Hello Dmitri,
Yes, the $ru is ok and contains the right domain name. Regards, Igor. De : sr-users [mailto:sr-users-boun...@lists.sip-router.org] De la part de Dmitri Savolainen Envoyé : vendredi 15 mai 2015 10:51 À : Kamailio (SER) - Users Mailing List Objet : Re: [SR-Users] Issue with Asterisk interconnection for VoiceMail And $ru is OK while sending to wrong (initial) IP? Did you try to set/check $du too? 2015-05-15 11:30 GMT+03:00 Igor Potjevlesch <igor.potjevle...@gmail.com <mailto:igor.potjevle...@gmail.com> >: Hello, I experienced a strange issue with some of VoIP accounts. When the INVITE comes into MANAGE_FAILURE, after timeout, the config identifies, with "dialplan", the right Asterisk instance that should handle the call for voicemail. This part is okay, and results in a new INVITE with the Request-URI formed with the right domain (eg. sip: <sip:%3cNUMBER%3e@asterisk3> <NUMBER>@asterisk3). Then, the request goes to RELAY. Here is the issue: sometimes, the request is forwarded to the IP of the UA (the one initially contacted) instead of the IP of Asterisk. I can't figure out the difference between a succeeded call and a failed one. If someone has an idea. Here is the config that handles the VoiceMail: failure_route[MANAGE_FAILURE] { […] if (isflagset(24)) { $avp(s:inv_timeout) = "5"; t_set_fr($avp(s:inv_timeout)*1000); if (avp_db_load("$to/username","$avp(s:vm_uri)/usr_vm")) { resetflag(24); avp_pushto("$ruri","$avp(s:vm_uri)"); # Dynamic routing if (avp_db_load("$ruri/username","$avp(s:client)/usr_fai")) { if (dp_translate("2","$avp(s:client)/$avp(s:dest)") == 1) { $ru = "sip:" + $rU + "@" + $avp(s:dest); } else { # Load default voicemail $avp(s:client) = "DEFAULT_VM"; dp_translate("2","$avp(s:client)/$avp(s:dest)"); $ru = "sip:" + $rU + "@" + $avp(s:dest); }; } else { # Load default voicemail $avp(s:client) = "DEFAULT_VM"; dp_translate("2","$avp(s:client)/$avp(s:dest)"); $ru = "sip:" + $rU + "@" + $avp(s:dest); } } else { xlog("L_WARN","time=[$Tf] call id=[$ci] call seq=[$cs] contact header=[$ct] from uri=[$fu] from tag=[$ft] request's method=[$rm] request's uri=[$ru] to uri=[$tu] to tag=[$tt] sip message id=[$mi] process id=[$pp] ip source=[$si] flags=[$mf], User have no mail box\n"); exit; }; prefix("710"); xlog("L_WARN","time=[$Tf] call id=[$ci] call seq=[$cs] contact header=[$ct] from uri=[$fu] from tag=[$ft] request's method=[$rm] request's uri=[$ru] to uri=[$tu] to tag=[$tt] sip message id=[$mi] process id=[$pp] ip source=[$si] flags=[$mf], failure route to Voice Mail\n"); route(RELAY); exit; } Regards, Igor. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Savolainen Dmitri
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