Re: [OpenSIPS-Users] Is there some sort of ring group implementation where users are dialled and just the first one to answer will get the cal?
Hi Chen-che. Thank you! I think I really should follow your suggestion. I'll check on t_relay () . Best Regards . RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 (Brasil) De: users-boun...@lists.opensips.org users-boun...@lists.opensips.org em nome de microx acmic...@gmail.com Enviado: quinta-feira, 23 de julho de 2015 04:43 Para: users@lists.opensips.org Assunto: Re: [OpenSIPS-Users] Is there some sort of ring group implementation where users are dialled and just the first one to answer will get the cal? Hi, In my own experience, one way to achieve ring group implementation is to make those users register use the same number. When someone calls the number, all the users will receive the INVITE request and only the first one answering the call will start a session. The others will receive CANCEL request from OpenSIPS. To enable this feature, just t_relay() the request in branch_route. B.R, Chen-Che -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Is-there-some-sort-of-ring-group-implementation-where-users-are-dialled-and-just-the-first-one-to-an-tp7598035p7598039.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Using STUN/TURN server with OpenSIPS
So OpenSIPS does not support use of a TURN server without installing an external module (Mediaproxy, RTPproxy)? Is that right? On 23 July 2015 at 20:11, Tito Cumpen t...@xsvoce.com wrote: There is a stun module you can use for opensips http://www.opensips.org/html/docs/modules/2.2.x/stun.html as far as TURN if you're using ICE a turn relay can be used and added to the sdp as a lower priority candidate or intermediary while ICE is in discovery . I am familiar with media proxy possessing this capability as well as rtpengine. I'd look into them. Thanks, Tito On Thu, Jul 23, 2015 at 2:27 PM, Nabeel nabeelshik...@gmail.com wrote: Hi, I have a STUN/TURN server set up in my SIP clients which also support ICE. However, OpenSIPS does not make use of this STUN/TURN server when attempting to make a call. How do I configure OpenSIPS to use the STUN/TURN server at a given port? Which is better to use: RTPproxy or STUN/TURN server, and why? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] ERROR:core:io_watch_del: BUG - trying to del fd 38 with flags 2 1
Hi Bogdan, Almost a month ago, I have raised the issue regarding the opensips log file filling up with the following message ERROR:core:io_watch_del: BUG - trying to del fd 38 with flags 2 1 Now I can reproduce it every time with the following scenario: 1) UAC and UAS are connected to opensips proxy with TCP transport 2) Connections looks good 3) Netstat shows the connection to UAC in ESTABLISHED state and the tcp_conn_lists of opensips also looks fine for that UAC 4) Now unplug the Ethernet cable on UAC 5) After tcp connection timeout (set to 5 mins in opensips.cfg), the tcp connection goes away from netstat as well as from tcp_conn_lists 6) UAS tries another tcp call to the UAC which is still unplugged 7) tcp_conn_lists shows the tcp connection to the UAC and netstat shows the connection in SYN_SENT state 8) After the tcp connection timeout (set to 5 mins in opensips.cfg), the connection goes away from netstat however it remains there in tcp_conn_lists and at that moment the BUG - trying to del fd 38 with flags 2 1 starts printing in infinite loop. I looked at the source code and observed the following 1) When the network cable is plugged in The io_watch_add happens with flag IO_WATCH_READ in tcp_main.c when the command is CONN_NEW 2) When the network cable is unplugged The io_watch_add happens with flag IO_WATCH_WRITE in tcp_main.c when the command is ASYNC_CONNECT 3) While doing io_watch_del after timeout, from handle_tcpconn_ev method, io_watch_del always uses IO_WATCH_READ to delete the fd, which gives this error in case of unplugged cable. Please look into this and suggest how can I fix this issue ? Thanks Rahul Gupta -- DISCLAIMER: This e-mail may contain information that is confidential, privileged or otherwise protected from disclosure. If you are not an intended recipient of this e-mail, do not duplicate or redistribute it by any means. Please delete it and any attachments and notify the sender that you have received it in error. Unintended recipients are prohibited from taking action on the basis of information in this e-mail.E-mail messages may contain computer viruses or other defects, may not be accurately replicated on other systems, or may be intercepted, deleted or interfered with without the knowledge of the sender or the intended recipient. If you are not comfortable with the risks associated with e-mail messages, you may decide not to use e-mail to communicate with IPC. IPC reserves the right, to the extent and under circumstances permitted by applicable law, to retain, monitor and intercept e-mail messages to and from its systems. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Using Jenkins to Build OpenSIPS
