Re: [SR-Users] Kamailio, CPL, SIP CGI and alternatives.
list (nil) Jul 8 13:16:24 kamailio /usr/local/sbin/kamailio[4246]: DEBUG: core [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=2 Jul 8 13:16:24 kamailio /usr/local/sbin/kamailio[4247]: DEBUG: core [xavp.c:448]: xavp_destroy_list(): destroying xavp list (nil) Jul 8 13:16:24 kamailio /usr/local/sbin/kamailio[4246]: DEBUG: core [parser/msg_parser.c:515]: parse_headers(): parse_headers: this is the first via Jul 8 13:16:24 kamailio /usr/local/sbin/kamailio[4247]: DEBUG: core [receive.c:296]: receive_msg(): receive_msg: cleaning up Jul 8 13:16:24 kamailio /usr/local/sbin/kamailio[4246]: DEBUG: core [receive.c:152]: receive_msg(): After parse_msg... Jul 8 13:16:24 kamailio /usr/local/sbin/kamailio[4246]: DEBUG: core [receive.c:193]: receive_msg(): preparing to run routing scripts... Jul 8 13:16:24 kamailio /usr/local/sbin/kamailio[4246]: DEBUG: core [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: tag=8661e39ff82910c16e76248ab1f6bbfe.08a3 Jul 8 13:16:24 kamailio /usr/local/sbin/kamailio[4246]: DEBUG: core [parser/parse_addr_spec.c:893]: parse_addr_spec(): end of header reached, state=29 Jul 8 13:16:24 kamailio /usr/local/sbin/kamailio[4246]: DEBUG: core [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: get_hdr_field: To [68]; uri=[sip:4111@192.168.0.197] Jul 8 13:16:24 kamailio /usr/local/sbin/kamailio[4246]: DEBUG: core [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body [ sip:4111@192.168.0.197] Jul 8 13:16:24 kamailio /usr/local/sbin/kamailio[4246]: DEBUG: maxfwd [mf_funcs.c:85]: is_maxfwd_present(): value = 70 Jul 8 13:16:24 kamailio /usr/local/sbin/kamailio[4246]: DEBUG: core [parser/msg_parser.c:170]: get_hdr_field(): get_hdr_field: cseq CSeq: 705 ACK Jul 8 13:16:24 kamailio /usr/local/sbin/kamailio[4246]: DEBUG: core [parser/msg_parser.c:204]: get_hdr_field(): DEBUG: get_hdr_body : content_length=0 Jul 8 13:16:24 kamailio /usr/local/sbin/kamailio[4246]: DEBUG: core [parser/msg_parser.c:106]: get_hdr_field(): found end of header Jul 8 13:16:24 kamailio /usr/local/sbin/kamailio[4246]: DEBUG: core [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: tag=olakr Jul 8 13:16:24 kamailio /usr/local/sbin/kamailio[4246]: DEBUG: core [parser/parse_addr_spec.c:893]: parse_addr_spec(): end of header reached, state=29 Jul 8 13:16:24 kamailio /usr/local/sbin/kamailio[4246]: DEBUG: sanity [mod_sanity.c:255]: w_sanity_check(): sanity checks result: 1 Jul 8 13:16:24 kamailio /usr/local/sbin/kamailio[4246]: DEBUG: core [msg_translator.c:204]: check_via_address(): check_via_address(192.168.0.167, 192.168.0.167, 0) Jul 8 13:16:24 kamailio /usr/local/sbin/kamailio[4246]: DEBUG: siputils [checks.c:106]: has_totag(): totag found Jul 8 13:16:24 kamailio /usr/local/sbin/kamailio[4246]: DEBUG: rr [loose.c:113]: find_first_route(): No Route headers found Jul 8 13:16:24 kamailio /usr/local/sbin/kamailio[4246]: DEBUG: rr [loose.c:929]: loose_route(): There is no Route HF Jul 8 13:16:24 kamailio /usr/local/sbin/kamailio[4246]: DEBUG: tm [t_lookup.c:1072]: t_check_msg(): DEBUG: t_check_msg: msg id=5 global id=4 T start=0x Jul 8 13:16:24 kamailio /usr/local/sbin/kamailio[4246]: DEBUG: tm [t_lookup.c:527]: t_lookup_request(): t_lookup_request: start searching: hash=14256, isACK=1 Jul 8 13:16:24 kamailio /usr/local/sbin/kamailio[4246]: DEBUG: tm [t_lookup.c:470]: matching_3261(): DEBUG: RFC3261 transaction matched, tid=assidktr Jul 8 13:16:24 kamailio /usr/local/sbin/kamailio[4246]: DEBUG: tm [t_lookup.c:726]: t_lookup_request(): DEBUG: t_lookup_request: transaction found (T=0xb35662a8) Jul 8 13:16:24 kamailio /usr/local/sbin/kamailio[4246]: DEBUG: tm [t_lookup.c:1141]: t_check_msg(): DEBUG: t_check_msg: msg id=5 global id=5 T end=0xb35662a8 Jul 8 13:16:24 kamailio /usr/local/sbin/kamailio[4246]: DEBUG: tm [t_reply.c:1663]: cleanup_uac_timers(): DEBUG: cleanup_uac_timers: RETR/FR timers reset Jul 8 13:16:24 kamailio /usr/local/sbin/kamailio[4246]: DEBUG: core [timer.c:595]: timer_add_safe(): timer_add called on an active timer 0xb35662f0 (0xb33b5888, 0xb33b5888), flags 201 Jul 8 13:16:24 kamailio /usr/local/sbin/kamailio[4246]: DEBUG: tm [t_funcs.c:180]: put_on_wait(): tm: put_on_wait: transaction 0xb35662a8 already on wait Jul 8 13:16:24 kamailio /usr/local/sbin/kamailio[4246]: DEBUG: core [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Jul 8 13:16:24 /usr/local/sbin/kamailio[4246]: last message repeated 5 times Jul 8 13:16:24 kamailio /usr/local/sbin/kamailio[4246]: DEBUG: core [xavp.c:448]: xavp_destroy_list(): destroying xavp list (nil) Jul 8 13:16:24 kamailio /usr/local/sbin/kamailio[4246]: DEBUG: core [receive.c:296]: receive_msg(): receive_msg: cleaning up root@kamailio:/var/log# 2014-07-07 7:49 GMT+02:00 Daniel-Constantin Mierla mico...@gmail.com: Hello, On 05/07/14 10:05, LAA wrote: Hi all, I'm looking for a mechanism to let users personalize the behaviour of their profile
[SR-Users] Kamailio, CPL, SIP CGI and alternatives.
Hi all, I'm looking for a mechanism to let users personalize the behaviour of their profile, uploading configuration changes to Kamailio. For example voice mail redirecion, time-based routing, etc. I'm testing CPL scripts (cpl-c module with Kamailio 4.1.3), but I have problems with the behaviour of some features. reject and redirect seem to be working fine, but proxy is not working. ¿Has anybody experienced the same problems? I could hardly find any post regarding CPL scripting with Kamailio. What are the most commonly used alternatives for this purpose with Kamailio? ¿SIP CGI? ¿any other? Best regards. Luis. ___ 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
Re: [SR-Users] Webrtc: Don't catch 488 between JSSIP and SIP UA's
... validate Origin - make sure the client is from an
# authorised website. For example,
#
# if ($hdr(Origin) != http://communicator.MY_DOMAIN;
# $hdr(Origin) != https://communicator.MY_DOMAIN;) {
# xlog(L_WARN, Unauthorised client $hdr(Origin)\n);
# xhttp_reply(403, Forbidden, , );
# exit;
# }
# Optional... perform HTTP authentication
# ws_handle_handshake() exits (no further configuration file
# processing of the request) when complete.
if (ws_handle_handshake())
{
# Optional... cache some information about the
# successful connection
exit;
}
}
xhttp_reply(404, Not Found, , );
}
event_route[websocket:closed] {
xlog(L_INFO, WebSocket connection from $si:$sp has closed\n);
}
#!endif
#!ifdef WITH_WEBRTCGW
onreply_route[REPLY_TO_WS] {
xlog(L_INFO, Reply from softphone: $rs);
if (t_check_status(183)) {
change_reply_status(180, Ringing);
remove_body();
exit;
}
if(!(status=~[12][0-9][0-9]))
return;
rtpproxy_manage(froc+SP);
route(NATMANAGE);
}
onreply_route[REPLY_FROM_WS] {
xlog(L_INFO, Reply from webrtc client: $rs);
if(status=~[12][0-9][0-9]) {
rtpproxy_manage(froc-sp);
route(NATMANAGE);
}
}
onreply_route[MANAGE_CLASSIC_REPLY] {
xlog(L_INFO, Boring reply from softphone: $rs);
if(status=~[12][0-9][0-9]) {
rtpproxy_manage(co);
route(NATMANAGE);
}
}
#!endif
2014-06-02 21:49 GMT+02:00 LAA ornitorrinco7...@gmail.com:
Apologize. Previous message was too long.
