Hi again, below is debug trace of * cli when i remove register string from sip.conf
*CLI> [May 12 19:33:06] <--- SIP read from 192.168.0.254:5060 ---> INVITE sip:[email protected] <sip%[email protected]>SIP/2.0 Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK56e3b44a;rport Max-Forwards: 70 From: "caller" <sip:[email protected]<sip%[email protected]> >;tag=as5b6db7a2 To: <sip:[email protected] <sip%[email protected]>> Contact: <sip:[email protected] <sip%[email protected]>> Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.0 Date: Wed, 12 May 2010 14:32:18 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 814806874 814806874 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 17632 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 19:33:06] --- (14 headers 13 lines) --- [May 12 19:33:06] Sending to 192.168.0.254 : 5060 (NAT) [May 12 19:33:06] Using INVITE request as basis request - [email protected] [May 12 19:33:06] Found no matching peer or user for '192.168.0.254:5060' [May 12 19:33:06] Found RTP audio format 0 [May 12 19:33:06] Found RTP audio format 3 [May 12 19:33:06] Found RTP audio format 101 [May 12 19:33:06] Peer audio RTP is at port 192.168.0.254:17632 [May 12 19:33:06] Found description format PCMU for ID 0 [May 12 19:33:06] Found description format GSM for ID 3 [May 12 19:33:06] Found description format telephone-event for ID 101 [May 12 19:33:06] Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x6 (gsm|ulaw)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw) [May 12 19:33:06] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [May 12 19:33:06] Peer audio RTP is at port 192.168.0.254:17632 [May 12 19:33:06] Looking for 17185594743 in default (domain nasir.server.com) [May 12 19:33:06] WARNING[4113]: chan_sip.c:3930 sip_new: setting callerid number to 12129887777 [May 12 19:33:06] list_route: hop: <sip:[email protected]<sip%[email protected]> > [May 12 19:33:06] <--- Transmitting (NAT) to 192.168.0.254:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.254:5060 ;branch=z9hG4bK56e3b44a;received=192.168.0.254;rport=5060 From: "caller" <sip:[email protected]<sip%[email protected]> >;tag=as5b6db7a2 To: <sip:[email protected] <sip%[email protected]>> Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[email protected]<sip%[email protected]> > Content-Length: 0 On Wed, May 12, 2010 at 7:26 PM, Nasir Javaid <[email protected]>wrote: > 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:[email protected]<sip%[email protected]>SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK3c63f272;rport > Max-Forwards: 70 > From: "caller" <sip:[email protected]<sip%[email protected]> > >;tag=as76623e31 > To: <sip:[email protected] <sip%[email protected]> > > > Contact: <sip:[email protected] <sip%[email protected]>> > Call-ID: [email protected] > 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 - > [email protected] > [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:[email protected]<sip%[email protected]> > >;tag=as76623e31 > To: <sip:[email protected] <sip%[email protected]> > >;tag=as0a721b3a > Call-ID: [email protected] > 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 ' > [email protected]' in 32000 ms (Method: > INVITE) > [May 12 19:21:14] > <--- SIP read from 192.168.0.254:5060 ---> > ACK sip:[email protected] > <sip%[email protected]>SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK3c63f272;rport > Max-Forwards: 70 > From: "caller" <sip:[email protected]<sip%[email protected]> > >;tag=as76623e31 > To: <sip:[email protected] <sip%[email protected]> > >;tag=as0a721b3a > Contact: <sip:[email protected] <sip%[email protected]>> > Call-ID: [email protected] > 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:[email protected]<sip%[email protected]>SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK05611806;rport > Max-Forwards: 70 > From: "caller" <sip:[email protected]<sip%[email protected]> > >;tag=as76623e31 > To: <sip:[email protected] <sip%[email protected]> > > > Contact: <sip:[email protected] <sip%[email protected]>> > Call-ID: [email protected] > CSeq: 103 INVITE > User-Agent: Asterisk PBX 1.6.2.0 > Proxy-Authorization: Digest username="abc", realm="asterisk", > algorithm=MD5, > uri="sip:[email protected]<sip%[email protected]>", > 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 - > [email protected] > [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:[email protected]<sip%[email protected]> > > > [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:[email protected]<sip%[email protected]> > >;tag=as76623e31 > To: <sip:[email protected] <sip%[email protected]> > > > Call-ID: [email protected] > CSeq: 103 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:[email protected]<sip%[email protected]> > > > Content-Length: 0 > > > > > On Wed, May 12, 2010 at 7:14 PM, Nasir Javaid > <[email protected]>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 <[email protected]> >> Subject: Re: [asterisk-users] asterisk-users Digest, Vol 70, Issue 24 >> To: [email protected] >> Message-ID: <[email protected]> >> 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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