[Freeswitch-users] Add Call-Info header

[Josh M. Patten]

Nevermind, this fixed it:

<action application="conference_set_auto_outcall" data="{alert_info=sipXpage}sofia/custom_dialplan/7001 at sipxpbx.bc.local;sipx-noroute=VoiceMail;sipx-userforward=false"/>

________________________________
From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Josh M. Patten [jpatten at co.brazos.tx.us]
Sent: Saturday, May 07, 2011 1:18 AM
To: freeswitch-users at 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 at 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 at 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 at 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 at 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 at 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 at 10.200.129.102>;tag=D3Q2t45cZvcar
   To: <sip:7001 at sipxpbx.bc.local;sipx-noroute=VoiceMail;sipx-userforward=false>
   Call-ID: 308e7013-f311-122e-448b-463bb5d10d94
   CSeq: 12037743 INVITE
   Contact: <sip:mod_sofia at 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 at 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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