Re: [OpenSIPS-Users] Media Proxy setup
Hi, On Jul 13, 2011, at 1:46 AM, Kurtis vel wrote: Does anyone have advice on the best way to arrange media proxy? Should it run on the same servers as opensips or should I offf load the cpu and ram onto its own box? Depends on your needs :-) The MediaProxy architecture allows you to gradually scale without any trouble. You may start with OpenSIPS and MediaProxy (both dispatcher and relay) on the same machine, and as load increases you may add more Relay nodes and the load (for RTP relaying) will be evenly distributed by the dispatcher. Regards, -- Saúl Ibarra Corretgé AG Projects ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] ICE and Mediaproxy break when using append_branch and serial_branches/next_branches
Hi,
On Jul 12, 2011, at 9:29 PM, duane.lar...@gmail.com wrote:
I am using append_branch and serial_branches/next_branches to implement a
FindMe/FollowMe feature. I just noticed that when I do this I am getting no
audio clients that don't have a public IP. If I bypass the FindMe/FollowMe
stuff audio works just fine. I am not sure what exactly is going on when it
breaks. The scenario I have right now is
90133XXX18 calls 90127X2XX9
modparam(mediaproxy, mediaproxy_socket,
/var/run/mediaproxy/dispatcher.sock)
modparam(mediaproxy, ice_candidate, low-priority)
OpenSIPS knows that when 90127X2XX9 is called to first set $ru to 90127X2XX9
and then append_branch. Then OpenSIPS sets $ru to an outbound number that has
to be reached via SIP trunk provider. Q-Values are set for both numbers so
that the outbound number is called first and then 90127X2XX9 is called. Then
I call serialize_branches(1) and then next_branches. I turn on Mediaproxy by
doing the following
if (method==INVITE !has_totag()) {
# We can also use a specific media relay if we need to
engage_media_proxy();
}
Then the call is made. I notice when doing a siptrace on the call that
sometimes my c=IN IP4 in the SDP never has the IP of the Mediaproxy when it
calls the outbound number and then the 90127X2XX9 number. Then other times it
does include the mediaproxy IP which is 173.XXX.XXX.111. It's just random
when I test a call. Engage_media_proxy is called when the call to the
outbound number is made and also when the call to the 90127X2XX9 number is
made. If I disable ICE on the Blink client it doesn't seem to make a
difference on this problem.
I am using a branch version of OpenSIPS that was posted yesterday and I just
upgraded Mediaproxy Dispatcher and Relay to the latest version without any
luck.
Its been a while since I haven't used serial forking, but since you are using
engage_media_proxy you may need to check how serial forking and the dialog
module work together. Are you calling engage_media_proxy for every new branch
that is appended? That is, is this always failing if the *first* endpoint
doesn't answer?
Regards,
--
Saúl Ibarra Corretgé
AG Projects
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Re: [OpenSIPS-Users] [OpenSIPS-Business] [RELEASE] OpenSIPS 1.7.0 - a new major release is out
Good morning Bogdan and everybody, I have build packages for Debian i386 / Debian amd64 / CentOS el5-i386 / CentOS x86_64 and modified the packaging files in the SVN branch and trunk so it will work for everybody, cf commit SVN 8175 These packages are available for testing on centos.leurent.eu and debian.leurent.eu as usual (cf opensips download page), feel free to import test or rebuild from source Have a nice day Le mardi 12 juillet 2011 19:57:33, Bogdan-Andrei Iancu a écrit : One more half an year, one more major release - OpenSIPS 1.7.0 beta release is out *OpenSIPS 1.7.0* comes with several major improvements (DB area, dialog support, TCP and debugging), but also with new functionalities (like Event Notification Interface, UAC Registrant module, scripting support, etc). A complete compilation with all the additions and improvements for *OpenSIPS 1.7.0 *release is available under: http://www.opensips.org/Main/Ver170 A comprehensive migration document is available also: http://www.opensips.org/Resources/DocsMigration164to170 Many thanks to all the people who got