Sorry, I was out for a while. Still have this issue? >From what I am seeing, asterisk is expecting for the password. Is the voicemail configured ? Check username and password.
Somewhere there it says that couldn't read username and password from the voicemail. Have the extensions.conf at asterisk dialplan configured properly? El mar 31, 2014 2:25 PM, "Slava Bendersky" <volga...@networklab.ca> escribió: > Hello Pedro, > > Here SDP from asterisk. Asterisk it just don't know where to send traffic. > Sip peer on asterisk connects no issue. > > [voice] > type=peer > host=kamailio ip > defaultuser=1300 > fromuser=1300 > user=1300 > secret=test > permit=local subnet > disallow=all > allow=ulaw > dtmfmode=rfc2833 > context=voicemailbox > canreinvite=no > insecure=port,invite > qualify=yes > directrtpsetup=no > > > > > -- Incorrect password '' for user '1200' (context = default) > -- <SIP/1200-00000004> Playing 'vm-incorrect-mailbox.gsm' (language > 'en') > Retransmitting #9 (no NAT) to 10.237.236.207:5060: > SIP/2.0 200 OK > Via: SIP/2.0/UDP > 10.237.236.207;branch=z9hG4bKf682.cc9d98383fa97727d9968596f24c2c0a.0;received=10.237.236.207 > Via: SIP/2.0/UDP 10.237.236.212:64609 > ;branch=z9hG4bK-d8754z-f319541e694ad32f-1---d8754z- > Record-Route: <sip:10.237.236.207;lr=on> > From: "Slava Bendersky"<sip:1...@networklab.loc > ;transport=UDP>;tag=6358d712 > To: <sip:1...@networklab.loc;transport=UDP>;tag=as3b53c4ae > Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. > CSeq: 2 INVITE > Server: Asterisk PBX 12.0.0 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH > Supported: replaces, timer > Session-Expires: 1800;refresher=uas > Contact: <sip:120@10.237.236.207:5062> > Content-Type: application/sdp > Require: timer > Content-Length: 183 > > v=0 > o=root 1990993471 1990993471 IN IP4 10.237.236.207 > s=Asterisk PBX 12.0.0 > c=IN IP4 10.237.236.207 > t=0 0 > m=audio 15070 RTP/AVP 0 > a=rtpmap:0 PCMU/8000 > a=ptime:20 > a=sendrecv > > --- > Retransmitting #10 (no NAT) to 10.237.236.207:5060: > SIP/2.0 200 OK > Via: SIP/2.0/UDP > 10.237.236.207;branch=z9hG4bKf682.cc9d98383fa97727d9968596f24c2c0a.0;received=10.237.236.207 > Via: SIP/2.0/UDP 10.237.236.212:64609 > ;branch=z9hG4bK-d8754z-f319541e694ad32f-1---d8754z- > Record-Route: <sip:10.237.236.207;lr=on> > From: "Slava Bendersky"<sip:1...@networklab.loc > ;transport=UDP>;tag=6358d712 > To: <sip:1...@networklab.loc;transport=UDP>;tag=as3b53c4ae > Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. > CSeq: 2 INVITE > Server: Asterisk PBX 12.0.0 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH > Supported: replaces, timer > Session-Expires: 1800;refresher=uas > Contact: <sip:120@10.237.236.207:5062> > Content-Type: application/sdp > Require: timer > Content-Length: 183 > > v=0 > o=root 1990993471 1990993471 IN IP4 10.237.236.207 > s=Asterisk PBX 12.0.0 > c=IN IP4 10.237.236.207 > t=0 0 > m=audio 15070 RTP/AVP 0 > a=rtpmap:0 PCMU/8000 > a=ptime:20 > a=sendrecv > > --- > [Mar 31 14:44:25] WARNING[1834]: chan_sip.c:4259 retrans_pkt: > Retransmission timeout reached on transmission > YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. for seqno 2 (Critical > Response) -- See > https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions > Packet timed out after 32000ms with no response > [Mar 31 14:44:25] WARNING[1834]: chan_sip.c:4288 retrans_pkt: Hanging up > call YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. - no reply to our > critical packet (see > https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). > [Mar 31 14:44:25] WARNING[2801][C-0000000e]: app_voicemail.c:10590 > vm_authenticate: Couldn't read username > Scheduling destruction of SIP dialog > 'YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI.' in 32000 ms (Method: INVITE) > set_destination: Parsing <sip:10.237.236.207;lr=on> for address/port to > send to > set_destination: set destination to 10.237.236.207:5060 > Reliably Transmitting (no NAT) to 10.237.236.207:5060: > BYE sip:1200@10.237.236.212:64609;transport=UDP SIP/2.0 > Via: SIP/2.0/UDP 10.237.236.207:5062;branch=z9hG4bK7b8eba54 > Route: <sip:10.237.236.207;lr=on> > Max-Forwards: 70 > From: <sip:1...@networklab.loc;transport=UDP>;tag=as3b53c4ae > To: "Slava Bendersky"<sip:1...