mention the complete scnario and your sip.conf. Regards,
Faisal (sent from phone) Rafael Visser <[email protected]> wrote: > >Hi Gurus.. >I use asterisk for just for ivr. >My issue is that when the switch changes it's host name from MSSASU1.MYDOMAIN >to MSSSASU1.MYDOMAiN.COM or MSSSASU1.MYDOMAiN.COM.PY the call is rejected with >"No matching peer" and the "handle_request_invite: Sending fake auth rejection >for device x". It doesn't match it's own default context. > >Also, it has somethig to do with the numbers of digits of the dialed number. >Few digits works ok, 14 to more works wrong. >Do you know what am i missing? >Thanks in advance. > > > > > > > > > >Debug with long hostname (B is considered as an '*') >================================ ><--- SIP read from TCP:10.146.9.70:6240 ---> >INVITE sip:[email protected];user=phone SIP/2.0 >From: <sip:[email protected];user=phone>;tag=3016589695 >To: <sip:[email protected];user=phone> >Max-Forwards: 70 >Via: SIP/2.0/TCP >MSSASU1.MYDOMAIN.COM.PY:5060;branch=z9hG4bK00000035391821780096 >Call-ID: [email protected] >CSeq: 7313 INVITE >P-Asserted-Identity: <sip:[email protected];user=phone> >Accept: application/sdp >Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,PRACK,UPDATE >P-Charging-Vector: >icid-value=6510081000-0826-16155907;icid-generated-at=MSSASU1.MYDOMAIN.COM.PY;orig-ioi=MSSASU1.MYDOMAIN.COM.PY >Supported: 100rel >Content-Type: application/sdp >Contact: <sip:MSSASU1.MYDOMAIN.COM.PY:5060;transport=UDP> >Content-Length: 414 > >v=0 >o=- 7530078 7530078 IN IP4 MSSASU1.MYDOMAIN.COM.PY >s=- >t=0 0 >a=sendrecv >m=audio 13802 RTP/AVP 8 96 18 97 >c=IN IP4 10.143.1.67 >b=RR:0 >b=RS:0 >a=rtpmap:8 PCMA/8000 >a=rtpmap:96 AMR/8000 >a=fmtp:96 >mode-set=0,2,4,7;mode-change-period=2;mode-change-capability=2;mode-change-neighbor=1;max-red=0 >a=rtpmap:18 G729/8000 >a=fmtp:18 annexb=yes >a=rtpmap:97 telephone-event/8000 >a=fmtp:97 0-15 >a=maxptime:40 ><-------------> >--- (15 headers 17 lines) --- >Sending to 10.146.9.70:5060 (no NAT) >Using INVITE request as basis request - >[email protected] >################ >No matching peer for '971200152' from '10.146.9.70:6240' >[Aug 26 16:15:55] NOTICE[6873]: chan_sip.c:21975 handle_request_invite: >Sending fake auth rej >ection for device ><sip:[email protected];user=phone>;tag=3016589695 >################# ><--- Reliably Transmitting (no NAT) to 10.146.9.70:5060 ---> >SIP/2.0 401 Unauthorized >Via: SIP/2.0/TCP >MSSASU1.MYDOMAIN.COM.PY:5060;branch=z9hG4bK00000035391821780096;received=10.146.9.70 >From: <sip:[email protected];user=phone>;tag=3016589695 >To: ><sip:[email protected];user=phone>;tag=as4cfd0d54 >Call-ID: [email protected] >CSeq: 7313 INVITE >Server: Asterisk PBX 1.8.7.0 >Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >PUBLISH >Supported: replaces, timer >WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="35ff0feb" >Content-Length: 0 > > > > >Short hostname on switch >=============== >Connected to Asterisk 1.8.7.0 currently running on fdosis-ims1 (pid = 25430) >fdosis-ims1*CLI> core set verbose 1 >Verbosity was 0 and is now 1 > ><--- SIP read from UDP:10.146.9.70:5060 ---> >INVITE sip:[email protected];user=phone SIP/2.0 >From: <sip:[email protected];user=phone>;tag=0046120455 >To: <sip:[email protected];user=phone> >Max-Forwards: 70 >Via: SIP/2.0/UDP MSSASU1.MYDOMAIN:5060;branch=z9hG4bK00000038670956791982 >Call-ID: [email protected] >CSeq: 14481 INVITE >P-Asserted-Identity: <sip:[email protected];user=phone> >Accept: application/sdp >llow: INVITE,ACK,OPTIONS,BYE,CANCEL,PRACK,UPDATE >P-Charging-Vector: >icid-value=A6D6B81000-0826-16440101;icid-generated-at=MSSASU1.MYDOMAIN;orig-ioi=MSSASU1.MYDOMAIN.COM.PY >Supported: 100rel >Content-Type: application/sdp >Contact: <sip:MSSASU1.MYDOMAIN:5060;transport=UDP> >Content-Length: 407 > >v=0 >o=- 8986991 8986991 IN IP4 MSSASU1.MYDOMAIN >s=- >t=0 0 >a=sendrecv >m=audio 30838 RTP/AVP 8 96 18 97 >c=IN IP4 10.143.1.68 >b=RR:0 >b=RS:0 >a=rtpmap:8 PCMA/8000 >a=rtpmap:96 AMR/8000 >a=fmtp:96 >mode-set=0,2,4,7;mode-change-period=2;mode-change-capability=2;mode-change-neighbor=1;max-red=0 >a=rtpmap:18 G729/8000 >a=fmtp:18 annexb=yes >a=rtpmap:97 telephone-event/8000 >a=fmtp:97 0-15 >a=maxptime:40 ><-------------> >--- (15 headers 17 lines) --- >Sending to 10.146.9.70:5060 (no NAT) >Using INVITE request as basis request - >[email protected] >Found peer 'sip.ericsson' for '971200152' from 10.146.9.70:5060 >Found RTP audio format 8 >Found RTP audio format 96 >Found RTP audio format 18 >Found RTP audio format 97 >Found audio description format PCMA for ID 8 >Found unknown media description format AMR for ID 96 >Found audio description format G729 for ID 18 >Found audio description format telephone-event for ID 97 >Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x108 >(alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) >Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 >(telephone-event|), combined - 0x0 (nothing) >Peer audio RTP is at port 10.143.1.68:30838 >Looking for B56510123456789012345 in incoming-sip-ericsson (domain >SISIVR03.MYDOMAIN.COM.PY) >list_route: hop: <sip:MSSASU1.MYDOMAIN:5060;transport=UDP> > ><--- Transmitting (no NAT) to 10.146.9.70:5060 ---> >SIP/2.0 100 Trying >Via: SIP/2.0/UDP >MSSASU1.MYDOMAIN:5060;branch=z9hG4bK00000038670956791982;received=10.146.9.70 >From: <sip:[email protected];user=phone>;tag=0046120455 >To: <sip:[email protected];user=phone> >Call-ID: [email protected] >CSeq: 14481 INVITE >Server: Asterisk PBX 1.8.7.0 >Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >PUBLISH >Supported: replaces, timer >Contact: <sip:[email protected]:5060> >Content-Length: 0 > > > >-- >_____________________________________________________________________ >-- Bandwidth and Colocation Provided by http://www.api-digital.com -- >New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
