effective_caller_id_name not working while calling multiple endpoints with :_:
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@mydomain.com:_:user/31@mydomain.com:_:user/32@mydomain.com:_:user/33@mydomain.com:_:user/34@mydomain.com"/> Freeswitch tries to set it: EXECUTE sofia/external/069xxxxxxxx@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@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@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@10.xx.xx.141>;tag=5evr6508K9S3K. To: <sip:31@10.xx.xx.14172:2048;line=hxbudrul>. Call-ID: ad359ddd-c7cc-122d-3683-001517c965a5. CSeq: 129803060 INVITE. Contact: <sip:mod_sofia@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@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 #
I think you should first thing update to latest GIT :) David Ponzone Direction Technique email: david.ponzone@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@mydomain.com:_:user/31@mydomain.com:_:user/32@mydomain.com :_:user/33@mydomain.com:_:user/34@mydomain.com"/>
Freeswitch tries to set it: EXECUTE sofia/external/069xxxxxxxx@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@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@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@10.xx.xx.141>;tag=5evr6508K9S3K. To: <sip:31@10.xx.xx.14172:2048;line=hxbudrul>. Call-ID: ad359ddd-c7cc-122d-3683-001517c965a5. CSeq: 129803060 INVITE. Contact: <sip:mod_sofia@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@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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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@bar.com,sofia/internal/foo2@bar.com:_:sofia/internal/foo3@bar3.com On Wed, Apr 21, 2010 at 5:27 AM, David Ponzone <david.ponzone@gmail.com>wrote:
I think you should first thing update to latest GIT :)
David Ponzone Direction Technique email: david.ponzone@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@mydomain.com:_:user/31@mydomain.com:_:user/32@mydomain.com :_:user/33@mydomain.com:_:user/34@mydomain.com"/>
Freeswitch tries to set it: EXECUTE sofia/external/069xxxxxxxx@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@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@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@10.xx.xx.141>;tag=5evr6508K9S3K. To: <sip:31@10.xx.xx.14172:2048;line=hxbudrul>. Call-ID: ad359ddd-c7cc-122d-3683-001517c965a5. CSeq: 129803060 INVITE. Contact: <sip:mod_sofia@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@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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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
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@my.domain:_:user/201@my.domain:_:user/205@my.domain:_:user/208@my.domain:_:user/211@my.domain:_:user/230@my.domain"/> When the dialplan is executed, it seems to be processed correctly: EXECUTE sofia/local/06912345678@192.168.178.218:5060 bridge(<effective_caller_id_name=My Name>{global_to_originate_1=true}user/200@my.domain:_:user/201@my.domain:_:user/205@my.domain:_:user/208@my.domain:_:user/211@my.domain:_:user/230@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@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@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@192.168.178.220>;tag=a8m8ccQcgjjUg. To: <sip:200@192.168.178.50:3072;line=v3bii5l2>. Call-ID: b6cafa23-c959-122d-4682-080027e51f59. CSeq: 129888323 INVITE. Contact: <sip:mod_sofia@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@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@bar.com <mailto:foo@bar.com>,sofia/internal/foo2@bar.com:_:sofia/internal/foo3@bar3.com <mailto:foo3@bar3.com>
On Wed, Apr 21, 2010 at 5:27 AM, David Ponzone <david.ponzone@gmail.com <mailto:david.ponzone@gmail.com>> wrote:
I think you should first thing update to latest GIT :)
David Ponzone Direction Technique email: david.ponzone@ipeva.fr <mailto:david.ponzone@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@mydomain.com <mailto:user/30@mydomain.com>:_:user/31@mydomain.com <mailto:user/31@mydomain.com>:_:user/32@mydomain.com <mailto:user/32@mydomain.com>:_:user/33@mydomain.com <mailto:user/33@mydomain.com>:_:user/34@mydomain.com <mailto:user/34@mydomain.com>"/>
Freeswitch tries to set it: EXECUTE sofia/external/069xxxxxxxx@10.xx.xx.1414 <mailto:sofia/external/069xxxxxxxx@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@10.xx.xx.1414 <mailto:sofia/external/069xxxxxxxx@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@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@10.xx.xx.141>;tag=5evr6508K9S3K. To: <sip:31@10.xx.xx.14172:2048;line=hxbudrul>. Call-ID: ad359ddd-c7cc-122d-3683-001517c965a5. CSeq: 129803060 INVITE. Contact: <sip:mod_sofia@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@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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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900 ------------------------------------------------------------------------
