Re: [OpenSIPS-Users] An error exists in recording the number of ongoing sessions at RTP Proxies

2013-02-27 Thread microx
Hi Bogdan-Andrei, I tried to use the get_static_lock and release_static_lock to set up the critical session. However, I still get an incorrect result (please refer to the attached log). It is likely that I misuse the locking mechanism and thus I also provide the corresponding code. Please

[OpenSIPS-Users] CDRTool Quota problem

2013-02-27 Thread HaGGarD
Hi @all, I'm trying to set up CDRTool together with kamailio, mediaproxy and freeradius. I'm using debian 6.0.6 with latest CDRTool (debian package from ag projects) and set up everything exactly how it's explained in the install docs. I get the calldata from freeradius rated correctly and showed

[OpenSIPS-Users] Routing requests when using GRUU

2013-02-27 Thread Saúl Ibarra Corretgé
Hi, Today I came across this problem again, and while I thought I could fix it by using match_dialog and fix_route_dialog that didn't work out, so I think we need to get something fixed so that GRUU routing works. Lets assume this scenario: - Bob has b...@biloxi.com URI and is registered on

Re: [OpenSIPS-Users] NAT

2013-02-27 Thread Adrian Georgescu
Using a media relay is the solution for your problem. You are asking for a solution to not use the solution which makes no sense. Adrian On Feb 25, 2013, at 7:24 PM, Roberto Spadim wrote: humm i got the same problem but didn't found a solution my solution was connect internet (public ip)

Re: [OpenSIPS-Users] Opensips and Callcontroll KeyError

2013-02-27 Thread Davide Dal Frà
Hi Saul, As we have discussed on IRC few day's ago, i've edited my config adding the avp_token of call-control module. After two day's of testing, now the detection of call with same callid is ok and the error no longer appear in log. So all work fine :) Thanks Davide On 02/25/2013

Re: [OpenSIPS-Users] Opensips and Callcontroll KeyError

2013-02-27 Thread Saúl Ibarra Corretgé
On Feb 27, 2013, at 2:48 PM, Davide Dal Frà wrote: Hi Saul, As we have discussed on IRC few day's ago, i've edited my config adding the avp_token of call-control module. After two day's of testing, now the detection of call with same callid is ok and the error no longer appear in

[OpenSIPS-Users] [RELEASE] 1.9 Major Release becomes stable GA today!

2013-02-27 Thread Bogdan-Andrei Iancu
After one month of successful beta testing, the 1.9.0 major released turned into a fully stable GA releasesuitable for production usage. Feel free to download and torture it - it will proudly do the job for you :) ! Once again, many thanks to all people who got involved in the testing,

Re: [OpenSIPS-Users] [RELEASE] 1.9 Major Release becomes stable GA today!

2013-02-27 Thread Seth Schultz
Hello, I just did a clean checkout of the 1.9.0 major release, but when I build, I am seeing these messages repeated for every file: command-line:0:0: warning: CFG_DIR redefined [enabled by default] command-line:0:0: note: this is the location of the previous definition

Re: [OpenSIPS-Users] [RELEASE] 1.9 Major Release becomes stable GA today!

2013-02-27 Thread Saúl Ibarra Corretgé
On Feb 27, 2013, at 7:13 PM, Bogdan-Andrei Iancu wrote: After one month of successful beta testing, the 1.9.0 major released turned into a fully stable GA releasesuitable for production usage. Feel free to download and torture it - it will proudly do the job for you :) ! Once again,

Re: [OpenSIPS-Users] [RELEASE] 1.9 Major Release becomes stable GA today!

2013-02-27 Thread Brett Nemeroff
Great job guys! This is one exciting build -Brett On Wed, Feb 27, 2013 at 12:13 PM, Bogdan-Andrei Iancu bog...@opensips.orgwrote: After one month of successful beta testing, the 1.9.0 major released turned into a fully stable GA releasesuitable for production usage. Feel free to

Re: [OpenSIPS-Users] [RELEASE] 1.9 Major Release becomes stable GA today!

2013-02-27 Thread Rudy
Fantastic! Another amazing release full of exciting features to explore. Thanks to everyone for their contributions. Best regards, --Rudy Dynamic Packet Toll-Free: 888.929.VOIP ( 8647 ) On Wed, Feb 27, 2013 at 1:48 PM, Brett Nemeroff br...@nemeroff.com wrote: Great job guys! This is one

[OpenSIPS-Users] Issue with From domain coming from Asterisk

2013-02-27 Thread Duane Larson
I wanted to see if I could get this answered on the OpenSIPS mailing list even though this kind of has to do with how Asterisk works. I am hoping someone has run into this and figured a way to resolve the issue. I have OpenSIPS set up to be a proxy for a cluster of Asterisk servers. When a call