involved in this release (and in the overall OpenSIPS project) and contributed with code, with testing and debugging, with patches or reports, with support on the lists, help with packaging and documentation. I will avoid listing names, not because they do not deserve it, but simply because it will impossible to list list everybody here and I do not what to be unfair with some of them (because I simply forgot a name or because of the limited space). But nevertheless, I want to thanks you all for out great job and not in my behalf, but in the behalf of people who will find this piece of software a useful tool. The full Changelog is available here: http://opensips.org/pub/opensips/1.7.0/src/ChangeLog To get the *OpenSIPS 1.7.0* version, see : * website page:http://www.opensips.org/Resources/Downloads * SF project: https://sourceforge.net/project/showfiles.php?group_id=232389package_id=281827release_id=670379 Note that for the moment only the source tarballs are available. The packages (debs,rpms, etc) will be generated starting now. If anybody can help in generating packages for different distros or architectures, please let me know and I will upload them on the website. Best regards, Bogdan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Manipulating sdp headers
Hi, Is there anyway to manipulate the sdp headers apart from the codec information. I have tried various methods in the textops module but they only seem to address the SIP part of the message. The problem I have is the in some circumstances I get 2 (c) records in the sdp of which rtpproxy only updates the last. Some downstream apps are only processing the first (c) record resulting in no speech as any rtp is bypassing the proxy. If I could strip the first (c) record it would be a quick fix. Any help you could offer would be greatly appreciated. Regards Chris ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Problem with siptrace module
Hi Duane, Dani,
I've tried adding the call to create_dialog() and still no messages I've
also added a call to sip_trace() at the beginning of route, and nothing.
This is my script now:
##Global Parameters #
debug=3
#log_facility=LOG_LOCAL6
fork=yes
children=4
/* uncomment the following lines to enable debugging */
#debug=6
#fork=no
#log_stderror=yes
/* uncomment the next line to disable TCP (default on) */
disable_tcp=yes
port=5060
/* uncomment and configure the following line if you want opensips to
bind on a specific interface/port/proto (default bind on all available)
*/
listen=udp:192.168.2.154:5060
# -- module loading --
mpath=/usr/local/lib/opensips/modules/
loadmodule maxfwd.so
loadmodule sl.so
loadmodule tm.so
loadmodule dispatcher.so
loadmodule mi_fifo.so
loadmodule signaling.so
loadmodule options.so
loadmodule textops.so
loadmodule db_mysql.so
loadmodule siptrace.so
loadmodule dialog.so
loadmodule rr.so
# - setting module-specific parameters ---
# -- dispatcher params --
modparam(mi_fifo, fifo_name, /tmp/opensips_fifo)
modparam(dispatcher, ds_ping_from, sip:proxy@192.1.8.2.154)
modparam(dispatcher, ds_ping_interval, 30)
modparam(dispatcher, ds_probing_threshhold, 2)
modparam(dispatcher, ds_probing_mode, 1)
modparam(dispatcher, list_file,
/usr/local/etc/opensips/dispatcher.list)
# modparam(dispatcher, force_dst, 1)
modparam(siptrace, db_url,
mysql://root:Viamonte1621@localhost/opensips)
#modparam(siptrace, enable_ack_trace, 1)
modparam(siptrace, trace_on, 1)
modparam(siptrace, table, sip_trace)
#modparam(siptrace, trace_flag, 22)
#modparam(acc, log_level, 1)
#modparam(acc, log_flag, 1)
#modparam(acc, db_url, mysql://root:Viamonte1621@localhost/opensips)
route{
sip_trace();
if ( !mf_process_maxfwd_header(10) )
{
sl_send_reply(483,To Many Hops);
drop();
};
if (is_method(OPTIONS)) {
options_reply();
exit;
}
if(is_method(INVITE)){
create_dialog();
trace_dialog();
}
ds_select_dst(1, 0);
if ($retcode0) {
xlog([Redmond] Service full\n);
sl_send_reply(500,Service full);
exit;
}
forward();
}
failure_route[1] {
if (t_check_status((408)|(5[0-9][0-9]))) {
ds_mark_dst();
if (ds_select_dst(1, 0)) {
forward();
} else {
t_reply(503, Service Unavailable);
}
}
}
Any other ideas?
