Re: [OpenSIPS-Users] New MediaProxy release 2.6.0
Hi, On Mar 4, 2014, at 8:47 PM, Nick Cameo wrote: Hello Saul, I was looking at a SIP trace on an active call as I type this. Actually we only need the public ip address advertised in the SDP payload for the device which is behind NAT (ie, listening on a private ip address with 1to1 port forwarded mapping done on the router for the public ip address). Ok, in that case it should work, since it's the scenario which is supposed to cover. The advertised address of the VIA and Contact headers are what shuffle between public and private IP addresses. This of course is the handled by OpenSIPS as it already does so eg: Via: SIP/2.0/UDP 74.71.33.154:5060;branch=z9hG4bK4.0. Contact: sip:74.71.33.154:5060;did=5f7.8d. Via: SIP/2.0/UDP 192.168.0.100:5060;branch=z9hG4bK596;rport. Contact: sip:192.168.0.200:5080;did=5f7.8d61 In summary, can you kindly direct us to the much needed documentation on how to install, configure, and run an OpenSIPS+MediaProxy instance from behind a natted 1to1 mapped instance. Only upon doing so will I resume doing jumping jacks which I have halted indefinitely until further clarity. There is only one parameter needed for this: advertised_ip, in the [Relay] section: ; The host IP address to return when a session is allocated in the relay. This ; could be of use in case the relay is behind NAT but it has a 1 to 1 mapping ; with a public IP address, like Amazon EC2, for example. ;advertised_ip = Just put your public IP address there and the OpenSIPS module will put that one on the SDP. Cheers, -- Saúl Ibarra Corretgé AG Projects ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips1.7 with MediaProxy
Hello,
The best options for you is to use dialog module with topology hiding.
This can be easily combined with any of the media relays (rtpproxy or
mediaproxy) for hiding the media path.
If you have any particular questions on the setup, I will try to help you.
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 25.02.2014 09:26, ?? ? ?? wrote:
Hi. Please help.
We have:
1.One server consists of: CenOS6.5 + Opensips1.7 + MediaProxy2.5
2.One MGW: Cisco AS5350
3.UserID=telephone number and registration on OpenSips through MySQL
4.Call to PSTN pass through MGW with prefix :
route[4] { prefix();
rewritehostport(192.168.0.3:5060 http://192.168.0.3:5060);
if (!t_relay()) { sl_reply_error(); };
exit;}
Now, such a scheme works:
(UAC )sip-Opensips 1.7---SIP---MGW Cisco
192.168.0.65 192.168.0.2 192.168.0.3
RTP---
MGW CiscoPSTN
In this topology visible
It's not safe, it's necessary to build a new wiring diagram:
(UAC )---sip,RTP(Opensips---rtp,SIP--)-MGW
Cisco---PSTN
85.85.85.85(85.85.85.2 192.168.0.2) 192.168.0.3
questions:
1. to hide the network topology from the users (can be used dialog
module, function: topology_hiding?)
2. hide RTP traffic to MGW for Opensips-server (can be used MediaProxy
or rtpproxy)?
Please, give examples opensips.cfg-file ?
___
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 as simple frontend to Asterisk to deal with NAT
Hello Rudy,
Be sure you use t_relay() when forwarding the REGISTER request (and
not forward() function).
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 23.02.2014 20:32, Rudy Eschauzier wrote:
Ok, I think I am getting there. I am able to forward the client registration to Asterisk,
and Twinkle reports registration succeeded (YES!). I am having some trouble
saving the location, however.
This is what I have:
onreply_route {
xlog(incoming reply\n);
xlog(L_INFO,\n\n$C(bc)[ Reply $rs ($rr) from $si concerning $rm
]$C(xx)\n$mb$C(bc)[ End of Reply ]$C(xx)\n);
if ($rs==200) {
save(location);
}
exit;
}
The forwarding is done in the main route block like this:
if(!t_relay())
sl_reply_error();
But the result that OpenSips gives is:
Feb 23 19:22:29 [26043] ERROR:registrar:save: Transaction not created on
Register - can not save on reply
That seems to make sense, as the transaction is created by the t_relay. How
else would OpenSips know where the response is coming from?
