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Hi Gurus..<br>I use asterisk for just for ivr.<br>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. <br><br>Also, it has somethig to do with the numbers of digits of the dialed number. Few digits works ok, 14 to more works wrong.<br>Do you know what am i missing?<br>Thanks in advance.<br><br><br><br><br><br><br><br><br><br>Debug with long hostname (B is considered as an '*')<br>================================<br><--- SIP read from TCP:10.146.9.70:6240 ---><br>INVITE sip:B56510123456789012345@SISIVR03.MYDOMAIN.COM.PY;user=phone SIP/2.0<br>From: <sip:971200152@MSSASU1.MYDOMAIN.COM.PY;user=phone>;tag=3016589695<br>To: <sip:B56510123456789012345@SISIVR03.MYDOMAIN.COM.PY;user=phone><br>Max-Forwards: 70<br>Via: SIP/2.0/TCP MSSASU1.MYDOMAIN.COM.PY:5060;branch=z9hG4bK00000035391821780096<br>Call-ID: 9caX8060616182201-AAAABOPA-@MSSASU1.MYDOMAIN.COM.PY<br>CSeq: 7313 INVITE<br>P-Asserted-Identity: <sip:971200152@MSSASU1.MYDOMAIN.COM.PY;user=phone><br>Accept: application/sdp<br>Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,PRACK,UPDATE<br>P-Charging-Vector: icid-value=6510081000-0826-16155907;icid-generated-at=MSSASU1.MYDOMAIN.COM.PY;orig-ioi=MSSASU1.MYDOMAIN.COM.PY<br>Supported: 100rel<br>Content-Type: application/sdp<br>Contact: <sip:MSSASU1.MYDOMAIN.COM.PY:5060;transport=UDP><br>Content-Length: 414<br><br>v=0<br>o=- 7530078 7530078 IN IP4 MSSASU1.MYDOMAIN.COM.PY<br>s=-<br>t=0 0<br>a=sendrecv<br>m=audio 13802 RTP/AVP 8 96 18 97<br>c=IN IP4 10.143.1.67<br>b=RR:0<br>b=RS:0<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:96 AMR/8000<br>a=fmtp:96 mode-set=0,2,4,7;mode-change-period=2;mode-change-capability=2;mode-change-neighbor=1;max-red=0<br>a=rtpmap:18 G729/8000<br>a=fmtp:18 annexb=yes<br>a=rtpmap:97 telephone-event/8000<br>a=fmtp:97 0-15<br>a=maxptime:40<br><-------------><br>--- (15 headers 17 lines) ---<br>Sending to 10.146.9.70:5060 (no NAT)<br>Using INVITE request as basis request - 9caX8060616182201-AAAABOPA-@MSSASU1.MYDOMAIN.COM.PY<br>################<br>No matching peer for '971200152' from '10.146.9.70:6240'<br>[Aug 26 16:15:55] NOTICE[6873]: chan_sip.c:21975 handle_request_invite: Sending fake auth rej<br>ection for device <sip:971200152@MSSASU1.MYDOMAIN.COM.PY;user=phone>;tag=3016589695<br>#################<br><--- Reliably Transmitting (no NAT) to 10.146.9.70:5060 ---><br>SIP/2.0 401 Unauthorized<br>Via: SIP/2.0/TCP MSSASU1.MYDOMAIN.COM.PY:5060;branch=z9hG4bK00000035391821780096;received=10.146.9.70<br>From: <sip:971200152@MSSASU1.MYDOMAIN.COM.PY;user=phone>;tag=3016589695<br>To: <sip:B56510123456789012345@SISIVR03.MYDOMAIN.COM.PY;user=phone>;tag=as4cfd0d54<br>Call-ID: 9caX8060616182201-AAAABOPA-@MSSASU1.MYDOMAIN.COM.PY<br>CSeq: 7313 INVITE<br>Server: Asterisk PBX 1.8.7.0<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>Supported: replaces, timer<br>WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="35ff0feb"<br>Content-Length: 0<br><br><br><br><br>Short hostname on switch<br>===============<br>Connected to Asterisk 1.8.7.0 