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 > >
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