L.
El 02/06/2014 20:25, LAA ornitorrinco7...@gmail.com escribió:
Hi all,
Another guy strugling his mind trying to get a configuration to enable
calls between WebRTC UA (JSSIP) to standard SIP UA (Twinkle or SjPhone)
I've been working with the examples that were shared by Carlos Ruiz Diaz
and Peter Dunkley (thanks to both).
http://www.slideshare.net/crocodilertc/webrtc-websockets
http://caruizdiaz.com/2014/02/26/webrtc-kamailio/
Kamailio is not running behind NAT. I'm using rtpproxy-ng module with
Kamailio 4.1.3, and Rtpengine.
I share a link with my current configuration, wich is based in Peters
example, with websocket support from websocket.cfg example.
- Calls between SIP standard UA's are working OK. I have some endpoint
behind nat.
- Calls between JSIP UA's are working OK. So, websocket support is
running.
- Calls from JSIP and Twinkle are NOT WORKING OK. sip UA send's back a
488 response, and Kamailio send it back to JSSIP (Incompatible SDP).
- Calls from Twinkle to Jsip are NOT WORKING OK: Kamailio sends an
INVITE to JSIP, and it returns an error. And Kamailio sends 488 to Twinkle.
It seems as if Kamailio is not catching 488. I share a snippet of my
config, and links to tcpdump captures:
https://www.dropbox.com/s/i7c9ty57oauujc4/fromws0.pcap
https://www.dropbox.com/s/q3q30pgzvdoswts/kamailio.cfg
https://www.dropbox.com/s/rqtjwcbgg1foaoq/tows0.pcap
What am I missing?
Best regards.
Luis.
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[SR-Users] Webrtc: Don't catch 488 between JSSIP and SIP UA's
Apologize. Previous message was too long. L. El 02/06/2014 20:25, LAA ornitorrinco7...@gmail.com escribió: Hi all, Another guy strugling his mind trying to get a configuration to enable calls between WebRTC UA (JSSIP) to standard SIP UA (Twinkle or SjPhone) I've been working with the examples that were shared by Carlos Ruiz Diaz and Peter Dunkley (thanks to both). http://www.slideshare.net/crocodilertc/webrtc-websockets http://caruizdiaz.com/2014/02/26/webrtc-kamailio/ Kamailio is not running behind NAT. I'm using rtpproxy-ng module with Kamailio 4.1.3, and Rtpengine. I share a link with my current configuration, wich is based in Peters example, with websocket support from websocket.cfg example. - Calls between SIP standard UA's are working OK. I have some endpoint behind nat. - Calls between JSIP UA's are working OK. So, websocket support is running. - Calls from JSIP and Twinkle are NOT WORKING OK. sip UA send's back a 488 response, and Kamailio send it back to JSSIP (Incompatible SDP). - Calls from Twinkle to Jsip are NOT WORKING OK: Kamailio sends an INVITE to JSIP, and it returns an error. And Kamailio sends 488 to Twinkle. It seems as if Kamailio is not catching 488. I share a snippet of my config, and links to tcpdump captures: https://www.dropbox.com/s/i7c9ty57oauujc4/fromws0.pcap https://www.dropbox.com/s/q3q30pgzvdoswts/kamailio.cfg https://www.dropbox.com/s/rqtjwcbgg1foaoq/tows0.pcap What am I missing? Best regards. Luis. ___ 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
[SR-Users] Problems migrating from rtpproxy/rtoproxy.so to rtpproxy-ng/rtpengine