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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@my.domain:_:user/201@my.domain:_:user/205@my.domain:_:user/208@my.domain:_:user/211@my.domain:_:user/230@my.domain"/>
When the dialplan is executed, it seems to be processed correctly: EXECUTE sofia/local/06912345678@192.168.178.218:5060 bridge(<effective_caller_id_name=My Name>{global_to_originate_1=true}user/200@my.domain:_:user/201@my.domain:_:user/205@my.domain:_:user/208@my.domain:_:user/211@my.domain:_:user/230@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@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@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@192.168.178.220>;tag=a8m8ccQcgjjUg. To: <sip:200@192.168.178.50:3072;line=v3bii5l2>. Call-ID: b6cafa23-c959-122d-4682-080027e51f59. CSeq: 129888323 INVITE. Contact: <sip:mod_sofia@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@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@bar.com <mailto:foo@bar.com>,sofia/internal/foo2@bar.com:_:sofia/internal/foo3@bar3.com <mailto:foo3@bar3.com>
On Wed, Apr 21, 2010 at 5:27 AM, David Ponzone <david.ponzone@gmail.com <mailto:david.ponzone@gmail.com>> wrote:
I think you should first thing update to latest GIT :)
David Ponzone Direction Technique email: david.ponzone@ipeva.fr <mailto:david.ponzone@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@mydomain.com <mailto:user/30@mydomain.com>:_:user/31@mydomain.com <mailto:user/31@mydomain.com>:_:user/32@mydomain.com <mailto:user/32@mydomain.com>:_:user/33@mydomain.com <mailto:user/33@mydomain.com>:_:user/34@mydomain.com <mailto:user/34@mydomain.com>"/>
Freeswitch tries to set it: EXECUTE sofia/external/069xxxxxxxx@10.xx.xx.1414 <mailto:sofia/external/069xxxxxxxx@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@10.xx.xx.1414 <mailto:sofia/external/069xxxxxxxx@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@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@10.xx.xx.141>;tag=5evr6508K9S3K. To: <sip:31@10.xx.xx.14172:2048;line=hxbudrul>. Call-ID: ad359ddd-c7cc-122d-3683-001517c965a5. CSeq: 129803060 INVITE. Contact: <sip:mod_sofia@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@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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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900 ------------------------------------------------------------------------
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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@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@my.domain:_:user/201@my.domain :_:user/205@my.domain:_:user/208@my.domain:_:user/211@my.domain :_:user/230@my.domain"/>
When the dialplan is executed, it seems to be processed correctly: EXECUTE sofia/local/06912345678@192.168.178.218:5060 bridge(<effective_caller_id_name=My Name>{global_to_originate_1=true}user/200@my.domain:_:user/201@my.domain :_:user/205@my.domain:_:user/208@my.domain:_:user/211@my.domain :_:user/230@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@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@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@192.168.178.220<sip%3A06912345678@192.168.178.220> ;tag=a8m8ccQcgjjUg. To: <sip:200@192.168.178.50:3072;line=v3bii5l2>. Call-ID: b6cafa23-c959-122d-4682-080027e51f59. CSeq: 129888323 INVITE. Contact: <sip:mod_sofia@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@192.168.178.220 <sip%3A06912345678@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@bar.com
<mailto:foo@bar.com>,sofia/internal/foo2@bar.com:_:sofia/internal/ foo3@bar3.com <mailto:foo3@bar3.com>
On Wed, Apr 21, 2010 at 5:27 AM, David Ponzone <david.ponzone@gmail.com <mailto:david.ponzone@gmail.com>> wrote:
I think you should first thing update to latest GIT :)
David Ponzone Direction Technique email: david.ponzone@ipeva.fr <mailto:david.ponzone@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@mydomain.com <mailto:user/30@mydomain.com>:_:user/31@mydomain.com <mailto:user/31@mydomain.com>:_:user/32@mydomain.com <mailto:user/32@mydomain.com>:_:user/33@mydomain.com <mailto:user/33@mydomain.com>:_:user/34@mydomain.com <mailto:user/34@mydomain.com>"/>
Freeswitch tries to set it: EXECUTE sofia/external/069xxxxxxxx@10.xx.xx.1414 <mailto:sofia/external/069xxxxxxxx@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@10.xx.xx.1414 <mailto:sofia/external/069xxxxxxxx@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@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@10.xx.xx.141
;tag=5evr6508K9S3K.