Thanks
Diego
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of
duane.lar...@gmail.com
Sent: martes, 12 de julio de 2011 06:35 p.m.
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Problem with siptrace module
Diego,
So I think your wanting to do Stateless routing. I just saw this posting of
Known Issues under version 1.7. Not sure if it applies to your version
also
http://www.opensips.org/html/docs/modules/1.7.x/siptrace.html#id250363
I see you are using sip_trace() in your script. I have personally never
messed with stateless and siptrace. Might need someone else to weigh in on
this.
On Jul 12, 2011 1:09pm, Diego Barberio diego.barbe...@redmondsoftware.com
wrote:
Hi All, I'm having a minnor configuration problem with the siptrace.I'm
trying to trace all the messages and their responses, but I'm only able to
log the methods (INVITE, ACK and BYE) but not the responses.This is my
configuration script. ##Global Parameters #
debug=3#log_facility=LOG_LOCAL6 fork=yeschildren=4 /* uncomment the
following lines to enable debugging */#debug=6#fork=no#log_stderror=yes /*
uncomment the next line to disable TCP (default on) */disable_tcp=yes
port=5060 /* uncomment and configure the following line if you want opensips
to bind on a specific interface/port/proto (default bind on all available)
*/listen=udp:192.168.2.154:5060 # -- module loading
--mpath=/usr/local/lib/opensips/modules/lo
admodule maxfwd.soloadmodule sl.soloadmodule tm.soloadmodule
dispatcher.soloadmodule mi_fifo.soloadmodule signaling.soloadmodule
options.soloadmodule textops.soloadmodule db_mysql.soloadmodule
siptrace.so#loadmodule acc.so# - setting module-specific
parameters ---# -- dispatcher params --modparam(mi_fifo,
fifo_name, /tmp/opensips_fifo) modparam(dispatcher, ds_ping_from,
sip:proxy@192.1.8.2.154)modparam(dispatcher,
[OpenSIPS-Users] Manipulating sdp headers2
Hi,
Okay I am now accessing the headers in sdp but I cannot seem to use a
variable in the textops methods.
If I do the following...
$var(line1)=$(rb{sdp.line,c,0});
$var(line2)=$(rb{sdp.line,c,1});
Xlog($var(line1));
If ($var(line2)!=){
Xlog($var(line2));
Replace_body($var(line2),);
}
I get line1 and line2 variables printout but replace_body does nothing.
If I just put in Replace_body(c=IN IP4,); it deletes the relevant
part so the function is working but it just won't except a variable as
the search term.
Any help would be greatly appreciated.
Regards
Chris
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Re: [OpenSIPS-Users] How to limit calls to specific number
HI, first aaa_radius_auth and specific sql procedure in sql server. the second asterisk/freeswitch load balncing Dani On 07/12/11 17:06, duane.lar...@gmail.com wrote: For your first question would this work? http://www.ag-projects.com/projects-products-96/535-call-control For your second question I hear that SEMS has better performance than Asterisk or Freeswitch, but I think you have to put a lot of work into it because it isn't as easy to work with as Asterisk. http://www.iptel.org/sems If you can't figure SEMS out then maybe your best bet for an IVR that can handle 1000 calls would be Asterisk Clustering. On Jul 12, 2011 8:03am, Akib Sayyed akibsay...@gmail.com wrote: hello guys i am creating billing system for premium number portal here i need to allow specific number of minutes to a DID. how can i do that any idea's also i need to handle 1000 call i know Opensips can handle it but i want to route those calls to IVR server tell me best server for IVR which can handle 1000 concurrent calls for ivr also server hardware needed if its asterisk -- Akib Sayyed Matrix-Shell akibsay...@gmail.com akibsay...@matrixshell.com Mob:- +91-966-514-2243 ___ 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] Manipulating sdp headers2
In the documentation I think it says the first argument you give
replace_body has to be a regular expression. So you can't give it a
variable.
