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
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