On 05/28/14 13:31, LAA wrote: * Hi all, * * I'm currently running a pilot with kamailio 4.1.3 stable, and I would ** like to test WebRTC Capabilities. Websockets Support is runnig OK, and ** now I'm trying to deal with calls between WebTRC and legacy softphones. ** I have installed rtpengine (as it a replacement for old mediaproxy-ng), ** and it is running: * * rtpengine --table=0 --ip=192.168.0.197 --listen-udp=127.0.0.1:1 http://127.0.0.1:1 * You need to use the --listen-ng=... option instead of listen-udp for this to work. cheers Thanks Richard, of course it works. Regards. Luis. ___ 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
[SR-Users] Problems migrating from rtpproxy/rtoproxy.so to rtpproxy-ng/rtpengine
Hi all, I'm currently running a pilot with kamailio 4.1.3 stable, and I would like to test WebRTC Capabilities. Websockets Support is runnig OK, and now I'm trying to deal with calls between WebTRC and legacy softphones. I have installed rtpengine (as it a replacement for old mediaproxy-ng), and it is running: rtpengine --table=0 --ip=192.168.0.197 --listen-udp=127.0.0.1:1 As a first step, I'm trying to migrate the standard NAT handling configuration to the new module. (...) #loadmodule rtpproxy.so loadmodule rtpproxy-ng.so (...) #modparam(rtpproxy, rtpproxy_sock, udp:127.0.0.1:7624) modparam(rtpproxy-ng, rtpproxy_sock, udp:127.0.0.1:1) NAT hanling is not working, and I catched this message from /var/log/syslog May 28 18:55:35 kamailio rtpengine[2474]: Failed to properly parse UDP command line '4547_3 d7:command4:pinge' from 127.0.0.1:60874, using fallback RE May 28 18:55:35 kamailio rtpengine[2474]: Failed to properly parse UDP command line '4545_4 d7:command4:pinge' from 127.0.0.1:57236, using fallback RE And from Kamailio's log: May 28 19:23:47 kamailio /usr/local/sbin/kamailio[4868]: ERROR: rtpproxy-ng [rtpproxy.c:1425]: rtpp_test(): proxy responded with invalid response May 28 19:23:47 kamailio /usr/local/sbin/kamailio[4869]: ERROR: rtpproxy-ng [rtpproxy.c:1425]: rtpp_test(): proxy responded with invalid response Is rtpproxy-ng module compatible with rtpengine? Best regards Luis. ___ 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
[SR-Users] Problem with forward on busy.
Hello Daniel, I think that debugger module was released with Kamailio version 3.1.0. wasn't it? As I'm running Kamailio 3.0.0, I have set up debug=9. Here you have my config file, the raw capture of the call and the lines in kamailio log file. I see some messages regarding cpl-c module, and I was experiencing some problems with it too. I tested without cpl-c module and I get the same trace but no messages in log file. Thanks in advance. Regards. Luis. https://www.dropbox.com/s/t79ic02b6h85ppj/onbusy_DCM_txt https://www.dropbox.com/s/20c5pqilocey052/kamailio.cfg LOG *** ul 28 23:24:43 kamailio /usr/local/sbin/kamailio[1618] : : tm [t_hooks.c:211]: BUG:tm:register_tmcb: no transaction found Jul 28 23:24:43 kamailio /usr/local/sbin/kamailio[1618]: ERROR: cpl-c [cpl_proxy.h:482]: failed to register TMCB_RESPONSE_OUT callback Jul 28 23:24:43 kamailio /usr/local/sbin/kamailio[1618]: ERROR: cpl-c [cpl_run.c:1040]: runtime error Jul 28 23:24:43 kamailio /usr/local/sbin/kamailio[1618]: ERROR: tm [tm.c:1168]: ERROR: t_reply: cannot send a t_reply to a message for which no T-state has been established ___ 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
Re: [SR-Users] Problem with forward on busy.
