Add Call-Info header
Hi there I'm trying to add add a Call-Info parameter to perform an auto-answer for phones that are not directly registered to FreeSWITCH (they are registered on a sipXecs server). None of the normal methods of auto-answer seem to work so I am attempting to copy the method that sipXecs uses to force phones to auto-answer. Upon analysis of how sipXecs performs an auto-answer, I find the following information in the SIP headers: Call-Info: <sip:7012@sipxpbx.bc.local;sipx-noroute=VoiceMail;sipx-userforward=false>;answer-after=0 The key here is the answer-after=0 part. What I've done in my dialplan is to add the following (In this example 7003 is initiating a call to a conference bridge which is then outcalling 7001): <action application="set" data="sip_h_Call-Info=answer-after=0"/> <action application="conference_set_auto_outcall" data="sofia/custom_dialplan/7001@sipxpbx.bc.local;sipx-noroute=VoiceMail;sipx-userforward=false"/> However in analysis of the SIP messages I never see any Call-Info header being set, though FreeSWITCH processes the entry: EXECUTE sofia/custom_dialplan/7003@sipxpbx.bc.local set(sip_h_Call-Info=answer-after=0) 2011-05-07 00:56:46.563506 [DEBUG] mod_dptools.c:1059 sofia/custom_dialplan/7003@sipxpbx.bc.local SET [sip_h_Call-Info]=[answer-after=0] However as you can see in the following SIP message, the header just isn't there: ------------------------------------------------------------------------ INVITE sip:7001@sipxpbx.bc.local;sipx-noroute=VoiceMail;sipx-userforward=false SIP/2.0 Via: SIP/2.0/UDP 10.200.129.102;rport;branch=z9hG4bK7aF5c90Dmecce Max-Forwards: 70 From: "FreeSWITCH" <sip:0000000000@10.200.129.102>;tag=D3Q2t45cZvcar To: <sip:7001@sipxpbx.bc.local;sipx-noroute=VoiceMail;sipx-userforward=false> Call-ID: 308e7013-f311-122e-448b-463bb5d10d94 CSeq: 12037743 INVITE Contact: <sip:mod_sofia@10.200.129.102:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-bfd0ba9 2011-03-07 13-02-41 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 239 X-FS-Support: update_display Remote-Party-ID: "FreeSWITCH" <sip:0000000000@10.200.129.102>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1304728354 1304728355 IN IP4 10.200.129.102 s=FreeSWITCH c=IN IP4 10.200.129.102 t=0 0 m=audio 19452 RTP/AVP 9 0 8 98 10 101 13 a=rtpmap:98 SPEEX/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ Thanks for your help :-)
Nevermind, this fixed it: <action application="conference_set_auto_outcall" data="{alert_info=sipXpage}sofia/custom_dialplan/7001@sipxpbx.bc.local;sipx-noroute=VoiceMail;sipx-userforward=false"/> ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org [freeswitch-users-bounces@lists.freeswitch.org] on behalf of Josh M. Patten [jpatten@co.brazos.tx.us] Sent: Saturday, May 07, 2011 1:18 AM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Add Call-Info header Hi there I'm trying to add add a Call-Info parameter to perform an auto-answer for phones that are not directly registered to FreeSWITCH (they are registered on a sipXecs server). None of the normal methods of auto-answer seem to work so I am attempting to copy the method that sipXecs uses to force phones to auto-answer. Upon analysis of how sipXecs performs an auto-answer, I find the following information in the SIP headers: Call-Info: <sip:7012@sipxpbx.bc.local;sipx-noroute=VoiceMail;sipx-userforward=false>;answer-after=0 The key here is the answer-after=0 part. What I've done in my dialplan is to add the following (In this example 7003 is initiating a call to a conference bridge which is then outcalling 7001): <action application="set" data="sip_h_Call-Info=answer-after=0"/> <action application="conference_set_auto_outcall" data="sofia/custom_dialplan/7001@sipxpbx.bc.local;sipx-noroute=VoiceMail;sipx-userforward=false"/> However in analysis of the SIP messages I never see any Call-Info header being set, though FreeSWITCH processes the entry: EXECUTE sofia/custom_dialplan/7003@sipxpbx.bc.local set(sip_h_Call-Info=answer-after=0) 2011-05-07 00:56:46.563506 [DEBUG] mod_dptools.c:1059 sofia/custom_dialplan/7003@sipxpbx.bc.local SET [sip_h_Call-Info]=[answer-after=0] However as you can see in the following SIP message, the header just isn't there: ------------------------------------------------------------------------ INVITE sip:7001@sipxpbx.bc.local;sipx-noroute=VoiceMail;sipx-userforward=false SIP/2.0 Via: SIP/2.0/UDP 10.200.129.102;rport;branch=z9hG4bK7aF5c90Dmecce Max-Forwards: 70 From: "FreeSWITCH" <sip:0000000000@10.200.129.102>;tag=D3Q2t45cZvcar To: <sip:7001@sipxpbx.bc.local;sipx-noroute=VoiceMail;sipx-userforward=false> Call-ID: 308e7013-f311-122e-448b-463bb5d10d94 CSeq: 12037743 INVITE Contact: <sip:mod_sofia@10.200.129.102:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-bfd0ba9 2011-03-07 13-02-41 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 239 X-FS-Support: update_display Remote-Party-ID: "FreeSWITCH" <sip:0000000000@10.200.129.102>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1304728354 1304728355 IN IP4 10.200.129.102 s=FreeSWITCH c=IN IP4 10.200.129.102 t=0 0 m=audio 19452 RTP/AVP 9 0 8 98 10 101 13 a=rtpmap:98 SPEEX/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ Thanks for your help :-)
Sounds like a wiki edit is in order! On Fri, May 6, 2011 at 11:58 PM, Josh M. Patten <jpatten@co.brazos.tx.us>wrote:
Nevermind, this fixed it:
<action application="conference_set_auto_outcall" data="{alert_info=sipXpage}sofia/custom_dialplan/7001@sipxpbx.bc.local ;sipx-noroute=VoiceMail;sipx-userforward=false"/>
------------------------------ *From:* freeswitch-users-bounces@lists.freeswitch.org [ freeswitch-users-bounces@lists.freeswitch.org] on behalf of Josh M. Patten [jpatten@co.brazos.tx.us] *Sent:* Saturday, May 07, 2011 1:18 AM *To:* freeswitch-users@lists.freeswitch.org *Subject:* [Freeswitch-users] Add Call-Info header
Hi there
I'm trying to add add a Call-Info parameter to perform an auto-answer for phones that are not directly registered to FreeSWITCH (they are registered on a sipXecs server). None of the normal methods of auto-answer seem to work so I am attempting to copy the method that sipXecs uses to force phones to auto-answer. Upon analysis of how sipXecs performs an auto-answer, I find the following information in the SIP headers:
Call-Info: <sip:7012@sipxpbx.bc.local ;sipx-noroute=VoiceMail;sipx-userforward=false>;answer-after=0
The key here is the answer-after=0 part. What I've done in my dialplan is to add the following (In this example 7003 is initiating a call to a conference bridge which is then outcalling 7001):
<action application="set" data="sip_h_Call-Info=answer-after=0"/> <action application="conference_set_auto_outcall" data="sofia/custom_dialplan/7001@sipxpbx.bc.local ;sipx-noroute=VoiceMail;sipx-userforward=false"/>
However in analysis of the SIP messages I never see any Call-Info header being set, though FreeSWITCH processes the entry: EXECUTE sofia/custom_dialplan/7003@sipxpbx.bc.localset(sip_h_Call-Info=answer-after=0) 2011-05-07 00:56:46.563506 [DEBUG] mod_dptools.c:1059 sofia/custom_dialplan/7003@sipxpbx.bc.local SET [sip_h_Call-Info]=[answer-after=0]
However as you can see in the following SIP message, the header just isn't there:
------------------------------------------------------------------------ INVITE sip:7001@sipxpbx.bc.local;sipx-noroute=VoiceMail;sipx-userforward=false SIP/2.0 Via: SIP/2.0/UDP 10.200.129.102;rport;branch=z9hG4bK7aF5c90Dmecce Max-Forwards: 70 From: "FreeSWITCH" <sip:0000000000@10.200.129.102>;tag=D3Q2t45cZvcar To: <sip:7001@sipxpbx.bc.local ;sipx-noroute=VoiceMail;sipx-userforward=false> Call-ID: 308e7013-f311-122e-448b-463bb5d10d94 CSeq: 12037743 INVITE Contact: <sip:mod_sofia@10.200.129.102:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-bfd0ba9 2011-03-07 13-02-41 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 239 X-FS-Support: update_display Remote-Party-ID: "FreeSWITCH" <sip:0000000000@10.200.129.102
;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1304728354 1304728355 IN IP4 10.200.129.102 s=FreeSWITCH c=IN IP4 10.200.129.102 t=0 0 m=audio 19452 RTP/AVP 9 0 8 98 10 101 13 a=rtpmap:98 SPEEX/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------
Thanks for your help :-)
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participants (2)
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Brad Mina
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Josh M. Patten