[SR-Users] Caller ID issue

Daniel-Constantin Mierla miconda at gmail.com
Tue Oct 12 20:23:20 CEST 2010


  Hello,

the INVITE comes with that Caller ID set from Asterisk. It was very 
unlikely Kamailio changes it unless you use uac module.

I guess Asterisk in matching on source IP and port and happens to select 
another (pretty much randomly) caller id.

Try to use type=user in sipusers table.

Another option is to get the caller id from incoming invite to asterisk 
and set it for outgoing invite from asterisk.

Let me know if any of these works.

Cheers,
Daniel


On 10/12/10 5:14 PM, Lucas Alvarez wrote:
> Hi Daniel-Constantin, thank for your quick response. This is the link 
> to the SIP trace:
>
> 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 <http://192.168.15.11:5060/> and 
> asterisk at 192.168.15.11:5080 
> <http://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 
> <http://192.168.15.11:5060>:
> INVITE sip:1020 at 192.168.15.11:5060 
> <http://sip:1020@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 
> <mailto:sip%3A1020 at 192.168.15.11>>;tag=as1a1d0e0e
> To: <sip:1020 at 192.168.15.11:5060 <http://sip:1020@192.168.15.11:5060>>
> Contact: <sip:1020 at 192.168.15.11:5080 
> <http://sip:1020@192.168.15.11:5080>>
> Call-ID: 7278984921bca2d55477817467d99103 at 192.168.15.11 
> <mailto: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 <mailto: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
>
>

-- 
Daniel-Constantin Mierla
http://www.asipto.com

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20101012/ffe4b79c/attachment.htm>


More information about the sr-users mailing list