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

Peter P GMX Prometheus001 at gmx.net
Fri Apr 23 02:39:10 PDT 2010


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>;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>;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
>>     #
>>
>>
>>
>>     _______________________________________________
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>
>
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>
> -- 
> Anthony Minessale II
>
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