[SR-Users] Kamailio with Jitsi - Presence trouble

Hello all!I tried sending this earlier too but since I was not subscribed to the mailing list, it bounced back.I am trying to setup a secure videoconferencing setup formy small office. After a lot of convincing, my bosses have allowed meto create a setup and have given me a live IP. I have used

Re: [SR-Users] Kamailio -PSTN Gateway

Morning Daniel: Thank you very much for your response! As for the routing explanation, I am afraid, I couldn't narrate the issue. I rephrase it, here below: 1: By default Kamailio listens on all interfaces (implying that it has knowledge of all interfaces and corresponding subnets, please corr

Re: [SR-Users] kamailio call limit

Tristan thanks for advice. I will try to shutdown pike module ant thy again. 2014-09-29 18:30 GMT+04:00 Tristan Mahé : > Seems like pike is activated ( antiflood ), check your config ! > > My 2 cents... > > Le 29/09/2014 09:55, Yuriy Gorlichenko a écrit : > > Hi. I tested with sipp my system. Wit

[SR-Users] Call Group versus MAX_BRANCHES limit

Hi folks, I have a question related with the limited number of branches being 12 in config.h #define MAX_BRANCHES12 /*!< maximum number of branches per transaction */ I am trying to implement a Call Group that consists in trying each member of the group (that can result in a para

Re: [SR-Users] R: Re: R: Re: RTPPROXY & BRANCH

On 09/29/14 14:51, Marino Mileti wrote: > Without rtpproxy: > > - A offers port a1,a2 (audio video) in INVITE to B,C (in case of no natted > client so no needs of rtpproxy) > - B offers port b1,b2 (183) > - C offers port c1,c2 (182). > - A starts to send audio/video RTP to B on port b1,b2 > - A s

Re: [SR-Users] R: Re: R: Re: RTPPROXY & BRANCH

On 09/29/14 14:23, Marino Mileti wrote: > The problem isn't on 183s but on the multiple INVITE that Kamailio sends to > clients behind rtpengine. Rtpengine open new ports for answer but on INVITE > the rtpengine ports are the same...This happens because for all these > clients the callid is still t

[SR-Users] R: Re: R: Re: RTPPROXY & BRANCH

Without rtpproxy: - A offers port a1,a2 (audio video) in INVITE to B,C (in case of no natted client so no needs of rtpproxy) - B offers port b1,b2 (183) - C offers port c1,c2 (182). - A starts to send audio/video RTP to B on port b1,b2 - A starts to send audio/video RTP to C on port c1,c2 With r

Re: [SR-Users] rtpproxy & extra_id_pv

On 09/29/14 14:29, Marino Mileti wrote: > Wow! Do you have an example of how to do that? How I have to modify my > kamailio.conf in order to instructs rtpproxy to user from-tag & to-tag in > this way? You don't have to do anything, tags are already included in all the messages. cheers __

[SR-Users] rtpproxy & extra_id_pv

Wow! Do you have an example of how to do that? How I have to modify my kamailio.conf in order to instructs rtpproxy to user from-tag & to-tag in this way? -Messaggio originale- Da: sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] Per conto di Richard Fuc

[SR-Users] R: Re: R: Re: RTPPROXY & BRANCH

The problem isn't on 183s but on the multiple INVITE that Kamailio sends to clients behind rtpengine. Rtpengine open new ports for answer but on INVITE the rtpengine ports are the same...This happens because for all these clients the callid is still the same..so for rtpengine there's no difference.

Re: [SR-Users] rtpproxy & extra_id_pv

On 09/29/14 14:08, Marino Mileti wrote: > But with from-tag and To-tag it's possible to instruct rtpengine to generate > new couple of ports for each branch of a call? In the source code of > rtpengine it seems that it check only the callid parameter Yes it will. The call-id is only a vague umbrel

[SR-Users] rtpproxy & extra_id_pv

But with from-tag and To-tag it's possible to instruct rtpengine to generate new couple of ports for each branch of a call? In the source code of rtpengine it seems that it check only the callid parameter > Hi guys, > I've seen that setting the parameter extra_id_pv, every branch > should be a d