Any suggestions?
Thanks,
Rudy.
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Problem with aaa_radius
Hello Aleksandr, The logs you posted does not show any error - are those the last logs during the start attempt ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 24.02.2014 10:28, Alexander Mustafin wrote: Hello. I’m trying to setup new server and just copy opensips.cfg from other server. This configuration tested and works successfully. But now, I’ve silent problem. OpenSIPS from yum-repo: version: opensips 1.10.0beta-tls (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, DISABLE_NAGLE, USE_MCAST, SHM_MEM, 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. @(#) $Id$ main.c compiled on 03:46:24 Aug 6 2013 with gcc 4.4.7 radiusclient-ng installed and config copied too. When I insert radius_send_auth(radius_auth,radius_auth_resp); in config - OpenSIPS can’t start and I have no errors in syslog. Sets are defined in module configuration. Syslog: === Feb 24 02:34:50 reg opensips: DBG:core:yyparse: loading module /usr/lib64/opensips/modules/aaa_radius.so Feb 24 02:34:50 reg opensips: DBG:core:set_mod_param_regex: aaa_radius matches module aaa_radius Feb 24 02:34:50 reg opensips: DBG:core:set_mod_param_regex: found radius_config in module aaa_radius [/usr/lib64/opensips/modules/] Feb 24 02:34:50 reg opensips: DBG:core:set_mod_param_regex: aaa_radius matches module aaa_radius Feb 24 02:34:50 reg opensips: DBG:core:set_mod_param_regex: found sets in module aaa_radius [/usr/lib64/opensips/modules/] Feb 24 02:34:50 reg opensips: DBG:core:__search_avp_map: looking for [username] avp - found -1 Feb 24 02:34:50 reg opensips: DBG:core:new_avp_alias: added alias username with id 1 Feb 24 02:34:50 reg opensips: DBG:core:__search_avp_map: looking for [username] avp - found 1 Feb 24 02:34:50 reg opensips: DBG:core:__search_avp_map: looking for [rpid] avp - found -1 Feb 24 02:34:50 reg opensips: DBG:core:new_avp_alias: added alias rpid with id 2 Feb 24 02:34:50 reg opensips: DBG:core:set_mod_param_regex: aaa_radius matches module aaa_radius Feb 24 02:34:50 reg opensips: DBG:core:set_mod_param_regex: found sets in module aaa_radius [/usr/lib64/opensips/modules/] Feb 24 02:34:50 reg opensips: DBG:core:__search_avp_map: looking for [credit_time] avp - found -1 Feb 24 02:34:50 reg opensips: DBG:core:new_avp_alias: added alias credit_time with id 3 Feb 24 02:34:50 reg opensips: DBG:core:__search_avp_map: looking for [return_code] avp - found -1 Feb 24 02:34:50 reg opensips: DBG:core:new_avp_alias: added alias return_code with id 4 Best regards, Alexander Mustafin mustafin.aleksa...@gmail.com mailto:mustafin.aleksa...@gmail.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] installing opensips on Debian / Ubuntu
Hi Tomasz, Let me escalate this with the person taking care of the deb repository for OpenSIPS. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 25.02.2014 15:07, Tomasz Chmielewski wrote: I'm trying to install opensips on Debian or Ubuntu. However, the provided deb packages seem to have wrong dependencies which prevent the installation: # apt-get install opensips opensips-console opensips-http-modules opensips-mysql-module opensips-identity-module Reading package lists... Done Building dependency tree Reading state information... Done Some packages could not be installed. This may mean that you have requested an impossible situation or if you are using the unstable distribution that some required packages have not yet been created or been moved out of Incoming. The following information may help to resolve the situation: The following packages have unmet dependencies: opensips-mysql-module : Depends: libmysqlclient16 (= 5.1.21-1) but it is not installable E: Unable to correct problems, you have held broken packages. I've tried using from deb http://apt.opensips.org/ stable110 main http://apt.opensips.org/ on Debian 7 and Ubuntu 12.04.4 LTS. Both have similarly broken dependencies. How to best install on Ubuntu or Debian? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Adding Proxy-Authorization header
Hi Stefano, Vlad
Thank you for your response I tried your suggestion but still doesn't work.