Here are some examples
http://opensips-open-sip-server.1449251.n2.nabble.com/Modify-SDP-td3945462.html#a4014981
http://opensips-open-sip-server.1449251.n2.nabble.com/Regular-expression-problems-with-replace-body-td4116208.html
On Jul 13, 2011 7:47am, Chris Martineau ch...@ghosttelecom.com wrote:
Hi, Okay I am now accessing the headers in sdp but I cannot seem to use a
variable in the textops methods. If I do the following...
$var(line1)=$(rb{sdp.line,c,0});$var(line2)=$(rb{sdp.line,c,1});Xlog(“$var(line1)”);
If ($var(line2)!=””){ Xlog(“$var(line2)”);
Replace_body(“$var(line2)”,””);} I get line1 and line2 variables printout
but replace_body does nothing. If I just put in Replace_body(“c=IN
IP4”,””); it deletes the relevant part so the function is working but it
just won't except a variable as the search term. Any help would be
greatly appreciated. Regards Chris
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Re: [OpenSIPS-Users] Manipulating sdp headers2
Thanks,
I have just figured that out and have got what I wanted to work.
Regards
Chris
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of
duane.lar...@gmail.com
Sent: 13 July 2011 14:54
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Manipulating sdp headers2
In the documentation I think it says the first argument you give
replace_body has to be a regular expression. So you can't give it a
variable.
Here are some examples
http://opensips-open-sip-server.1449251.n2.nabble.com/Modify-SDP-td39454
62.html#a4014981
http://opensips-open-sip-server.1449251.n2.nabble.com/Regular-expression
-problems-with-replace-body-td4116208.html
On Jul 13, 2011 7:47am, Chris Martineau ch...@ghosttelecom.com wrote:
Hi, Okay I am now accessing the headers in sdp but I cannot seem to
use a variable in the textops methods. If I do the following...
$var(line1)=$(rb{sdp.line,c,0});$var(line2)=$(rb{sdp.line,c,1});Xlog($v
ar(line1)); If ($var(line2)!=){Xlog($var(line2));
Replace_body($var(line2),);} I get line1 and line2 variables
printout but replace_body does nothing. If I just put in
Replace_body(c=IN IP4,); it deletes the relevant part so the
function is working but it just won't except a variable as the search
term. Any help would be greatly appreciated. Regards Chris
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Re: [OpenSIPS-Users] 1.7 and rtpproxy issues
Hi Chris, The error is pretty obvious: one (or more) of your RTPProxy node doesn't support media playback. Regarding the RTPProxy broken pipe errors, how often are you getting those errors? Regards, Razvan Crainea OpenSIPS Developer On 13.07.2011 18:02, Chris Martineau wrote: Hi, Since loading the latest 1.7 I have had a problem with the playback on rtpproxy. I startup opensips with 8 rtpproxy instances 4 each on 2 other servers and all works okay. After a random period opensips starts throwing an error as follows ERROR:rtpproxy:rtpproxy_stream: required functionality is not supported by the version of the rtpproxy running on the selected node. Please upgrade the rtpproxy and try again. This seems to happen on one of the servers which gives no playback while the other seems to work fine playing back as normal. The only errors I see on rtpproxy are Broken pipe errors which I have raised previously but haven't had a response yet. Restart the instances on the effected server and it starts working again. The servers were built at the same time with the same git version of rtpproxy and were working fine. Any ideas? Many thanks Chris ___ 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] ICE and Mediaproxy break when using append_branch and serial_branches/next_
I set the Q value of the 90127X2XX9 device so that it gets called first and
the outbound number gets called second. When I do this I am able to get
two-way audio when ringing the 90127X2XX9 client and also when the client
picks up. So it appears to work as normal when I do that.
