here i am attaching debug trace of sip in case of sccessfull call when using register string...
*CLI> [May 12 19:21:14] <--- SIP read from 192.168.0.254:5060 ---> INVITE sip:17185594...@nasir.server.com <sip%3a17185594...@nasir.server.com>SIP/2.0 Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK3c63f272;rport Max-Forwards: 70 From: "caller" <sip:12129887...@192.168.0.254<sip%3a12129887...@192.168.0.254> >;tag=as76623e31 To: <sip:17185594...@nasir.server.com <sip%3a17185594...@nasir.server.com>> Contact: <sip:12129887...@192.168.0.254 <sip%3a12129887...@192.168.0.254>> Call-ID: 245c407103141a6841c0ac106bd5a...@192.168.0.254 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.0 Date: Wed, 12 May 2010 14:20:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 284 v=0 o=root 618893758 618893758 IN IP4 192.168.0.254 s=Asterisk PBX 1.6.2.0 c=IN IP4 192.168.0.254 t=0 0 m=audio 11026 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [May 12 19:21:14] --- (14 headers 13 lines) --- [May 12 19:21:14] Sending to 192.168.0.254 : 5060 (NAT) [May 12 19:21:14] Using INVITE request as basis request - 245c407103141a6841c0ac106bd5a...@192.168.0.254 [May 12 19:21:14] Found peer 'abc' [May 12 19:21:14] <--- Reliably Transmitting (NAT) to 192.168.0.254:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.254:5060 ;branch=z9hG4bK3c63f272;received=192.168.0.254;rport=5060 From: "caller" <sip:12129887...@192.168.0.254<sip%3a12129887...@192.168.0.254> >;tag=as76623e31 To: <sip:17185594...@nasir.server.com <sip%3a17185594...@nasir.server.com> >;tag=as0a721b3a Call-ID: 245c407103141a6841c0ac106bd5a...@192.168.0.254 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7bc52d0a" Content-Length: 0 <------------> [May 12 19:21:14] Scheduling destruction of SIP dialog ' 245c407103141a6841c0ac106bd5a...@192.168.0.254' in 32000 ms (Method: INVITE) [May 12 19:21:14] <--- SIP read from 192.168.0.254:5060 ---> ACK sip:17185594...@nasir.server.com <sip%3a17185594...@nasir.server.com>SIP/2.0 Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK3c63f272;rport Max-Forwards: 70 From: "caller" <sip:12129887...@192.168.0.254<sip%3a12129887...@192.168.0.254> >;tag=as76623e31 To: <sip:17185594...@nasir.server.com <sip%3a17185594...@nasir.server.com> >;tag=as0a721b3a Contact: <sip:12129887...@192.168.0.254 <sip%3a12129887...@192.168.0.254>> Call-ID: 245c407103141a6841c0ac106bd5a...@192.168.0.254 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.2.0 Content-Length: 0 <-------------> [May 12 19:21:14] --- (10 headers 0 lines) --- [May 12 19:21:14] <--- SIP read from 192.168.0.254:5060 ---> INVITE sip:17185594...@nasir.server.com <sip%3a17185594...@nasir.server.com>SIP/2.0 Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK05611806;rport Max-Forwards: 70 From: "caller" <sip:12129887...@192.168.0.254<sip%3a12129887...@192.168.0.254> >;tag=as76623e31 To: <sip:17185594...@nasir.server.com <sip%3a17185594...@nasir.server.com>> Contact: <sip:12129887...@192.168.0.254 <sip%3a12129887...@192.168.0.254>> Call-ID: 245c407103141a6841c0ac106bd5a...@192.168.0.254 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.6.2.0 Proxy-Authorization: Digest username="abc", realm="asterisk", algorithm=MD5, uri="sip:17185594...@nasir.server.com <sip%3a17185594...@nasir.server.com>", nonce="7bc52d0a", response="f138ecd92bb706207a7b8d00f1c1bed7" Date: Wed, 12 May 2010 14:20:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 284 v=0 o=root 618893758 618893759 IN IP4 192.168.0.254 s=Asterisk PBX 1.6.2.0 c=IN IP4 192.168.0.254 t=0 0 m=audio 11026 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [May 12 19:21:14] --- (15 headers 13 lines) --- [May 12 19:21:14] Sending to 192.168.0.254 : 5060 (NAT) [May 12 19:21:14] Using INVITE request as basis request - 245c407103141a6841c0ac106bd5a...