Problem about displayname of a routing call
Why the Caller-Username is "97719006" but the Caller-Caller-ID-Name is "Extension 97730002"?? EVENT DUMP: Channel-State: [CS_ROUTING] Channel-State-Number: [2] Channel-Name: [sofia/internal/sip:97730001@210.68.184.192:62101 ;rinstance=16b8076934af7da9] Unique-ID: [342618e3-84cd-494b-b745-760b60639924] Call-Direction: [outbound] Answer-State: [ringing] Caller-Username: [97719006] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [Extension 97730002] Caller-Caller-ID-Number: [97730002] Caller-Network-Addr: [163.28.32.51] Caller-Destination-Number: [sip:97730001@210.68.184.192:62101 ;rinstance=16b80769 34af7da9]
Chances are that is what you set it to on the user. Verify the users settings in the directory. /b On Jun 2, 2009, at 6:28 AM, Brad Tuan wrote:
Why the Caller-Username is "97719006" but the Caller-Caller-ID-Name is "Extension 97730002"??
Brian West brian@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com
I don't have a 97719006 User in my FS. It was passed from another sip proxy. 2009/6/2 Brian West <brian@freeswitch.org>
Chances are that is what you set it to on the user. Verify the users settings in the directory. /b
On Jun 2, 2009, at 6:28 AM, Brad Tuan wrote:
Why the Caller-Username is "97719006" but the Caller-Caller-ID-Name is "Extension 97730002"??
Brian West brian@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
_______________________________________________ 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
Then it had to be passed in from the proxy. /b On Jun 2, 2009, at 8:11 AM, Brad Tuan wrote:
I don't have a 97719006 User in my FS.
It was passed from another sip proxy.
Brian West brian@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com
When send 100 Trying: From: 97719006 <sip:97719006@61.61.162.130:5060
;tag=124388224932run00 But when send INVITE: From: "Extension 97730002" <sip:97730002@203.64.215.209<sip%3A97730002@203.64.215.209> ;tag=U3QF8QUp1F3tQ What happened between sending Trying and sending INVITE ??
------------------------------------------------------------------------ send 556 bytes to udp/[163.28.32.51]:5070 at 13:03:05.218750: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 163.28.32.51:5070;branch=z9hG4bK5b64.83b3e3d3.0 Via: SIP/2.0/UDP 61.61.162.130:5060;branch=z9hG4bKrun12439477847278735900 Via: SIP/2.0/UDP 192.168.0.2:5260;branch=z9hG4bKrun124394778472787359 Record-Route: <sip:163.28.32.51:5070;lr=on;ftag=124388224932run00> From: 97719006 <sip:97719006@61.61.162.130:5060>;tag=124388224932run00 To: <sip:97730001@163.28.32.51:5070> Call-ID: i58YWNjMDU3ZWJhN2M1YzVlYjMzOTgxMjk4OWZiNTU0Yzc.00 CSeq: 7359 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.3-12163 Content-Length: 0 ------------------------------------------------------------------------ 2009-06-02 21:03:05 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/97719006@61.61.162.13 -b28d-fd1ecda44f18] 2009-06-02 21:03:05 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 97719006->97730001 in context default 2009-06-02 21:03:05 [INFO] switch_ivr_async.c:1730 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx 2009-06-02 21:03:05 [INFO] switch_ivr_async.c:1730 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 record_session::C:SipG -06-02-21-03-05.wav 2009-06-02 21:03:05 [INFO] switch_ivr_async.c:1730 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf 2009-06-02 21:03:05 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/sip:97730001@210.68.1 d6224d69efcf1 [e8ef86f8-e5aa-d246-8ffb-bf9e0f9dc160] send 1347 bytes to udp/[210.68.184.192]:62113 at 13:03:05.812500: ------------------------------------------------------------------------ INVITE sip:97730001@210.68.184.192:62113;rinstance=d0ed6224d69efcf1 SIP/2.0 Via: SIP/2.0/UDP 203.64.215.209;rport;branch=z9hG4bKg01ymegmSyp5c Max-Forwards: 67 From: "Extension 97730002" <sip:97730002@203.64.215.209<sip%3A97730002@203.64.215.209>
;tag=U3QF8QUp1F3tQ To: <sip:97730001@210.68.184.192:62113;rinstance=d0ed6224d69efcf1> Call-ID: 8dcc44c8-ca18-122c-2780-39a48cb53b8d CSeq: 115855556 INVITE
2009/6/2 Brad Tuan <brad.tuan@gmail.com>
I don't have a 97719006 User in my FS.
