Re: [SR-Users] SIP Analysis Testing tools

2011-09-05 Thread Daniel-Constantin Mierla
Hello, On 9/2/11 5:03 PM, Vladimir Broz wrote: Hi, for sake of Vaclav's courage, let me introduce the SIP Analysis Testing tools. This is a set of very powerful utilities that may help you to do SIP call flows analysis. It can do SIP dialogs and transactions matching and analysis in

Re: [SR-Users] [sr-dev] SIP Analysis Testing tools

2011-09-05 Thread Juha Heinanen
Daniel-Constantin Mierla writes: thank you for sharing it with us. Are the diagrams generated as image, or does also kind of html where can browse the content of SIP messages? Maye uploading some screenshots on the wiki of github will make it more appealing to try. i tried it and got a

[SR-Users] Pass all rtp to RTPProxy

2011-09-05 Thread Stas Bakulin
Hello! What is it right way to pass all rtp traffic through RTPProxy? How this can be configured in Kamailio? Stas Bakulin -- og...@kvant-telecom.ru | www.kvant-telecom.ru Tel: +7 473 233 0330 (128)

Re: [SR-Users] Check disconnected users

2011-09-05 Thread Daniel-Constantin Mierla
Hello, On 9/2/11 8:40 PM, Anto wrote: Hello I want to store in a file access and disconnections to the system. To access (registrar), I thought scheduling a perl script and executed with the perl module, the problem I find with disconnections. Since the user can be disconnected for various

Re: [SR-Users] new developer: Sven Knoblich

2011-09-05 Thread marius zbihlei
On 09/05/2011 10:58 AM, Daniel-Constantin Mierla wrote: Hello, On 9/1/11 3:17 PM, Henning Westerholt wrote: Hi all, I'm happy to announce a new developer for the sip-router project: Sven Knoblich. Sven is an experienced C/C++ developer in our offices in Karlsruhe and since about a year part

[SR-Users] 10 Years SER - Follow UP

2011-09-05 Thread Daniel-Constantin Mierla
Hello, first I want to thank to all participants at 10 Years SER event in Berlin last Friday. We spent great time, at least I did! We accepted 50 visiting participants to match the maximum capacity of the conference room, being full booked about 2 weeks before. Just few people didn't show

Re: [SR-Users] [sr-dev] 10 Years SER - Follow UP

2011-09-05 Thread Olle E. Johansson
5 sep 2011 kl. 10:57 skrev Daniel-Constantin Mierla: Many thanks again to our sponsors for free drinks and food at this event: Sipgate (http://www.sipgate.de), Amooma (http://www.amooma.de), NG-Voice (http://www.ng-voice.com), Asipto (http://www.asipto.com), Frafos (http://www.frafos.com)

[SR-Users] Carrier module for kamailio

2011-09-05 Thread Mino Haluz
Hi, I need a module which could allow me to send traffic to various carriers and it has to support some important features. So some basic ones: - possibility to re-route the call in case the original route fails - peak/offpeak conditions (time-based) - route traffic according to prefix I found

[SR-Users] Carrier module for kamailio

2011-09-05 Thread Juha Heinanen
Mino Haluz writes: I found LCR and carrierroute module, but it does not have peak/offpeak feature. Correct me if I am wrong.. run from cron a script that modifies rules based on your needs. -- juha ___ SIP Express Router (SER) and Kamailio

[SR-Users] tls + tcp problems

2011-09-05 Thread MÉSZÁROS Mihály
Hi, I am experiencing two problem: First: I am experiencing that tls read buffer overflow, if i use default value of tcp_rd_buf_size If i doubled the size of read buffer and i use tcp_rd_buf_size=8192 then the problem seems to be solved. Second: I am experiencing that TLS connection is

[SR-Users] tls + tcp problems

2011-09-05 Thread Juha Heinanen
MÉSZÁROS Mihály writes: Second: I am experiencing that TLS connection is dropped by sip proxy very frequently! And i think the tcp connection shouldn't dropped! So my guess is that TLS communication is not restarting the timeout counters! So in every 2 minutes the sip-router is restarting

[SR-Users] Accounting only the 2nd branch of missed serial forked call

2011-09-05 Thread Ozren Lapcevic
Hi, I'm having some problems accounting missed serial forked calls to mysql database. I have following setup. Each user can have up to two contacts: telephone number (routed to asterisk) and SIP URI. Users can specify which contact has higher priority - which one should ring first. There is also

Re: [SR-Users] db_mysql Commands out of sync; you can't run this command now

2011-09-05 Thread MÉSZÁROS Mihály
Hello Daniel, Thank You, and many thanks to Alex too! The fix you made is working fine for me. Cheers, Misi ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org

Re: [SR-Users] question

2011-09-05 Thread Carsten Bock
Hi, you should look at the following things: - dispatcher module (for loadbalancing between destinations) - rtpproxy module if you want to have NAT-relay for clients behind NAT To what GUI do you refer? Siremis? Probably you should look at the webpage: http://siremis.asipto.com/ Kind regards,

Re: [SR-Users] Kamailio + rtpproxy talking to multiple carrier gateways

2011-09-05 Thread Sarat C. Vemuri
Alex, Sorry about this clumsy way of replying, but I am stuck with Outlook 2010 and Exchange. You stated the problem correctly. Carrier GWs (UAS) and internal clients (UAC) cannot reach each other directly (TCP/UDP). They have to go through Kamailio and rtpproxy. I am goint to reply to

Re: [SR-Users] Kamailio + rtpproxy talking to multiple carrier gateways

2011-09-05 Thread Alex Balashov
On 09/05/2011 08:07 PM, Sarat C. Vemuri wrote: How do I remove the public IP entry from the route set before forwarding the reply to Internal UAC? You don't -- at least, not in a protocol-compliant way. You can, of course, do remove_hf(Route); if you want to, but with effects that are