This is a snippet from my script:
modparam(uac_auth,credential,268:192.168.2.98:password)
t_on_failure(2);
t_relay();
failure_route[2] {
if(t_check_status(407)){
uac_auth();
xlog(In failure route 2\n);
}
}
According to the log, the uac_auth function is being called but the
following INVITEs doesn't include the Proxy-Authorization header
What am I missing?
Thanks
Diego
On Mon, Feb 24, 2014 at 2:12 PM, Vlad Paiu vladp...@opensips.org wrote:
Hello,
The registrant module is to be used only for generating REGISTER requests
( with auth included ).
For proxied calls, you need to use the uac and uac_auth modules ( [1] )
for adding the auth headers - call uac_auth() ( [2] ) function within
failure route when receiving a challenge.
[1] http://www.opensips.org/html/docs/modules/1.11.x/uac_auth.html
[2] http://www.opensips.org/html/docs/modules/1.11.x/uac.html#id250288
Best Regards
Vlad Paiu
OpenSIPS Developerhttp://www.opensips-solutions.com
On 24.02.2014 17:33, Stefano Pisani wrote:
You can use module UAC_AUTH
Il 24/02/2014 16.18, Diego Barberio ha scritto:
Hi all,
I have opensips registered to an IP-PBX using registrant module and I
want to make an outbound call to that PBX through the proxy.
I'm sending and INVITE from my application to the proxy with a From that
is actually registered by the proxy, however OpenSIPs is not adding the
Proxy-Authorization header so the INVITE is rejected with a 401
Unauthorized and that response is forwarded to my application.
I just want opensips to add the Proxy-Authorization header so the call is
not rejected by the IP-PBX. Is it possible to achieve this?
Thanks
Diego
___
Users mailing
listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
Users mailing
listUsers@lists.opensips.orghttp://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] Renegotiation
Hello Jorge, You need to use the failure route - http://www.opensips.org/Documentation/Script-Routes-1-10#toc3 . Use the t_on_failure() function (from TM module) to arm a failure route before sending the INVITE out (before the t_relay()). The failure route will be triggered when receiving the negative reply - use t_check_status() [http://www.opensips.org/html/docs/modules/1.10.x/tm.html#id294673] to check the reply code and if 415, simply change the RURI and send the INVITE out again. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 04.03.2014 13:13, Jorge Ortea wrote: Hi all, Anyone know how to do this? Thanks. Regards. From: dar...@hotmail.com To: stefano.pis...@omnianet.it; users@lists.opensips.org Date: Fri, 28 Feb 2014 13:12:16 +0100 Subject: Re: [OpenSIPS-Users] Renegotiation Hi Stefano, How I can do that? Very thanks. Regards. Date: Thu, 27 Feb 2014 17:04:29 +0100 From: stefano.pis...@omnianet.it To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] Renegotiation You can trap the 415 response from the called peer and send the call through asterisk to get transcoding. Il 27/02/2014 16.51, Jorge Ortea ha scritto: Hi all, I have a scenario with OpenSIPS 1.8 and Asterisks 1.4. Proxy SIP has two ways to manage a call, the first is B2BUA and second is be relay between UAC and Asterisk. I have a problem, when OpenSIPS works as B2BUA and both UAC can't negotiate codec then this call failed. I would like restart this same call at second way (with Asterisk). Is that possible? Thanks. Regards. ___ 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Looking for OpenSips consultancy service
Hello Jerry, I have to apology, but we simply cannot answer to all emails we get in a really short time frame. I personally was off the office last week traveling to Mobile World Congress and off the emails. Sometimes patience is a virtue :) If your inquiry is still open, just drop me an email. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 27.02.2014 05:03, jerrynguyen75 wrote: Hi Ovidiu, Yes, i did ... and select the first one OpenSips Solution (try to find the best)...but there are no reply from them. Best regards Jerry -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Looking-for-OpenSips-consultancy-service-tp7589776p7589804.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] Opensips 1.9.1 and REGISTER process
Hello Denis,
Have you tried to use the E option in the save() function? see:
http://www.opensips.org/html/docs/modules/1.9.x/registrar.html#id250454
It should have the same effect (setting the max expire), but is
per REGISTER bases. Just to see if this works for you.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 03.03.2014 15:39, dpa wrote:
Hello!
