Re: [SR-Users] Issue using Kamailio as a proxy in front of Asterisk

2012-06-06 Thread Stoyan Mihaylov
We use Jitsi as SIP client, and openxcap along with camailio to handle presence. Then jitsi know if account is online or offline. Our Asterisk dont know nothing about accounts (it accepts all calls from kamailio). There I run AGI scripts, which can check kamailio tables - and I can know if account

Re: [SR-Users] Issue using Kamailio as a proxy in front of Asterisk

2012-06-06 Thread Dominik Mauritz
I have already tried that. I defined SIP-Accounts in Asterisk with host= (instead of host=dynamic). This solves the described problem but it also has side effects: - You don't have the correct presence status on your phone (e. g. xlite) indicating wether the account is online or offline - Aste

Re: [SR-Users] kamailio as SBC

2012-06-06 Thread Alexey Mechanoshin
Simple example integrate: Kamailio 3.1.x and FreeSWITCH 1.0.6+ for Media Services and SBC 2012/6/6 Olle E. Johansson > > 6 jun 2012 kl. 15:16 skrev Daniel-Constantin Mierla: > > > Hello, > > > > if you look to do topology hi

Re: [SR-Users] : Capturing a users call (SIP+RTP+T.38)

2012-06-06 Thread Anton Kvashenkin
You can try homer sip capture server, just google it. 06.06.2012 15:08 пользователь "Klaus Darilion" написал: > Hi all! > > Do you know any comfortable tools to filter out a certain users call? e.g. > searching in SIP packets for the user pattern, get the media port out of > SDP and capture also

Re: [SR-Users] Issue using Kamailio as a proxy in front of Asterisk

2012-06-06 Thread Stoyan Mihaylov
We use also Kamailio in front of Asterisk - but I forward only calls to Asterisk - register/unregister stay in Kamailio. Asterisk dont know which device is registered, and which is not. On Wed, Jun 6, 2012 at 8:20 PM, Dominik Mauritz wrote: > Hi All, > > some days ago I installed Kamailio as a f

[SR-Users] Issue using Kamailio as a proxy in front of Asterisk

2012-06-06 Thread Dominik Mauritz
Hi All, some days ago I installed Kamailio as a front end for Asterisk following this tutorial: http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb I added Multi Domain support and adjusted some other things to fit my environment. Almost everything is working perfectly

Re: [SR-Users] Making dialog module work for me !!

2012-06-06 Thread SamyGo
Hi Sir, I've used the funcation as you informed. *$dlg_var(incall) = $fU; * It do insert my desired value into the dialog_var table but it don't gets deleted on call hangup ! I'm executing this code in my ON_REPLY route: onreply_route[REPLY_ONE] { xdbg("incoming reply\n"); if(is_method("B

Re: [SR-Users] Dispatcher alg:7 not working accordingly

2012-06-06 Thread SamyGo
Sorry for late reply: this wasn't very helpful. I think Hashing algo code needs to get bit smarter. If there is any possibility can you please let me know. ! On Mon, Jun 4, 2012 at 3:33 PM, Daniel-Constantin Mierla wrote: > To make the life easier to spot the hash code in such case, I just > com

Re: [SR-Users] Capturing a users call (SIP+RTP+T.38)

2012-06-06 Thread SamyGo
Hi, Once I tried and pretty much succeeded in getting wireshark to capture packets on the go from a server. I could see in wireshark capture interface "Remote://" - But AFAIR that needs some libraries to be installed on remote server and 2-years ago it wasn't very stable/reliable as well. Regards,

Re: [SR-Users] Capturing a users call (SIP+RTP+T.38)

2012-06-06 Thread Daniel-Constantin Mierla
Hello, if you look to a command line tool, there is tshark that comes with wireshark and takes a lot of parameters. I just remembered that once I saw it can do Lua scripting and there is an example of dumping sip calls in different files, see: http://wiki.wireshark.org/Lua/Examples Cheers,

Re: [SR-Users] Capturing a users call (SIP+RTP+T.38)

2012-06-06 Thread Klaus Darilion
Wireshark is nice for "later analysis". But IMO it is not suitable to capture. I will have a look at the suggested tools. thanks Klaus On 06.06.2012 15:17, Daniel-Constantin Mierla wrote: Hello, afaik, wireshark can be used also to store the signaling and media for sip calls -- it can reply

Re: [SR-Users] kamailio as SBC

2012-06-06 Thread Olle E. Johansson
6 jun 2012 kl. 15:16 skrev Daniel-Constantin Mierla: > Hello, > > if you look to do topology hiding, look at topoh module. If you need a full > sbc, it might be better to use dedicated software for it -- e.g., SEMS has an > explicit SBC module, other software such as asterisk or freeswitch can

Re: [SR-Users] Capturing a users call (SIP+RTP+T.38)

2012-06-06 Thread Daniel-Constantin Mierla
Hello, afaik, wireshark can be used also to store the signaling and media for sip calls -- it can reply the audio, if it is such call. Cheers, Daniel On 6/6/12 3:09 PM, Sebastian Ferguson wrote: Hi: I've used this one http://www.voipmonitor.org/ I don't think it can record T.38 but it's a

Re: [SR-Users] kamailio as SBC

2012-06-06 Thread Daniel-Constantin Mierla
Hello, if you look to do topology hiding, look at topoh module. If you need a full sbc, it might be better to use dedicated software for it -- e.g., SEMS has an explicit SBC module, other software such as asterisk or freeswitch can do it -- with them you will get specific features such as tra

Re: [SR-Users] kamailio as SBC

2012-06-06 Thread SamyGo
Hi, Can you be more specific on what you are trying to achieve in terms of servers topology !? See TOPOLOGY-HIDINGmodule as well. Regards. Sammy G. On Wed, Jun 6, 2012 at 5:45 PM, Mino Haluz wrote: > Hi, > > I know that kamailio is

Re: [SR-Users] Capturing a users call (SIP+RTP+T.38)

2012-06-06 Thread Sebastian Ferguson
Hi: I've used this one http://www.voipmonitor.org/ I don't think it can record T.38 but it's a great tool. You can save a pcap file for every call, you can have the MOS score of each call and you can save RTP as well as a wav file. You can also try this one http://oreka.sourceforge.net/ I think t

[SR-Users] kamailio as SBC

2012-06-06 Thread Mino Haluz
Hi, I know that kamailio is SIP proxy, but is there any way how to implement kamailio as SBC like OpenSIPS with B2BUA module ? I tried OpenSIPS with this module, but it does not work with mediaproxy module. Mino ___ SIP Express Router (SER) and Kamaili

[SR-Users] Capturing a users call (SIP+RTP+T.38)

2012-06-06 Thread Klaus Darilion
Hi all! Do you know any comfortable tools to filter out a certain users call? e.g. searching in SIP packets for the user pattern, get the media port out of SDP and capture also RTP+T.38 I found pcapsipdump but have not tried it yet. Any suggestions or do I have to write my own tool? thanks

Re: [SR-Users] Kamailio 3.2 + RTP proxy - no errors - no media

2012-06-06 Thread Klaus Darilion
For "briding" you have to isntruct rtpproxy when to use which interface. Thus, in kamailio.cfg, when processing a message, find out if the message goes from v4 to v6, or vice versa , or v4-v4 or v6-v6. Then instruct rtpproxy about the direction using the 'e' and 'i' flag. i stands for interna