Hi folks!! I have an incoming call being bridged to an outgoing call ann i'm trying to pass the phone number of the incoming caller as the caller id on the leg B. But it doesn't work. Here is my dialplan: <extension name="incoming"> <condition field="destination_number" expression="^(3055908934)$"> <action application="info"/> <action application="set" data="effective_callee_id_number=${caller_id_number}"/> <action application="bridge" data="{sip_cid_type=rpid}sofia/gateway/xcast/17862065658"/> </condition> </extension> Here is the Log and SIP trace: http://pastebin.freeswitch.org/12368 This is the INVITE on the bLeg: INVITE sip:17862065658@wholesale.xcastlabs.com<sip%3A17862065658@wholesale.xcastlabs.com>SIP/2.0 Via: SIP/2.0/UDP 66.231.242.173:5080;rport;branch=z9hG4bKS6jFBtp9crtFp Max-Forwards: 68 From: "Javier" <sip:xcast@wholesale.xcastlabs.com<sip%3Axcast@wholesale.xcastlabs.com> ;transport=udp>;tag=K6U13Hy2rpgBe To: <sip:17862065658@wholesale.xcastlabs.com<sip%3A17862065658@wholesale.xcastlabs.com>
Call-ID: 0c42bf79-a59c-122d-b98c-0014384f7ec2 CSeq: 127923447 INVITE Contact: <sip:gw+xcast@66.231.242.173:5080;transport=udp;gw=xcast> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16921 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 250 X-FS-Support: update_display Remote-Party-ID: "Javier" <sip:3052606062@wholesale.xcastlabs.com<sip%3A3052606062@wholesale.xcastlabs.com>
;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1268060050 1268060051 IN IP4 66.231.242.173 s=FreeSWITCH c=IN IP4 66.231.242.173 t=0 0 m=audio 23516 RTP/AVP 18 101 13 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 And i'm on FreeSWITCH Version 1.0.trunk (16921) Many thanks /Javier
Try using the 'export' app rather than 'set'. Dan 2010/3/8 Javier Aristizábal <javieraristizabal@gmail.com>
Hi folks!! I have an incoming call being bridged to an outgoing call ann i'm trying to pass the phone number of the incoming caller as the caller id on the leg B. But it doesn't work.
Here is my dialplan:
<extension name="incoming"> <condition field="destination_number" expression="^(3055908934)$"> <action application="info"/> <action application="set" data="effective_callee_id_number=${caller_id_number}"/> <action application="bridge" data="{sip_cid_type=rpid}sofia/gateway/xcast/17862065658"/> </condition> </extension>
Here is the Log and SIP trace: http://pastebin.freeswitch.org/12368
This is the INVITE on the bLeg:
INVITE sip:17862065658@wholesale.xcastlabs.com<sip%3A17862065658@wholesale.xcastlabs.com>SIP/2.0 Via: SIP/2.0/UDP 66.231.242.173:5080;rport;branch=z9hG4bKS6jFBtp9crtFp Max-Forwards: 68 From: "Javier" <sip:xcast@wholesale.xcastlabs.com<sip%3Axcast@wholesale.xcastlabs.com> ;transport=udp>;tag=K6U13Hy2rpgBe To: <sip:17862065658@wholesale.xcastlabs.com<sip%3A17862065658@wholesale.xcastlabs.com>
Call-ID: 0c42bf79-a59c-122d-b98c-0014384f7ec2 CSeq: 127923447 INVITE Contact: <sip:gw+xcast@66.231.242.173:5080;transport=udp;gw=xcast> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16921 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 250 X-FS-Support: update_display Remote-Party-ID: "Javier" <sip:3052606062@wholesale.xcastlabs.com<sip%3A3052606062@wholesale.xcastlabs.com>
;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1268060050 1268060051 IN IP4 66.231.242.173 s=FreeSWITCH c=IN IP4 66.231.242.173 t=0 0 m=audio 23516 RTP/AVP 18 101 13 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20
And i'm on FreeSWITCH Version 1.0.trunk (16921)
Many thanks
/Javier
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HI Dan. Thanks. I did it with the same result. /Javier On Mon, Mar 8, 2010 at 4:50 PM, Dan Le <dule.maillist@gmail.com> wrote:
Try using the 'export' app rather than 'set'.
Dan
2010/3/8 Javier Aristizábal <javieraristizabal@gmail.com>
Hi folks!! I have an incoming call being bridged to an outgoing call ann i'm trying to pass the phone number of the incoming caller as the caller id on the leg B. But it doesn't work.