OK, Thanks Daniel. I will upgrade. Regards. Luis. 2013/7/29 Daniel-Constantin Mierla mico...@gmail.com Hello, if you run release 3.0.0 then you should update at least to latest in its series 3.0.x. But I strongly recommend to go for a newer release, the best is 4.0.x, or at least 3.3.x. Anyhow, always upgrade in the releases series, so whenever you run x.y.z, be sure that you have the highest .z version, because a change only of z number means a release with fixes for x.y. Overall, 3.0.0 is really old to remember if there were any issues -- note that was the first version of kamailio+ser code together, and we had to tune it in following 3.0.x versions to be compatible with what everyone expected from kamailio or ser. Cheers, Daniel On 7/29/13 10:52 PM, LAA wrote: Hello Daniel, I think that debugger module was released with Kamailio version 3.1.0. wasn't it? As I'm running Kamailio 3.0.0, I have set up debug=9. Here you have my config file, the raw capture of the call and the lines in kamailio log file. I see some messages regarding cpl-c module, and I was experiencing some problems with it too. I tested without cpl-c module and I get the same trace but no messages in log file. Thanks in advance. Regards. Luis. https://www.dropbox.com/s/t79ic02b6h85ppj/onbusy_DCM_txt https://www.dropbox.com/s/20c5pqilocey052/kamailio.cfg LOG *** ul 28 23:24:43 kamailio /usr/local/sbin/kamailio[1618] : : tm [t_hooks.c:211]: BUG:tm:register_tmcb: no transaction found Jul 28 23:24:43 kamailio /usr/local/sbin/kamailio[1618]: ERROR: cpl-c [cpl_proxy.h:482]: failed to register TMCB_RESPONSE_OUT callback Jul 28 23:24:43 kamailio /usr/local/sbin/kamailio[1618]: ERROR: cpl-c [cpl_run.c:1040]: runtime error Jul 28 23:24:43 kamailio /usr/local/sbin/kamailio[1618]: ERROR: tm [tm.c:1168]: ERROR: t_reply: cannot send a t_reply to a message for which no T-state has been established -- Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ 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
Re: [SR-Users] Problem with forward on busy
Hello Daniel,
I have tried without append_branch(); and it just don't create the new
branch and it sends back the 486 message to the UAC that originated the
call:
if (t_check_status(486|408)) {
revert_uri();
prefix(voicemail);
remove_hf(P-App-Name);
append_hf(P-App-Name: voicemail\r\n);
append_hf(P-App-Param: mod=box;usr= $rU;dom=sipproxy.a.com
;uid=$rU;did=sipproxy.a.com;\r\n);
rewritehostport(192.168.0.197:5080);
$du = $null;
#$du = sip:192.168.0.197;
#append_branch();
t_relay();
}
}
|Time | 192.168.3.20 |
192.168.0.167 |
| | | 192.168.0.197 |
|3,415| INVITE SDP ( telephone-event)
| |SIP From: sip:4095@192.168.0.197
To:sip:4440@192.168.0.197
| |(5060) -- (5060) | |
|3,420| 407 Proxy Authentication Required
| |SIP Status
| |(5060) -- (5060) | |
|3,422| ACK | | |SIP
Request
| |(5060) -- (5060) | |
|3,422| INVITE SDP ( telephone-event)
| |SIP From: sip:4095@192.168.0.197
To:sip:4440@192.168.0.197
| |(5060) -- (5060) | |
|3,435| 100 trying -- your call is important to us
| |SIP Status
| |(5060) -- (5060) | |
|3,436| | INVITE SDP (
telephone-event) |SIP Request
| | |(5060) -- (5060) |
|3,437| | 100 Trying| |SIP
Status
| | |(5060) -- (5060) |
|3,437| | 486 Busy Here |SIP
Status
| | |(5060) -- (5060) |
|3,441| | ACK | |SIP
Request
| | |(5060) -- (5060) |
|3,459| 486 Busy Here | |SIP
Status
| |(5060) -- (5060) | |
|3,461| ACK | | |SIP
Request
| |(5060) -- (5060) | |
Regards
Luis
2013/7/25 Daniel Tryba dan...@pocos.nl
On Wednesday 24 July 2013 20:41:04 LAA wrote:
May be I'm loosing something. I have changed my config as you
suggested (I thing so...):
if (t_check_status(486|408)) {
...
$du = $null;
#$du = sip:192.168.0.197;
append_branch();
t_relay();
Did you try without the append_branch()?
--
POCOS B.V. - Croy 9c - 5653 LC Eindhoven
Telefoon: 040 293 8661 - Fax: 040 293 8658
http://www.pocos.nl/ - http://www.sipo.nl/
K.v.K. Eindhoven 17097024
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Re: [SR-Users] Problem with forward on busy
OK, Daniel and thanks for your help,
I see that you don't append brach but you are calling route(RELAY) instead
of t_relay() directly. I have tryed with this configuration within failure
route:
if (t_check_status(486|408)) {
#revert_uri();
prefix(voicemail);
remove_hf(P-App-Name);
append_hf(P-App-Name: voicemail\r\n);
append_hf(P-App-Param: mod=box;usr= $rU;dom=sipproxy.a.com
;uid=$rU;did=sipproxy.a.com;\r\n);
rewritehostport(192.168.0.197:5080);
$du = $null;
#append_branch();
route(RELAY);
#t_relay();
}
}
And kamailio gets into a strange behavior
|Time | 192.168.3.20 |
192.168.0.167 |
| | | 192.168.0.197 |
|3,366| INVITE SDP ( telephone-event)
| |SIP From: sip:4095@192.168.0.197
To:sip:4440@192.168.0.197
| |(5060) -- (5060) | |
|3,370| 407 Proxy Authentication Required
| |SIP Status
| |(5060) -- (5060) | |
|3,380| ACK | | |SIP
Request
| |(5060) -- (5060) | |
|3,382| INVITE SDP ( telephone-event)
| |SIP From: sip:4095@192.168.0.197
To:sip:4440@192.168.0.197
| |(5060) -- (5060) | |
|3,393| 100 trying -- your call is important to us
| |SIP Status
| |(5060) -- (5060) | |
|3,394| | INVITE SDP (
telephone-event) |SIP Request
| | |(5060) -- (5060) |
|3,395| | 100 Trying| |SIP
Status
| | |(5060) -- (5060) |
|3,395| | 486 Busy Here |SIP
Status
| | |(5060) -- (5060) |
|3,398| | ACK | |SIP
Request
| | |(5060) -- (5060) |
|3,416| 500 I'm terribly sorry, server error occurred
...SL) | |SIP Status
| |(5060) -- (5060) | |
|3,416| 486 Busy Here | |SIP
Status
| |(5060) -- (5060) | |
|3,418| ACK | | |SIP
Request
| |(5060) -- (5060) | |
|3,418| ACK | | |SIP
Request
| |(5060) -- (5060) | |
|3,872| 486 Busy Here | |SIP
Status
| |(5060) -- (5060) | |
|3,873| ACK | | |SIP
Request
| |(5060) -- (5060) | |
|4,875| 486 Busy Here | |SIP
Status
| |(5060) -- (5060) | |
|4,876| ACK | | |SIP
Request
| |(5060) -- (5060) | |
Are you using this sequence within failure route? or in the call routing
section? I'm using this sequence in the route section that is working OK:
if ($rU=~^voicemail.*) {
remove_hf(P-App-Name);
append_hf(P-App-Name: voicemail\r\n);
append_hf(P-App-Param: mod=box;usr=$rU;dom=sipproxy.a.com
;uid=$rU;did=sipproxy.a.com;\r\n);
$ru = sip: + $rU + @ + 192.168.0.197:5080;
route(RELAY);
exit;
}
The problem is when I try to get a call forwarded by kamailio to voice mail
when it gets a busy message to the destination message. In your
implementation are you expecting a 302 (temporary unavailable) message from
the destination UAC?
Regards.
L.
2013/7/25 Daniel Tryba dan...@pocos.nl
On Thursday 25 July 2013 16:30:21 you wrote:
if (t_check_status(486|408)) {
revert_uri();
prefix(voicemail);
remove_hf(P-App-Name);
append_hf(P-App-Name: voicemail\r\n);
append_hf(P-App-Param: mod=box;usr= $rU;dom=sipproxy.a.com
;uid=$rU;did=sipproxy.a.com;\r\n);
rewritehostport(192.168.0.197:5080);
$du = $null;
#$du = sip:192.168.0.197;
#append_branch();
t_relay();
Taking a look at my config which I found to work after the long struggle
you
are experiencing right now.
if($avp(dst_voicemail))
{
$du=$null;
$ru = sip:tovm- + $avp(dst_voicemail) + @ +
Re: [SR-Users] Problem with forward on busy
I have checked that I'm , experiencing the same problem when the
redirection to voicemail is originated by the destination UAC via 302
message. Kamailio sends the packet to the destination UAC, even when I set
$du to null.
