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>;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>;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

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