[Freeswitch-users] effective_caller_id_name not working while calling multiple endpoints with :_:

Anthony Minessale anthony.minessale at gmail.com
Wed Apr 28 13:26:54 PDT 2010


effective_caller_id_name does not go in the brackets
you want origination_caller_id_name and origination_caller_id_number inside
the <> and/or {}



On Wed, Apr 28, 2010 at 11:05 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> Nobody has an idea how to solve this? Shall I open a JIRA?
>
> Best regards
> Peter
>
> Peter P GMX schrieb:
> > Hello Anthony,
> >
> > I upgraded to newest GIT and tried it
> >
> > The dialplan now contains the fowllowing:
> > <action application="bridge" data="<effective_caller_id_name=My
> > Name>{global_to_originate_1=true}user/200 at my.domain:_:user/201 at my.domain
> :_:user/205 at my.domain:_:user/208 at my.domain:_:user/211 at my.domain
> :_:user/230 at my.domain"/>
> >
> > When the dialplan is executed, it seems to be processed correctly:
> > EXECUTE sofia/local/06912345678 at 192.168.178.218:5060
> > bridge(<effective_caller_id_name=My
> > Name>{global_to_originate_1=true}user/200 at my.domain:_:user/201 at my.domain
> :_:user/205 at my.domain:_:user/208 at my.domain:_:user/211 at my.domain
> :_:user/230 at my.domain)
> > 2010-04-23 10:59:12.479598 [DEBUG] switch_ivr_originate.c:1394 variable
> > string 0 = [effective_caller_id_name=My Name]
> > 2010-04-23 10:59:12.501255 [DEBUG] switch_ivr_originate.c:1885 variable
> > string 0 = [global_to_originate_1=true]
> > 2010-04-23 10:59:12.590269 [DEBUG] switch_ivr_originate.c:1885 variable
> > string 0 = [presence_id=200 at my.domain]
> > 2010-04-23 10:59:12.590269 [DEBUG] switch_ivr_originate.c:1885 variable
> > string 1 = [transfer_fallback_extension=200]
> >
> > However the INVITE message does not contain the caller_id_name, see below
> >
> >
> > What am I doing wrong?
> >
> > Best regards
> > Peter
> >
> > U 192.168.178.220:5060 -> 192.168.178.50:3072
> > INVITE sip:200 at 192.168.178.50:3072;line=v3bii5l2 SIP/2.0.
> > Via: SIP/2.0/UDP 192.168.178.220;rport;branch=z9hG4bK4jUFH215p85tr.
> > Max-Forwards: 70.
> > From: "06912345678" <sip:06912345678 at 192.168.178.220<sip%3A06912345678 at 192.168.178.220>
> >;tag=a8m8ccQcgjjUg.
> > To: <sip:200 at 192.168.178.50:3072;line=v3bii5l2>.
> > Call-ID: b6cafa23-c959-122d-4682-080027e51f59.
> > CSeq: 129888323 INVITE.
> > Contact: <sip:mod_sofia at 192.168.178.220:5060>.
> > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-.
> > 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: 348.
> > X-FS-Support: update_display.
> > Remote-Party-ID: "06912345678"
> > <sip:06912345678 at 192.168.178.220 <sip%3A06912345678 at 192.168.178.220>
> >;party=calling;screen=yes;privacy=off.
> > .
> > v=0.
> > o=FreeSWITCH 1272001222 1272001223 IN IP4 192.168.178.220.
> > s=FreeSWITCH.
> > c=IN IP4 192.168.178.220.
> > t=0 0.
> > m=audio 12096 RTP/AVP 9 0 8 99 3 101 13.
> > a=rtpmap:9 G722/8000.
> > a=rtpmap:0 PCMU/8000.
> > a=rtpmap:8 PCMA/8000.
> > a=rtpmap:99 SPEEX/8000.
> > a=rtpmap:3 GSM/8000.
> > a=rtpmap:101 telephone-event/8000.
> > a=fmtp:101 0-16.
> > a=rtpmap:13 CN/8000.
> > a=ptime:20.
> >
> > Anthony Minessale schrieb:
> >
> >> when using enterprise_originate you must use the special leading <>
> >> brackets to insert global variables meant for each tier 1 originate
> >>
> >>
> <global_to_all_originates=true>{global_to_originate_1=true}sofia/internal/
> foo at bar.com
> >> <mailto:foo at bar.com>,sofia/internal/foo2 at bar.com:_:sofia/internal/
> foo3 at bar3.com
> >> <mailto:foo3 at bar3.com>
> >>
> >>
> >> On Wed, Apr 21, 2010 at 5:27 AM, David Ponzone
> >> <david.ponzone at gmail.com <mailto:david.ponzone at gmail.com>> wrote:
> >>
> >>     I think you should first thing update to latest GIT :)
> >>
> >>     David Ponzone  Direction Technique
> >>     email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> >>     tel:      01 74 03 18 97
> >>     gsm:   06 66 98 76 34
> >>
> >>     Service Client IPeva
> >>     tel:      0811 46 26 26
> >>     www.ipeva.fr  -   www.ipeva-studio.com
> >>
> >>     /Ce message et toutes les pièces jointes sont confidentiels et
> >>     établis à l'intention exclusive de ses destinataires. Toute
> >>     utilisation ou diffusion non autorisée est interdite. Tout message
> >>     électronique est susceptible d'altération. /*/IPeva/*/ décline
> >>     toute responsabilité au titre de ce message s'il a été altéré,
> >>     déformé ou falsifié. Si vous n'êtes pas destinataire de ce
> >>     message, merci de le détruire immédiatement et d'avertir
> >>     l'expéditeur./
> >>     /
> >>     /
> >>
> >>
> >>
> >>     Le 21/04/2010 à 12:14, Peter P GMX a écrit :
> >>
> >>
> >>>     Setting the effective_caller_id_name when dialing multiple
> endpoints