It was passed from another sip proxy.
2009/6/2 Brian West <brian@freeswitch.org>
Chances are that is what you set it to on the user. Verify the users settings in the directory. /b
On Jun 2, 2009, at 6:28 AM, Brad Tuan wrote:
Why the Caller-Username is "97719006" but the Caller-Caller-ID-Name is "Extension 97730002"??
Brian West brian@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
_______________________________________________ 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
I would update if I were you! :) Anyway something had to have changed it it won't magically do it. /b On Jun 2, 2009, at 8:28 AM, Brad Tuan wrote:
User-Agent: FreeSWITCH-mod_sofia/1.0.3-12163
Brian West brian@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com
So I need to new a User(97719006) in directory\default ?? 2009/6/2 Brian West <brian@freeswitch.org>
I would update if I were you! :) Anyway something had to have changed it it won't magically do it. /b
On Jun 2, 2009, at 8:28 AM, Brad Tuan wrote:
User-Agent: FreeSWITCH-mod_sofia/1.0.3-12163
Brian West brian@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
_______________________________________________ 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
How to update FreeSWITCH-mod_sofia/1.0.3-12163?? 2009/6/2 Brian West <brian@freeswitch.org>
I would update if I were you! :) Anyway something had to have changed it it won't magically do it. /b
On Jun 2, 2009, at 8:28 AM, Brad Tuan wrote:
User-Agent: FreeSWITCH-mod_sofia/1.0.3-12163
Brian West brian@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
_______________________________________________ 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
On Tue, Jun 2, 2009 at 6:41 AM, Brad Tuan <brad.tuan@gmail.com> wrote:
How to update FreeSWITCH-mod_sofia/1.0.3-12163??
Your best bet is to use SVN trunk. It is the most stable version available, even more stable than the latest 1.0.4pre8 release candidate. Back up your entire freeswitch folder in case there's an issue. Hopefully you're running in Linux, so you could do: mv /usr/local/freeswitch /usr/local/freeswitch.bak Then use the quick and dirty install from the wiki: http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install When the install is finished you will have a fresh copy of FS and a brand new default configuration. You'll need to go back and enable and build any modules you need that aren't done by default. You will also need to re-apply any changes you made to the default configuration from your previous install. Hopefully you didn't have to edit any of the files or maybe just a few, like vars.xml. In any case, I recommend editing as few of the default config files as possible. Let us know how it goes... -MC
......I've update my FS by SVN.......... but the User-Agent became to FreeSWITCH-mod_sofia/1.0.trunk-"UNKNOWN" Is that right?? And the displayname is still "97730002"....... What i confused is why "97730002" ?? ( I have users from 97730000~97739999,but when I call them from 97710006 , the display name is always "97730002"(it should be "97710006".....) ) recv 883 bytes from udp/[163.28.32.51]:5070 at 04:25:58.812500: ------------------------------------------------------------------------ INVITE sip:97730009@203.64.215.209:5060;transport=udp SIP/2.0 Record-Route: <sip:163.28.32.51:5070;lr=on;ftag=124393762732run00> Via: SIP/2.0/UDP 163.28.32.51:5070;branch=z9hG4bK27d.b8e8b5b.0 Via: SIP/2.0/UDP 61.61.162.130:5060;branch=z9hG4bKrun12440031628377850000 Via: SIP/2.0/UDP 192.168.0.2:5260;branch=z9hG4bKrun124400316283778500 From: 97719006 <sip:97719006@61.61.162.130:5060>;tag=124393762732run00 To: <sip:97730009@163.28.32.51:5070> Contact: <sip:97719006@61.61.162.130:5060> Call-ID: i99OGJlYTY4MDQ1NWY3YmUwMGUxZjkzZmRjZTRmNWM0ODU.00 CSeq: 8500 INVITE Max-Forwards: 68 Content-Type: application/sdp Content-Length: 237 v=0 o=169 0 0 IN IP4 61.61.162.130 s=ots c=IN IP4 61.61.162.130 t=0 0 m=audio 5158 RTP/AVP 18 8 0 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:18 G729/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 ------------------------------------------------------------------------ send 560 bytes to udp/[163.28.32.51]:5070 at 04:25:58.859375: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 