[SR-Users] Are there any good arguments (except political ones) for not using Kamailio within Docker?

Gents, I've been loving OpenVZ for a multitude of deployments in the last 24 months, but recently I stumbled over Docker, and I really like it! It's simplicity is very appealing. I was trying to find good reasons not to use Docker in production environments. So far I couldn't get any further tha

Re: [SR-Users] Dispatcher - failure_route to specific node

On 16/09/14 15:55, Federico Cabiddu wrote: looking at the trace, it seems to me that the 407 is not sent by Kamailio but by the FS1 server ("User-Agent: IMX" in the 407, as for the 1st 407 sent by FS2), and that you are probably missing in the trace the messages between Kamailio and FS1. This s

Re: [SR-Users] Dispatcher - failure_route to specific node

Hi, looking at the trace, it seems to me that the 407 is not sent by Kamailio but by the FS1 server ("User-Agent: IMX" in the 407, as for the 1st 407 sent by FS2), and that you are probably missing in the trace the messages between Kamailio and FS1. If this is the case you could either: - disable t

[SR-Users] Difference between modules_k and modules_s

In the source code of kamailio, I found some modules code in modules_k folder. Code of same modules is also given modules_s. Examples of such modules are registrar module and usrloc module. If we want to add somethings like log statements etc. Where the change is supposed to be done ? Thanks. Regar

Re: [SR-Users] Print callee domain in logs

What if in INVITE packet, callee's domain is domain1. For example user1@domain1 > calls > user2@domain1 but user2 is registered with domain2. its aor is user2@domain2 in location table. Now I want to somehow route my Invite packet to domain2. How is it possible ?? -- View this m

[SR-Users] routing SIP REGISTER with app_perl

Hello, I'm learning Kamailio. My ultimate goal is to be able to route SIP messages from User Agents to specific FS boxes. I'm starting with routing REGISTER messages. I used app_perl to randomly lookup one of the FS box's IP Address from postgres database, then I stored the IP in avp and then in h

Re: [SR-Users] Dispatcher - failure_route to specific node

On 15/09/14 16:03, Federico Cabiddu wrote: What happens if you call t_relay() after setting $du? Same result. Kamailio sends 407 to caller, which then creates a new INVITE. I suspect my configuration doesn't reflect the intent, so I'm adding it below along with other particulars of the test en

[SR-Users] NO_DTLS_FINGERPRINT when calling from webrtc to webrtc client

Hi. Another topic about WebRTC, websockets with kamailio and rtpengine ;-) My problem is how to distinguish a call to WS UAC and how to SIP UAC in scenarios: 1) WS client1 -> WS kamailio -> SIP kamailio -> SIP UAC 2) WS client1 -> WS kamailio -> SIP kamailio -> WS kamailio -> WS client2 WS k

Re: [SR-Users] How do I translate rtpproxy bridge mode config to mediaproxy-ng/rtpengine?

On Sep 15, 2014, at 1:32 PM, Richard Fuchs wrote: > On 08/25/14 19:25, Alex Villací­s Lasso wrote: >> I have a rtpproxy configuration that spawns several rtpproxy instances, >> using bridge mode. An example is shown below: >> >> /usr/bin/rtpproxy -p /var/run/rtpproxy.pid-7723 -u rtpproxy -s >>

Re: [SR-Users] How do I translate rtpproxy bridge mode config to mediaproxy-ng/rtpengine?

On 08/25/14 19:25, Alex Villací­s Lasso wrote: > I have a rtpproxy configuration that spawns several rtpproxy instances, > using bridge mode. An example is shown below: > > /usr/bin/rtpproxy -p /var/run/rtpproxy.pid-7723 -u rtpproxy -s > udp:127.0.0.1 7723 192.168.2.18/127.0.0.1 -m 1 -M 2

[SR-Users] mimic ds_select_dst with app_perl

I like dispatcher but i need something more custom to my setup so I want to attempt to emulate some basic behavior of dispatcher. I have a simple perl script kamailio calls, it does a database query to get an IP address and port similar to an item listed into the dispatcher list. my question is h