??¿?¿?¿¿??¿
if ($rU=~^voicemail.*) {
$du = $null;
remove_hf(P-App-Name);
append_hf(P-App-Name: voicemail\r\n);
append_hf(P-App-Param: mod=box;usr=$rU;dom=sipproxy.a.com
;uid=$rU;did=sipproxy.a.com;\r\n);
$ru = sip: + $rU + @ + 192.168.0.197:5080;
$du = $null;
route(RELAY);
exit;
}
Conv.| Time| 192.168.3.20 |
192.168.0.167 |
| | | 192.168.0.197 |
0|3,574| INVITE SDP ( telephone-event)
| |SIP From: sip:4095@192.168.0.197
To:sip:4440@192.168.0.197
| |(5060) -- (5060) | |
0|3,575| 407 Proxy Authentication Required
| |SIP Status
| |(5060) -- (5060) | |
0|3,577| ACK | |
|SIP Request
| |(5060) -- (5060) | |
0|3,577| INVITE SDP ( telephone-event)
| |SIP From: sip:4095@192.168.0.197
To:sip:4440@192.168.0.197
| |(5060) -- (5060) | |
0|3,584| 100 trying -- your call is important to
us | |SIP Status
| |(5060) -- (5060) | |
0|3,585| | INVITE SDP (
telephone-event) |SIP Request
| | |(5060) -- (5060) |
0|3,587| | 100 Trying|
|SIP Status
| | |(5060) -- (5060) |
0|3,587| | 302 Moved Temporarily
|SIP Status
| | |(5060) -- (5060) |
0|3,588| | ACK |
|SIP Request
| | |(5060) -- (5060) |
0|3,592| 302 Moved Temporarily |
|SIP Status
| |(5060) -- (5060) | |
0|3,594| ACK | |
|SIP Request
| |(5060) -- (5060) | |
-
1|3,596| INVITE SDP ( telephone-event)
| |SIP From: sip:4095@192.168.0.197
To:sip:voicemail4440@192.168.0.167:5060
| |(5060) -- (5060) | |
1|3,596| 407 Proxy Authentication Required
| |SIP Status
| |(5060) -- (5060) | |
1|3,600| ACK | |
|SIP Request
| |(5060) -- (5060) | |
1|3,601| INVITE SDP ( telephone-event)
| |SIP From: sip:4095@192.168.0.197
To:sip:voicemail4440@192.168.0.167:5060
| |(5060) -- (5060) | |
1|3,608| 100 trying -- your call is important to
us | |SIP Status
| |(5060) -- (5060) | |
1|3,608| | INVITE SDP (
telephone-event) |SIP Request
| | |(5060) -- (5060) |
1|3,608| | 404 Not Found
|SIP Status
| | |(5060) -- (5060) |
1|3,609| | ACK |
|SIP Request
| | |(5060) -- (5060) |
1|3,614| 404 Not Found |
|SIP Status
| |(5060) -- (5060) | |
1|3,615| ACK | |
|SIP Request
| |(5060) -- (5060) | |
2013/7/25 LAA ornitorrinco7...@gmail.com
OK, Daniel and thanks for your help,
I see that you don't append brach but you are calling route(RELAY) instead
of t_relay() directly. I have tryed with this configuration within failure
route:
if (t_check_status(486|408)) {
#revert_uri();
prefix(voicemail);
remove_hf(P-App-Name);
append_hf(P-App-Name: voicemail\r\n);
append_hf(P-App-Param: mod=box;usr= $rU;dom=sipproxy.a.com
;uid=$rU;did=sipproxy.a.com;\r\n);
rewritehostport(192.168.0.197:5080);
$du = $null;
#append_branch();
route(RELAY);
#t_relay
[SR-Users] Problem with forward on busy
Hi Carsten, I forgot exit!! Anyway, aas this was the last part of failure route it don't make any differente. Many Thanks. L. ___ 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
Re: [SR-Users] Problem with forward on busy
Hello Hero,
Thanks for your help.