There is one question.
A little part of opensips.cfg
.
modparam("registrar", "default_expires", 60)
modparam("registrar", "max_expires", 60)
modparam("registrar", "min_expires", 0)
..
If I enter register timeout on my SIP UA to
1600, for example, Opensips will return to SIP UA 1600
timeout.
In 1.6.4-2 there were no problem with it. If I
enter 1600 timeout Opensips returned 60 and after 60 s there
was another attempt to register to Opensips.
What did I miss?
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] codec_delete_except_re SDP Corruption
Hello, Is the codec deletion the only change you do over the SDP (codec related) ? What opensips version are you using ? Also, could you post (privately is needed) the exact original and modified SDP ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 28.02.2014 23:09, Kneeoh wrote: I am using the codec codec_delete_except_re function to remove unwanted codecs: codec_delete_except_re(PCMU|G729|telephone-event); Apparently it is placing or leaving a space (hex 20) at the end of the media description line if the codec at the end of the description is removed by this function. Which I'm told is not syntactically correct. The net result is that upstream vendors 488 the calls on which codecs are removed and have this space at the end of the media description line. Has anyone else encountered this issue and resolved it? Thanks in advance ___ 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
Hello,
The best options for you is to use dialog module with topology hiding.
This can be easily combined with any of the media relays (rtpproxy or
mediaproxy) for hiding the media path.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 28.02.2014 10:14, ? ?? wrote:
Hi. Please help.
We have:
One MGW: Cisco AS5350
UserID=telephone number and registration on OpenSips through MySQL
Call to PSTN pass through MGW with prefix :
Now, such a scheme works:
(UAC )sip-Opensips 1.7---SIP---MGW Cisco
85.85.85.95 85.85.85.85 85.85.85.11
RTP---MGW
CiscoPSTN
Here is an example CFG-file that works now:
The message 183 prefix and visible IP gateway. And that could be a
threat of fraud.
Here: if you use the function topology_hiding (); it does not happen a
fair exchange:
BYE comes to the message 404, Not here rather than 200 OK
I use client_nat_test to cut off all requests for registration are
NAT, but it does not work!