Hi guys! Up until now I been using menuconfig to build opensips. I would like to try to use something different to build OpenSIPS. Is there any way to build OpenSIPS in an automated fashion? Something like severak automated tasks that downloads 2.1 tag, run a configure and a make, make install? I did a diff of a before and after menuconfig run on a vanilla 2.1 directory, hoping to find a single file with the changes, but found several *.d files modified. Is there any way to have a full blown menuconfig saved and restored? on a new checked out directory to do a new build? I even change the default prefix. Thanks in advanced. -- Víctor E. Medina M. Platform Architect / Chief Infrastructure +58424 291 4561 BB #79A8AFA2 @VMCibersys ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Dropped request on branch route causes error message
Hi list, Sometimes my routing logic decides not to forward an Invite request and then I drop() this request. The problem is that if I drop the request from the branch_route the following error message appears: ERROR:tm:t_forward_nonack: failure to add branches I found this thread ( http://opensips.org/pipermail/users/2014-October/030155.html) in which Bogdan states that 'IF no branch is actually sent out, you will get the error logs from t_forward_nonack (as you have)'. But is it some bad practice or design error to have no branch sent out at all? Everything seems to work fine but this error message bugs me Regards, Patrick ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Using Jenkins to Build OpenSIPS
Patrick, thanks you SO much! 2015-07-23 9:32 GMT-04:30 Patrick Wakano pwak...@gmail.com: You can build an installer script by putting a Makefile.conf into your Opensips source files dir and then run make all make install. You just have to run the make menuconfig once to get the Makefile.conf you desire and then use it in your installer. Patrick On Thu, Jul 23, 2015 at 10:48 AM, Victor Medina victor.med...@cibersys.com wrote: Hi guys! Up until now I been using menuconfig to build opensips. I would like to try to use something different to build OpenSIPS. Is there any way to build OpenSIPS in an automated fashion? Something like severak automated tasks that downloads 2.1 tag, run a configure and a make, make install? I did a diff of a before and after menuconfig run on a vanilla 2.1 directory, hoping to find a single file with the changes, but found several *.d files modified. Is there any way to have a full blown menuconfig saved and restored? on a new checked out directory to do a new build? I even change the default prefix. Thanks in advanced. -- Víctor E. Medina M. Platform Architect / Chief Infrastructure +58424 291 4561 BB #79A8AFA2 @VMCibersys ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Víctor E. Medina M. Platform Architect / Chief Infrastructure +58424 291 4561 BB #79A8AFA2 @VMCibersys ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Using Jenkins to Build OpenSIPS
You can build an installer script by putting a Makefile.conf into your Opensips source files dir and then run make all make install. You just have to run the make menuconfig once to get the Makefile.conf you desire and then use it in your installer. Patrick On Thu, Jul 23, 2015 at 10:48 AM, Victor Medina victor.med...@cibersys.com wrote: Hi guys! Up until now I been using menuconfig to build opensips. I would like to try to use something different to build OpenSIPS. Is there any way to build OpenSIPS in an automated fashion? Something like severak automated tasks that downloads 2.1 tag, run a configure and a make, make install? I did a diff of a before and after menuconfig run on a vanilla 2.1 directory, hoping to find a single file with the changes, but found several *.d files modified. Is there any way to have a full blown menuconfig saved and restored? on a new checked out directory to do a new build? I even change the default prefix. Thanks in advanced. -- Víctor E. Medina M. Platform Architect / Chief Infrastructure +58424 291 4561 BB #79A8AFA2 @VMCibersys ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] topology hiding not accepting BYE before 200 OK
Hi Guys,
I seem to be having some trouble with the new topology_hiding module in
opensips 2.1
here is the call scenario
UAC -- Opensips -- UAS
UAC Sends Invite to UAS with topology hiding module
UAS sends 180 with to-tag
UAC sends BYE
When the Bye is sent opensips loops the call till max forwards is reached
from what i can see from the debugs the Bye from UAC is accepted and
matches the topology_hiding_match function but does not rewrite the
destination IP so when the message passes t_relay() its sending the Bye
to Itself from Itself.
I have tried this without topology hiding and the BYE is relayed as it
should
.
My route looks as follows
route{
script_trace( 3, $rm from $si, ruri=$ru, me);
if (!mf_process_maxfwd_header(10)) {
sl_send_reply(483,Too Many Hops);
exit;
}
if ( check_source_address(1,$avp(trunk_attrs)) ) {
# request comes from trunks
setflag(IS_TRUNK);
} else if ( is_from_gw() ) {
# request comes from GWs
} else {
#send_reply(403,Forbidden);
xlog(Message is not from Trunk or GW $si);
#exit;
}
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
#if (loose_route()) {
if(topology_hiding_match()) {
# validate the sequential request against dialog
if ( $DLG_status!=NULL !validate_dialog() ) {
xlog(In-Dialog $rm from $si (callid=$ci) is not valid
according to dialog\n);
## exit;
}
if (is_method(BYE)) {
setflag(ACC_DO); # do accounting ...
