Hi Users,
I'm Setting UP the Voicemails by integrating with Asterisk and OpenSER,
After 32 sec or 6 ring, it has to go the Voicemail server of Asterisk,
In openser.cfg ........... is not hiiting the Asterisk server ............. ... any one help me ........ .... .... modparam("tm","fr_timer",6) modparam("tm","fr_inv_timer",24) modparam("tm","wt_timer",1) #mrodparam("tm", "ruri_matching", 0) #modparam("tm", "via1_matching", 0) modparam("avpops","avp_url","mysql://root:passwd@192.168.2.75/openser") modparam("avpops", "avp_table", "usr_preferences") modparam("avpops","avp_aliases","inv=i:15") ............... route { ...... if (loose_route()) { t_relay(); exit; }; if(is_method("INVITE") && !has_totag()) { xdbg("user [$ruri] has voicemail redirection enabled\n"); # backup R-URI avp_write("$ruri","$avp(inv)"); setflag(2); }; .............. .... route(1); } route[1] { if(isflagset(2)) { t_on_failure("1"); }; } failure_route[1] { log("----------------------------------------- \n"); if (t_was_cancelled()) { xdbg("transaction was cancelled by UAC\n"); return; } # restore initial uri avp_pushto("$ruri", "$avp(inv)"); prefix("9"); # route to Asterisk Media Server rewritehostport("192.168.2.75:5060"); append_branch(); t_relay("192.168.2.75:5060"); resetflag(2); }
.........................
Hello,
watch the network traffic with "ngrep -d any -qt port 5060" on your sip server system and see if the INVITE is sent to asterisk. You can plug some xlog("__message__") in your configuration to see how the request is processed and if the INVITE hit the failure_route.
Cheers, Daniel
On 01/05/07 14:57, raviprakash sunkara wrote:
Hi Users,
I'm Setting UP the Voicemails by integrating with Asterisk and OpenSER,
After 32 sec or 6 ring, it has to go the Voicemail server of Asterisk,
In openser.cfg ........... is not hiiting the Asterisk server ............. ... any one help me ........ .... .... modparam("tm","fr_timer",6) modparam("tm","fr_inv_timer",24) modparam("tm","wt_timer",1) #mrodparam("tm", "ruri_matching", 0) #modparam("tm", "via1_matching", 0) modparam("avpops","avp_url","mysql://root:passwd@192.168.2.75/openser") modparam("avpops", "avp_table", "usr_preferences") modparam("avpops","avp_aliases","inv=i:15") ............... route { ...... if (loose_route()) { t_relay(); exit; }; if(is_method("INVITE") && !has_totag()) { xdbg("user [$ruri] has voicemail redirection enabled\n"); # backup R-URI avp_write("$ruri","$avp(inv)"); setflag(2); }; .............. .... route(1); } route[1] { if(isflagset(2)) { t_on_failure("1"); }; } failure_route[1] { log("----------------------------------------- \n"); if (t_was_cancelled()) { xdbg("transaction was cancelled by UAC\n"); return; } # restore initial uri avp_pushto("$ruri", "$avp(inv)"); prefix("9"); # route to Asterisk Media Server rewritehostport(" 192.168.2.75:5060 http://192.168.2.75:5060"); append_branch(); t_relay("192.168.2.75:5060 http://192.168.2.75:5060"); resetflag(2); }
.........................
-- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara@hyperion-tech.com mailto:ravi.sunkara@hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara@hyperion-tech.com mailto:ravi.sunkara@hyperion-tech.com www.hyperion-tech.com http://www.hyperion-tech.com
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
In the route[1] you do not show a t_relay. Is this just be a typo or is the t_relay actually missing from this route block? If it is missing how do you initiate the relay which you hope to end up in voicemail?
Daniel-Constantin Mierla wrote:
Hello,
watch the network traffic with "ngrep -d any -qt port 5060" on your sip server system and see if the INVITE is sent to asterisk. You can plug some xlog("__message__") in your configuration to see how the request is processed and if the INVITE hit the failure_route.
Cheers, Daniel
On 01/05/07 14:57, raviprakash sunkara wrote:
Hi Users,
I'm Setting UP the Voicemails by integrating with Asterisk and OpenSER,
After 32 sec or 6 ring, it has to go the Voicemail server of Asterisk,
In openser.cfg ........... is not hiiting the Asterisk server ............. ... any one help me ........ .... .... modparam("tm","fr_timer",6) modparam("tm","fr_inv_timer",24) modparam("tm","wt_timer",1) #mrodparam("tm", "ruri_matching", 0) #modparam("tm", "via1_matching", 0) modparam("avpops","avp_url","mysql://root:passwd@192.168.2.75/openser") modparam("avpops", "avp_table", "usr_preferences") modparam("avpops","avp_aliases","inv=i:15") ............... route { ...... if (loose_route()) { t_relay(); exit; }; if(is_method("INVITE") && !has_totag()) { xdbg("user [$ruri] has voicemail redirection enabled\n"); # backup R-URI avp_write("$ruri","$avp(inv)"); setflag(2); }; .............. .... route(1); } route[1] { if(isflagset(2)) { t_on_failure("1"); }; } failure_route[1] { log("----------------------------------------- \n"); if (t_was_cancelled()) { xdbg("transaction was cancelled by UAC\n"); return; } # restore initial uri avp_pushto("$ruri", "$avp(inv)"); prefix("9"); # route to Asterisk Media Server rewritehostport(" 192.168.2.75:5060 http://192.168.2.75:5060"); append_branch(); t_relay("192.168.2.75:5060 http://192.168.2.75:5060"); resetflag(2); }
.........................
-- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara@hyperion-tech.com mailto:ravi.sunkara@hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara@hyperion-tech.com mailto:ravi.sunkara@hyperion-tech.com www.hyperion-tech.com http://www.hyperion-tech.com
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
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Daniel-Constantin Mierla -
raviprakash sunkara -
Steve Blair