163.28.32.51:5070;branch=z9hG4bK27d.b8e8b5b.0 Via: SIP/2.0/UDP 61.61.162.130:5060;branch=z9hG4bKrun12440031628377850000 Via: SIP/2.0/UDP 192.168.0.2:5260;branch=z9hG4bKrun124400316283778500 Record-Route: <sip:163.28.32.51:5070;lr=on;ftag=124393762732run00> From: 97719006 <sip:97719006@61.61.162.130:5060>;tag=124393762732run00 To: <sip:97730009@163.28.32.51:5070> Call-ID: i99OGJlYTY4MDQ1NWY3YmUwMGUxZjkzZmRjZTRmNWM0ODU.00 CSeq: 8500 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Content-Length: 0 ------------------------------------------------------------------------ 2009-06-03 12:25:58 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel sofia/internal/97719006@61.61.162.130:5060[4caf176f-efdd-2a4d-99c9-cc62 f34cc3da] 2009-06-03 12:25:59 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing 977 19006->97730009 in context default 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 switch_ivr_bind_dtmf_meta_ses sion() Bound B-Leg: 1 execute_extension::dx XML features 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 switch_ivr_bind_dtmf_meta_ses sion() Bound B-Leg: 2 record_session::C:\SipGo/recordings/97719006.2009-06-03-12 -25-59.wav 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 switch_ivr_bind_dtmf_meta_ses sion() Bound B-Leg: 3 execute_extension::cf XML features 2009-06-03 12:25:59 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel sofia/internal/sip:97730009@210.68.184.192:62807 ;rinstance=89358e5ea9aaa 80e [86bb74ab-c3c8-c24d-b5c6-0707423f93d1] 2009-06-03 12:25:59 [INFO] switch_ivr_originate.c:1664 switch_ivr_originate() Se nding early media 2009-06-03 12:25:59 [INFO] mod_sofia.c:1487 sofia_receive_message() Ring SDP: v=0 o=FreeSWITCH 1243986137 1243986138 IN IP4 203.64.215.209 s=FreeSWITCH c=IN IP4 203.64.215.209 t=0 0 m=audio 17022 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-06-03 12:25:59 [NOTICE] mod_sofia.c:1490 sofia_receive_message() Pre-Answer sofia/internal/97719006@61.61.162.130:5060! send 1402 bytes to udp/[210.68.184.192]:62807 at 04:25:59.234375: ------------------------------------------------------------------------ INVITE sip:97730009@210.68.184.192:62807;rinstance=89358e5ea9aaa80e SIP/2.0 Via: SIP/2.0/UDP 203.64.215.209;rport;branch=z9hG4bKyX6NyKUygQ9ta Max-Forwards: 67 From: "Extension 97730002" <sip:97730002@203.64.215.209<sip%3A97730002@203.64.215.209>
;tag=F9rteQHjgS52m To: <sip:97730009@210.68.184.192:62807;rinstance=89358e5ea9aaa80e> Call-ID: 7aede8c6-ca99-122c-3f82-7d6e23531ace CSeq: 115883243 INVITE Contact: <sip:mod_sofia@203.64.215.209:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-descrip tion, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 449 Remote-Party-ID: "Extension 97730002" <sip:97730002@203.64.215.209<sip%3A97730002@203.64.215.209> ;party=cal ling;screen=yes;privacy=off v=0 o=FreeSWITCH 1293924755044965733 4031320830041280168 IN IP4 203.64.215.209 s=FreeSWITCH c=IN IP4 203.64.215.209 t=0 0 m=audio 21978 RTP/AVP 8 115 107 9 0 3 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:107 G7221/16000 a=fmtp:107 bitrate=32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/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
2009/6/3 Michael Collins <msc@freeswitch.org>
On Tue, Jun 2, 2009 at 6:41 AM, Brad Tuan <brad.tuan@gmail.com> wrote:
How to update FreeSWITCH-mod_sofia/1.0.3-12163??
Your best bet is to use SVN trunk. It is the most stable version available, even more stable than the latest 1.0.4pre8 release candidate. Back up your entire freeswitch folder in case there's an issue. Hopefully you're running in Linux, so you could do: mv /usr/local/freeswitch /usr/local/freeswitch.bak
Then use the quick and dirty install from the wiki: http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install
When the install is finished you will have a fresh copy of FS and a brand new default configuration. You'll need to go back and enable and build any modules you need that aren't done by default. You will also need to re-apply any changes you made to the default configuration from your previous install. Hopefully you didn't have to edit any of the files or maybe just a few, like vars.xml. In any case, I recommend editing as few of the default config files as possible.