currently running on fdosis-ims1 (pid = 25430)<br>fdosis-ims1*CLI> core set verbose 1<br>Verbosity was 0 and is now 1<br><br><--- SIP read from UDP:10.146.9.70:5060 ---><br>INVITE sip:B56510123456789012345@SISIVR03.MYDOMAIN.COM.PY;user=phone SIP/2.0<br>From: <sip:971200152@MSSASU1.MYDOMAIN;user=phone>;tag=0046120455<br>To: <sip:B56510123456789012345@SISIVR03.MYDOMAIN.COM.PY;user=phone><br>Max-Forwards: 70<br>Via: SIP/2.0/UDP MSSASU1.MYDOMAIN:5060;branch=z9hG4bK00000038670956791982<br>Call-ID: qDaQ1240646182201-AAAAAKDE-@MSSASU1.MYDOMAIN<br>CSeq: 14481 INVITE<br>P-Asserted-Identity: <sip:971200152@MSSASU1.MYDOMAIN;user=phone><br>Accept: application/sdp<br>llow: INVITE,ACK,OPTIONS,BYE,CANCEL,PRACK,UPDATE<br>P-Charging-Vector: icid-value=A6D6B81000-0826-16440101;icid-generated-at=MSSASU1.MYDOMAIN;orig-ioi=MSSASU1.MYDOMAIN.COM.PY<br>Supported: 100rel<br>Content-Type: application/sdp<br>Contact: <sip:MSSASU1.MYDOMAIN:5060;transport=UDP><br>Content-Length: 407<br><br>v=0<br>o=- 8986991 8986991 IN IP4 MSSASU1.MYDOMAIN<br>s=-<br>t=0 0<br>a=sendrecv<br>m=audio 30838 RTP/AVP 8 96 18 97<br>c=IN IP4 10.143.1.68<br>b=RR:0<br>b=RS:0<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:96 AMR/8000<br>a=fmtp:96 mode-set=0,2,4,7;mode-change-period=2;mode-change-capability=2;mode-change-neighbor=1;max-red=0<br>a=rtpmap:18 G729/8000<br>a=fmtp:18 annexb=yes<br>a=rtpmap:97 telephone-event/8000<br>a=fmtp:97 0-15<br>a=maxptime:40<br><-------------><br>--- (15 headers 17 lines) ---<br>Sending to 10.146.9.70:5060 (no NAT)<br>Using INVITE request as basis request - qDaQ1240646182201-AAAAAKDE-@MSSASU1.MYDOMAIN<br>Found peer 'sip.ericsson' for '971200152' from 10.146.9.70:5060<br>Found RTP audio format 8<br>Found RTP audio format 96<br>Found RTP audio format 18<br>Found RTP audio format 97<br>Found audio description format PCMA for ID 8<br>Found unknown media description format AMR for ID 96<br>Found audio description format G729 for ID 18<br>Found audio description format telephone-event for ID 97<br>Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x108 (alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)<br>Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing)<br>Peer audio RTP is at port 10.143.1.68:30838<br>Looking for B56510123456789012345 in incoming-sip-ericsson (domain SISIVR03.MYDOMAIN.COM.PY)<br>list_route: hop: <sip:MSSASU1.MYDOMAIN:5060;transport=UDP><br><br><--- Transmitting (no NAT) to 10.146.9.70:5060 ---><br>SIP/2.0 100 Trying<br>Via: SIP/2.0/UDP MSSASU1.MYDOMAIN:5060;branch=z9hG4bK00000038670956791982;received=10.146.9.70<br>From: <sip:971200152@MSSASU1.MYDOMAIN;user=phone>;tag=0046120455<br>To: <sip:B56510123456789012345@SISIVR03.MYDOMAIN.COM.PY;user=phone><br>Call-ID: qDaQ1240646182201-AAAAAKDE-@MSSASU1.MYDOMAIN<br>CSeq: 14481 INVITE<br>Server: Asterisk PBX 1.8.7.0<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>Supported: replaces, timer<br>Contact: <sip:B56510123456789012345@10.146.9.132:5060><br>Content-Length: 0<br><br><br> </div></body>
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