> >>>     with :_: do not seem to work.
> >>>     See example:
> >>>      <action application="set" data="effective_caller_id_name=MyName"/>
> >>>      <action application="bridge"
> >>>
> >>>     data="user/30 at mydomain.com
> >>>     <mailto:user/30 at mydomain.com>:_:user/31 at mydomain.com
> >>>     <mailto:user/31 at mydomain.com>:_:user/32 at mydomain.com
> >>>     <mailto:user/32 at mydomain.com>:_:user/33 at mydomain.com
> >>>     <mailto:user/33 at mydomain.com>:_:user/34 at mydomain.com
> >>>     <mailto:user/34 at mydomain.com>"/>
> >>>
> >>>     Freeswitch tries to set it:
> >>>       EXECUTE sofia/external/069xxxxxxxx at 10.xx.xx.1414
> >>>     <mailto:sofia/external/069xxxxxxxx at 10.xx.xx.1414>:5060
> >>>     set(effective_caller_id_name=MyName)
> >>>       2010-04-21 11:11:48.642571 [DEBUG] mod_dptools.c:816
> >>>     sofia/external/069xxxxxxxx at 10.xx.xx.1414
> >>>     <mailto:sofia/external/069xxxxxxxx at 10.xx.xx.1414>:5060 SET
> >>>     [effective_caller_id_name]=[MyName]
> >>>
> >>>     But the SIP messages do not contain the effective_caller_id_name.
> >>>
> >>>     If we change the ":_:" sperator to "," then the
> >>>     effective_caller_id_name
> >>>     is correctly submittted (hower I cannot call
> >>>     multiple-registrations on
> >>>     one number then).
> >>>
> >>>     We are on
> >>>      FreeSWITCH Version 1.0.head (svn-17188)
> >>>
> >>>     Any ideas how to overcome this? Or shall I open a JIRA?
> >>>
> >>>     Best regards
> >>>     Peter
> >>>
> >>>     See example SIP message:
> >>>
> >>>     U 10.xx.xx.141:5060 -> 10.xx.xx.14172:2048
> >>>     INVITE sip:31 at 10.xx.xx.14172:2048;line=hxbudrul SIP/2.0.
> >>>     Via: SIP/2.0/UDP 10.xx.xx.141;rport;branch=z9hG4bKZD66c84339SHH.
> >>>     Max-Forwards: 70.
> >>>     From: "069xxxxxxxx" <sip:069xxxxxxxx at 10.xx.xx.141
> >;tag=5evr6508K9S3K.
> >>>     To: <sip:31 at 10.xx.xx.14172:2048;line=hxbudrul>.
> >>>     Call-ID: ad359ddd-c7cc-122d-3683-001517c965a5.
> >>>     CSeq: 129803060 INVITE.
> >>>     Contact: <sip:mod_sofia at 10.xx.xx.141:5060>.
> >>>     User-Agent: FreeSWITCH-mod_sofia/1.0.head-svn-17188.
> >>>     Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >>>     REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
> >>>     Supported: timer, precondition, path, replaces.
> >>>     Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
> >>>     include-session-description, presence.winfo, message-summary,
> refer.
> >>>     Content-Type: application/sdp.
> >>>     Content-Disposition: session.
> >>>     Content-Length: 920.
> >>>     X-FS-Support: update_display.
> >>>     Remote-Party-ID: "069xxxxxxxx"
> >>>     <sip:069xxxxxxxx at 10.xx.xx.141
> >;party=calling;screen=yes;privacy=off.
> >>>     .
> >>>     v=0.
> >>>     o=FreeSWITCH 1271830560 1271830561 IN IP4 10.xx.xx.141.
> >>>     s=FreeSWITCH.
> >>>     c=IN IP4 10.xx.xx.141.
> >>>     t=0 0.
> >>>     m=audio 12232 RTP/AVP 115 96 99 18 4 7 124 8 6 123 0 10 98 122 9
> >>>     121 3
> >>>     101 13.
> >>>     a=rtpmap:115 G7221/32000.
> >>>     a=fmtp:115 bitrate=48000.
> >>>     a=rtpmap:96 AMR/8000.
> >>>     a=fmtp:96 octet-align=0.
> >>>     a=rtpmap:99 SPEEX/32000.
> >>>     a=rtpmap:18 G729/8000.
> >>>     a=rtpmap:4 G723/8000.
> >>>     a=rtpmap:7 LPC/8000.
> >>>     a=rtpmap:124 G726-16/8000.
> >>>     a=rtpmap:8 PCMA/8000.
> >>>     a=rtpmap:6 DVI4/16000.
> >>>     a=rtpmap:123 G726-24/8000.
> >>>     a=rtpmap:0 PCMU/8000.
> >>>     a=rtpmap:10 L16/22050.
> >>>     a=rtpmap:98 iLBC/8000.
> >>>     a=fmtp:98 mode
> >>>     #
> >>>
> >>>
> >>>
> >>>     _______________________________________________
> >>>     FreeSWITCH-users mailing list
> >>>     FreeSWITCH-users at lists.freeswitch.org
> >>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>     http://www.freeswitch.org
> >>>
> >>     _______________________________________________
> >>     FreeSWITCH-users mailing list
> >>     FreeSWITCH-users at lists.freeswitch.org
> >>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>     http://www.freeswitch.org
> >>
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> >> <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> >> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> >> <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> >> pstn:+19193869900
> >> ------------------------------------------------------------------------
> >>
> >> _______________________________________________
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> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
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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 at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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