I am calling engage_media_proxy every time since if (method==INVITE
!has_totag()) always is true.
In this scenario the 90127X2XX9 gets called first and ICE makes it so that
both my clients talk directly to each other since they are on the same
subnet. Then when the outbound number is called all media goes through my
MediaRelay.
And to be sure I disabled ICE on one of the clients and called again. This
time Mediaproxy was used when both of the clients talked to each other. And
I called a second time and let the first call to the client fail and when
the outbound number was called I had two-way audio I am guessing because it
is to a Public SIP gateway.
So there does seem to be an issue with serial forking. I think it only
fails if the *first* endpoint doesn't answer and the second endpoint is a
Private IP address client.
On Jul 13, 2011 2:28am, Saul Ibarra Corretge s...@ag-projects.com wrote:
Hi,
On Jul 12, 2011, at 9:29 PM, duane.lar...@gmail.com wrote:
I am using append_branch and serial_branches/next_branches to implement
a FindMe/FollowMe feature. I just noticed that when I do this I am
getting no audio clients that don't have a public IP. If I bypass the
FindMe/FollowMe stuff audio works just fine. I am not sure what exactly
is going on when it breaks. The scenario I have right now is
90133XXX18 calls 90127X2XX9
modparam(mediaproxy, mediaproxy_socket, /var/run/mediaproxy/dispatcher.sock)
modparam(mediaproxy, ice_candidate, low-priority)
OpenSIPS knows that when 90127X2XX9 is called to first set $ru to
90127X2XX9 and then append_branch. Then OpenSIPS sets $ru to an outbound
number that has to be reached via SIP trunk provider. Q-Values are set
for both numbers so that the outbound number is called first and then
90127X2XX9 is called. Then I call serialize_branches(1) and then
next_branches. I turn on Mediaproxy by doing the following
if (method==INVITE !has_totag()) {
# We can also use a specific media relay if we need to
engage_media_proxy();
}
Then the call is made. I notice when doing a siptrace on the call that
sometimes my c=IN IP4 in the SDP never has the IP of the Mediaproxy
when it calls the outbound number and then the 90127X2XX9 number. Then
other times it does include the mediaproxy IP which is 173.XXX.XXX.111.
It's just random when I test a call. Engage_media_proxy is called when
the call to the outbound number is made and also when the call to the
90127X2XX9 number is made. If I disable ICE on the Blink client it
doesn't seem to make a difference on this problem.
I am using a branch version of OpenSIPS that was posted yesterday and I
just upgraded Mediaproxy Dispatcher and Relay to the latest version
without any luck.
Its been a while since I haven't used serial forking, but since you are
using engage_media_proxy you may need to check how serial forking and the
dialog module work together. Are you calling engage_media_proxy for every
new branch that is appended? That is, is this always failing if the
*first* endpoint doesn't answer?
Regards,
--
Saúl Ibarra Corretgé
AG Projects
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[OpenSIPS-Users] Variable length mask on permissions module
Hello all. Is there any restriction on permissions module of OpenSIPS 1.5.x for handling multiple entries with different lengths of subnet mask? Also, is there any report of missbehavior of the module when inconsistent entries (hosts with subnet mask) are loaded into? Thanks in advance for your answer. -- Sergio Gutiérrez ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] avp_db_load - ERROR:core:parse_uri: uri too short
I am starting to configure a concurrent call limit in opensips 1.6.4 and
so far have the following configuration :-
$var(maxchannels) = 1;
if (avp_db_load($au/username,s)) {
$var(maxchannels) = $avp(s:maxchannels);
}
xlog(L_NOTICE: User $au max channels = $var(maxchannels)\n);
In the database I have :-
INSERT INTO `usr_preferences` (`id`, `uuid`, `username`, `domain`,
`attribute`, `type`, `value`, `last_modified`) VALUES
(1, '', '1000', '', 'maxchannels', 0, '2', '2011-07-13 00:00:01');
The problem I am having is than when the call goes out I get the
following error :-
Jul 13 16:24:25 vmopensips1 /sbin/opensips[1828]: ERROR:core:parse_uri:
uri too short: 1000 (4)
Jul 13 16:24:25 vmopensips1 /sbin/opensips[1828]:
ERROR:avpops:ops_dbload_avps: failed to parse uri
Jul 13 16:24:25 vmopensips1 /sbin/opensips[1828]: L_NOTICE: User 1000
max channels = 1
I dont understand why it is thinking its a uri and not username.
