Zen Kato
2006-Jul-31 20:15 UTC
[asterisk-users] FC4(2.6.16-1.2108_FC4smp) meetme 404 Not Found
Hi, I installed asterisk-1.2.10, zaptel-1.2.7 on 2.6.16-1.2108_FC4smp. When I dial '0033', which is a meetme number, but '404 Not Found' comes back. I checked zaptel(ztdummy) on FC4, it seems work fine. Meetme has been working on FC3. Can someone tell me why this happens on FC4? My extensions.conf is; exten => 0033,1,Meetme(|qM) exten => 0033,2,Hangup ngrep shows as follows; U 192.168.0.103:5060 -> 192.168.0.3:5070 INVITE sip:0033@192.168.0.3:5070 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.103;br anch=z9hG4bKa854c86267e80f96..From: <sip:0303@192.168.0.3:5070>;tag=c7a5ee3 fa865dcc1..To: <sip:0033@192.168.0.3:5070>..Contact: <sip:0303@192.168.0.10 3>..Supported: replaces..Call-ID: 1c59a92f2174f5ca@192.168.0.103..CSeq: 589 86 INVITE..User-Agent: Grandstream BT100 1.0.6.8..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Type: ap plication/sdp..Content-Length: 354....v=0..o=0303 8000 8000 IN IP4 192.168. 0.103..s=SIP Call..c=IN IP4 192.168.0.103..t=0 0..m=audio 5004 RTP/AVP 0 8 4 18 2 15 97 9..a=sendrecv..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=r tpmap:4 G723/8000..a=rtpmap:18 G729/8000..a=rtpmap:2 G726-32/8000..a=rtpmap :15 G728/8000..a=rtpmap:97 iLBC/8000..a=fmtp:97 mode=20..a=rtpmap:9 G722/16 000..a=ptime:20.. # U 192.168.0.3:5070 -> 192.168.0.103:5060 SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP 192.168.0.103;b ranch=z9hG4bKa854c86267e80f96;received=192.168.0.103..From: <sip:0303@192.1 68.0.3:5070>;tag=c7a5ee3fa865dcc1..To: <sip:0033@192.168.0.3:5070>;tag=as01 593a47..Call-ID: 1c59a92f2174f5ca@192.168.0.103..CSeq: 58986 INVITE..User-A gent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCR IBE, NOTIFY..Contact: <sip:0033@192.168.0.3:5070>..Proxy-Authenticate: Dige st algorithm=MD5, realm="asterisk", nonce="72494d6d"..Content-Length: 0.... # U 192.168.0.103:5060 -> 192.168.0.3:5070 ACK sip:0033@192.168.0.3:5070 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.103;branc h=z9hG4bKa854c86267e80f96..From: <sip:0303@192.168.0.3:5070>;tag=c7a5ee3fa8 65dcc1..To: <sip:0033@192.168.0.3:5070>;tag=as01593a47..Contact: <sip:0303@ 192.168.0.103>..Call-ID: 1c59a92f2174f5ca@192.168.0.103..CSeq: 58986 ACK..U ser-Agent: Grandstream BT100 1.0.6.8..Max-Forwards: 70..Allow: INVITE,ACK,C ANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Length: 0.... # U 192.168.0.103:5060 -> 192.168.0.3:5070 INVITE sip:0033@192.168.0.3:5070 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.103;br anch=z9hG4bK6e9ddb4b834276ef..From: <sip:0303@192.168.0.3:5070>;tag=c7a5ee3 fa865dcc1..To: <sip:0033@192.168.0.3:5070>..Contact: <sip:0303@192.168.0.10 3>..Supported: replaces..Proxy-Authorization: Digest username="0303", realm ="asterisk", algorithm=MD5, uri="sip:0033@192.168.0.3:5070", nonce="72494d6 d", response="35378e1d15e71946d8ca187b102d0087"..Call-ID: 1c59a92f2174f5ca@ 192.168.0.103..CSeq: 58987 INVITE..User-Agent: Grandstream BT100 1.0.6.8..M ax-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUB SCRIBE..Content-Type: application/sdp..Content-Length: 354....v=0..o=0303 8 000 8001 IN IP4 192.168.0.103..s=SIP Call..c=IN IP4 192.168.0.103..t=0 0..m =audio 5004 RTP/AVP 0 8 4 18 2 15 97 9..a=sendrecv..a=rtpmap:0 PCMU/8000..a =rtpmap:8 PCMA/8000..a=rtpmap:4 G723/8000..a=rtpmap:18 G729/8000..a=rtpmap: 2 G726-32/8000..a=rtpmap:15 G728/8000..a=rtpmap:97 iLBC/8000..a=fmtp:97 mod e=20..a=rtpmap:9 G722/16000..a=ptime:20.. # U 192.168.0.3:5070 -> 192.168.0.103:5060 SIP/2.0 404 Not Found..Via: SIP/2.0/UDP 192.168.0.103;branch=z9hG4bK6e9ddb4 b834276ef;received=192.168.0.103..From: <sip:0303@192.168.0.3:5070>;tag=c7a 5ee3fa865dcc1..To: <sip:0033@192.168.0.3:5070>;tag=as01593a47..Call-ID: 1c5 9a92f2174f5ca@192.168.0.103..CSeq: 58987 INVITE..User-Agent: Asterisk PBX.. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Contact : <sip:0033@192.168.0.3:5070>..Content-Length: 0.... # U 192.168.0.103:5060 -> 192.168.0.3:5070 ACK sip:0033@192.168.0.3:5070 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.103;branc h=z9hG4bK6e9ddb4b834276ef..From: <sip:0303@192.168.0.3:5070>;tag=c7a5ee3fa8 65dcc1..To: <sip:0033@192.168.0.3:5070>;tag=as01593a47..Contact: <sip:0303@ 192.168.0.103>..Proxy-Authorization: Digest username="0303", realm="asteris k", algorithm=MD5, uri="sip:0033@192.168.0.3:5070", nonce="72494d6d", respo nse="9bea041787bf296bcd1c5d730733f615"..Call-ID: 1c59a92f2174f5ca@192.168.0 .103..CSeq: 58987 ACK..User-Agent: Grandstream BT100 1.0.6.8..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Cont ent-Length: 0.... ########################################################exit 74 received, 0 dropped Regards, Zen
Zen Kato
2006-Aug-01 16:43 UTC
[asterisk-users] FC4(2.6.16-1.2108_FC4smp) meetme 404 Not Found
Hi, I could found out why the phone received '404 Not Found'. The reason was this part is not parsed and not Added extensions after that. Because there was not at least one space after ; in front of the line of exten => 0033,1,Meetme(|qM). Regards, Zen From: Zen Kato <zenkato@zm.commufa.jp> Subject: [asterisk-users] FC4(2.6.16-1.2108_FC4smp) meetme 404 Not Found Date: Tue, 01 Aug 2006 12:15:04 +0900 (JST)> Hi, > > I installed asterisk-1.2.10, zaptel-1.2.7 on 2.6.16-1.2108_FC4smp. > > When I dial '0033', which is a meetme number, but '404 Not Found' > comes back. I checked zaptel(ztdummy) on FC4, it seems work fine. > Meetme has been working on FC3. > > Can someone tell me why this happens on FC4? > > My extensions.conf is; > > exten => 0033,1,Meetme(|qM) > exten => 0033,2,Hangup > > ngrep shows as follows; > > U 192.168.0.103:5060 -> 192.168.0.3:5070 > INVITE sip:0033@192.168.0.3:5070 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.103;br > anch=z9hG4bKa854c86267e80f96..From: <sip:0303@192.168.0.3:5070>;tag=c7a5ee3 > fa865dcc1..To: <sip:0033@192.168.0.3:5070>..Contact: <sip:0303@192.168.0.10 > 3>..Supported: replaces..Call-ID: 1c59a92f2174f5ca@192.168.0.103..CSeq: 589 > 86 INVITE..User-Agent: Grandstream BT100 1.0.6.8..Max-Forwards: 70..Allow: > INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Type: ap > plication/sdp..Content-Length: 354....v=0..o=0303 8000 8000 IN IP4 192.168. > 0.103..s=SIP Call..c=IN IP4 192.168.0.103..t=0 0..m=audio 5004 RTP/AVP 0 8 > 4 18 2 15 97 9..a=sendrecv..