Let us know how it goes... -MC
_______________________________________________ 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
okay, you will need to use pastebin and post your configuration. anything you changed from the default config, especially in the dialplan, but also vars.xml, sip_profiles, etc. Turn on debug level logging (F8 or "console loglevel 7") and also do the SIP trace. Make a few test calls and capture all the output. -MC On Tue, Jun 2, 2009 at 9:38 PM, Brad Tuan <brad.tuan@gmail.com> wrote:
......I've update my FS by SVN..........
but the User-Agent became to FreeSWITCH-mod_sofia/1.0.trunk-"UNKNOWN"
Is that right??
And the displayname is still "97730002".......
What i confused is why "97730002" ??
( I have users from 97730000~97739999,but when I call them from 97710006 , the display name is always "97730002"(it should be "97710006".....) )
recv 883 bytes from udp/[163.28.32.51]:5070 at 04:25:58.812500: ------------------------------------------------------------------------ INVITE sip:97730009@203.64.215.209:5060;transport=udp SIP/2.0 Record-Route: <sip:163.28.32.51:5070;lr=on;ftag=124393762732run00> Via: SIP/2.0/UDP 163.28.32.51:5070;branch=z9hG4bK27d.b8e8b5b.0 Via: SIP/2.0/UDP 61.61.162.130:5060 ;branch=z9hG4bKrun12440031628377850000 Via: SIP/2.0/UDP 192.168.0.2:5260;branch=z9hG4bKrun124400316283778500 From: 97719006 <sip:97719006@61.61.162.130:5060>;tag=124393762732run00 To: <sip:97730009@163.28.32.51:5070> Contact: <sip:97719006@61.61.162.130:5060> Call-ID: i99OGJlYTY4MDQ1NWY3YmUwMGUxZjkzZmRjZTRmNWM0ODU.00 CSeq: 8500 INVITE Max-Forwards: 68 Content-Type: application/sdp Content-Length: 237 v=0 o=169 0 0 IN IP4 61.61.162.130 s=ots c=IN IP4 61.61.162.130 t=0 0 m=audio 5158 RTP/AVP 18 8 0 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:18 G729/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 ------------------------------------------------------------------------ send 560 bytes to udp/[163.28.32.51]:5070 at 04:25:58.859375: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 163.28.32.51:5070;branch=z9hG4bK27d.b8e8b5b.0 Via: SIP/2.0/UDP 61.61.162.130:5060 ;branch=z9hG4bKrun12440031628377850000 Via: SIP/2.0/UDP 192.168.0.2:5260;branch=z9hG4bKrun124400316283778500 Record-Route: <sip:163.28.32.51:5070;lr=on;ftag=124393762732run00> From: 97719006 <sip:97719006@61.61.162.130:5060>;tag=124393762732run00 To: <sip:97730009@163.28.32.51:5070> Call-ID: i99OGJlYTY4MDQ1NWY3YmUwMGUxZjkzZmRjZTRmNWM0ODU.00 CSeq: 8500 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Content-Length: 0 ------------------------------------------------------------------------ 2009-06-03 12:25:58 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel sofia/internal/97719006@61.61.162.130:5060[4caf176f-efdd-2a4d-99c9-cc62 f34cc3da] 2009-06-03 12:25:59 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing 977 19006->97730009 in context default 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 switch_ivr_bind_dtmf_meta_ses sion() Bound B-Leg: 1 execute_extension::dx XML features 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 switch_ivr_bind_dtmf_meta_ses sion() Bound B-Leg: 2 record_session::C:\SipGo/recordings/97719006.2009-06-03-12 -25-59.wav 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 switch_ivr_bind_dtmf_meta_ses sion() Bound B-Leg: 3 execute_extension::cf XML features 2009-06-03 12:25:59 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel sofia/internal/sip:97730009@210.68.184.192:62807 ;rinstance=89358e5ea9aaa 80e [86bb74ab-c3c8-c24d-b5c6-0707423f93d1] 2009-06-03 12:25:59 [INFO] switch_ivr_originate.c:1664 switch_ivr_originate() Se nding early media 2009-06-03 12:25:59 [INFO] mod_sofia.c:1487 sofia_receive_message() Ring SDP: v=0 o=FreeSWITCH 1243986137 1243986138 IN IP4 203.64.215.209 s=FreeSWITCH c=IN IP4 203.64.215.209 t=0 0 m=audio 17022 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-06-03 12:25:59 [NOTICE] mod_sofia.c:1490 sofia_receive_message() Pre-Answer sofia/internal/97719006@61.61.162.130:5060! send 1402 bytes to udp/[210.68.184.192]:62807 at 04:25:59.234375: ------------------------------------------------------------------------ INVITE sip:97730009@210.68.184.192:62807;rinstance=89358e5ea9aaa80e SIP/2.0 Via: SIP/2.0/UDP 203.64.215.209;rport;branch=z9hG4bKyX6NyKUygQ9ta Max-Forwards: 67 From: "Extension 97730002" <sip:97730002@203.64.215.209<sip%3A97730002@203.64.215.209>
;tag=F9rteQHjgS52m To: <sip:97730009@210.68.184.192:62807;rinstance=89358e5ea9aaa80e> Call-ID: 7aede8c6-ca99-122c-3f82-7d6e23531ace CSeq: 115883243 INVITE Contact: <sip:mod_sofia@203.64.215.209:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-descrip tion, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 449 Remote-Party-ID: "Extension 97730002" <sip:97730002@203.64.215.209<sip%3A97730002@203.64.215.209> ;party=cal ling;screen=yes;privacy=off v=0 o=FreeSWITCH 1293924755044965733 4031320830041280168 IN IP4 203.64.215.209 s=FreeSWITCH c=IN IP4 203.64.215.209 t=0 0 m=audio 21978 RTP/AVP 8 115 107 9 0 3 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:107 G7221/16000 a=fmtp:107 bitrate=32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/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
2009/6/3 Michael Collins <msc@freeswitch.org>
On Tue, Jun 2, 2009 at 6:41 AM, Brad Tuan <brad.tuan@gmail.com> wrote:
How to update FreeSWITCH-mod_sofia/1.0.3-12163??
Your best bet is to use SVN trunk. It is the most stable version available, even more stable than the latest 1.0.4pre8 release candidate. Back up your entire freeswitch folder in case there's an issue. Hopefully you're running in Linux, so you could do: mv /usr/local/freeswitch /usr/local/freeswitch.bak
Then use the quick and dirty install from the wiki: http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install
When the install is finished you will have a fresh copy of FS and a brand new default configuration. You'll need to go back and enable and build any modules you need that aren't done by default. You will also need to re-apply any changes you made to the default configuration from your previous install. Hopefully you didn't have to edit any of the files or maybe just a few, like vars.xml. In any case, I recommend editing as few of the default config files as possible.
Let us know how it goes... -MC
_______________________________________________ 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
_______________________________________________ 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
I only change freeSWITCH\conf\dialplan\default.xml <extension name="Local_Extension"> <condition field="destination_number" expression="^(9773\d{4} | 10[01][0-9])$"> and add user xml from 97730000~97739999 in freeSWITCH\conf\directory\default <include> <user id="97730000"> <params> <param name="password" value="$${default_password}"/> <param name="vm-password" value="97730000"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="97730000"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 97730000"/> <variable name="effective_caller_id_number" value="97730000"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="techsupport"/> </variables> </user> </include> 2009/6/3 Michael Collins <msc@freeswitch.org>
okay, you will need to use pastebin and post your configuration. anything you changed from the default config, especially in the dialplan, but also vars.xml, sip_profiles, etc. Turn on debug level logging (F8 or "console loglevel 7") and also do the SIP trace. Make a few test calls and capture all the output.
-MC
On Tue, Jun 2, 2009 at 9:38 PM, Brad Tuan <brad.tuan@gmail.com> wrote:
......I've update my FS by SVN..........
but the User-Agent became to FreeSWITCH-mod_sofia/1.0.trunk-"UNKNOWN"
Is that right??
And the displayname is still "97730002".......
What i confused is why "97730002" ??