port=5060
listen=udp:85.85.85.85:5060 http://85.85.85.85:5060 #Opensips-server
route{
if (has_totag()) {
if (loose_route()) {
if (is_method(BYE)) {
setflag(1);
setflag(3);}
else if (is_method(INVITE)) {
#topology_hiding();
record_route();}
route(1);}
else {
if ( is_method(ACK) ) {
if ( t_check_trans() ) {
t_relay();
exit;}
else {
exit;
}}
sl_send_reply(404,Not here);
}
exit;
}
#initial requests
if (is_method(CANCEL)){
if (t_check_trans())
t_relay();
exit;}
t_check_trans();
# authenticate if from local subscriber (uncomment to enable auth)
# authenticate all initial non-REGISTER request that pretend to be
# generated by local subscriber (domain from FROM URI is local)
if (!(method==REGISTER) from_uri==myself) #/*no multidomain version*/
{if (!proxy_authorize(, subscriber))
{proxy_challenge(, 0);
exit;}
if (!db_check_from())
{sl_send_reply(403,Forbidden auth ID);
exit;}
consume_credentials();
}
# 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
if (!is_method(REGISTER|MESSAGE)) record_route();
# account only INVITEsif (is_method(INVITE))
{
# if (!src_ip==85.85.85.11) #CISCO MGW IP
#{
#topology_hiding();
#}
setflag(1); # do accounting
}
if (!uri==myself)## replace with following line if multi-domain
support is used
{
route(1);}
# requests for my domain
if (is_method(PUBLISH)){
sl_send_reply(503, Service Unavailable);
exit;}
if (is_method(REGISTER)){
#if(client_nat_test(3))
#{
#sl_send_reply(403, Not working NAT);
#exit;
#}
# authenticate the REGISTER requests (uncomment to enable auth)
if (!www_authorize(, subscriber)){
www_challenge(, 0);
exit;}
if (!db_check_to()) {
sl_send_reply(403,Forbidden auth ID);
exit;}
if (!save(location))
sl_reply_error();
exit;
}
if ($rU==NULL) {
# request with no Username in RURI
sl_send_reply(484,Address Incomplete);
exit;
}
# do lookup with method filtering
if ((src_ip==85.85.85.11) (!lookup(location)))
{
switch ($retcode) {
case -1:
case -3:
t_newtran();
t_reply(404, Not Found);
exit;
case -2:
sl_send_reply(405, Method Not Allowed);
exit;
}}
# when routing via usrloc, log the missed calls also
setflag(2);
if (src_ip==85.85.85.11) {
route(1);}
route(3);
}
route[1] {
# for INVITEs enable some additional helper routes
if (is_method(INVITE)) {
t_on_branch(2);
t_on_reply(2);
t_on_failure(1);}
if (!t_relay()) {
sl_reply_error();};
exit;}
route[3] {
prefix();
rewritehostport(85.85.85.11:5060 http://85.85.85.11:5060);
if (!t_relay()) {
sl_reply_error();
};exit;
}
branch_route[2] { xlog(new branch at $ru\n);}
onreply_route[2] { xlog(incoming reply\n); }
failure_route[1] {
if (t_was_cancelled()) {exit;}}
It's not safe, it's necessary to build a new wiring diagram:
(UAC )---sip,RTP(Opensips---rtp,SIP--)-MGW Cisco---PSTN
85.85.85.95(85.85.85.85 192.168.0.2) 192.168.0.3
questions:
1. to hide the network topology from the users (can be used dialog
module, function: topology_hiding?)
2. hide RTP traffic to MGW for Opensips-server (can be used MediaProxy
or rtpproxy)?
3. Cut off all who are NAT!!!
Please, give examples opensips.cfg-file ?
