Re: [SR-Users] SOLVED: Re: Proper way to make Asterisk recognize SIP trunk of incoming INVITE when IP is not available

2014-05-02 Thread Daniel-Constantin Mierla
On 02/05/14 19:39, Alex Villací­s Lasso wrote: El 02/05/14 10:40, Alex Villací­s Lasso escribió: El 24/04/14 19:09, Alex Villací­s Lasso escribió: I am currently preparing a kamailio-asterisk combination. The asterisk installation uses realtime for SIP. The kamailio configuration (attached) w

[SR-Users] SOLVED: Re: Proper way to make Asterisk recognize SIP trunk of incoming INVITE when IP is not available

2014-05-02 Thread Alex Villací­s Lasso
El 02/05/14 10:40, Alex Villací­s Lasso escribió: El 24/04/14 19:09, Alex Villací­s Lasso escribió: I am currently preparing a kamailio-asterisk combination. The asterisk installation uses realtime for SIP. The kamailio configuration (attached) was based on the reference at http://kb.asipto.com/a

Re: [SR-Users] Kamailio v4.0.5 crash - transformations related ?

2014-05-02 Thread Daniel-Constantin Mierla
Hello, do you know if it was an ACK for a negative response? I looked a bit over the code and the issue could be with the lifetime of the dlg variable. Cheers, Daniel On 30/04/14 13:39, Dragos Oancea wrote: Hi We experimented a crash with kamailio 4.0.5 , it looks like a memory corruption

Re: [SR-Users] Frequent Crash in Kamailio 4.1.3 in the file data_lump.c see gdb backtrace in mail

2014-05-02 Thread Daniel-Constantin Mierla
Hello, can you give the output from gdb of next commands: bt full p *l Cheers, Daniel On 02/05/14 18:22, varun pratapsingh wrote: HI All/ Daniel, According to the recommendation I have migrated my Kamailio 4.1.2 to the Kamailio 4.1.3. But very frequent crashes are coming in Kamailio 4.1.3

[SR-Users] Frequent Crash in Kamailio 4.1.3 in the file data_lump.c see gdb backtrace in mail

2014-05-02 Thread varun pratapsingh
HI All/ Daniel, According to the recommendation I have migrated my Kamailio 4.1.2 to the Kamailio 4.1.3. But very frequent crashes are coming in Kamailio 4.1.3 which was not coming on the 4.1.2. The gdb backtrace of the crash is below: I have seen many crashes in couple of days but the "data_lump

Re: [SR-Users] Proper way to make Asterisk recognize SIP trunk of incoming INVITE when IP is not available

2014-05-02 Thread Alex Villací­s Lasso
El 24/04/14 19:09, Alex Villací­s Lasso escribió: I am currently preparing a kamailio-asterisk combination. The asterisk installation uses realtime for SIP. The kamailio configuration (attached) was based on the reference at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-ast

[SR-Users] Kamalio

2014-05-02 Thread Akash Ashok
Hi, I have just started using Kamalio and We are using it for a VOIP app that I am trying to build. Here are the things that i have done 1. I have setup kamalio server 2. I could get it running using linphone app 3. I could add users using kamctl command Since I am building a smartphone app I wan

Re: [SR-Users] What does this script do?.

2014-05-02 Thread VOIP Tests
Thank you Daniel. On Fri, May 2, 2014 at 4:29 AM, Daniel-Constantin Mierla wrote: > Loading all integer-named attributes from domain_preferences table (which > must have same structure as usr_preference table), using only domain for > selecting the records. > > Cheers, > Daniel > > > On 30/04/1

Re: [SR-Users] Limit the number of requests per second based on the user

2014-05-02 Thread Sebastian Damm
Hi Daniel, my last mail was probably not really clear. I don't want a pipe for all users together, but one for each user. What I wanted to say is, I don't need different limits for some users, they all share the same limits. I have implemented the htable version now, and it's working as expected (

[SR-Users] Dispatcher ping source ip:port?

2014-05-02 Thread Daniel Tryba
Is it possible to manipulate the pings (OPTIONS in my case) to force a source ip:port combo per destination? dispartcher.list: 1 sip:192.168.0.1:5060 2 sip:192.168.1.1:5070 3 sip:192.168.2.1:5050 The kamailio (4.1) is listening on 192.168.3.1 5060 and 5070. dispatcher 2 cares about the source

Re: [SR-Users] Voice is breaking when using Kamailio 4.1.3

2014-05-02 Thread varun pratapsingh
Hi Daniel, Yes I agree with you that kamailio is not involved in the rtp handling and forwarding but RTP Proxy does this. So I have mentioned that when my call is established then my RTP passes through the RTPProxy. The issue is that RTP is coming proper at UAC Side so no Voice Problem but may be

Re: [SR-Users] How can I implement Asterisk as Media server?

2014-05-02 Thread Daniel-Constantin Mierla
Hello, if you use one of latest asterisk version as media server, it should have support for webrtc media handling, so just forward the calls to it. For a media gateway, you can use rtpproxy enginge module and application along with kamailio (for stable version 4.1, rtpproxy-ng module). Che

Re: [SR-Users] What does this script do?.

2014-05-02 Thread Daniel-Constantin Mierla
Loading all integer-named attributes from domain_preferences table (which must have same structure as usr_preference table), using only domain for selecting the records. Cheers, Daniel On 30/04/14 23:38, VOIP Tests wrote: Can someone let me know what this script does? avp_db_load("$ru/domain"

Re: [SR-Users] Voice is breaking when using Kamailio 4.1.3

2014-05-02 Thread Daniel-Constantin Mierla
Hello, kamailio is not involved in the rtp handling/forwarding. So either is a bad connection somewhere in the path or an issue in the client sides. Can you try with different phones? Cheers, Daniel On 02/05/14 07:49, varun pratapsingh wrote: HI All/ Daniel, We are using the Kamailio+ RTPPr