setflag(ACC_FAILED); # ... even if the transaction fails
} else if (is_method(INVITE)) {
# even if in most of the cases is useless, do RR for
# re-INVITEs alos, as some buggy clients do change route set
# during the dialog.
record_route();
}
# lets handle re-invites and offer proxy
if (has_body(application/sdp)) {
# Begin rtp session update gyrations
if (method == INVITE) {
# INVITE w/ SDP, so early neg
# This is offer, reply is answer
rtpproxy_offer(iewlz20);
t_on_reply(1);
} else if (method == ACK) {
# ACK w/ SDP, so late neg (done now)
# This is answer
rtpproxy_answer(iewlz20);
}
}
# route it out to whatever destination was set by loose_route()
# in $du (destination URI).
route(RELAY);
} else {
if ( is_method(ACK) ) {
if ( t_check_trans() ) {
# non loose-route, but stateful ACK; must be an ACK
after
# a 487 or e.g. 404 from upstream server
t_relay();
exit;
} else {
# ACK without matching transaction -
# ignore and discard
exit;
}
}
sl_send_reply(404,Not here);
}
exit;
}
INITIAL REQUESTS
if ( !isflagset(IS_TRUNK) ) {
## accept new calls only from trunks
send_reply(403,Not from trunk);
exit;
}
# CANCEL processing
if (is_method(CANCEL)) {
if (t_check_trans())
t_relay();
exit;
} else if (!is_method(INVITE)) {
send_reply(405,Method Not Allowed);
exit;
}
if ($rU==NULL) {
# request with no Username in RURI
sl_send_reply(484,Address Incomplete);
exit;
}
t_check_trans();
# preloaded route checking
if (loose_route()) {
xlog(L_ERR,
Attempt to route with preloaded Route's [$fu/$tu/$ru/$ci]);
if (!is_method(ACK))
sl_send_reply(403,Preload Route denied);
exit;
}
# record routing
record_route();
setflag(ACC_DO); # do accounting
# create dialog with timeout
if ( !create_dialog(B) ) {
send_reply(500,Internal Server Error);
exit;
}
if (is_avp_set($avp(trunk_attrs)) $avp(trunk_attrs)=~^[0-9]+$) {
get_profile_size(trunkCalls,$si,$var(size));
if ( $(var(size){s.int}) = $(avp(trunk_attrs){s.int}) ) {
send_reply(486,Busy Here);
exit;
}
}
set_dlg_profile(trunkCalls,$si);
# apply transformations from dialplan table
dp_translate(0,$rU/$rU);
# route calls based on prefix
if ( !do_routing(1) ) {
send_reply(404,No Route found);
exit;
}
t_on_failure(GW_FAILOVER);
if (is_method(INVITE)) {
force_send_socket(udp:XXX.XXX.XXX.XXX:5060);
#rtpproxy_engage('ierz20');
Re: [OpenSIPS-Users] Using STUN/TURN server with OpenSIPS
There is a stun module you can use for opensips http://www.opensips.org/html/docs/modules/2.2.x/stun.html as far as TURN if you're using ICE a turn relay can be used and added to the sdp as a lower priority candidate or intermediary while ICE is in discovery . I am familiar with media proxy possessing this capability as well as rtpengine. I'd look into them. Thanks, Tito On Thu, Jul 23, 2015 at 2:27 PM, Nabeel nabeelshik...@gmail.com wrote: Hi, I have a STUN/TURN server set up in my SIP clients which also support ICE. However, OpenSIPS does not make use of this STUN/TURN server when attempting to make a call. How do I configure OpenSIPS to use the STUN/TURN server at a given port? Which is better to use: RTPproxy or STUN/TURN server, and why? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Using STUN/TURN server with OpenSIPS
Hi, I have a STUN/TURN server set up in my SIP clients which also support ICE. However, OpenSIPS does not make use of this STUN/TURN server when attempting to make a call. How do I configure OpenSIPS to use the STUN/TURN server at a given port? Which is better to use: RTPproxy or STUN/TURN server, and why? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpenSIPS work behind COMCAST Business Class?
We are looking to switch everything from one of our datacenters to our corporate which runs COMCAST Business Class. Googling people have all kinds of problems with other solutions on COMCAST (But I think they are all residential setups). We have a Cisco DPC3939B COMCAST Modem running in non-bridge mode, with a large group of static IPs - 1 is earmarked for this install. The design is: Possibly OpenSIPS in VA NAT as 10.1.x.y - PUBLIC IP via DPC3939B Has two SIP phones here (Cisco 7940) - all 10.1.x.x addresses internally In GA I have one SIP phone (7940) In FL I have three locations with one SIP phone each (7940) In PA I have one SIP phone (7940) I have enough bandwidth if I need OpenSIPS to literally be the proxy for all points (e.g. no REDIRECT of the RTP). Total of 8 DIDs. I think that covers config questions. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips taking 100% cpu
Hi Bogdan, I only looked at 3 PIDs and all three showed the same bt. Thanks Rahul -- DISCLAIMER: This e-mail may contain information that is confidential, privileged or otherwise protected from disclosure. If you are not an intended recipient of this e-mail, do not duplicate or redistribute it by any means. Please delete it and any attachments and notify the sender that you have received it in error. Unintended recipients are prohibited from taking action on the basis of information in this e-mail.E-mail messages may contain computer viruses or other defects, may not be accurately replicated on other systems, or may be intercepted, deleted or interfered with without the knowledge of the sender or the intended recipient. If you are not comfortable with the risks associated with e-mail messages, you may decide not to use e-mail to communicate with IPC. IPC reserves the right, to the extent and under circumstances permitted by applicable law, to retain, monitor and intercept e-mail messages to and from its systems. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] changing $rU number
Hello! Opensips 1.10. I am using DROUTING module from making routing. But some SIP UA sends to Opensips tel. number with some unnecessary characters, such as %20 and -. The question is how can I delete these characters from request user? Thank you for any help. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] changing $rU number
$rU is read/write so you can use regexp and just rewrite the variable On Thu, Jul 23, 2015 at 10:33 AM, dpa denis7...@mail.ru wrote: Hello! Opensips 1.10. I am using DROUTING module from making routing. But some SIP UA sends to Opensips tel. number with some unnecessary characters, such as “%20” and “-”. The question is how can I delete these characters from request user? Thank you for any help. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Is there some sort of ring group implementation where users are dialled and just the first one to answer will get the cal?