___
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] B2BUA REFER scenario
Hello, I currently have this configuration: PSTN SIP/ALG Router OpenSips 1.10 IVR OpenSips has a single IP on the private network. I have configured opensips using top hiding in the dialog module and it works fine for calls to ptsn and calls from pstn. I have also configured opensips using B2BUA top hiding and it also works fine for calls to ptsn and calls from pstn. Now I want to test B2BUA REFER scenario (where calls from PSTN are answered by IVR, then IVR does a REFER to another PSTN number). When the IVR sends REFER the call is dropped after .6 seconds. The flow that I've seen in the trace is below: PSTN opensips IVR invite+SDP (Call1) | - Trying (Call1) | | invite+SDP (Call2)- | - OK+SDP (Call2) - OK+SDP (Call1) | Ack (Call1) | | ACK (Call2) - Ivr dialog take place here | - REFER (Call2) - Invite (Call1) | | Accepted (Call2) - | BYE (Call2) - Trying (Call1) | OK+SDP (Call1) | - Invite+SDP(Call3)| | - OK (Call2) Trying (Call3) - | OK+SDP (Call1) - | OK+SDP (Call1) - | 0.6 seconds elapse here Bye (Call1) - | - OK (Call1) | - Cancel (Call3) | OK (Call3) - | Req Termd (Call3) -| - Ack (Call3) | It looks as though the PSTN times out waiting for an ACK after sending OK+SDP(Call1) a couple times and then waiting .6 seconds. The question is - what should the flow look like? According to this post: http://lists.opensips.org/pipermail/users/2012-April/021352.html, things appear to be working as expected up to the point where we receive Trying (Call3). Should I be seeing the OK+SDP from call 3 next? I'd like to troubleshoot further but I'm not sure where to look. Thanks! Tony ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] codec_delete_except_re SDP Corruption
Sure, I'll get that for you. I've got some additional data that may answer your question. The problem seems to occur when the codec being removed is at the end of the SDP offer. So you'll get something like 0 18 101 19 in the media description line and 19 is the one you remove. It leaves the space after the 101, which is causing the problem. On Wednesday, March 5, 2014 12:57 PM, Bogdan-Andrei Iancu bog...@opensips.org wrote: Hello, Is the codec deletion the only change you do over the SDP (codec related) ? What opensips version are you using ? Also, could you post (privately is needed) the exact original and modified SDP ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 28.02.2014 23:09, Kneeoh wrote: I am using the codec codec_delete_except_re function to remove unwanted codecs: codec_delete_except_re(PCMU|G729|telephone-event); Apparently it is placing or leaving a space (hex 20) at the end of the media description line if the codec at the end of the description is removed by this function. Which I'm told is not syntactically correct. The net result is that upstream vendors 488 the calls on which codecs are removed and have this space at the end of the media description line. Has anyone else encountered this issue and resolved it? Thanks in advance ___ 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] CDR in flat file
Hi,
I am trying to configure the OpenSIPS v.1.10 to make it write the CDRs to flat
files.
But no luck yet.
The opensips.cfg looks like this:
...
loadmodule dialog.so
modparam(dialog, dlg_match_mode, 1)
loadmodule acc.so
modparam(acc, detect_direction, 1)
modparam(acc, failed_transaction_flag, ACC_FAILED)
modparam(acc, db_url, flatstore:/var/log/acc)
modparam(acc, log_flag, LOG_FLAG)
modparam(acc, log_facility, LOG_LOCAL0)
modparam(acc, cdr_flag, CDR_FLAG)
modparam(acc, db_flag, DB_FLAG)
loadmodule db_flatstore.so
modparam(db_flatstore, flush, 0)
modparam(db_flatstore, suffix, $time(%H))
...
route[relay]{
if (is_method(INVITE)){
rewritehostport(54.84.239.100:5080);
create_dialog();
setflag(LOG_FLAG);
setflag(DB_FLAG);
setflag(CDR_FLAG);
if (!t_relay()) {
send_reply(500,Internal Error);
}
}
}
...
Am I missing something or doing something wrong?
