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 [1...@longdistance:1] NoOp("SIP/1090-00000037", "Callerid number: 1090 Name: Lucas Voice ") in new stack -- Executing [1...@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:1...@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:1...@192.168.15.11 <sip%3a1...@192.168.15.11> >;tag=as1a1d0e0e To: <sip:1...@192.168.15.11:5060> Contact: <sip:1...@192.168.15.11:5080> Call-ID: 7278984921bca2d55477817467d99...@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 <mico...@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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