Re: [SR-Users] R: Re: R: Re: RTPPROXY & BRANCH

On 09/29/14 13:29, Frank Carmickle wrote: > > On Sep 29, 2014, at 1:24 PM, Richard Fuchs wrote: > >> On 09/29/14 13:19, Frank Carmickle wrote: >>> >>> On Sep 29, 2014, at 1:14 PM, Richard Fuchs wrote: This may work with rtpengine, as it will open new ports for answers come from d

Re: [SR-Users] R: Re: R: Re: RTPPROXY & BRANCH

On Sep 29, 2014, at 1:24 PM, Richard Fuchs wrote: > On 09/29/14 13:19, Frank Carmickle wrote: >> >> On Sep 29, 2014, at 1:14 PM, Richard Fuchs wrote: >>> >>> This may work with rtpengine, as it will open new ports for answers come >>> from different endpoints. But the final two-way associatio

Re: [SR-Users] R: Re: R: Re: RTPPROXY & BRANCH

On 09/29/14 13:19, Frank Carmickle wrote: > > On Sep 29, 2014, at 1:14 PM, Richard Fuchs wrote: >> >> This may work with rtpengine, as it will open new ports for answers come >> from different endpoints. But the final two-way association for the >> actual call may still end up broken, as it has n

Re: [SR-Users] rtpproxy & extra_id_pv

On 09/26/14 16:57, Marino Mileti wrote: > Hello, > >> On Friday 26 September 2014 16:44:44 Marino Mileti wrote: >>> Hi guys, >>> I've seen that setting the parameter extra_id_pv, every branch should >>> be a different callid.. >>> How can i set this parameter? I've tried with : >>> modparam("rtpp

Re: [SR-Users] R: Re: R: Re: RTPPROXY & BRANCH

On Sep 29, 2014, at 1:14 PM, Richard Fuchs wrote: > > This may work with rtpengine, as it will open new ports for answers come > from different endpoints. But the final two-way association for the > actual call may still end up broken, as it has no way of knowing which > client ends up receiving

Re: [SR-Users] rtpengine with rejected re-invites to new RTP ports

On 09/25/14 12:05, Jeff Pyle wrote: > Hello, > > Given the following scenario with Kamailio and rtpengine in the middle: > > - call establishes with G.711 RTP > - b-leg re-invites to T.38, indicating a different port number then he > is using for G.711 > - a-leg refuses the re-invite with a 48

Re: [SR-Users] R: Re: R: Re: RTPPROXY & BRANCH

On 09/25/14 10:41, Marino Mileti wrote: > > No no. The video will be sent by the caller user to all the callees. > > I'l try to explain better. My scenario is: > > - A make a call to a group... B & C are group member...so Kamailio is > able to call them in parallel using alias.. > > - B & C r

Re: [SR-Users] R: Re: RTPPROXY & BRANCH

On 09/29/14 13:03, Richard Fuchs wrote: > On 09/25/14 10:22, Frank Carmickle wrote: >> >> On Sep 25, 2014, at 10:09 AM, Marino Mileti > > wrote: >> >>> Because I've more than 1 client behind NAT (1,2,3 mobile phones) and I >>> would like to reach all of them in paral

Re: [SR-Users] R: Re: RTPPROXY & BRANCH

On 09/25/14 10:22, Frank Carmickle wrote: > > On Sep 25, 2014, at 10:09 AM, Marino Mileti > wrote: > >> Because I've more than 1 client behind NAT (1,2,3 mobile phones) and I would >> like to reach all of them in parallel mode. I can't use for all of them same >>

Re: [SR-Users] rtpproxy_offer and fix_nated_sdp in one route

On 09/24/14 09:16, Sebastian Damm wrote: > Hi, > > I switched from rtpproxy module to the rtpproxy-ng module lately, and > noticed a strange behavior. In my branch route to the device, I have two > statements: > > fix_nated_sdp("1"); > rtpproxy_offer(); > > The first command appends a line with

Re: [SR-Users] kamailio call limit

Seems like pike is activated ( antiflood ), check your config ! My 2 cents... Le 29/09/2014 09:55, Yuriy Gorlichenko a écrit : > Hi. I tested with sipp my system. With asterisk sipp gets more than 500 > calls, with kamailio 68 at maximum with same settings. > > What I did: > > run sipp though k

