Sounds like a wiki edit is in order!<br><br><div class="gmail_quote">On Fri, May 6, 2011 at 11:58 PM, Josh M. Patten <span dir="ltr"><<a href="mailto:jpatten@co.brazos.tx.us">jpatten@co.brazos.tx.us</a>></span> wrote:<br>
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<div style="direction:ltr;font-family:Tahoma;color:#000000;font-size:10pt">Nevermind, this fixed it:<br>
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<action application="conference_set_auto_outcall" data="{alert_info=sipXpage}sofia/custom_dialplan/7001@sipxpbx.bc.local;sipx-noroute=VoiceMail;sipx-userforward=false"/><br>
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<div style="direction:ltr"><font color="#000000" size="2" face="Tahoma"><b>From:</b> <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a> [<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a>] on behalf of Josh M. Patten [<a href="mailto:jpatten@co.brazos.tx.us" target="_blank">jpatten@co.brazos.tx.us</a>]<br>
<b>Sent:</b> Saturday, May 07, 2011 1:18 AM<br>
<b>To:</b> <a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br>
<b>Subject:</b> [Freeswitch-users] Add Call-Info header<br>
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Hi there<br>
<br>
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:<br>
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Call-Info: <sip:7012@sipxpbx.bc.local;sipx-noroute=VoiceMail;sipx-userforward=false>;answer-after=0<br>
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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):<br>
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<action application="set" data="sip_h_Call-Info=answer-after=0"/><br>
<action application="conference_set_auto_outcall" data="sofia/custom_dialplan/7001@sipxpbx.bc.local;sipx-noroute=VoiceMail;sipx-userforward=false"/><br>
<br>
However in analysis of the SIP messages I never see any Call-Info header being set, though FreeSWITCH processes the entry:<br>
EXECUTE sofia/custom_dialplan/7003@sipxpbx.bc.local set(sip_h_Call-Info=answer-after=0)<br>
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]<br>
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However as you can see in the following SIP message, the header just isn't there:<br>
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------------------------------------------------------------------------<br>
INVITE sip:7001@sipxpbx.bc.local;sipx-noroute=VoiceMail;sipx-userforward=false SIP/2.0<br>
Via: SIP/2.0/UDP 10.200.129.102;rport;branch=z9hG4bK7aF5c90Dmecce<br>
Max-Forwards: 70<br>
From: "FreeSWITCH" <<a href="mailto:sip%3A0000000000@10.200.129.102" target="_blank">sip:0000000000@10.200.129.102</a>>;tag=D3Q2t45cZvcar<br>
To: <sip:7001@sipxpbx.bc.local;sipx-noroute=VoiceMail;sipx-userforward=false><br>
Call-ID: 308e7013-f311-122e-448b-463bb5d10d94<br>
CSeq: 12037743 INVITE<br>
Contact: <<a href="http://sip:mod_sofia@10.200.129.102:5060" target="_blank">sip:mod_sofia@10.200.129.102:5060</a>><br>
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-bfd0ba9 2011-03-07 13-02-41 -0600<br>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE<br>
Supported: timer, precondition, path, replaces<br>
Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer<br>
Content-Type: application/sdp<br>
Content-Disposition: session<br>
Content-Length: 239<br>
X-FS-Support: update_display<br>
Remote-Party-ID: "FreeSWITCH" <<a href="mailto:sip%3A0000000000@10.200.129.102" target="_blank">sip:0000000000@10.200.129.102</a>>;party=calling;screen=yes;privacy=off<br>
<br>
v=0<br>
o=FreeSWITCH 1304728354 1304728355 IN IP4 10.200.129.102<br>
s=FreeSWITCH<br>
c=IN IP4 10.200.129.102<br>
t=0 0<br>
m=audio 19452 RTP/AVP 9 0 8 98 10 101 13<br>
a=rtpmap:98 SPEEX/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=ptime:20<br>
------------------------------------------------------------------------<br>
<br>
Thanks for your help :-)<br>
<br>
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