@192.168.0.254 [May 12 19:21:14] Found peer 'abc' [May 12 19:21:14] Found RTP audio format 0 [May 12 19:21:14] Found RTP audio format 3 [May 12 19:21:14] Found RTP audio format 101 [May 12 19:21:14] Peer audio RTP is at port 192.168.0.254:11026 [May 12 19:21:14] Found description format PCMU for ID 0 [May 12 19:21:14] Found description format GSM for ID 3 [May 12 19:21:14] Found description format telephone-event for ID 101 [May 12 19:21:14] Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x6 (gsm|ulaw)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw) [May 12 19:21:14] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [May 12 19:21:14] Peer audio RTP is at port 192.168.0.254:11026 [May 12 19:21:14] Looking for 17185594743 in payasyougo (domain nasir.server.com) [May 12 19:21:14] WARNING[3785]: chan_sip.c:3930 sip_new: setting callerid number to 12129339037 [May 12 19:21:14] list_route: hop: <sip:12129887...@192.168.0.254<sip%3a12129887...@192.168.0.254> > [May 12 19:21:14] <--- Transmitting (NAT) to 192.168.0.254:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.254:5060 ;branch=z9hG4bK05611806;received=192.168.0.254;rport=5060 From: "caller" <sip:12129887...@192.168.0.254<sip%3a12129887...@192.168.0.254> >;tag=as76623e31 To: <sip:17185594...@nasir.server.com <sip%3a17185594...@nasir.server.com>> Call-ID: 245c407103141a6841c0ac106bd5a...@192.168.0.254 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:17185594...@nasir.server.com<sip%3a17185594...@nasir.server.com> > Content-Length: 0 On Wed, May 12, 2010 at 7:14 PM, Nasir Javaid <nasirjavaidna...@gmail.com>wrote: > Hi Vardan > > I did same as you told and deleted the SIP information in Astdb and > restarted asterisk. but the result was same. > > as you said there might be mistake in sip.conf so i am pasting both servers > configuration here.. > > 1- nasir.server.com > > [abc] > username=abc > type=friend > secret=mysecret > nat=yes > mailbox=12234568 > incominglimit=2 > outgoinglimit=2 > host=dynamic > dtmfmode=rfc2833 > context=payasyougo > canreinvite=yes > callerid="Nasir Qazi" <12234> > accountcode=6:0:abc > amaflags=default > > disallow=all > allow=ulaw > allow=alaw > allow=g729 > allow=gsm > > > 2- 192.168.0.254 (client system) > > > [abc] > type=peer > username=abc > secret=mysecret > host=nasir.server.com > > context=default > dtmfmode=rfc2833 > canreinvite=yes > insecure=very > disallow=all > allow=ulaw > allow=alaw > allow=g729 > allow=gsm > nat=yes > ;qualify=yes > > [caller] > type=friend > secret=123456 > host=dynamic > callerid="caller <12129887777>" > context=out > nat=yes > dtmfmode=rfc2833 > canreinvite=yes > insecure=no > disallow=all > allow=ulaw > allow=alaw > allow=g729 > allow=gsm > t38_udptl=yes > qualify=yes > > > I have registered [caller] on xlite at client system and dialing following > context in local system that will dial [abc] > > [out] > exten=> _X.,1,Dial(SIP/${ext...@abc,30,1) > exten=> _X.,n,Hangup > > > as you can see above *highlighted that context of abc is payasyougo.*problem > is that i want the call to land in that context on > nasir.server.com, which works if i use register string. but without > register string call goes to default context on nasir.server.com > > regards, > > Nasir Javaid > > > Message: 19 > Date: Tue, 11 May 2010 20:54:30 +0500 > From: Vardan <hvarda...@gmail.com> > Subject: Re: [asterisk-users] asterisk-users Digest, Vol 70, Issue 24 > To: asterisk-users@lists.digium.com > Message-ID: <hsbujk$qk...@dough.gmane.org> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hello Nasir > > I have some please. > Do so, it help. > Find all records about interexchange beetwen this two server and delete > all records in sip.conf for this both server (first make backup > sip.conf, or any another conf file that you use). > restart asterisk. > look in astdb about this old records, if any found, delete him > Next, create new record in sip.conf on both servers, without > registration string, reload sip.conf. > give him right context from extensions.conf. > > Can you do this? > > I think is some mistake about configuration in sip.conf, you have I > think two same records (peer or friend). > > Vardan >
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