( I have users from 97730000~97739999,but when I call them from 97710006 , the display name is always "97730002"(it should be "97710006".....) )
recv 883 bytes from udp/[163.28.32.51]:5070 at 04:25:58.812500:
------------------------------------------------------------------------ INVITE sip:97730009@203.64.215.209:5060;transport=udp SIP/2.0 Record-Route: <sip:163.28.32.51:5070;lr=on;ftag=124393762732run00> Via: SIP/2.0/UDP 163.28.32.51:5070;branch=z9hG4bK27d.b8e8b5b.0 Via: SIP/2.0/UDP 61.61.162.130:5060 ;branch=z9hG4bKrun12440031628377850000 Via: SIP/2.0/UDP 192.168.0.2:5260;branch=z9hG4bKrun124400316283778500 From: 97719006 <sip:97719006@61.61.162.130:5060>;tag=124393762732run00 To: <sip:97730009@163.28.32.51:5070> Contact: <sip:97719006@61.61.162.130:5060> Call-ID: i99OGJlYTY4MDQ1NWY3YmUwMGUxZjkzZmRjZTRmNWM0ODU.00 CSeq: 8500 INVITE Max-Forwards: 68 Content-Type: application/sdp Content-Length: 237 v=0 o=169 0 0 IN IP4 61.61.162.130 s=ots c=IN IP4 61.61.162.130 t=0 0 m=audio 5158 RTP/AVP 18 8 0 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:18 G729/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15
------------------------------------------------------------------------ send 560 bytes to udp/[163.28.32.51]:5070 at 04:25:58.859375:
------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 163.28.32.51:5070;branch=z9hG4bK27d.b8e8b5b.0 Via: SIP/2.0/UDP 61.61.162.130:5060 ;branch=z9hG4bKrun12440031628377850000 Via: SIP/2.0/UDP 192.168.0.2:5260;branch=z9hG4bKrun124400316283778500 Record-Route: <sip:163.28.32.51:5070;lr=on;ftag=124393762732run00> From: 97719006 <sip:97719006@61.61.162.130:5060>;tag=124393762732run00 To: <sip:97730009@163.28.32.51:5070> Call-ID: i99OGJlYTY4MDQ1NWY3YmUwMGUxZjkzZmRjZTRmNWM0ODU.00 CSeq: 8500 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Content-Length: 0
------------------------------------------------------------------------ 2009-06-03 12:25:58 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel sofia/internal/97719006@61.61.162.130:5060[4caf176f-efdd-2a4d-99c9-cc62 f34cc3da] 2009-06-03 12:25:59 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing 977 19006->97730009 in context default 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 switch_ivr_bind_dtmf_meta_ses sion() Bound B-Leg: 1 execute_extension::dx XML features 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 switch_ivr_bind_dtmf_meta_ses sion() Bound B-Leg: 2 record_session::C:\SipGo/recordings/97719006.2009-06-03-12 -25-59.wav 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 switch_ivr_bind_dtmf_meta_ses sion() Bound B-Leg: 3 execute_extension::cf XML features 2009-06-03 12:25:59 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel sofia/internal/sip:97730009@210.68.184.192:62807 ;rinstance=89358e5ea9aaa 80e [86bb74ab-c3c8-c24d-b5c6-0707423f93d1] 2009-06-03 12:25:59 [INFO] switch_ivr_originate.c:1664 switch_ivr_originate() Se nding early media 2009-06-03 12:25:59 [INFO] mod_sofia.c:1487 sofia_receive_message() Ring SDP: v=0 o=FreeSWITCH 1243986137 1243986138 IN IP4 203.64.215.209 s=FreeSWITCH c=IN IP4 203.64.215.209 t=0 0 m=audio 17022 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-06-03 12:25:59 [NOTICE] mod_sofia.c:1490 sofia_receive_message() Pre-Answer sofia/internal/97719006@61.61.162.130:5060! send 1402 bytes to udp/[210.68.184.192]:62807 at 04:25:59.234375:
------------------------------------------------------------------------ INVITE sip:97730009@210.68.184.192:62807;rinstance=89358e5ea9aaa80e SIP/2.0 Via: SIP/2.0/UDP 203.64.215.209;rport;branch=z9hG4bKyX6NyKUygQ9ta Max-Forwards: 67 From: "Extension 97730002" <sip:97730002@203.64.215.209<sip%3A97730002@203.64.215.209>
;tag=F9rteQHjgS52m To: <sip:97730009@210.68.184.192:62807;rinstance=89358e5ea9aaa80e> Call-ID: 7aede8c6-ca99-122c-3f82-7d6e23531ace CSeq: 115883243 INVITE Contact: <sip:mod_sofia@203.64.215.209:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-descrip tion, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 449 Remote-Party-ID: "Extension 97730002" <sip:97730002@203.64.215.209<sip%3A97730002@203.64.215.209> ;party=cal ling;screen=yes;privacy=off v=0 o=FreeSWITCH 1293924755044965733 4031320830041280168 IN IP4 203.64.215.209 s=FreeSWITCH c=IN IP4 203.64.215.209 t=0 0 m=audio 21978 RTP/AVP 8 115 107 9 0 3 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:107 G7221/16000 a=fmtp:107 bitrate=32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/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
2009/6/3 Michael Collins <msc@freeswitch.org>
On Tue, Jun 2, 2009 at 6:41 AM, Brad Tuan <brad.tuan@gmail.com> wrote:
How to update FreeSWITCH-mod_sofia/1.0.3-12163??
