here i am attaching debug trace of sip in case of sccessfull call when using register string...<br><br><br>*CLI> [May 12 19:21:14] <br><--- SIP read from <a href="http://192.168.0.254:5060">192.168.0.254:5060</a> ---><br>
INVITE <a href="mailto:sip%3A17185594743@nasir.server.com">sip:17185594743@nasir.server.com</a> SIP/2.0<br>Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK3c63f272;rport<br>Max-Forwards: 70<br>From: "caller" <<a href="mailto:sip%3A12129887777@192.168.0.254">sip:12129887777@192.168.0.254</a>>;tag=as76623e31<br>
To: <<a href="mailto:sip%3A17185594743@nasir.server.com">sip:17185594743@nasir.server.com</a>><br>Contact: <<a href="mailto:sip%3A12129887777@192.168.0.254">sip:12129887777@192.168.0.254</a>><br>Call-ID: <a href="mailto:245c407103141a6841c0ac106bd5a53d@192.168.0.254">245c407103141a6841c0ac106bd5a53d@192.168.0.254</a><br>
CSeq: 102 INVITE<br>User-Agent: Asterisk PBX 1.6.2.0<br>Date: Wed, 12 May 2010 14:20:25 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>Supported: replaces, timer<br>Content-Type: application/sdp<br>
Content-Length: 284<br><br>v=0<br>o=root 618893758 618893758 IN IP4 192.168.0.254<br>s=Asterisk PBX 1.6.2.0<br>c=IN IP4 192.168.0.254<br>t=0 0<br>m=audio 11026 RTP/AVP 0 3 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:3 GSM/8000<br>
a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br>a=ptime:20<br>a=sendrecv<br><br><-------------><br>[May 12 19:21:14] --- (14 headers 13 lines) ---<br>[May 12 19:21:14] Sending to 192.168.0.254 : 5060 (NAT)<br>
[May 12 19:21:14] Using INVITE request as basis request - <a href="mailto:245c407103141a6841c0ac106bd5a53d@192.168.0.254">245c407103141a6841c0ac106bd5a53d@192.168.0.254</a><br>[May 12 19:21:14] Found peer 'abc'<br>
[May 12 19:21:14] <br><--- Reliably Transmitting (NAT) to <a href="http://192.168.0.254:5060">192.168.0.254:5060</a> ---><br>SIP/2.0 407 Proxy Authentication Required<br>Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK3c63f272;received=192.168.0.254;rport=5060<br>
From: "caller" <<a href="mailto:sip%3A12129887777@192.168.0.254">sip:12129887777@192.168.0.254</a>>;tag=as76623e31<br>To: <<a href="mailto:sip%3A17185594743@nasir.server.com">sip:17185594743@nasir.server.com</a>>;tag=as0a721b3a<br>
Call-ID: <a href="mailto:245c407103141a6841c0ac106bd5a53d@192.168.0.254">245c407103141a6841c0ac106bd5a53d@192.168.0.254</a><br>CSeq: 102 INVITE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
Supported: replaces<br>Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7bc52d0a"<br>Content-Length: 0<br><br><br><------------><br>[May 12 19:21:14] Scheduling destruction of SIP dialog '<a href="mailto:245c407103141a6841c0ac106bd5a53d@192.168.0.254">245c407103141a6841c0ac106bd5a53d@192.168.0.254</a>' in 32000 ms (Method: INVITE)<br>
[May 12 19:21:14] <br><--- SIP read from <a href="http://192.168.0.254:5060">192.168.0.254:5060</a> ---><br>ACK <a href="mailto:sip%3A17185594743@nasir.server.com">sip:17185594743@nasir.server.com</a> SIP/2.0<br>Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK3c63f272;rport<br>
Max-Forwards: 70<br>From: "caller" <<a href="mailto:sip%3A12129887777@192.168.0.254">sip:12129887777@192.168.0.254</a>>;tag=as76623e31<br>To: <<a href="mailto:sip%3A17185594743@nasir.server.com">sip:17185594743@nasir.server.com</a>>;tag=as0a721b3a<br>
Contact: <<a href="mailto:sip%3A12129887777@192.168.0.254">sip:12129887777@192.168.0.254</a>><br>Call-ID: <a href="mailto:245c407103141a6841c0ac106bd5a53d@192.168.0.254">245c407103141a6841c0ac106bd5a53d@192.168.0.254</a><br>