Hi, In my own experience, one way to achieve ring group implementation is to make those users register use the same number. When someone calls the number, all the users will receive the INVITE request and only the first one answering the call will start a session. The others will receive CANCEL request from OpenSIPS. To enable this feature, just t_relay() the request in branch_route. B.R, Chen-Che -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Is-there-some-sort-of-ring-group-implementation-where-users-are-dialled-and-just-the-first-one-to-an-tp7598035p7598039.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] changing $rU number
Thank you. But what can I use for do it? From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Schneur Rosenberg Sent: Thursday, July 23, 2015 10:40 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] changing $rU number $rU is read/write so you can use regexp and just rewrite the variable On Thu, Jul 23, 2015 at 10:33 AM, dpa denis7...@mail.ru wrote: Hello! Opensips 1.10. I am using DROUTING module from making routing. But some SIP UA sends to Opensips tel. number with some unnecessary characters, such as “%20” and “-”. The question is how can I delete these characters from request user? Thank you for any help. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] changing $rU number
Documentation can be found here http://www.opensips.org/html/docs/modules/1.10.x/dialplan.html For egsample to remove all - characters in $rU do the following Add the following into the dialplan table mysql select * from dialplan\G; *** 1. row *** id: 6 dpid: 1 pr: 98 match_op: 0 match_exp: - match_flags: 0 subst_exp: repl_exp: timerec: disabled: 0 attrs: *** 2. row *** Then call dp_translate(1,$rU/$rU); in your script. Regards Trevor Steyn On 23/07/2015 10:51, dpa wrote: And how dialplan helps me to do it, if ,for example, one time I have such characters 8%2089-0987-09, and in another time I have 987-89%20908-1? *From:*users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] *On Behalf Of *Trevor Steyn *Sent:* Thursday, July 23, 2015 11:27 AM *To:* users@lists.opensips.org *Subject:* Re: [OpenSIPS-Users] changing $rU number Hi Denis You can use the dialplan module to rewrite variables as below dp_translate(1,$rU/$rU); Then insert you regexp into the dialplan tables, read the dialplan module documentation for more info. Regards Trevor On 23/07/2015 09:49, dpa wrote: Thank you. But what can I use for do it? *From:*users-boun...@lists.opensips.org mailto:users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] *On Behalf Of *Schneur Rosenberg *Sent:* Thursday, July 23, 2015 10:40 AM *To:* OpenSIPS users mailling list *Subject:* Re: [OpenSIPS-Users] changing $rU number $rU is read/write so you can use regexp and just rewrite the variable On Thu, Jul 23, 2015 at 10:33 AM, dpa denis7...@mail.ru mailto:denis7...@mail.ru wrote: Hello! Opensips 1.10. I am using DROUTING module from making routing. But some SIP UA sends to Opensips tel. number with some unnecessary characters, such as “%20” and “-”. The question is how can I delete these characters from request user? Thank you for any help. ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] changing $rU number
I understand, thank you. I will try. -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Trevor Steyn Sent: Thursday, July 23, 2015 12:46 PM To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] changing $rU number Documentation can be found here http://www.opensips.org/html/docs/modules/1.10.x/dialplan.html For egsample to remove all - characters in $rU do the following Add the following into the dialplan table mysql select * from dialplan\G; *** 1. row *** id: 6 dpid: 1 pr: 98 match_op: 0 match_exp: - match_flags: 0 subst_exp: repl_exp: timerec: disabled: 0 attrs: *** 2. row *** Then call dp_translate(1,$rU/$rU); in your script. Regards Trevor Steyn On 23/07/2015 10:51, dpa wrote: And how dialplan helps me to do it, if ,for example, one time I have such characters 8%2089-0987-09, and in another time I have 987-89%20908-1? *From:*users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] *On Behalf Of *Trevor Steyn *Sent:* Thursday, July 23, 2015 11:27 AM *To:* users@lists.opensips.org *Subject:* Re: [OpenSIPS-Users] changing $rU number Hi Denis You can use the dialplan module to rewrite variables as below dp_translate(1,$rU/$rU); Then insert you regexp into the dialplan tables, read the dialplan module documentation for more info. Regards Trevor On 23/07/2015 09:49, dpa wrote: Thank you. But what can I use for do it? *From:*users-boun...