Your best bet is to use SVN trunk. It is the most stable version available, even more stable than the latest 1.0.4pre8 release candidate. Back up your entire freeswitch folder in case there's an issue. Hopefully you're running in Linux, so you could do: mv /usr/local/freeswitch /usr/local/freeswitch.bak
Then use the quick and dirty install from the wiki: http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install
When the install is finished you will have a fresh copy of FS and a brand new default configuration. You'll need to go back and enable and build any modules you need that aren't done by default. You will also need to re-apply any changes you made to the default configuration from your previous install. Hopefully you didn't have to edit any of the files or maybe just a few, like vars.xml. In any case, I recommend editing as few of the default config files as possible.
Let us know how it goes... -MC
_______________________________________________ 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
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I only change freeSWITCH\conf\dialplan\default.xml <extension name="Local_Extension"> <condition field="destination_number" expression="^(9773\d{4} | 10[01][0-9])$"> and add user xml from 97730000~97739999 in freeSWITCH\conf\directory\default <include> <user id="97730000"> <params> <param name="password" value="$${default_password}"/> <param name="vm-password" value="97730000"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="97730000"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 97730000"/> <variable name="effective_caller_id_number" value="97730000"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="techsupport"/> </variables> </user> </include> 2009/6/3 Michael Collins <msc@freeswitch.org>
okay, you will need to use pastebin and post your configuration. anything you changed from the default config, especially in the dialplan, but also vars.xml, sip_profiles, etc. Turn on debug level logging (F8 or "console loglevel 7") and also do the SIP trace. Make a few test calls and capture all the output.
-MC
On Tue, Jun 2, 2009 at 9:38 PM, Brad Tuan <brad.tuan@gmail.com> wrote:
......I've update my FS by SVN..........
but the User-Agent became to FreeSWITCH-mod_sofia/1.0.trunk-"UNKNOWN"
Is that right??
And the displayname is still "97730002".......
What i confused is why "97730002" ??
( I have users from 97730000~97739999,but when I call them from 97710006 , the display name is always "97730002"(it should be "97710006".....) )
recv 883 bytes from udp/[163.28.32.51]:5070 at 04:25:58.812500:
------------------------------------------------------------------------ INVITE sip:97730009@203.64.215.209:5060;transport=udp SIP/2.0 Record-Route: <sip:163.28.32.51:5070;lr=on;ftag=124393762732run00> Via: SIP/2.0/UDP 163.28.32.51:5070;branch=z9hG4bK27d.b8e8b5b.0 Via: SIP/2.0/UDP 61.61.162.130:5060 ;branch=z9hG4bKrun12440031628377850000 Via: SIP/2.0/UDP 192.168.0.2:5260;branch=z9hG4bKrun124400316283778500 From: 97719006 <sip:97719006@61.61.162.130:5060>;tag=124393762732run00 To: <sip:97730009@163.28.32.51:5070> Contact: <sip:97719006@61.61.162.130:5060> Call-ID: i99OGJlYTY4MDQ1NWY3YmUwMGUxZjkzZmRjZTRmNWM0ODU.00 CSeq: 8500 INVITE Max-Forwards: 68 Content-Type: application/sdp Content-Length: 237 v=0 o=169 0 0 IN IP4 61.61.162.130 s=ots c=IN IP4 61.61.162.130 t=0 0 m=audio 5158 RTP/AVP 18 8 0 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:18 G729/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15
------------------------------------------------------------------------ send 560 bytes to udp/[163.28.32.51]:5070 at 04:25:58.859375:
------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 163.28.32.51:5070;branch=z9hG4bK27d.b8e8b5b.0 Via: SIP/2.0/UDP 61.61.162.130:5060 ;branch=z9hG4bKrun12440031628377850000 Via: SIP/2.0/UDP 192.168.0.2:5260;branch=z9hG4bKrun124400316283778500 Record-Route: <sip:163.28.32.51:5070;lr=on;ftag=124393762732run00> From: 97719006 <sip:97719006@61.61.162.130:5060>;tag=124393762732run00 To: <sip:97730009@163.28.32.51:5070> Call-ID: i99OGJlYTY4MDQ1NWY3YmUwMGUxZjkzZmRjZTRmNWM0ODU.00 CSeq: 8500 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Content-Length: 0
