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Sorry, the last config was not clear. <br>I replaced for the following sip.conf<br><br><br>[general]<br>context=default ; Default context for incoming calls<br>allowguest=no ; Allow or reject guest calls -sin password- (default is yes)<br>allowoverlap=no ; Disable overlap dialing support. (Default is yes)<br>udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)<br>tcpenable=yes ; Enable server for incoming TCP connections (default is no)<br>tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)<br>srvlookup=yes ; Enable DNS SRV lookups on outbound calls<br>relaxdtmf=yes<br>dtmfmode=inband<br>;rfc2833compensate=yes<br><br><br>[sip.ericsson]<br>;cambios allowguest hosts<br>allowguest=no ; Allow or reject guest calls -sin password- (default is yes)<br>type=friend<br>calllimit=200<br>fromuser=ivr1<br>dtmfmode=inband<br>username=administrador<br>context=incoming-sip-ericsson<br>host=10.146.9.70<br>host=ericsson<br>host=MSSASU1.MYDOMAIN.COM.PY<br>port=5060<br>disallow=all<br>allow=alaw<br>allow=gsm<br>allow=ulaw<br>qualify=yes<br>insecure=no<br><br><br><div><div id="SkyDrivePlaceholder"></div><hr id="stopSpelling">From: rafael_visser@hotmail.com<br>To: asterisk-users@lists.digium.com<br>Date: Sun, 26 Aug 2012 19:52:43 -0400<br>Subject: Re: [asterisk-users] the lenght of the uri affects on dialplan?<br><br>
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Ok...<br><br><br>sip.conf<br>[general]<br>context=default ; Default context for incoming calls<br>allowguest=no ; Allow or reject guest calls -sin password- (default is yes)<br>allowoverlap=no ; Disable overlap dialing support. (Default is yes)<br>udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)<br>tcpenable=yes ; Enable server for incoming TCP connections (default is no)<br>tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)<br>srvlookup=yes ; Enable DNS SRV lookups on outbound calls<br>relaxdtmf=yes<br>dtmfmode=inband<br>;rfc2833compensate=yes<br><br><br>users.conf<br>[general]<br>fullname = New User<br>userbase = 6000<br>hasvoicemail = yes<br>vmsecret = 1234<br>hassip = yes<br>hasiax = no<br>hash323 = no<br>hasmanager = no<br>callwaiting = yes<br>threewaycalling = yes<br>callwaitingcallerid = yes<br>transfer = yes<br>canpark = yes<br>cancallforward = yes<br>callreturn = yes<br>callgroup = 1<br>pickupgroup = 1<br>allowguest=no ; Allow or reject guest calls -sin password- (default is yes)<br><br>[sip.ericsson]<br>;cambios allowguest hosts<br>;allowguest=no ; Allow or reject guest calls -sin password- (default is yes)<br>type=friend<br>calllimit=200<br>fromuser=ivr1<br>dtmfmode=inband<br>username=administrador<br>context=incoming-sip-ericsson<br>host=10.146.9.70<br>host=ericsson<br>host=MSSASU1.MYDOMAIN.COM.PY<br>port=5060<br>disallow=all<br>allow=alaw<br>allow=gsm<br>allow=ulaw<br>qualify=yes<br>insecure=no<br><br><div><div id="ecxSkyDrivePlaceholder"></div>> Date: Mon, 27 Aug 2012 03:42:51 +0500<br>> From: faisal@vopium.com<br>> To: asterisk-users@lists.digium.com<br>> Subject: Re: [asterisk-users] the lenght of the uri affects on dialplan?<br>> <br>> mention the complete scnario and your sip.conf.<br>> <br>> Regards,<br>> <br>> Faisal <br>> (sent from phone)<br>> <br>> Rafael Visser <rafael_visser@hotmail.com> wrote:<br>> <br>> ><br>> >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>> > <br>> >--<br>> >_____________________________________________________________________<br>> >-- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> >New to Asterisk? Join us for a live introductory webinar every Thurs:<br>> > http://www.asterisk.org/hello<br>> ><br>> >asterisk-users mailing list<br>> >To UNSUBSCRIBE or update options visit:<br>> > http://lists.digium.com/mailman/listinfo/asterisk-users<br>> --<br>> _____________________________________________________________________<br>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>> http://www.asterisk.org/hello<br>> <br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-users<br></div> </div>
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