Sorry, the last config was not clear. I replaced for the following sip.conf
[general] context=default ; Default context for incoming calls allowguest=no ; Allow or reject guest calls -sin password- (default is yes) allowoverlap=no ; Disable overlap dialing support. (Default is yes) udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all) tcpenable=yes ; Enable server for incoming TCP connections (default is no) tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces) srvlookup=yes ; Enable DNS SRV lookups on outbound calls relaxdtmf=yes dtmfmode=inband ;rfc2833compensate=yes [sip.ericsson] ;cambios allowguest hosts allowguest=no ; Allow or reject guest calls -sin password- (default is yes) type=friend calllimit=200 fromuser=ivr1 dtmfmode=inband username=administrador context=incoming-sip-ericsson host=10.146.9.70 host=ericsson host=MSSASU1.MYDOMAIN.COM.PY port=5060 disallow=all allow=alaw allow=gsm allow=ulaw qualify=yes insecure=no From: rafael_vis...@hotmail.com To: asterisk-users@lists.digium.com Date: Sun, 26 Aug 2012 19:52:43 -0400 Subject: Re: [asterisk-users] the lenght of the uri affects on dialplan? Ok... sip.conf [general] context=default ; Default context for incoming calls allowguest=no ; Allow or reject guest calls -sin password- (default is yes) allowoverlap=no ; Disable overlap dialing support. (Default is yes) udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all) tcpenable=yes ; Enable server for incoming TCP connections (default is no) tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces) srvlookup=yes ; Enable DNS SRV lookups on outbound calls relaxdtmf=yes dtmfmode=inband ;rfc2833compensate=yes users.conf [general] fullname = New User userbase = 6000 hasvoicemail = yes vmsecret = 1234 hassip = yes hasiax = no hash323 = no hasmanager = no callwaiting = yes threewaycalling = yes callwaitingcallerid = yes transfer = yes canpark = yes cancallforward = yes callreturn = yes callgroup = 1 pickupgroup = 1 allowguest=no ; Allow or reject guest calls -sin password- (default is yes) [sip.ericsson] ;cambios allowguest hosts ;allowguest=no ; Allow or reject guest calls -sin password- (default is yes) type=friend calllimit=200 fromuser=ivr1 dtmfmode=inband username=administrador context=incoming-sip-ericsson host=10.146.9.70 host=ericsson host=MSSASU1.MYDOMAIN.COM.PY port=5060 disallow=all allow=alaw allow=gsm allow=ulaw qualify=yes insecure=no > Date: Mon, 27 Aug 2012 03:42:51 +0500 > From: fai...@vopium.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] the lenght of the uri affects on dialplan? > > mention the complete scnario and your sip.conf. > > Regards, > > Faisal > (sent from phone) > > Rafael Visser <rafael_vis...@hotmail.com> 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:b56510123456789012...@sisivr03.mydomain.com.py;user=phone SIP/2.0 > >From: <sip:971200...@mssasu1.mydomain.com.py;user=phone>;tag=3016589695 > >To: <sip:b56510123456789012...@sisivr03.mydomain.com.py;user=phone> > >Max-Forwards: 70 > >Via: SIP/2.0/TCP > >MSSASU1.MYDOMAIN.COM.PY:5060;branch=z9hG4bK00000035391821780096 > >Call-ID: 9cax8060616182201-aaaabo...@mssasu1.mydomain.com.py > >CSeq: 7313 INVITE > >P-Asserted-Identity: <sip:971200...@mssasu1.mydomain.com.py;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 - > >9cax8060616182201-aaaabo...@mssasu1.mydomain.com.py > >################ > >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:971200...@mssasu1.mydomain.com.py;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:971200...@mssasu1.mydomain.com.py;user=phone>;tag=3016589695 > >To: > ><sip:b56510123456789012...@sisivr03.mydomain.com.py;user=phone>;tag=as4cfd0d54 > >Call-ID: 9cax8060616182201-aaaabo...@mssasu1.mydomain.com.py > >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:b56510123456789012...@sisivr03.mydomain.com.py;user=phone SIP/2.0 > >From: <sip:971200152@MSSASU1.MYDOMAIN;user=phone>;tag=0046120455 > >To: <sip:b56510123456789012...@sisivr03.mydomain.com.py;user=phone> > >Max-Forwards: 70 > >Via: SIP/2.0/UDP MSSASU1.MYDOMAIN:5060;branch=z9hG4bK00000038670956791982 > >Call-ID: qDaQ1240646182201-AAAAAKDE-@MSSASU1.MYDOMAIN > >CSeq: 14481 INVITE > >P-Asserted-Identity: <sip:971200152@MSSASU1.MYDOMAIN;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 - > >qDaQ1240646182201-AAAAAKDE-@MSSASU1.MYDOMAIN > >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:971200152@MSSASU1.MYDOMAIN;user=phone>;tag=0046120455 > >To: <sip:b56510123456789012...@sisivr03.mydomain.com.py;user=phone> > >Call-ID: qDaQ1240646182201-AAAAAKDE-@MSSASU1.MYDOMAIN > >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:B56510123456789012345@10.146.9.132:5060> > >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 -- _____________________________________________________________________ -- 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