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=r > tpmap:4 G723/8000..a=rtpmap:18 G729/8000..a=rtpmap:2 G726-32/8000..a=rtpmap > :15 G728/8000..a=rtpmap:97 iLBC/8000..a=fmtp:97 mode=20..a=rtpmap:9 G722/16 > 000..a=ptime:20.. > # > U 192.168.0.3:5070 -> 192.168.0.103:5060 > SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP 192.168.0.103;b > ranch=z9hG4bKa854c86267e80f96;received=192.168.0.103..From: <sip:0303@192.1 > 68.0.3:5070>;tag=c7a5ee3fa865dcc1..To: <sip:0033@192.168.0.3:5070>;tag=as01 > 593a47..Call-ID: 1c59a92f2174f5ca@192.168.0.103..CSeq: 58986 INVITE..User-A > gent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCR > IBE, NOTIFY..Contact: <sip:0033@192.168.0.3:5070>..Proxy-Authenticate: Dige > st algorithm=MD5, realm="asterisk", nonce="72494d6d"..Content-Length: 0.... > # > U 192.168.0.103:5060 -> 192.168.0.3:5070 > ACK sip:0033@192.168.0.3:5070 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.103;branc > h=z9hG4bKa854c86267e80f96..From: <sip:0303@192.168.0.3:5070>;tag=c7a5ee3fa8 > 65dcc1..To: <sip:0033@192.168.0.3:5070>;tag=as01593a47..Contact: <sip:0303@ > 192.168.0.103>..Call-ID: 1c59a92f2174f5ca@192.168.0.103..CSeq: 58986 ACK..U > ser-Agent: Grandstream BT100 1.0.6.8..Max-Forwards: 70..Allow: INVITE,ACK,C > ANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Length: 0.... > # > U 192.168.0.103:5060 -> 192.168.0.3:5070 > INVITE sip:0033@192.168.0.3:5070 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.103;br > anch=z9hG4bK6e9ddb4b834276ef..From: <sip:0303@192.168.0.3:5070>;tag=c7a5ee3 > fa865dcc1..To: <sip:0033@192.168.0.3:5070>..Contact: <sip:0303@192.168.0.10 > 3>..Supported: replaces..Proxy-Authorization: Digest username="0303", realm > ="asterisk", algorithm=MD5, uri="sip:0033@192.168.0.3:5070", nonce="72494d6 > d", response="35378e1d15e71946d8ca187b102d0087"..Call-ID: 1c59a92f2174f5ca@ > 192.168.0.103..CSeq: 58987 INVITE..User-Agent: Grandstream BT100 1.0.6.8..M > ax-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUB > SCRIBE..Content-Type: application/sdp..Content-Length: 354....v=0..o=0303 8 > 000 8001 IN IP4 192.168.0.103..s=SIP Call..c=IN IP4 192.168.0.103..t=0 0..m > =audio 5004 RTP/AVP 0 8 4 18 2 15 97 9..a=sendrecv..a=rtpmap:0 PCMU/8000..a > =rtpmap:8 PCMA/8000..a=rtpmap:4 G723/8000..a=rtpmap:18 G729/8000..a=rtpmap: > 2 G726-32/8000..a=rtpmap:15 G728/8000..a=rtpmap:97 iLBC/8000..a=fmtp:97 mod > e=20..a=rtpmap:9 G722/16000..a=ptime:20.. > # > U 192.168.0.3:5070 -> 192.168.0.103:5060 > SIP/2.0 404 Not Found..Via: SIP/2.0/UDP 192.168.0.103;branch=z9hG4bK6e9ddb4 > b834276ef;received=192.168.0.103..From: <sip:0303@192.168.0.3:5070>;tag=c7a > 5ee3fa865dcc1..To: <sip:0033@192.168.0.3:5070>;tag=as01593a47..Call-ID: 1c5 > 9a92f2174f5ca@192.168.0.103..CSeq: 58987 INVITE..User-Agent: Asterisk PBX.. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Contact > : <sip:0033@192.168.0.3:5070>..Content-Length: 0.... > # > U 192.168.0.103:5060 -> 192.168.0.3:5070 > ACK sip:0033@192.168.0.3:5070 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.103;branc > h=z9hG4bK6e9ddb4b834276ef..From: <sip:0303@192.168.0.3:5070>;tag=c7a5ee3fa8 > 65dcc1..To: <sip:0033@192.168.0.3:5070>;tag=as01593a47..Contact: <sip:0303@ > 192.168.0.103>..Proxy-Authorization: Digest username="0303", realm="asteris > k", algorithm=MD5, uri="sip:0033@192.168.0.3:5070", nonce="72494d6d", respo > nse="9bea041787bf296bcd1c5d730733f615"..Call-ID: 1c59a92f2174f5ca@192.168.0 > .103..CSeq: 58987 ACK..User-Agent: Grandstream BT100 1.0.6.8..Max-Forwards: > 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Cont > ent-Length: 0.... > ########################################################exit > 74 received, 0 dropped > > Regards, > > Zen > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >