[SR-Users] Caller ID issue

[Lucas Alvarez]

Hi Daniel-Constantin, thank for your quick response. This is the link to the
SIP trace:

http://www.euscorp.com/images/Wedpooi321989812j1DD9ddd9PaHw8XCa
<http://www.euscorp.com/images/Wedpooi321989812j1DD9ddd9PaHw8XCa>
I didn't send it through the list cause the body size needed approval.
The trace is a call from the extension 1090 to 1020. Kamailio is listening
at 192.168.15.11:5060 and asterisk at 192.168.15.11:5080. Additionally I
have pasted below a short CLI trace on asterisk showing up a NoOp with the
caller id followed by the dial and the first invite.
I really appreciate you help. Regards.

Lucas


CLI trace:


    -- Executing [1020 at longdistance:1] NoOp("SIP/1090-00000037", "Callerid
number: 1090      Name: Lucas Voice ") in new stack
    -- Executing [1020 at longdistance:2] Dial("SIP/1090-00000037", "SIP/1020")
in new stack
[Oct 12 10:44:26] DEBUG[17631]: chan_sip.c:3462 update_call_counter: Call to
peer '1020' is 1 out of 10
Audio is at 192.168.15.11 port 18106
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.15.11:5060:
INVITE sip:1020 at 192.168.15.11:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.15.11:5080;branch=z9hG4bK7c1bd27b;rport
From: "Lucas Voice" <sip:1020 at 192.168.15.11 <sip%3A1020 at 192.168.15.11>
>;tag=as1a1d0e0e
To: <sip:1020 at 192.168.15.11:5060>
Contact: <sip:1020 at 192.168.15.11:5080>
Call-ID: 7278984921bca2d55477817467d99103 at 192.168.15.11
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 12 Oct 2010 14:44:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 287






On Mon, Oct 11, 2010 at 7:36 PM, Daniel-Constantin Mierla <miconda at gmail.com
> wrote:

>  Hello,
>
>
> On 10/11/10 11:28 PM, Lucas Alvarez wrote:
>
>> Hi, I'm having a problem with the caller ID, I have implemented an
>> integration between asterisk and kamailio following this tutorial:
>> http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb
>> and the problem is that when I call from extension, let's say 1000, to
>> another extension, let's say 2000, the callerid number is always the
>> number I'm calling, in this case 2000. Using xlog and printing $fu,
>> $fU variables I realize that when the call came from asterisk to the
>> destination number,  kamailio changes the "From" headers. I will
>> appreciate any kind of help.
>> Regards.
>>
>>  can you take a SIP trace of such case on kamailio server? preferably with
> ngrep:
>
> ngrep -d any -qt -W byline port 5060
>
> Cheers,
> Daniel
>
> --
> Daniel-Constantin Mierla
> http://www.asipto.com
>
>
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