Thanks
Gareth
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Re: [OpenSIPS-Users] ICE and Mediaproxy break when using append_branch and serial_branches/next_branches
Here is a debug=5 showing the call to outbound number then 90127X2XX9.
http://pastebin.com/ZpsR5WG8
Here is a siptrace of the same
http://pastebin.com/Rvg6nNGa
Not sure if it really helps
On Wed, Jul 13, 2011 at 2:28 AM, Saul Ibarra Corretge
s...@ag-projects.comwrote:
Hi,
On Jul 12, 2011, at 9:29 PM, duane.lar...@gmail.com wrote:
I am using append_branch and serial_branches/next_branches to implement a
FindMe/FollowMe feature. I just noticed that when I do this I am getting no
audio clients that don't have a public IP. If I bypass the FindMe/FollowMe
stuff audio works just fine. I am not sure what exactly is going on when it
breaks. The scenario I have right now is
90133XXX18 calls 90127X2XX9
modparam(mediaproxy, mediaproxy_socket,
/var/run/mediaproxy/dispatcher.sock)
modparam(mediaproxy, ice_candidate, low-priority)
OpenSIPS knows that when 90127X2XX9 is called to first set $ru to
90127X2XX9 and then append_branch. Then OpenSIPS sets $ru to an outbound
number that has to be reached via SIP trunk provider. Q-Values are set for
both numbers so that the outbound number is called first and then 90127X2XX9
is called. Then I call serialize_branches(1) and then next_branches. I turn
on Mediaproxy by doing the following
if (method==INVITE !has_totag()) {
# We can also use a specific media relay if we need to
engage_media_proxy();
}
Then the call is made. I notice when doing a siptrace on the call that
sometimes my c=IN IP4 in the SDP never has the IP of the Mediaproxy when
it calls the outbound number and then the 90127X2XX9 number. Then other
times it does include the mediaproxy IP which is 173.XXX.XXX.111. It's
just random when I test a call. Engage_media_proxy is called when the call
to the outbound number is made and also when the call to the 90127X2XX9
number is made. If I disable ICE on the Blink client it doesn't seem to make
a difference on this problem.
I am using a branch version of OpenSIPS that was posted yesterday and I
just upgraded Mediaproxy Dispatcher and Relay to the latest version without
any luck.
Its been a while since I haven't used serial forking, but since you are
using engage_media_proxy you may need to check how serial forking and the
dialog module work together. Are you calling engage_media_proxy for every
new branch that is appended? That is, is this always failing if the *first*
endpoint doesn't answer?
