[Freeswitch-users] Problem about displayname of a routing call

Michael Collins msc at freeswitch.org
Tue Jun 2 21:59:10 PDT 2009


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 at 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 at 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 at 61.61.162.130:5060>;tag=124393762732run00
>    To: <sip:97730009 at 163.28.32.51:5070>
>    Contact: <sip:97719006 at 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 at 61.61.162.130:5060>;tag=124393762732run00
>    To: <sip:97730009 at 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 at 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 at 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 at 61.61.162.130:5060!
> send 1402 bytes to udp/[210.68.184.192]:62807 at 04:25:59.234375:
>    ------------------------------------------------------------------------
>    INVITE sip:97730009 at 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 at 203.64.215.209<sip%3A97730002 at 203.64.215.209>
> >;tag=F9rteQHjgS52m
>    To: <sip:97730009 at 210.68.184.192:62807;rinstance=89358e5ea9aaa80e>
>    Call-ID: 7aede8c6-ca99-122c-3f82-7d6e23531ace
>    CSeq: 115883243 INVITE
>    Contact: <sip:mod_sofia at 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 at 203.64.215.209<sip%3A97730002 at 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 at freeswitch.org>
>
>>
>>
>> On Tue, Jun 2, 2009 at 6:41 AM, Brad Tuan <brad.tuan at 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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>>
>>
>
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