CSeq: 102 ACK<br>User-Agent: Asterisk PBX 1.6.2.0<br>Content-Length: 0<br><br><br><-------------><br>[May 12 19:21:14] --- (10 headers 0 lines) ---<br>[May 12 19:21:14] <br><--- SIP read from <a href="http://192.168.0.254:5060">192.168.0.254:5060</a> ---><br>
INVITE <a href="mailto:sip%3A17185594743@nasir.server.com">sip:17185594743@nasir.server.com</a> SIP/2.0<br>Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK05611806;rport<br>Max-Forwards: 70<br>From: "caller" <<a href="mailto:sip%3A12129887777@192.168.0.254">sip:12129887777@192.168.0.254</a>>;tag=as76623e31<br>
To: <<a href="mailto:sip%3A17185594743@nasir.server.com">sip:17185594743@nasir.server.com</a>><br>Contact: <<a href="mailto:sip%3A12129887777@192.168.0.254">sip:12129887777@192.168.0.254</a>><br>Call-ID: <a href="mailto:245c407103141a6841c0ac106bd5a53d@192.168.0.254">245c407103141a6841c0ac106bd5a53d@192.168.0.254</a><br>
CSeq: 103 INVITE<br>User-Agent: Asterisk PBX 1.6.2.0<br>Proxy-Authorization: Digest username="abc", realm="asterisk", algorithm=MD5, uri="<a href="mailto:sip%3A17185594743@nasir.server.com">sip:17185594743@nasir.server.com</a>", nonce="7bc52d0a", response="f138ecd92bb706207a7b8d00f1c1bed7"<br>
Date: Wed, 12 May 2010 14:20:25 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>Supported: replaces, timer<br>Content-Type: application/sdp<br>Content-Length: 284<br><br>v=0<br>o=root 618893758 618893759 IN IP4 192.168.0.254<br>
s=Asterisk PBX 1.6.2.0<br>c=IN IP4 192.168.0.254<br>t=0 0<br>m=audio 11026 RTP/AVP 0 3 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:3 GSM/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br>
a=ptime:20<br>a=sendrecv<br><br><-------------><br>[May 12 19:21:14] --- (15 headers 13 lines) ---<br>[May 12 19:21:14] Sending to 192.168.0.254 : 5060 (NAT)<br>[May 12 19:21:14] Using INVITE request as basis request - <a href="mailto:245c407103141a6841c0ac106bd5a53d@192.168.0.254">245c407103141a6841c0ac106bd5a53d@192.168.0.254</a><br>
[May 12 19:21:14] Found peer 'abc'<br>[May 12 19:21:14] Found RTP audio format 0<br>[May 12 19:21:14] Found RTP audio format 3<br>[May 12 19:21:14] Found RTP audio format 101<br>[May 12 19:21:14] Peer audio RTP is at port <a href="http://192.168.0.254:11026">192.168.0.254:11026</a><br>
[May 12 19:21:14] Found description format PCMU for ID 0<br>[May 12 19:21:14] Found description format GSM for ID 3<br>[May 12 19:21:14] Found description format telephone-event for ID 101<br>[May 12 19:21:14] Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x6 (gsm|ulaw)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw)<br>
[May 12 19:21:14] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)<br>[May 12 19:21:14] Peer audio RTP is at port <a href="http://192.168.0.254:11026">192.168.0.254:11026</a><br>
[May 12 19:21:14] Looking for 17185594743 in payasyougo (domain <a href="http://nasir.server.com">nasir.server.com</a>)<br>[May 12 19:21:14] WARNING[3785]: chan_sip.c:3930 sip_new: setting callerid number to 12129339037<br>
[May 12 19:21:14] list_route: hop: <<a href="mailto:sip%3A12129887777@192.168.0.254">sip:12129887777@192.168.0.254</a>><br>[May 12 19:21:14] <br><--- Transmitting (NAT) to <a href="http://192.168.0.254:5060">192.168.0.254:5060</a> ---><br>
SIP/2.0 100 Trying<br>Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK05611806;received=192.168.0.254;rport=5060<br>From: "caller" <<a href="mailto:sip%3A12129887777@192.168.0.254">sip:12129887777@192.168.0.254</a>>;tag=as76623e31<br>
To: <<a href="mailto:sip%3A17185594743@nasir.server.com">sip:17185594743@nasir.server.com</a>><br>Call-ID: <a href="mailto:245c407103141a6841c0ac106bd5a53d@192.168.0.254">245c407103141a6841c0ac106bd5a53d@192.168.0.254</a><br>
CSeq: 103 INVITE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces<br>Contact: <<a href="mailto:sip%3A17185594743@nasir.server.com">sip:17185594743@nasir.server.com</a>><br>