Regards,
--
Saúl Ibarra Corretgé
AG Projects
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--
--
*--*--*--*--*--*
Duane
*--*--*--*--*--*
--
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Re: [OpenSIPS-Users] 1.7 and rtpproxy issues
Hi Razvan, All the rtpproxy instances are pulled from the latest git and they were all working okay. All the instances do work when you first start them but for some reason it starts throwing this error. It can last quite a while before this happens so it seems to support playback when you start it but at some time during running it stops? No errors seem to be thrown at the rtpproxy end although I only have logging at the ERR level. What determines if the rtpproxy supports media playback or not? What could change that would cause it to throw this error mid way through running. It may be something local but I have no idea where to start looking seeing as nothing has changed on the media servers between 1.6 and 1.7! The broken pipe issue seems to only kick in when the call to notify of timeout occurs. Of the 8 instances it happens on all of them except the first one started which would point to some sort of clash or error at the opensips receiving end because it is across both servers. i.e server1: works,broken,broken,broken server2: broken,broken,broken,broken. Any help you could offer would be greatly appreciated. Regards Chris From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea Sent: 13 July 2011 16:12 To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] 1.7 and rtpproxy issues Hi Chris, The error is pretty obvious: one (or more) of your RTPProxy node doesn't support media playback. Regarding the RTPProxy broken pipe errors, how often are you getting those errors? Regards, Razvan Crainea OpenSIPS Developer On 13.07.2011 18:02, Chris Martineau wrote: Hi, Since loading the latest 1.7 I have had a problem with the playback on rtpproxy. I startup opensips with 8 rtpproxy instances 4 each on 2 other servers and all works okay. After a random period opensips starts throwing an error as follows ERROR:rtpproxy:rtpproxy_stream: required functionality is not supported by the version of the rtpproxy running on the selected node. Please upgrade the rtpproxy and try again. This seems to happen on one of the servers which gives no playback while the other seems to work fine playing back as normal. The only errors I see on rtpproxy are Broken pipe errors which I have raised previously but haven't had a response yet. Restart the instances on the effected server and it starts working again. The servers were built at the same time with the same git version of rtpproxy and were working fine. Any ideas? Many thanks Chris ___ 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] serialize_branch breaking $du
This week just isn't my week when it comes to serialize_branch. I just noticed that when I set $ru then $du then append_branch and then set $ru again with a different uri and serialize the call that should be forwarded to the $du isn't being forwarded. Instead the call goes directly to the ip that is in $ru. This can be seen in the following debugs that I already posted to Saul for my Mediaproxy issue. Syslog Debug http://pastebin.com/ZpsR5WG8 Do a search for serialize_branches: Branch information and you will see that the branch uri and duri are all set. Then in this siptrace you see that opensips doesn't send it to the $du destination but instead sends it directly to the uri in $ru. http://pastebin.com/Rvg6nNGa 1. U 2011/07/13 10:49:04.650882 173.XXX.XXX.107:5060 - 64.136.174.30:5060 2. INVITE sip:9014XX7XX9@64.136.174.30:5060 SIP/2.0. 3. Record-Route: sip:173.XXX.XXX.107;lr;ftag=ca22d05c94774292b9249854dada78de;did=3b5.b8577b82. 4. Via: SIP/2.0/UDP 173.XXX.XXX.107;branch=z9hG4bK34d7.0f26b417.0. 5. Via: SIP/2.0/UDP 64.132.245.122:53383 ;received=64.132.245.122;rport=53383;branch=z9hG4bKPj26a970ba1b3f44f9b823cbbe4c063bd6. 6. Max-Forwards: 69. 7. From: SecondDell sip:90133xx...@irock.com ;tag=ca22d05c94774292b9249854dada78de. 8. To: sip:90127x2...@irock.com. 9. Contact: sip:zdcrgabl@64.132.245.122:53383. 10. Call-ID: a46fcbf5977c40c6aa23bac44aa05ff4. 11. CSeq: 3377 INVITE. 12. Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER. 13. Supported: 100rel, norefersub. 14. User-Agent: Blink 0.2.7 (Windows). 15. Content-Type: application/sdp. 16. Content-Length: 693. 17. P-hint: inbound-inbound . 18. P-hint: Route[6]: mediaproxy . I just upgraded to 1.7 and that didn't matter. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] (no subject)