@networklab.loc;transport=UDP>;tag=6358d712 > Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. > CSeq: 102 BYE > User-Agent: Asterisk PBX 12.0.0 > X-Asterisk-HangupCause: No user responding > X-Asterisk-HangupCauseCode: 18 > Content-Length: 0 > > > --- > > <--- SIP read from UDP:10.237.236.207:5060 ---> > SIP/2.0 481 Call/Transaction Does Not Exist > Via: SIP/2.0/UDP 10.237.236.207:5062;branch=z9hG4bK7b8eba54 > To: "Slava Bendersky"<sip:1...@networklab.loc;transport=UDP>;tag=6358d712 > From: <sip:1...@networklab.loc;transport=UDP>;tag=as3b53c4ae > Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. > CSeq: 102 BYE > Accept-Language: en > Content-Length: 0 > > <-------------> > --- (8 headers 0 lines) --- > Really destroying SIP dialog > 'YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI.' Method: INVITE > Reliably Transmitting (no NAT) to 10.237.236.207:5060: > OPTIONS sip:10.237.236.207 SIP/2.0 > Via: SIP/2.0/UDP 10.237.236.207:5062;branch=z9hG4bK51a7f1ef > Max-Forwards: 70 > From: "asterisk" <sip:1...@networklab.loc>;tag=as7232ca20 > To: <sip:10.237.236.207> > Contact: <sip:1300@10.237.236.207:5062> > Call-ID: 46ea55704ee7005705c98d9106904...@networklab.loc > CSeq: 102 OPTIONS > User-Agent: Asterisk PBX 12.0.0 > Date: Mon, 31 Mar 2014 18:44:35 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH > Supported: replaces, timer > Content-Length: 0 > > Slava. > > ------------------------------ > *From: *"Pedro Niño" <nino.pe...@gmail.com> > *To: *"Kamailio (SER) - Users Mailing List" <sr-users@lists.sip-router.org > > > *Sent: *Monday, March 31, 2014 9:51:11 AM > *Subject: *Re: [SR-Users] message 484 > > So, the problem is that calls made from a direct connected user, falls to > voicemail? Even if the other user is online? > > All the users are on the same asterisk server? Or using a trunk outside? > > As a test, tried to register to the asterisk server directly and test the > call? > > That's why I was asking to elaborate, and show a bit more about the call > flow behavior... A small text diagram and desired behavior would be useful > > El mar 31, 2014 8:13 AM, "Slava Bendersky" <volga...@networklab.ca> > escribió: > >> Hello Olle, >> Overlap is disabled on asterisk. I more wonder about this message. >> >> Mar 31 08:40:20 dsm01 /usr/sbin/kamailio[6101]: WARNING: sanity >> [sanity.c:833]: check_parse_uris(): sanity_check(): check_parse_uris(): >> failed to parse From uri >> >> Because from direct connected network, call failing to voicemail. >> >> Slva. >> ------------------------------ >> *From: *"Olle E. Johansson" <o...@edvina.net> >> *To: *"Kamailio (SER) - Users Mailing List" < >> sr-users@lists.sip-router.org> >> *Sent: *Monday, March 31, 2014 3:33:11 AM >> *Subject: *Re: [SR-Users] message 484 >> >> Hi! >> I guess this is a poorly configured Asterisk server that has >> "Allowoverlap" enabled. >> A 484 is used for overlap dialing. The server says "I need more digits to >> complete this call". >> >> /O >> >> On 31 Mar 2014, at 02:30, Pedro Niño <nino.pe...@gmail.com> wrote: >> >> I think this is the correct behavior, as asterisk server is complaining >> about the address/request not containing all the necesary data to process >> the message >> >> Can you please elaborate with a bit more of detail? Also can use tools >> like sngrep, tcpdump (or wireshark) to have a better view of the complete >> call flow. >> >> Maybe that way we can help. >> El mar 29, 2014 1:59 AM, "Slava Bendersky" <volga...@networklab.ca> >> escribió: >> >>> Hello Everyone, >>> How to correct message 484 >>> Is need use txt module to fill string with correct information ? >>> >>> <--- SIP read from UDP:192.168.100.145:5060 ---> >>> SIP/2.0 484 Address Incomplete >>> Via: SIP/2.0/UDP 192.168.100.145:5062;branch=z9hG4bK5ec564e6 >>> From: "asterisk" <sip:1...@networklab.loc>;tag=as0a530a8d >>> To: <sip:192.168.100.145>;tag=b27e1a1d33761e85846fc98f5f3a7e58.93df >>> ---> This line ins question. >>> Call-ID: 631e893f75da720865e8468132884...@networklab.loc >>> CSeq: 102 OPTIONS >>> Contact: <sip:1300@192.168.100.145:5062>;expires=3600 >>> Server: kamailio (4.1.2 (x86_64/linux)) >>> Content-Length: 0 >>> >>> >>> Slava. >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users@lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users@lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users@lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users@lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >
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