bridge doesn't respect bypass_media=true over the socket
Hello everyone, PB here: http://pastebin.freeswitch.org/11482 FS rev 15909. The relevant bits from the log are here (starting around line 135): # 2009-12-11 12:06:12.875158 [DEBUG] switch_ivr.c:548 sofia/pjsip/nobody@192.168.4.253 Command Execute bridge({originate_timeout=30,bypass_media=true,origination_caller_id_number=1111,origination_caller_id_name=Tara Ousley}sofia/voalte/huttoj@192.168.4.17) # EXECUTE sofia/pjsip/nobody@192.168.4.253 bridge({originate_timeout=30,bypass_media=true,origination_caller_id_number=1111,origination_caller_id_name=Tara Ousley}sofia/voalte/huttoj@192.168.4.17) # 2009-12-11 12:06:12.875158 [DEBUG] switch_ivr_originate.c:1735 variable string 0 = [originate_timeout=30] # 2009-12-11 12:06:12.875158 [DEBUG] switch_ivr_originate.c:1735 variable string 1 = [bypass_media=true] # 2009-12-11 12:06:12.875158 [DEBUG] switch_ivr_originate.c:1735 variable string 2 = [origination_caller_id_number=1111] # 2009-12-11 12:06:12.875158 [DEBUG] switch_ivr_originate.c:1735 variable string 3 = [origination_caller_id_name=Tara Ousley] bypass_media=true yet the SDP of the outgoing INVITE looks like this: # send 1032 bytes to udp/[192.168.4.17]:5060 at 12:06:12.876994: # ------------------------------------------------------------------------ # INVITE sip:huttoj@192.168.4.17 SIP/2.0 # Via: SIP/2.0/UDP 192.168.2.10:5062;rport;branch=z9hG4bK1K6mc3NcmmaNr # Max-Forwards: 69 # From: "Tara Ousley" <sip:1111@192.168.2.10>;tag=0tU8SN9pvejNK # To: <sip:huttoj@192.168.4.17> # Call-ID: 802f4045-4215-42a2-91a6-ff9cf18b1aa8 # CSeq: 124148250 INVITE # Contact: <sip:mod_sofia@192.168.2.10:5062> # User-Agent: Voalte Voice # Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY # Supported: timer, precondition, path, replaces # Allow-Events: talk, refer # Content-Type: application/sdp # Content-Disposition: session # Content-Length: 271 # X-voalte-call-id: 898ef33c-50f2-487c-9e8d-8c6fcee15ab8 # Remote-Party-ID: "Tara Ousley" <sip:1111@192.168.2.10>;party=calling;screen=yes;privacy=off # # v=0 # o=FreeSWITCH 1260508000 1260508001 IN IP4 192.168.2.10 # s=FreeSWITCH # c=IN IP4 192.168.2.10 # t=0 0 # m=audio 25172 RTP/AVP 9 0 101 # a=rtpmap:9 G722/8000 # a=rtpmap:0 PCMU/8000 # a=rtpmap:101 telephone-event/8000 # a=fmtp:101 0-16 # a=silenceSupp:off - - - - # a=ptime:20 192.168.2.10 is the address of my FS box... -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
Hey, You can't set bypass_media=true in {} or it will not take effect unless that b leg itself becomes an a leg some day. you need to execute set on bypass_media=true on the leg before you call bridge to trigger it. Alternatively you could set {bypass_media_after_bridge=true} or set it on A leg as described above on either leg and it will do the bypass once the audio is flowing. On Fri, Dec 11, 2009 at 11:14 AM, Kristian Kielhofner < kristian.kielhofner@gmail.com> wrote:
Hello everyone,
PB here:
http://pastebin.freeswitch.org/11482
FS rev 15909. The relevant bits from the log are here (starting around line 135):
# 2009-12-11 12:06:12.875158 [DEBUG] switch_ivr.c:548 sofia/pjsip/nobody@192.168.4.253 Command Execute
bridge({originate_timeout=30,bypass_media=true,origination_caller_id_number=1111,origination_caller_id_name=Tara Ousley}sofia/voalte/huttoj@192.168.4.17) # EXECUTE sofia/pjsip/nobody@192.168.4.253
bridge({originate_timeout=30,bypass_media=true,origination_caller_id_number=1111,origination_caller_id_name=Tara Ousley}sofia/voalte/huttoj@192.168.4.17) # 2009-12-11 12:06:12.875158 [DEBUG] switch_ivr_originate.c:1735 variable string 0 = [originate_timeout=30] # 2009-12-11 12:06:12.875158 [DEBUG] switch_ivr_originate.c:1735 variable string 1 = [bypass_media=true] # 2009-12-11 12:06:12.875158 [DEBUG] switch_ivr_originate.c:1735 variable string 2 = [origination_caller_id_number=1111] # 2009-12-11 12:06:12.875158 [DEBUG] switch_ivr_originate.c:1735 variable string 3 = [origination_caller_id_name=Tara Ousley]
bypass_media=true yet the SDP of the outgoing INVITE looks like this:
# send 1032 bytes to udp/[192.168.4.17]:5060 at 12:06:12.876994: # ------------------------------------------------------------------------ # INVITE sip:huttoj@192.168.4.17 <sip%3Ahuttoj@192.168.4.17> SIP/2.0 # Via: SIP/2.0/UDP 192.168.2.10:5062;rport;branch=z9hG4bK1K6mc3NcmmaNr # Max-Forwards: 69 # From: "Tara Ousley" <sip:1111@192.168.2.10 <sip%3A1111@192.168.2.10>
;tag=0tU8SN9pvejNK # To: <sip:huttoj@192.168.4.17 <sip%3Ahuttoj@192.168.4.17>> # Call-ID: 802f4045-4215-42a2-91a6-ff9cf18b1aa8 # CSeq: 124148250 INVITE # Contact: <sip:mod_sofia@192.168.2.10:5062> # User-Agent: Voalte Voice # Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY # Supported: timer, precondition, path, replaces # Allow-Events: talk, refer # Content-Type: application/sdp # Content-Disposition: session # Content-Length: 271 # X-voalte-call-id: 898ef33c-50f2-487c-9e8d-8c6fcee15ab8 # Remote-Party-ID: "Tara Ousley" <sip:1111@192.168.2.10 <sip%3A1111@192.168.2.10> ;party=calling;screen=yes;privacy=off #
# v=0 # o=FreeSWITCH 1260508000 1260508001 IN IP4 192.168.2.10 # s=FreeSWITCH # c=IN IP4 192.168.2.10 # t=0 0 # m=audio 25172 RTP/AVP 9 0 101 # a=rtpmap:9 G722/8000 # a=rtpmap:0 PCMU/8000 # a=rtpmap:101 telephone-event/8000 # a=fmtp:101 0-16 # a=silenceSupp:off - - - - # a=ptime:20
192.168.2.10 is the address of my FS box...
-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Thanks, that was it! On Fri, Dec 11, 2009 at 12:31 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Hey,
You can't set bypass_media=true in {} or it will not take effect unless that b leg itself becomes an a leg some day. you need to execute set on bypass_media=true on the leg before you call bridge to trigger it.
Alternatively you could set {bypass_media_after_bridge=true} or set it on A leg as described above on either leg and it will do the bypass once the audio is flowing.
-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
participants (2)
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Anthony Minessale
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Kristian Kielhofner