Bridge an incoming call to an external SIP address
Hi everyone, I am trying to get FS to route inbound calls to an outside sip provider but I am getting the call hung up on the FS side just upon the first or second ring on the B-leg. The diaplan action being executed is: <action application="bridge" data="sofia/internal/ XXX@sip.linphone.org"/> I have a softphone registered to this provider sip.linphone.org and it works well when calling XXX from some other IP phone. FS reports this in the logs: [DEBUG] switch_ivr_originate.c:1274 Raw Codec Activation Success L16@48000hz 1 channel 20ms [DEBUG] switch_core_codec.c:221 sofia/internal/ anonymous@webrtc.sip.mydomain.co Push codec L16:100 [DEBUG] switch_ivr_originate.c:1343 Play Ringback Tone [%(2000,4000,440,480)] [DEBUG] sofia.c:6760 Channel sofia/internal/anonymous@webrtc.sip.mydomain.co entering state [terminated][500] [NOTICE] sofia.c:7779 Hangup sofia/internal/anonymous@webrtc.sip.mydomain.co [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] [DEBUG] switch_core_codec.c:246 sofia/internal/ anonymous@webrtc.sip.mydomain.co Restore previous codec opus:116. [NOTICE] switch_ivr_originate.c:3523 Hangup sofia/external/ XXX@sip.linphone.org [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] Result is the same if I try to call any other sip destination. I appreciate any input anyone may give: have been working on this for a week already. Thanks, Cato
Are you sure 'internal' is the sofia profile you want to be using to reach an external address? On 21 November 2016 at 13:35, Cato Gonzalez <catogonzalez@gmail.com> wrote:
Hi everyone,
I am trying to get FS to route inbound calls to an outside sip provider but I am getting the call hung up on the FS side just upon the first or second ring on the B-leg. The diaplan action being executed is:
<action application="bridge" data="sofia/internal/XXX@sip. linphone.org"/>
I have a softphone registered to this provider sip.linphone.org and it works well when calling XXX from some other IP phone. FS reports this in the logs:
[DEBUG] switch_ivr_originate.c:1274 Raw Codec Activation Success L16@48000hz 1 channel 20ms [DEBUG] switch_core_codec.c:221 sofia/internal/anonymous@ webrtc.sip.mydomain.co Push codec L16:100 [DEBUG] switch_ivr_originate.c:1343 Play Ringback Tone [%(2000,4000,440,480)] [DEBUG] sofia.c:6760 Channel sofia/internal/anonymous@ webrtc.sip.mydomain.co entering state [terminated][500] [NOTICE] sofia.c:7779 Hangup sofia/internal/anonymous@ webrtc.sip.mydomain.co [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] [DEBUG] switch_core_codec.c:246 sofia/internal/anonymous@ webrtc.sip.mydomain.co Restore previous codec opus:116. [NOTICE] switch_ivr_originate.c:3523 Hangup sofia/external/XXX@sip. linphone.org [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
Result is the same if I try to call any other sip destination. I appreciate any input anyone may give: have been working on this for a week already.
Thanks,
Cato
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Hi Steven, thanks for your reply. Yes, you are right, I had it all backwards :( I solved this by bridging to a gateway but *also* had to place this before bridging the call: <action application="set" data="bypass_media=true"/> Thanks again for replying. On Tue, Nov 22, 2016 at 6:14 PM, Steven Ayre <steveayre@gmail.com> wrote:
Are you sure 'internal' is the sofia profile you want to be using to reach an external address?
On 21 November 2016 at 13:35, Cato Gonzalez <catogonzalez@gmail.com> wrote:
Hi everyone,
I am trying to get FS to route inbound calls to an outside sip provider but I am getting the call hung up on the FS side just upon the first or second ring on the B-leg. The diaplan action being executed is:
<action application="bridge" data="sofia/internal/XXX@sip.l inphone.org"/>
I have a softphone registered to this provider sip.linphone.org and it works well when calling XXX from some other IP phone. FS reports this in the logs:
[DEBUG] switch_ivr_originate.c:1274 Raw Codec Activation Success L16@48000hz 1 channel 20ms [DEBUG] switch_core_codec.c:221 sofia/internal/anonymous@webrt c.sip.mydomain.co Push codec L16:100 [DEBUG] switch_ivr_originate.c:1343 Play Ringback Tone [%(2000,4000,440,480)] [DEBUG] sofia.c:6760 Channel sofia/internal/anonymous@webrt c.sip.mydomain.co entering state [terminated][500] [NOTICE] sofia.c:7779 Hangup sofia/internal/anonymous@webrt c.sip.mydomain.co [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] [DEBUG] switch_core_codec.c:246 sofia/internal/anonymous@webrt c.sip.mydomain.co Restore previous codec opus:116. [NOTICE] switch_ivr_originate.c:3523 Hangup sofia/external/ XXX@sip.linphone.org [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
Result is the same if I try to call any other sip destination. I appreciate any input anyone may give: have been working on this for a week already.
Thanks,
Cato
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (2)
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Cato Gonzalez
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Steven Ayre