having this error how to solve i have libxml_rpc_xmlparse.so.3 in my /usr/local/lib Jul 13 16:29:58 localhost opensips: ERROR:core:sr_load_module: could not open module /usr/local/lib/opensips/modules/mi_xmlrpc.so: libxmlrpc_xmlparse.so.3: cannot open shared object file: No such file or directory Jul 13 16:29:58 localhost opensips: CRITICAL:core:yyerror: parse error in config file, line 80, column 13-14: failed to load module Jul 13 16:29:58 localhost opensips: ERROR:core:set_mod_param_regex: no module matching mi_xmlrpc found Jul 13 16:29:58 localhost opensips: CRITICAL:core:yyerror: parse error in config file, line 198, column 22-23: Parameter port not found in module mi_xmlrpc - can't set Jul 13 16:29:58 localhost opensips: ERROR:core:main: bad config file (2 errors) -- Akib Sayyed Matrix-Shell akibsay...@gmail.com akibsay...@matrixshell.com Mob:- +91-966-514-2243 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] (no subject)
did you have the libxmlrpc installed on your system ? 2011/7/13 Akib Sayyed akibsay...@gmail.com having this error how to solve i have libxml_rpc_xmlparse.so.3 in my /usr/local/lib Jul 13 16:29:58 localhost opensips: ERROR:core:sr_load_module: could not open module /usr/local/lib/opensips/modules/mi_xmlrpc.so: libxmlrpc_xmlparse.so.3: cannot open shared object file: No such file or directory Jul 13 16:29:58 localhost opensips: CRITICAL:core:yyerror: parse error in config file, line 80, column 13-14: failed to load module Jul 13 16:29:58 localhost opensips: ERROR:core:set_mod_param_regex: no module matching mi_xmlrpc found Jul 13 16:29:58 localhost opensips: CRITICAL:core:yyerror: parse error in config file, line 198, column 22-23: Parameter port not found in module mi_xmlrpc - can't set Jul 13 16:29:58 localhost opensips: ERROR:core:main: bad config file (2 errors) -- Akib Sayyed Matrix-Shell akibsay...@gmail.com akibsay...@matrixshell.com Mob:- +91-966-514-2243 ___ 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] Dialog Ping
When the call is initially set up between the two clients should I see sip OPTIONS messages being sent to both clients every X seconds? I have this set up and I don't see any OPTIONS messages being sent at all during the duration of the good call. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Dialog-Ping-tp6484704p6580315.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] OpenSIPS behind Static NAT (Amazon EC2)
Hi Brendon, I am trying to setup OPENSIPS on Amazon EC2 server, as a load balancer for my two working Asterisk servers. I was able to install it by using the following guide: http://www.opensips.org/Resources/DocsTutRedhat5 After that I have no idea about how to route calls to this opensips server or how to load balance two Asterisk server using this opensips server. Any help is appreciated. Thank you. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-behind-Static-NAT-Amazon-EC2-tp4970920p6580977.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] (no subject)
yes i have installed to latest version On Wed, Jul 13, 2011 at 11:48 PM, Mike Tesliuk m...@ultra.net.br wrote: did you have the libxmlrpc installed on your system ? 2011/7/13 Akib Sayyed akibsay...@gmail.com having this error how to solve i have libxml_rpc_xmlparse.so.3 in my /usr/local/lib Jul 13 16:29:58 localhost opensips: ERROR:core:sr_load_module: could not open module /usr/local/lib/opensips/modules/mi_xmlrpc.so: libxmlrpc_xmlparse.so.3: cannot open shared object file: No such file or directory Jul 13 16:29:58 localhost opensips: CRITICAL:core:yyerror: parse error in config file, line 80, column 13-14: failed to load module Jul 13 16:29:58 localhost opensips: ERROR:core:set_mod_param_regex: no module matching mi_xmlrpc found Jul 13 16:29:58 localhost opensips: CRITICAL:core:yyerror: parse error in config file, line 198, column 22-23: Parameter port not found in module mi_xmlrpc - can't set Jul 13 16:29:58 localhost opensips: ERROR:core:main: bad config file (2 errors) -- Akib Sayyed Matrix-Shell akibsay...@gmail.com akibsay...@matrixshell.com Mob:- +91-966-514-2243 ___ 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 -- Akib Sayyed Matrix-Shell akibsay...@gmail.com akibsay...@matrixshell.com Mob:- +91-966-514-2243 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