Content-Length: 0<br><br><br><br><div class="gmail_quote">On Wed, May 12, 2010 at 7:14 PM, Nasir Javaid <span dir="ltr"><<a href="mailto:nasirjavaidnasir@gmail.com">nasirjavaidnasir@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Hi Vardan<br><br>I did same as you told and deleted the SIP information in Astdb and restarted asterisk. but the result was same.<br><br>as you said there might be mistake in sip.conf so i am pasting both servers configuration here..<br>
<br>1- <a href="http://nasir.server.com" target="_blank">nasir.server.com</a><br><br>[abc]<br>username=abc<br>type=friend<br>secret=mysecret<br>nat=yes<br>mailbox=12234568<br>incominglimit=2<br>outgoinglimit=2<br>host=dynamic<br>
dtmfmode=rfc2833<br>
<span style="color: rgb(51, 204, 0);">context=payasyougo</span><br>canreinvite=yes<br>callerid="Nasir Qazi" <12234><br>accountcode=6:0:abc<br>amaflags=default<div class="im"><br>disallow=all<br>allow=ulaw<br>
allow=alaw<br>
allow=g729<br>allow=gsm<br><br><br></div>2- 192.168.0.254 (client system)<div class="im"><br><br>[abc] <br>type=peer <br>username=abc<br>secret=mysecret<br></div>host=<a href="http://nasir.server.com" target="_blank">nasir.server.com</a><div class="im">
<br>context=default<br>dtmfmode=rfc2833<br>
canreinvite=yes <br></div>insecure=very <br><div class="im">disallow=all<br>allow=ulaw <br>allow=alaw<br>allow=g729<br>allow=gsm <br>nat=yes<br>;qualify=yes<br><br></div>[caller] <br>type=friend <br>secret=123456 <br>
host=dynamic <br>callerid="caller <12129887777>"<br>
context=out<br>nat=yes <br>dtmfmode=rfc2833<br>canreinvite=yes<br>insecure=no <br><div class="im">disallow=all<br>allow=ulaw <br>allow=alaw<br>allow=g729<br>allow=gsm<br></div>t38_udptl=yes <br>qualify=yes<br><br><br>I have registered [caller] on xlite at client system and dialing following context in local system that will dial [abc]<br>
<br>[out]<br>exten=> _X.,1,Dial(SIP/${EXTEN}@abc,30,1)<br>exten=> _X.,n,Hangup<br><br><br>as you can see above <b>highlighted that context of abc is payasyougo.</b> problem is that i want the call to land in that context on <a href="http://nasir.server.com" target="_blank">nasir.server.com</a>, which works if i use register string. but without register string call goes to default context on <a href="http://nasir.server.com" target="_blank">nasir.server.com</a><br>
<br>regards,<br><font color="#888888"><br>Nasir Javaid</font><div><div></div><div class="h5"><br><br>Message: 19<br>
Date: Tue, 11 May 2010 20:54:30 +0500<br>
From: Vardan <<a href="mailto:hvardan71@gmail.com" target="_blank">hvardan71@gmail.com</a>><br>
Subject: Re: [asterisk-users] asterisk-users Digest, Vol 70, Issue 24<br>
To: <a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a><br>
Message-ID: <hsbujk$qk9$<a href="mailto:1@dough.gmane.org" target="_blank">1@dough.gmane.org</a>><br>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed<br>
<br>
Hello Nasir<br>
<br>
I have some please.<br>
Do so, it help.<br>
Find all records about interexchange beetwen this two server and delete<br>
all records in sip.conf for this both server (first make backup<br>
sip.conf, or any another conf file that you use).<br>
restart asterisk.<br>
look in astdb about this old records, if any found, delete him<br>
Next, create new record in sip.conf on both servers, without<br>
registration string, reload sip.conf.<br>
give him right context from extensions.conf.<br>
<br>
Can you do this?<br>
<br>
I think is some mistake about configuration in sip.conf, you have I<br>
think two same records (peer or friend).<br>
<br>
Vardan
</div></div></blockquote></div><br>