@lists.opensips.org mailto:users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] *On Behalf Of *Schneur Rosenberg *Sent:* Thursday, July 23, 2015 10:40 AM *To:* OpenSIPS users mailling list *Subject:* Re: [OpenSIPS-Users] changing $rU number $rU is read/write so you can use regexp and just rewrite the variable On Thu, Jul 23, 2015 at 10:33 AM, dpa denis7...@mail.ru mailto:denis7...@mail.ru wrote: Hello! Opensips 1.10. I am using DROUTING module from making routing. But some SIP UA sends to Opensips tel. number with some unnecessary characters, such as %20 and -. The question is how can I delete these characters from request user? Thank you for any help. ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips taking 100% cpu
Hi Rahul, Do all opensips processes with 100% cpu usage, show the same bt ? (the lock in TCP layer) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 21.07.2015 21:16, Gupta, Rahul wrote: Hi Bogdan, Performance testing is really getting hit because of this issue. I will appreciate any feedback on this issue. Thanks Rahul *From:*Gupta, Rahul *Sent:* Friday, July 17, 2015 2:36 PM *To:* users@lists.opensips.org *Subject:* opensips taking 100% cpu Hi, I am using 1.11.3-tls version and opensips processes are taking 100% CPU. PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 12635 root 20 0 335m 1876 1068 R 100.0 0.0 335:02.96 opensips 12636 root 20 0 335m 1944 1140 R 100.0 0.0 335:03.71 opensips 12641 root 20 0 335m 88m 87m R 100.0 0.1 333:26.96 opensips 12645 root 20 0 335m 63m 62m R 100.0 0.1 333:23.76 opensips 12632 root 20 0 335m 1196 408 R 99.8 0.0 38:22.68 opensips 12647 root 20 0 335m 19m 18m R 99.8 0.0 561:16.07 opensips 12634 root 20 0 335m 1876 1068 R 99.5 0.0 333:22.10 opensips None of the commands from opensipsctl are working, so I can’t use opensipsctl trap to generate gbd info. I ran the gdb on couple of the PID and got the following back trace. Seems like opensips is stuck in some lock. [root@sa-z2-ccm1 ~]# gdb /usr/sbin/opensips 12633 GNU gdb (GDB) Red Hat Enterprise Linux (7.2-60.el6_4.1) Copyright (C) 2010 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later http://gnu.org/licenses/gpl.html This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type show copying and show warranty for details. This GDB was configured as x86_64-redhat-linux-gnu. For bug reporting instructions, please see: http://www.gnu.org/software/gdb/bugs/... Reading symbols from /usr/sbin/opensips...done. Attaching to program: /usr/sbin/opensips, process 12633 Reading symbols from /lib64/snoopy.so...done. Loaded symbols for /lib64/snoopy.so Reading symbols from /lib64/libdl.so.2...(no debugging symbols found)...done. Loaded symbols for /lib64/libdl.so.2 Reading symbols from /lib64/libresolv.so.2...(no debugging symbols found)...done. Loaded symbols for /lib64/libresolv.so.2 Reading symbols from /usr/lib64/libssl.so.10...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libssl.so.10 Reading symbols from /usr/lib64/libcrypto.so.10...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libcrypto.so.10 Reading symbols from /lib64/libc.so.6...(no debugging symbols found)...done. Loaded symbols for /lib64/libc.so.6 Reading symbols from /lib64/ld-linux-x86-64.so.2...(no debugging symbols found)...done. Loaded symbols for /lib64/ld-linux-x86-64.so.2 Reading symbols from /lib64/libgssapi_krb5.so.2...(no debugging symbols found)...done. Loaded symbols for /lib64/libgssapi_krb5.so.2 Reading symbols from /lib64/libkrb5.so.3...(no debugging symbols found)...done. Loaded symbols for /lib64/libkrb5.so.3 Reading symbols from /lib64/libcom_err.so.2...(no debugging symbols found)...done. Loaded symbols for /lib64/libcom_err.so.2 Reading symbols from /lib64/libk5crypto.so.3...(no debugging symbols found)...done. Loaded symbols for /lib64/libk5crypto.so.3 Reading symbols from /lib64/libz.so.1...(no debugging symbols found)...done. Loaded symbols for /lib64/libz.so.1 Reading symbols from /lib64/libkrb5support.so.0...(no debugging symbols found)...done. Loaded symbols for /lib64/libkrb5support.so.0 Reading symbols from /lib64/libkeyutils.so.1...(no debugging symbols found)...done. Loaded symbols for /lib64/libkeyutils.so.1 Reading symbols from /lib64/libpthread.so.0...