[Freeswitch-users] Add Call-Info header

[Brad Mina]

Sounds like a wiki edit is in order!

On Fri, May 6, 2011 at 11:58 PM, Josh M. Patten <jpatten at co.brazos.tx.us>wrote:

>  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.localset(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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