------------------------------------------------------------------------ 2009-06-03 12:25:58 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel sofia/internal/97719006@61.61.162.130:5060[4caf176f-efdd-2a4d-99c9-cc62 f34cc3da] 2009-06-03 12:25:59 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing 977 19006->97730009 in context default 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 switch_ivr_bind_dtmf_meta_ses sion() Bound B-Leg: 1 execute_extension::dx XML features 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 switch_ivr_bind_dtmf_meta_ses sion() Bound B-Leg: 2 record_session::C:\SipGo/recordings/97719006.2009-06-03-12 -25-59.wav 2009-06-03 12:25:59 [INFO] switch_ivr_async.c:1770 switch_ivr_bind_dtmf_meta_ses sion() Bound B-Leg: 3 execute_extension::cf XML features 2009-06-03 12:25:59 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel sofia/internal/sip:97730009@210.68.184.192:62807 ;rinstance=89358e5ea9aaa 80e [86bb74ab-c3c8-c24d-b5c6-0707423f93d1] 2009-06-03 12:25:59 [INFO] switch_ivr_originate.c:1664 switch_ivr_originate() Se nding early media 2009-06-03 12:25:59 [INFO] mod_sofia.c:1487 sofia_receive_message() Ring SDP: v=0 o=FreeSWITCH 1243986137 1243986138 IN IP4 203.64.215.209 s=FreeSWITCH c=IN IP4 203.64.215.209 t=0 0 m=audio 17022 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-06-03 12:25:59 [NOTICE] mod_sofia.c:1490 sofia_receive_message() Pre-Answer sofia/internal/97719006@61.61.162.130:5060! send 1402 bytes to udp/[210.68.184.192]:62807 at 04:25:59.234375:
------------------------------------------------------------------------ INVITE sip:97730009@210.68.184.192:62807;rinstance=89358e5ea9aaa80e SIP/2.0 Via: SIP/2.0/UDP 203.64.215.209;rport;branch=z9hG4bKyX6NyKUygQ9ta Max-Forwards: 67 From: "Extension 97730002" <sip:97730002@203.64.215.209<sip%3A97730002@203.64.215.209>
;tag=F9rteQHjgS52m To: <sip:97730009@210.68.184.192:62807;rinstance=89358e5ea9aaa80e> Call-ID: 7aede8c6-ca99-122c-3f82-7d6e23531ace CSeq: 115883243 INVITE Contact: <sip:mod_sofia@203.64.215.209:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-descrip tion, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 449 Remote-Party-ID: "Extension 97730002" <sip:97730002@203.64.215.209<sip%3A97730002@203.64.215.209> ;party=cal ling;screen=yes;privacy=off v=0 o=FreeSWITCH 1293924755044965733 4031320830041280168 IN IP4 203.64.215.209 s=FreeSWITCH c=IN IP4 203.64.215.209 t=0 0 m=audio 21978 RTP/AVP 8 115 107 9 0 3 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:107 G7221/16000 a=fmtp:107 bitrate=32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/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
2009/6/3 Michael Collins <msc@freeswitch.org>
On Tue, Jun 2, 2009 at 6:41 AM, Brad Tuan <brad.tuan@gmail.com> wrote:
How to update FreeSWITCH-mod_sofia/1.0.3-12163??
Your best bet is to use SVN trunk. It is the most stable version available, even more stable than the latest 1.0.4pre8 release candidate. Back up your entire freeswitch folder in case there's an issue. Hopefully you're running in Linux, so you could do: mv /usr/local/freeswitch /usr/local/freeswitch.bak
Then use the quick and dirty install from the wiki: http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install
When the install is finished you will have a fresh copy of FS and a brand new default configuration. You'll need to go back and enable and build any modules you need that aren't done by default. You will also need to re-apply any changes you made to the default configuration from your previous install. Hopefully you didn't have to edit any of the files or maybe just a few, like vars.xml. In any case, I recommend editing as few of the default config files as possible.
Let us know how it goes... -MC
_______________________________________________ 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
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also press f8 before you take the console log to get the debugging info and paste the resulting trace in http://pastebin.freeswitch.org rather than right in the email -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
I know why the display name is wrong.......... in conf\directory\97730002.xml <include> <user id="97730002" mailbox="97730002" cidr="163.28.32.51/32"> I forgot this setting........... but if I don't set cidr here ,the call from 163.28.32.51 can't come into my FS. How to make some setting for this?? 2009/6/3 Anthony Minessale <anthony.minessale@gmail.com>
also press f8 before you take the console log to get the debugging info and paste the resulting trace in http://pastebin.freeswitch.org rather than right in the email
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
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> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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participants (4)
-
Anthony Minessale
-
Brad Tuan
-
Brian West
-
Michael Collins