(no debugging symbols found)...done. [Thread debugging using libthread_db enabled] Loaded symbols for /lib64/libpthread.so.0 Reading symbols from /lib64/libselinux.so.1...(no debugging symbols found)...done. Loaded symbols for /lib64/libselinux.so.1 Reading symbols from /usr/lib64/opensips/modules/signaling.so...done. Loaded symbols for /usr/lib64/opensips/modules/signaling.so Reading symbols from /usr/lib64/opensips/modules/sl.so...done. Loaded symbols for /usr/lib64/opensips/modules/sl.so Reading symbols from /usr/lib64/opensips/modules/tm.so...done. Loaded symbols for /usr/lib64/opensips/modules/tm.so Reading symbols from /usr/lib64/opensips/modules/rr.so...done. Loaded symbols for /usr/lib64/opensips/modules/rr.so Reading symbols from /usr/lib64/opensips/modules/maxfwd.so...done. Loaded symbols for /usr/lib64/opensips/modules/maxfwd.so Reading symbols from /usr/lib64/opensips/modules/sipmsgops.so...done. Loaded symbols for /usr/lib64/opensips/modules/sipmsgops.so Reading symbols from /usr/lib64/opensips/modules/mi_fifo.so...done. Loaded symbols for /usr/lib64/opensips/modules/mi_fifo.so Reading symbols
Re: [OpenSIPS-Users] RTP Delay when changing RTP Source port
HI Rik,
I removed the all flags and it worked i got audio straight away i then
re-added flags one by one to find the culprit seems the
re-packetization was the issue when i add z20 there is that delay in
audio,
The debugs dont show much see below
The issue is i really need RTP going to UAS to be at 20ms payload due to
a vendor restriction, at least know i know where to look at i will dig
into the re-packetization flag to see if i can learn more on why it
would do this
if you have any ideas please let me know.
DBUG:get_command: received command 16653_6 UIEZ20c18,4,8,100,118
SDurfbb02-1ceee8fa17f07cf66cdcc9b4c851f49c-ctvvfv3 UAC_IP 53266
SDurfbb02-55B0B333-1D600C7-0ADE2C1B;1
INFO:handle_command: new session
SDurfbb02-1ceee8fa17f07cf66cdcc9b4c851f49c-ctvvfv3, tag
SDurfbb02-55B0B333-1D600C7-0ADE2C1B;1 requested, type strong
INFO:handle_command: new session on a port 57596 created, tag
SDurfbb02-55B0B333-1D600C7-0ADE2C1B;1
INFO:handle_command: pre-filling caller's address with 10.10.16.34:53266
INFO:handle_command: RTP packets from caller will be resized to 20
milliseconds
DBUG:doreply: sending reply 16653_6 57596 EXT_IP
DBUG:get_command: received command 16663_8 LEIZ20c18
SDurfbb02-1ceee8fa17f07cf66cdcc9b4c851f49c-ctvvfv3 UAS_IP 54274
SDurfbb02-55B0B333-1D600C7-0ADE2C1B;1 g56jcvtppqn2wxef.i;1
INFO:handle_command: lookup on ports 57596/33858, session timer restarted
INFO:handle_command: pre-filling callee's address with 196.2.126.52:54274
INFO:handle_command: RTP packets from callee will be resized to 20
milliseconds
DBUG:doreply: sending reply 16663_8 33858 INT_IP
DBUG:get_command: received command 16660_5 LEIZ20c18,100
SDurfbb02-1ceee8fa17f07cf66cdcc9b4c851f49c-ctvvfv3 UAS_IP 53680
SDurfbb02-55B0B333-1D600C7-0ADE2C1B;1 g56jcvtppqn2wxef.i;1
INFO:handle_command: lookup on ports 57596/33858, session timer restarted
INFO:handle_command: pre-filling callee's address with 196.2.126.52:53680
INFO:handle_command: RTP packets from callee will be resized to 20
milliseconds
DBUG:doreply: sending reply 16660_5 33858 INT_IP
DBUG:get_command: received command 16663_9 LEIZ20c18,8,100
SDurfbb02-1ceee8fa17f07cf66cdcc9b4c851f49c-ctvvfv3 UAS_IP 36606
SDurfbb02-55B0B333-1D600C7-0ADE2C1B;1 g56jcvtppqn2wxef.i;1
INFO:handle_command: lookup on ports 57596/33858, session timer restarted
INFO:handle_command: pre-filling callee's address with 196.2.126.52:36606
INFO:handle_command: RTP packets from callee will be resized to 20
milliseconds
DBUG:doreply: sending reply 16663_9 33858 INT_IP
DBUG:get_command: received command 16658_5 D
SDurfbb02-1ceee8fa17f07cf66cdcc9b4c851f49c-ctvvfv3
SDurfbb02-55B0B333-1D600C7-0ADE2C1B g56jcvtppqn2wxef.i
INFO:handle_delete: forcefully deleting session 1 on ports 57596/33858
INFO:remove_session: RTP stats: 991 in from callee, 388 in from caller,
1781 relayed, 0 dropped
INFO:remove_session: RTCP stats: 2 in from callee, 5 in from caller, 7
relayed, 0 dropped
INFO:remove_session: session on ports 57596/33858 is cleaned up
DBUG:doreply: sending reply 16658_5 0
Regards
Trevor Steyn
On 23/07/2015 10:28, Rik Broers wrote:
Try it with only the ie flags, wz20 only adds more complexity in
troubleshooting.
