001? and the callee (for which I don't have a trace here) receives 67287000? That seems to be your telco not liking our caller id (or where we send it) and putting the default (your PRI line number).
May be your telco wants the caller id coming in a facility message?
If you have a working pcap (if working from Asterisk, take the pcap when dialing from Asterisk) to see how the caller id is being sent, that would be useful.
Please open a jira ticket to keep working on your issue. ( http://jira.freeswitch.org/)
In the jira ticket add a debug log and matching pcap (by matching I mean the debug log and the pcap must be for the same call), and do not cut the debug log until the call is completely done.
Thanks!
Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. moy@sangoma.com
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On Thu, Jan 6, 2011 at 3:14 AM, Sam <u2nsam@gmail.com> wrote:
Tried installing the latest git-4272dcb 2011-01-05 20-12-19 -0600 it giving the sangoma codec error while instaling.
Huh?? What does your PRI issue has to do, at all, with Sangoma codec module? Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. moy@sangoma.com
participants (3)
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Michael Collins
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Moises Silva
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Sam