May be I'm loosing something. I have changed my config as you
suggested (I thing so...):
if (t_check_status(486|408)) {
revert_uri();
prefix(voicemail);
remove_hf(P-App-Name);
append_hf(P-App-Name: voicemail\r\n);
append_hf(P-App-Param: mod=box;usr= $rU;dom=sipproxy.a.com;uid=$
rU;did=sipproxy.a.com;\r\n);
rewritehostport(192.168.0.197:5080);
$du = $null;
#$du = sip:192.168.0.197;
append_branch();
t_relay();
}
}
Kamailio sends back 200 OK to the UAC that originated the call, but it
never sends the new INVITE
|Time | 192.168.3.20
| 192.168.0.167 |
| | | 192.168.0.197 |
|3,151| INVITE SDP ( telephone-event)
| |SIP From: sip:4095@192.168.0.197
To:sip:4440@192.168.0.197
| |(5060) -- (5060) | |
|3,159| 407 Proxy Authentication Required
| |SIP Status
| |(5060) -- (5060) | |
|3,161| ACK | | |SIP
Request
| |(5060) -- (5060) | |
|3,161| INVITE SDP ( telephone-event)
| |SIP From: sip:4095@192.168.0.197
To:sip:4440@192.168.0.197
| |(5060) -- (5060) | |
|3,174| 100 trying -- your call is important to us
| |SIP Status
| |(5060) -- (5060) | |
|3,174| | INVITE SDP (
telephone-event) |SIP Request
| | |(5060) -- (5060) |
|3,176| | 100 Trying| |SIP
Status
| | |(5060) -- (5060) |
|3,177| | 486 Busy Here |SIP
Status
| | |(5060) -- (5060) |
|3,180| | ACK | |SIP
Request
| | |(5060) -- (5060) |
|3,195| 200 OK SDP ( telephone-event)
| |SIP Status
| |(5060) -- (5060) | |
|3,200| ACK | | |SIP
Request
| |(5060) -- (5060) | |
|3,213| RTP (GSM) | | |RTP
Num packets:204 Duration:4.069s SSRC:0x8494958
| |(49222) -- (10028) | |
|7,288| BYE | | |SIP
Request
| |(5060) -- (5060) | |
|7,295| 200 OK| | |SIP
Status
| |(5060) -- (5060) | |
what am I loosing?
Regards
LAA
*
had the same issue here. you have to manually set $du=$null, else it
doesn't get reset for the failure branch.
On 7/23/13, LAA ornitorrinco7424 at gmail.com
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
wrote:
* Hi all, I'm running Kamailio 3.0.0, with SEMS integration as Media
Server for Voice** mail. I'm trying to get a configuration to forward calls
on busy to voice** mail. I have followed without success some examples. I'm
using** revert_uri(), rewritehostport() and append_branch(), within
failure_route.** It seems to be modifying R-URI properly, and generating the
new branch, but** Kamailio is sending the new invite packet to the IP address
of the original** destination UAC, and not to the IP address of the
voicemail, that was** indicated in the R-URI. Here you can see the packet
flow: |Time | 192.168.3.20** | 192.168.0.167
|** | | | 192.168.0.197 |** |5,069
| INVITE SDP ( telephone-event)** | |SIP From:
sip:4095 at 192.168.0.197
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users** To:sip:4440
at 192.168.0.197
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users** |
|(5060) -- (5060) | |** |5,071|
407 Proxy Authentication Required** | |SIP Status**
| |(5060) -- (5060) | |**
|5,074| ACK | | |SIP**
Request** | |(5060) -- (5060) |
|** |5,076| INVITE SDP ( telephone-event)** |
|SIP From: sip:4095 at 192.168.0.197
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users** To:sip:4440
at 192.168.0.197
http://lists.sip
[SR-Users] Problem with forward on busy
# redirect based on 3xx replies.
##if (t_check_status(3[0-9][0-9])
) {
##t_reply(404,Not found);
##exit;
##}
# uncomment the following lines if you want to redirect the failed
# calls to a different new destination
if (t_check_status(486|408)) {
revert_uri();
prefix(voicemail);
remove_hf(P-App-Name);
append_hf(P-App-Name: voicemail\r\n);
append_hf(P-App-Param: mod=box;usr= $rU;dom=sipproxy.a.com
;uid=$rU;did=sipproxy.a.com;\r\n);
$ru = sip: + $rU + @ + 192.168.0.197:5080;
#rewritehostport(192.168.0.197:5080);
#append_branch(sip:4888@192.168.0.102);
append_branch();
# do not set the missed call flag again
t_relay();
}
}
Has anybody experienced this problem? Any help would be wellcome
Best Regards
LAA
___
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
