Hello everyone! I need to relay 486/408/... other failure codes back to initial INVITE server. Here http://lists.sip-router.org/pipermail/sr-users/2010-November/066382.html is recommended just to exit failure_route, but that didn't work for me. I need that to let Freeswitch know which cause has ended the call. Now Kamailio just sends ACK to endpoint on receiving 486 BUSY. Would you kindly tell me how to achieve that? Thanks in advance
I use sngrep to track view the flow and I'm pretty sure it's accurate enough to tell me that. Here's relay route: route[RELAY] {
# enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; }
and here's reply routes
# Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); }
# Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); }
# Manage failure routing cases failure_route[MANAGE_FAILURE] {
if (t_check_status("486")) { exit; } route(NATMANAGE);
if (t_is_canceled()) { exit; }
}
However when endpoint replies with 486 BUSY I can't see that on FS, Kamailio just sends 408 REQ TERM after some amount of time
2015-12-15 13:34 GMT+02:00 Alex Balashov abalashov@evaristesys.com:
From what you describe, the reply should be going back to the sender. Are you absolutely sure that it's not? If so, are there any other actions you could be taking somewhere to drop it, such as in an onreply_route?
ACKs to negative final replies are hop-by-hop, so the ACK you're seeing directly from the proxy to the UAS is normal.
-- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States
Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
Sent from my BlackBerry. *From: *Alexandru Covalschi *Sent: *Tuesday, December 15, 2015 05:03 *To: *Kamailio (SER) - Users Mailing List *Reply To: *Kamailio (SER) - Users Mailing List *Subject: *[SR-Users] Relaying failure codes back to initial server
Hello everyone! I need to relay 486/408/... other failure codes back to initial INVITE server. Here http://lists.sip-router.org/pipermail/sr-users/2010-November/066382.html is recommended just to exit failure_route, but that didn't work for me. I need that to let Freeswitch know which cause has ended the call. Now Kamailio just sends ACK to endpoint on receiving 486 BUSY. Would you kindly tell me how to achieve that? Thanks in advance -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/
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
Maybe there is a parallel forking and one branch gets timed out (in this case 408 is selected against 486). How many INVITE requests do you see being sent out? Or you can eventually make the sip trace available for viewing on this mailing list or some web site/pastebin out there.
Cheers, Daniel
On 15/12/15 12:54, Alexandru Covalschi wrote:
I use sngrep to track view the flow and I'm pretty sure it's accurate enough to tell me that. Here's relay route: route[RELAY] {
# enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route"))t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; }
and here's reply routes
# Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); }
# Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); }
# Manage failure routing cases failure_route[MANAGE_FAILURE] {
if (t_check_status("486")) { exit; } route(NATMANAGE);
if (t_is_canceled()) { exit; }}
However when endpoint replies with 486 BUSY I can't see that on FS, Kamailio just sends 408 REQ TERM after some amount of time
2015-12-15 13:34 GMT+02:00 Alex Balashov <abalashov@evaristesys.com mailto:abalashov@evaristesys.com>:
From what you describe, the reply should be going back to the sender. Are you absolutely sure that it's not? If so, are there any other actions you could be taking somewhere to drop it, such as in an onreply_route? ACKs to negative final replies are hop-by-hop, so the ACK you're seeing directly from the proxy to the UAS is normal. -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 <tel:%2B1-800-250-5920> (toll-free) / +1-678-954-0671 <tel:%2B1-678-954-0671> (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ Sent from my BlackBerry. *From: *Alexandru Covalschi *Sent: *Tuesday, December 15, 2015 05:03 *To: *Kamailio (SER) - Users Mailing List *Reply To: *Kamailio (SER) - Users Mailing List *Subject: *[SR-Users] Relaying failure codes back to initial server Hello everyone! I need to relay 486/408/... other failure codes back to initial INVITE server. Here http://lists.sip-router.org/pipermail/sr-users/2010-November/066382.html is recommended just to exit failure_route, but that didn't work for me. I need that to let Freeswitch know which cause has ended the call. Now Kamailio just sends ACK to endpoint on receiving 486 BUSY. Would you kindly tell me how to achieve that? Thanks in advance -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 <tel:%2B37367398493> web: http://abs-telecom.com/ _______________________________________________ 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-- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/
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
Yes, sometimes there are more than one INVITE (custom platform specific behavior), in case of 1 INVITE per-dialogue it works nice. Can you suggest something to cover such cases?
2015-12-15 14:12 GMT+02:00 Daniel-Constantin Mierla miconda@gmail.com:
Maybe there is a parallel forking and one branch gets timed out (in this case 408 is selected against 486). How many INVITE requests do you see being sent out? Or you can eventually make the sip trace available for viewing on this mailing list or some web site/pastebin out there.
Cheers, Daniel
On 15/12/15 12:54, Alexandru Covalschi wrote:
I use sngrep to track view the flow and I'm pretty sure it's accurate enough to tell me that. Here's relay route: route[RELAY] {
# enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route"))t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; }
and here's reply routes
# Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); }
# Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); }
# Manage failure routing cases failure_route[MANAGE_FAILURE] {
if (t_check_status("486")) { exit; } route(NATMANAGE);
if (t_is_canceled()) { exit; }}
However when endpoint replies with 486 BUSY I can't see that on FS, Kamailio just sends 408 REQ TERM after some amount of time
2015-12-15 13:34 GMT+02:00 Alex Balashov abalashov@evaristesys.com:
From what you describe, the reply should be going back to the sender. Are you absolutely sure that it's not? If so, are there any other actions you could be taking somewhere to drop it, such as in an onreply_route?
ACKs to negative final replies are hop-by-hop, so the ACK you're seeing directly from the proxy to the UAS is normal.
-- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States
Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ http://www.csrpswitch.com/
Sent from my BlackBerry. *From: *Alexandru Covalschi *Sent: *Tuesday, December 15, 2015 05:03 *To: *Kamailio (SER) - Users Mailing List *Reply To: *Kamailio (SER) - Users Mailing List *Subject: *[SR-Users] Relaying failure codes back to initial server
Hello everyone! I need to relay 486/408/... other failure codes back to initial INVITE server. Here http://lists.sip-router.org/pipermail/sr-users/2010-November/066382.html http://lists.sip-router.org/pipermail/sr-users/2010-November/066382.html is recommended just to exit failure_route, but that didn't work for me. I need that to let Freeswitch know which cause has ended the call. Now Kamailio just sends ACK to endpoint on receiving 486 BUSY. Would you kindly tell me how to achieve that? Thanks in advance -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/
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
-- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu
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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Alex Balashov -
Alexandru Covalschi -
Daniel-Constantin Mierla