What do the rtpproxy logs tell you?
-Oorspronkelijk bericht-
Van: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] Namens Trevor Steyn
Verzonden: woensdag 22 juli 2015 21:04
Aan: users@lists.opensips.org
Onderwerp: Re: [OpenSIPS-Users] RTP Delay when changing RTP Source port
Hi Rik
I have tried using rtpproxy_offer/answer functions with the same results,
if (has_body(application/sdp)) {
if (rtpproxy_offer('iewz20')) {
t_on_reply(1);
} else {
t_on_reply(2);
}
}
t_on_failure(GW_FAILOVER);
route(RELAY);
}
onreply_route[1]
{
if (has_body(application/sdp))
rtpproxy_answer('eiwz20');
}
onreply_route[2]
{
if (has_body(application/sdp))
rtpproxy_offer('iewz20');
}
below you can see that Signalling and RTP flows.
You will see rtpproxy only starts relaying packets ~10seconds later after the
200OK even though Callee is sending RTP
http://salamander.iburst.co.za:8000/personal/signalling.txt
Seems the issue here is when RTP stream is already established (183 with
SDP) and the 200OK comes along with different source port it takes some time
before RTP proxy relays the packets to caller.
Regards
Trevor
On 22/07/2015 13:30, Rik Broers wrote:
I think rtpproxy_engage doesnt work correct with the fact that you bridge
internal to external. Also says in docs:
... Note that when used in bridge mode, this function might advertise wrong
interfaces in SDP (due to the fact that OpenSIPS is not aware of the
RTPProxy configuration), so you might face an undefined behavior.
You could try and use the rtpproxy_offer and answer functions. put in the
reply route an if
Re: [OpenSIPS-Users] RTP Delay when changing RTP Source port
Try it with only the ie flags, wz20 only adds more complexity in
troubleshooting.
What do the rtpproxy logs tell you?
-Oorspronkelijk bericht-
Van: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org]
Namens Trevor Steyn
Verzonden: woensdag 22 juli 2015 21:04
Aan: users@lists.opensips.org
Onderwerp: Re: [OpenSIPS-Users] RTP Delay when changing RTP Source port
Hi Rik
I have tried using rtpproxy_offer/answer functions with the same results,
if (has_body(application/sdp)) {
if (rtpproxy_offer('iewz20')) {
t_on_reply(1);
} else {
t_on_reply(2);
}
}
t_on_failure(GW_FAILOVER);
route(RELAY);
}
onreply_route[1]
{
if (has_body(application/sdp))
rtpproxy_answer('eiwz20');
}
onreply_route[2]
{
if (has_body(application/sdp))
rtpproxy_offer('iewz20');
}
below you can see that Signalling and RTP flows.
You will see rtpproxy only starts relaying packets ~10seconds later after the
200OK even though Callee is sending RTP
http://salamander.iburst.co.za:8000/personal/signalling.txt
Seems the issue here is when RTP stream is already established (183 with
SDP) and the 200OK comes along with different source port it takes some time
before RTP proxy relays the packets to caller.
Regards
Trevor
On 22/07/2015 13:30, Rik Broers wrote:
I think rtpproxy_engage doesnt work correct with the fact that you bridge
internal to external. Also says in docs:
... Note that when used in bridge mode, this function might advertise wrong
interfaces in SDP (due to the fact that OpenSIPS is not aware of the RTPProxy
configuration), so you might face an undefined behavior.
You could try and use the rtpproxy_offer and answer functions. put in the
reply route an if (has_body(application/sdp)) to also catch the 183 with
sdp .The docs have examples on how to use them and how to trigger on reply
routes.
Regards,
Rik
-Oorspronkelijk bericht-
Van: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] Namens Trevor Steyn
Verzonden: woensdag 22 juli 2015 9:52
Aan: users@lists.opensips.org
Onderwerp: [OpenSIPS-Users] RTP Delay when changing RTP Source port
Hi, All
Still quite new to opensips I have the following configuration running
on
version: opensips 2.1.0 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC,
FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE
262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support:
poll, epoll_lt, epoll_et, sigio_rt, select.
main.c compiled on 06:22:03 May 8 2015 with gcc 4.4.7
(Topology Hiding)
UAC --- Opensips(Internal)Opensips(External) UAS
(RTP PROXY BRIDGE)
what i am experiencing is the following call is setup between UAC and UAS
through opensips UAS sets up RTP with a 183 Session Progress message with SDP
Shortly after we get a 180 ringing (i understand this is not correct but
cannot be changed), When a 200OK is eventually sent the Source Port is
different to what was in the SDP on the 183 message.
Media starts flowing from UAS to opensips External from the new source port
but for the first 7 seconds or so opensips/rtpproxy does not pass on this
media to UAC from Internal.