To: <sip:31@10.xx.xx.14172:2048;line=hxbudrul>. Call-ID: ad359ddd-c7cc-122d-3683-001517c965a5. CSeq: 129803060 INVITE. Contact: <sip:mod_sofia@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@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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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> <mailto:MSN%3Aanthony_minessale@hotmail.com<MSN%253Aanthony_minessale@hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> <mailto:PAYPAL%3Aanthony.minessale@gmail.com<PAYPAL%253Aanthony.minessale@gmail.com>
IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> <mailto:sip%3A888@conference.freeswitch.org<sip%253A888@conference.freeswitch.org>
googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org<googletalk%253Aconf%252B888@conference.freeswitch.org>
pstn:+19193869900 ------------------------------------------------------------------------
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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@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
Thanks Anthony, origination_caller_id_name did the trick! Anthony Minessale schrieb:
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@gmx.net <mailto:Prometheus001@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@my.domain:_:user/201@my.domain:_:user/205@my.domain:_:user/208@my.domain:_:user/211@my.domain:_:user/230@my.domain"/> > > When the dialplan is executed, it seems to be processed correctly: > EXECUTE sofia/local/06912345678@192.168.178.218:5060 <http://06912345678@192.168.178.218:5060> > bridge(<effective_caller_id_name=My > Name>{global_to_originate_1=true}user/200@my.domain:_:user/201@my.domain:_:user/205@my.domain:_:user/208@my.domain:_:user/211@my.domain:_:user/230@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@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 <http://192.168.178.220:5060> -> 192.168.178.50:3072 <http://192.168.178.50:3072> > INVITE sip:200@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@192.168.178.220 <mailto:sip%3A06912345678@192.168.178.220>>;tag=a8m8ccQcgjjUg. > To: <sip:200@192.168.178.50:3072;line=v3bii5l2>. > Call-ID: b6cafa23-c959-122d-4682-080027e51f59. > CSeq: 129888323 INVITE. > Contact: <sip:mod_sofia@192.168.178.220:5060 <http://sip:mod_sofia@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@192.168.178.220 <mailto:sip%3A06912345678@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@bar.com <mailto:foo@bar.com> >> <mailto:foo@bar.com <mailto:foo@bar.com>>,sofia/internal/foo2@bar.com:_:sofia/internal/foo3@bar3.com <mailto:foo3@bar3.com> >> <mailto:foo3@bar3.com <mailto:foo3@bar3.com>> >> >> >> On Wed, Apr 21, 2010 at 5:27 AM, David Ponzone >> <david.ponzone@gmail.com <mailto:david.ponzone@gmail.com> <mailto:david.ponzone@gmail.com <mailto:david.ponzone@gmail.com>>> wrote: >> >> I think you should first thing update to latest GIT :) >> >> David Ponzone Direction Technique >> email: david.ponzone@ipeva.fr <mailto:david.ponzone@ipeva.fr> <mailto:david.ponzone@ipeva.fr <mailto:david.ponzone@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 <http://www.ipeva.fr> - www.ipeva-studio.com <http://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@mydomain.com <mailto:30@mydomain.com> >>> <mailto:user <mailto:user>/30@mydomain.com <mailto:30@mydomain.com>>:_:user/31@mydomain.com <mailto:31@mydomain.com> >>> <mailto:user <mailto:user>/31@mydomain.com <mailto:31@mydomain.com>>:_:user/32@mydomain.com <mailto:32@mydomain.com> >>> <mailto:user <mailto:user>/32@mydomain.com <mailto:32@mydomain.com>>:_:user/33@mydomain.com <mailto:33@mydomain.com> >>> <mailto:user <mailto:user>/33@mydomain.com <mailto:33@mydomain.com>>:_:user/34@mydomain.com <mailto:34@mydomain.com> >>> <mailto:user <mailto:user>/34@mydomain.com <mailto:34@mydomain.com>>"/> >>> >>> Freeswitch tries to set it: >>> EXECUTE sofia/external/069xxxxxxxx@10.xx.xx.1414 >>> <mailto:sofia <mailto:sofia>/external/069xxxxxxxx@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@10.xx.xx.1414 >>> <mailto:sofia <mailto:sofia>/external/069xxxxxxxx@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@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@10.xx.xx.141>;tag=5evr6508K9S3K. >>> To: <sip:31@10.xx.xx.14172:2048;line=hxbudrul>. >>> Call-ID: ad359ddd-c7cc-122d-3683-001517c965a5. >>> CSeq: 129803060 INVITE. >>> Contact: <sip:mod_sofia@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@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@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> >>> <mailto:FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@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@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> >> <mailto:FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@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@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> >> <mailto:MSN%3Aanthony_minessale@hotmail.com <mailto:MSN%253Aanthony_minessale@hotmail.com>> >> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> >> <mailto:PAYPAL%3Aanthony.minessale@gmail.com <mailto:PAYPAL%253Aanthony.minessale@gmail.com>> >> IRC: irc.freenode.net <http://irc.freenode.net> <http://irc.freenode.net> #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> >> <mailto:sip%3A888@conference.freeswitch.org <mailto:sip%253A888@conference.freeswitch.org>> >> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> >> <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org <mailto:googletalk%253Aconf%252B888@conference.freeswitch.org>> >> pstn:+19193869900 >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@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@lists.freeswitch.org <mailto:FreeSWITCH-users@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@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900 ------------------------------------------------------------------------
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participants (3)
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Anthony Minessale
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David Ponzone
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Peter P GMX