Here is my dialplan:
<extension name="incoming"> <condition field="destination_number" expression="^(3055908934)$"> <action application="info"/> <action application="set" data="effective_callee_id_number=${caller_id_number}"/> <action application="bridge" data="{sip_cid_type=rpid}sofia/gateway/xcast/17862065658"/> </condition> </extension>
Here is the Log and SIP trace: http://pastebin.freeswitch.org/12368
This is the INVITE on the bLeg:
INVITE sip:17862065658@wholesale.xcastlabs.com<sip%3A17862065658@wholesale.xcastlabs.com>SIP/2.0 Via: SIP/2.0/UDP 66.231.242.173:5080;rport;branch=z9hG4bKS6jFBtp9crtFp Max-Forwards: 68 From: "Javier" <sip:xcast@wholesale.xcastlabs.com<sip%3Axcast@wholesale.xcastlabs.com> ;transport=udp>;tag=K6U13Hy2rpgBe To: <sip:17862065658@wholesale.xcastlabs.com<sip%3A17862065658@wholesale.xcastlabs.com>
Call-ID: 0c42bf79-a59c-122d-b98c-0014384f7ec2 CSeq: 127923447 INVITE Contact: <sip:gw+xcast@66.231.242.173:5080;transport=udp;gw=xcast> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16921 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 250 X-FS-Support: update_display Remote-Party-ID: "Javier" <sip:3052606062@wholesale.xcastlabs.com<sip%3A3052606062@wholesale.xcastlabs.com>
;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1268060050 1268060051 IN IP4 66.231.242.173 s=FreeSWITCH c=IN IP4 66.231.242.173 t=0 0 m=audio 23516 RTP/AVP 18 101 13 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20
And i'm on FreeSWITCH Version 1.0.trunk (16921)
Many thanks
/Javier
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No we by default already pass it... no need to export vs set.. plus effective_caller_id_number/name is set on the A-LEG and auto exported for you to the B-LEG when created. His idiot provider wants it in the from field I suspect. /b On Mar 8, 2010, at 3:50 PM, Dan Le wrote:
Try using the 'export' app rather than 'set'.
Dan
chances are your provider wants the caller id in the from field. /b On Mar 8, 2010, at 3:38 PM, Javier Aristizábal wrote:
Hi folks!! I have an incoming call being bridged to an outgoing call ann i'm trying to pass the phone number of the incoming caller as the caller id on the leg B. But it doesn't work.
Hi Brian. Thanks, yeah it is possible. Is there a way to do that? /Javier On Mon, Mar 8, 2010 at 4:52 PM, Brian West <brian@freeswitch.org> wrote:
chances are your provider wants the caller id in the from field.
/b
On Mar 8, 2010, at 3:38 PM, Javier Aristizábal wrote:
Hi folks!! I have an incoming call being bridged to an outgoing call ann i'm trying to pass the phone number of the incoming caller as the caller id on the leg B. But it doesn't work.
_______________________________________________ FreeSWITCH-users mailing list 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
if all else fails try reading do_invite in sofia_glue.c you can really change just about ANYTHING to do with the invite with variables... be careful some let you totally break the packet where it won't even work. :P /b On Mar 8, 2010, at 3:59 PM, Javier Aristizábal wrote:
Hi Brian. Thanks, yeah it is possible. Is there a way to do that?
/Javier
Thanks Brian, let me try your ideas. /Javier On Mon, Mar 8, 2010 at 5:04 PM, Brian West <brian@freeswitch.org> wrote:
if all else fails try reading do_invite in sofia_glue.c you can really change just about ANYTHING to do with the invite with variables... be careful some let you totally break the packet where it won't even work. :P
/b
On Mar 8, 2010, at 3:59 PM, Javier Aristizábal wrote:
Hi Brian. Thanks, yeah it is possible. Is there a way to do that?
/Javier
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Brian. i did with the sip_from_uri, and it is working.. Many thanks /Javier 2010/3/8 Javier Aristizábal <javieraristizabal@gmail.com>
Thanks Brian, let me try your ideas.
/Javier
On Mon, Mar 8, 2010 at 5:04 PM, Brian West <brian@freeswitch.org> wrote:
if all else fails try reading do_invite in sofia_glue.c you can really change just about ANYTHING to do with the invite with variables... be careful some let you totally break the packet where it won't even work. :P
/b
On Mar 8, 2010, at 3:59 PM, Javier Aristizábal wrote:
Hi Brian. Thanks, yeah it is possible. Is there a way to do that?
/Javier
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2010/3/8 Javier Aristizábal <javieraristizabal@gmail.com>
Brian. i did with the sip_from_uri, and it is working..
Many thanks
/Javier
FYI, I couldn't find this specific tip in the wiki so I added it here: http://wiki.freeswitch.org/wiki/Variable_sip_from_uri#sip_from_uri -MC
Many Thanks Michael :) /Javier On Mon, Mar 8, 2010 at 5:35 PM, Michael Collins <msc@freeswitch.org> wrote:
2010/3/8 Javier Aristizábal <javieraristizabal@gmail.com>
Brian. i did with the sip_from_uri, and it is working..
Many thanks
/Javier
FYI,
I couldn't find this specific tip in the wiki so I added it here: http://wiki.freeswitch.org/wiki/Variable_sip_from_uri#sip_from_uri
-MC
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On Mon, Mar 8, 2010 at 1:52 PM, Brian West <brian@freeswitch.org> wrote:
chances are your provider wants the caller id in the from field.
/b
This sounds more likely. Javar tried the export already and that didn't help. What is the proper procedure for setting the From field? Can you just export sip_from_user when doing the bridge? -MC
On Mar 8, 2010, at 3:38 PM, Javier Aristizábal wrote:
Hi folks!! I have an incoming call being bridged to an outgoing call ann i'm trying to pass the phone number of the incoming caller as the caller id on the leg B. But it doesn't work.
_______________________________________________ FreeSWITCH-users mailing list 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
Try sip_from_uri /b On Mar 8, 2010, at 4:00 PM, Michael Collins wrote:
On Mon, Mar 8, 2010 at 1:52 PM, Brian West <brian@freeswitch.org> wrote: chances are your provider wants the caller id in the from field.
/b This sounds more likely. Javar tried the export already and that didn't help. What is the proper procedure for setting the From field? Can you just export sip_from_user when doing the bridge? -MC
participants (4)
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Brian West
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Dan Le
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Javier Aristizábal
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Michael Collins