[SR-Users] Kamailio with Jitsi - Presence trouble

Hello all!I am trying to setup a secure videoconferencing setup for my small office. After a lot of convincing, my bosses have allowed me to create a setup and have given me a live IP. I have used it on a Ubuntu 12.04 setup and want to setup a SIP server for very few users (less than 10, at most

[SR-Users] event_route[core:receive-parse-error] is not executed 4.1.6

Hello, I try to use an event_route[core:receive-parse-error] { xlog("L_WARN", "Event-parse-error: $rm from $avp(inc_carrier)/n$mb/n"); } corelog=1 debug=0 kamailio 4.1.6 from source Wrong header "From" is simulated by SIPP. In log I recei

[SR-Users] event_route[core:receive-parse-error] is not executed 4.1.6

Hello, I try to use an event_route[core:receive-parse-error] { xlog("L_WARN", "Parse-error: $rm from $avp(inc_carrier)/n$mb/n"); } ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users ma

Re: [SR-Users] Does kamailio reloads the modules per sip message ?

On 29 Sep 2014, at 10:56, Rahul MathuR wrote: > Hello, > > We have observed a strange behavior in corex module that it gets loaded at > every sip packet which arrives to kamailio. > We put a static variable and saw that it gets re-initialized to 0 everytime > any sip packet comes to it. >

[SR-Users] Does kamailio reloads the modules per sip message ?

Hello, We have observed a strange behavior in corex module that it gets loaded at every sip packet which arrives to kamailio. We put a static variable and saw that it gets re-initialized to 0 everytime any sip packet comes to it. Could you please tell me how to stop it and load it just once. Th

Re: [SR-Users] kamailio call limit

Hi. I tested with sipp my system. With asterisk sipp gets more than 500 calls, with kamailio 68 at maximum with same settings. What I did: run sipp though kamailio and look for active channels at asterisk. It give me result with 68 calls at maximum (with limit 500 calls and with generating with 1

Re: [SR-Users] Kamailio -PSTN Gateway

Hello, On 24/09/14 08:44, Zaka Ul Isam wrote: Hello Folks: Please help with above, I have browsed and tried various suggestions on this list without much luck! I think problem can be reduced to three questions ;) 1: Apart from DEFINE WITH PSTN directive, do I need certain modules to be comp

Re: [SR-Users] DBTEXT Engine & ALIAS_DB

Hello, be aware that db_text caches the records in memory, so if you change data inside the file at runtime, is not visible immediately. You can start kamailio with debug=3 in config and look at the logs to see if you get any hint on what happens there. As an alternative, db_sqlite is also

Re: [SR-Users] Unconditional Call forwarding

Hello, On 25/09/14 10:49, Rajesh Sharma wrote: Hi . I have installed Kamailio 4.1.6 and basic registration and proxy server functionalities are working fine. Now I want to simulate a Unconditional call forwarding scenario with 181-Call is being forwarded is reported to originator from serve

Re: [SR-Users] tm.t_uac_wait documentation?

On 25/09/14 20:43, Juha Heinanen wrote: Daniel-Constantin Mierla writes: The rpc commands supposed to have documentation in the code, so you can do: kamcmd help tm.t_uac_wait i did figure that out, but help text is not comprehensive. for example, it appears that host !! of contact uri in

Re: [SR-Users] kamailio call limit

Hello, how do you check the active calls number? Cheers, Daniel On 26/09/14 03:42, Yuriy Gorlichenko wrote: Hello. I try to test with SIPp my stak of kamailio->asterisk. I run SIPp with 200 calls/sec and see only 68 at maximum active calls at server. When I set 500 calls/sec with limit 1000 I

Re: [SR-Users] tm.t_uac_wait does not use advertise address

For the records, the issue has been fixed. Cheers, Daniel On 26/09/14 08:52, Juha Heinanen wrote: based on more tests, it appears that tm.t_uac_wait does not use advertise address (if given on listen line) when it substitutes !! (SUBST_CHARs) in request headers. i'll open a bug report on it.