Thanks in advance
Gary
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] MediaProxy scalability FAQ updated
Some people reportedly saw at least two calls, but they had a different sense of humor. -- Adrian On 04 Mar 2014, at 14:27, david da...@styleflare.com wrote: Mediaproxy can handle at least one simultaneous call, regardless of the hardware resources available providing no other program competes with the same resources on that machine. Bigger scalability can be achieved by adding more hardware. ??? Mediaproxy can handle at least one simultaneous call? On 3/4/14 11:15 AM, a...@ag-projects.com wrote: http://mediaproxy.ag-projects.com/projects/mediaproxy/wiki/Scalability Adrian ___ 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] MediaProxy scalability FAQ updated
truly, i saw it, it can handle it least one call! On Thu, Mar 6, 2014 at 12:40 AM, Adrian Georgescu a...@ag-projects.comwrote: Some people reportedly saw at least two calls, but they had a different sense of humor. -- Adrian On 04 Mar 2014, at 14:27, david da...@styleflare.com wrote: Mediaproxy can handle at least one simultaneous call, regardless of the hardware resources available providing no other program competes with the same resources on that machine. Bigger scalability can be achieved by adding more hardware. ??? Mediaproxy can handle at least one simultaneous call? On 3/4/14 11:15 AM, a...@ag-projects.com wrote: http://mediaproxy.ag-projects.com/projects/mediaproxy/wiki/Scalability Adrian ___ Users mailing listUsers@lists.opensips.orghttp://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 -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] MediaProxy scalability FAQ updated
Wait wait wait Can you please clarify.. does one simultaneous call mean one call, or two? Certainly this means two? On Tue, Mar 4, 2014 at 10:15 AM, a...@ag-projects.com wrote: http://mediaproxy.ag-projects.com/projects/mediaproxy/wiki/Scalability Adrian ___ 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 1.9.1 and REGISTER process
Hello Bogdan, Yes E flags is working. But without in doesn`t From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Wednesday, March 05, 2014 9:56 PM To: OpenSIPS users mailling list; Denis Putyato Subject: Re: [OpenSIPS-Users] Opensips 1.9.1 and REGISTER process Hello Denis, Have you tried to use the E option in the save() function? see: http://www.opensips.org/html/docs/modules/1.9.x/registrar.html#id250454 It should have the same effect (setting the max expire), but is per REGISTER bases. Just to see if this works for you. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 03.03.2014 15:39, dpa wrote: Hello! There is one question. A little part of opensips.cfg .. modparam(registrar, default_expires, 60) modparam(registrar, max_expires, 60) modparam(registrar, min_expires, 0) ... If I enter register timeout on my SIP UA to 1600, for example, Opensips will return to SIP UA 1600 timeout. In 1.6.4-2 there were no problem with it. If I enter 1600 timeout Opensips returned 60 and after 60 s there was another attempt to register to Opensips. What did I miss? Thank you for any help. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users image001.gif___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips1.7 with MediaProxy
In the incoming message to the UAC 183 can be seen: the prefix and IP gateway. Gateway waits INVITE from everyone. Therefore, it is necessary at least to hide this information!!! There is a suggestion - use the full proxy SIP and RTP traffic from the UAC. Necessary: MGW hide behind opensips-server: UAC(85.85.85.95)OPENSIPS(85.85.85.85, 192.168.0.85)--MGW(192.168.0.11) 1. What better use for the full RTP-traffic hiding: rtpproxy or mediaproxy? 2. I need help setting: mediaproxy or rtpproxy. 3. Example configuration file opensips.cfg such interaction: opensips-mediaproxy or opensips-rtpproxy. that is for today: [root@x ~]# ps ax | grep rtpp 17445 ?Ssl0:00 rtpproxy 17451 pts/2S+ 0:00 grep rtpp 30335 ?Ssl0:13 rtpproxy -u rtpproxy [root@x ~]# ps ax | grep disp 17456 pts/2S+ 0:00 grep disp 30008 ?SL 0:00 python ./media-dispatcher [root@x ~]# ps ax | grep rel 17458 pts/2S+ 0:00 grep rel 30016 ?SL 0:35 python ./media-relay and opensips 1.7 eth.0 ip=85.85.85.85 eth.1 ip=192.168.0.85 4. if we use rtpproxy, then iptables configuration is needed .. 5. if we use mediaproxy then where are the config files? and how to set them for the dispatcher and relay.? 6. In logic OPENSIPS not need to use checks: if the client is behind NAT on the remote end, and vice versa - do not allow the registration of such users! 05.03.2014 20:48, Bogdan-Andrei Iancu writes: If you have any particular questions on the setup, I will try to help you. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