Re: [SR-Users] app_perl persistent database connection

The answer to the question was: yes, after your question, in a dedicated fhrase/paragraph. I expected you wanted to know if the perl script is loaded at startup and then run from memory (precompiled form) at runtime -- sparing another email to ask what you really mean by "constantly run in back

Re: [SR-Users] app_perl persistent database connection

Interesting but could you elaborate on how that will answer my question? On Mon, Sep 15, 2014 at 11:44 AM, Daniel-Constantin Mierla < mico...@gmail.com> wrote: > Hello, > > On 15/09/14 17:03, Vik Killa wrote: > > hello, does kamailio start/run the perl script defined by > modparam("app_perl",

Re: [SR-Users] Kamailio 4.0.4 slow memory leak

Thank you! I will test this on my 4.0.4 setup and report back how it works. Thank you, Tim Heenan Engineer I - VoIP Wholesale | Windstream timothy.hee...@windstream.com | windstreambusiness.com o: 847.348.1338 | f: 847.963.0116

Re: [SR-Users] app_perl persistent database connection

Hello, On 15/09/14 17:03, Vik Killa wrote: hello, does kamailio start/run the perl script defined by modparam("app_perl", "filename", "/usr/local/test/bin/sips.pl ") at startup and constantly run in background? yes. The upcoming 4.2 has the option to reload the script (and r

[SR-Users] app_perl persistent database connection

hello, does kamailio start/run the perl script defined by modparam("app_perl", "filename", "/usr/local/test/bin/sips.pl") at startup and constantly run in background? I want a persistent database connection rather than having to constantly re-connect to database on each transaction. Is this possib

Re: [SR-Users] No audio/video transmission over different networks

Hello, the reply code indicates that the media type is not supported, thus there has been no gatewaying between webrtc and classic rtp. Just replacing rtpproxy with rtpengine is not enough, there are different parameters that have to be provided. Searching on web, I see that Carlos has publi

Re: [SR-Users] No audio/video transmission over different networks

Hi, I have successfully setup rtpproxy-ng kamailio module and mediaproxy-ng package on my ubuntu box. As suggested here: http://kamailio.org/docs/modules/devel/modules/rtpproxy-ng.html I have kept rtpproxy-ng's configuration same as the rtpproxy module, but still not able to connect the webrtc ca

Re: [SR-Users] howto synchronize ruid across multiple proxy instances

Hello, On 12/09/14 15:42, Andrew Pogrebennyk wrote: Hi, let's say I'm running two proxies/registrars that need to access Shared location DB in db_mode=1 (all changes to usrloc are immediately reflected in database too). I have observed that if the UAC re-registers before the previous registratio

Re: [SR-Users] Foward REGISTER to all dispatcher members

Hello, On 15/09/14 08:33, Kees Meijs - SIGNET B.V. wrote: Dear list members, Currently I'm using the dispatcher module to distribute traffic towards media servers (being Asterisk, in my case). I'd like to forward REGISTER requests to all (active) dispatcher members, e.g. using a short and si

Re: [SR-Users] Extract domain of a user from location list table

You should use 4.1 instead of 3.1, because 3.1 is really old (and no longer supported) version. Location table keeps attributes per username@domain. To get the attributes, you need the username@domain. The typical use case is when a SIP requests comes to the proxy targeting username@domain, yo

Re: [SR-Users] Print callee domain in logs

Hello, callee is in the sip signaling, respectively the r-uri. its domain is $rd, so you can print that with xlog(). Cheers, Daniel On 13/09/14 10:33, aawaise wrote: Hello, I have multidomain scenario with parameters (both domains (servers) using common database): modparam("usrloc", "use_do

Re: [SR-Users] No audio/video transmission over different networks

Hi, It appears that my last two messages have gone in moderation. Anyways, Can you please tell me, how can i setup rtpengine on Ubuntu machine? After installation - What configurations will i have to change? I have lurked the internet a lot but did not find any tutorial on this. Would appreciate