I run rtp proxy as follows
rtpproxy -l Internal IP/External IP -s udp:127.0.0.1:12221 -m
25000 -M 65000 -F -d DBUG:LOCAL1
route{
#script_trace( 3, $rm from $si, ruri=$ru, me);
if (!mf_process_maxfwd_header(10)) {
sl_send_reply(483,Too Many Hops);
exit;
}
if ( check_source_address(1,$avp(trunk_attrs)) ) {
# request comes from trunks
setflag(IS_TRUNK);
} else if ( is_from_gw() ) {
# request comes from GWs
} else {
send_reply(403,Forbidden);
exit;
}
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if(topology_hiding_match()) {
# validate the sequential request against dialog
if ( $DLG_status!=NULL !validate_dialog() ) {
xlog(In-Dialog $rm from $si (callid=$ci) is not valid
according to dialog\n);
## exit;
}
if (is_method(BYE)) {
setflag(ACC_DO); # do accounting ...
setflag(ACC_FAILED); # ... even if the transaction fails
} else if (is_method(INVITE)) {
# even if in most of the cases is useless, do RR for
# re-INVITEs alos, as some buggy clients do change route set
# during the dialog.
record_route();
}
route(RELAY);
} else {
if (
Re: [OpenSIPS-Users] changing $rU number
And how dialplan helps me to do it, if ,for example, one time I have such characters 8%2089-0987-09, and in another time I have 987-89%20908-1? From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Trevor Steyn Sent: Thursday, July 23, 2015 11:27 AM To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] changing $rU number Hi Denis You can use the dialplan module to rewrite variables as below dp_translate(1,$rU/$rU); Then insert you regexp into the dialplan tables, read the dialplan module documentation for more info. Regards Trevor On 23/07/2015 09:49, dpa wrote: Thank you. But what can I use for do it? From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Schneur Rosenberg Sent: Thursday, July 23, 2015 10:40 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] changing $rU number $rU is read/write so you can use regexp and just rewrite the variable On Thu, Jul 23, 2015 at 10:33 AM, dpa denis7...@mail.ru wrote: Hello! Opensips 1.10. I am using DROUTING module from making routing. But some SIP UA sends to Opensips tel. number with some unnecessary characters, such as %20 and -. The question is how can I delete these characters from request user? Thank you for any help. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] changing $rU number
http://www.opensips.org/html/docs/modules/1.10.x/regex.html On Thu, Jul 23, 2015 at 10:49 AM, dpa denis7...@mail.ru wrote: Thank you. But what can I use for do it? *From:* users-boun...@lists.opensips.org [mailto: users-boun...@lists.opensips.org] *On Behalf Of *Schneur Rosenberg *Sent:* Thursday, July 23, 2015 10:40 AM *To:* OpenSIPS users mailling list *Subject:* Re: [OpenSIPS-Users] changing $rU number $rU is read/write so you can use regexp and just rewrite the variable On Thu, Jul 23, 2015 at 10:33 AM, dpa denis7...@mail.ru wrote: Hello! Opensips 1.10. I am using DROUTING module from making routing. But some SIP UA sends to Opensips tel. number with some unnecessary characters, such as “%20” and “-”. The question is how can I delete these characters from request user? Thank you for any help. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] changing $rU number
Hi Denis You can use the dialplan module to rewrite variables as below dp_translate(1,$rU/$rU); Then insert you regexp into the dialplan tables, read the dialplan module documentation for more info. Regards Trevor On 23/07/2015 09:49, dpa wrote: Thank you. But what can I use for do it? *From:*users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] *On Behalf Of *Schneur Rosenberg *Sent:* Thursday, July 23, 2015 10:40 AM *To:* OpenSIPS users mailling list *Subject:* Re: [OpenSIPS-Users] changing $rU number $rU is read/write so you can use regexp and just rewrite the variable On Thu, Jul 23, 2015 at 10:33 AM, dpa denis7...@mail.ru mailto:denis7...@mail.ru wrote: Hello! Opensips 1.10. I am using DROUTING module from making routing. But some SIP UA sends to Opensips tel. number with some unnecessary characters, such as “%20” and “-”. The question is how can I delete these characters from request user? Thank you for any help. ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] changing $rU number
As I understand regex module just matches string against regexp. but not makes any manipulation. From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Schneur Rosenberg Sent: Thursday, July 23, 2015 11:24 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] changing $rU number http://www.opensips.org/html/docs/modules/1.10.x/regex.html On Thu, Jul 23, 2015 at 10:49 AM, dpa denis7...@mail.ru wrote: Thank you. But what can I use for do it? From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Schneur Rosenberg Sent: Thursday, July 23, 2015 10:40 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] changing $rU number $rU is read/write so you can use regexp and just rewrite the variable On Thu, Jul 23, 2015 at 10:33 AM, dpa denis7...@mail.ru wrote: Hello! Opensips 1.10. I am using DROUTING module from making routing. But some SIP UA sends to Opensips tel. number with some unnecessary characters, such as “%20” and “-”. The question is how can I delete these characters from request user? Thank you for any help. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
