Re: [SR-Users] Core Dump during shutdown on SPARC64

Hi Daniel, I haven’t yet but I will try to test shortly. Here is the value of the block variable: (gdb) p /s block $1 = (unsigned char *) 0x76e5ded4 “" Thanks, Spencer On Jan 13, 2016, at 11:07 PM, Daniel-Constantin Mierla > wrote: Hello,

Re: [SR-Users] Fwd: Kamailio and NAT

Yes it is possible, but is there an easy way to workaround the issue using Kamailio. Because I have the port because vendor is sending that info in Trying: 2016/01/13 20:10:15.842055 VENDOR-IP:5060 -> PRIVATE-IP-KAMAILIO:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP

Re: [SR-Users] Fwd: Kamailio and NAT

Not really up to date with all Asterisk features -- do you know if you can append a custom header to a SIP response that is going to be generated by Asterisk? Eventually the reply for an OPTIONS request. Cheers, Daniel On 14/01/16 17:19, Nelson Migliaro wrote: > Yes, I manage all devices, even

Re: [SR-Users] dispatcher state x and the return of the probing mode

> > So if I set ds_probing_mode to 0 ("only the gateways with state PROBING > are tested"), do I have to set state PROBING manually at startup for all > gateways? Or are gateways set to probing by default? in ds_probing_mode=0 not any PROBING by default. if you just want to test your node after

Re: [SR-Users] dispatcher state x and the return of the probing mode

Hi Dmitri, Am 13.01.2016 um 16:08 schrieb Dmitri Savolainen: > 1. Have you set modparam("dispatcher", "ds_probing_mode", 1)? If so, it > will be moved to PROBING mode after ds_ping_interval Yes, I have it set to 1. So if I set ds_probing_mode to 0 ("only the gateways with state PROBING are

[SR-Users] Different rtpproxy for different media type

Hi everybody, Is there a way to use different rtpproxies for different media types? I want to use one rtpproxy set for audio, and use another one for video. Is this possible in kamailio? Regards, Koray ___ SIP Express Router (SER) and Kamailio

[SR-Users] Fwd: Kamailio and NAT

Thank you Daniel for your answer, As you mention, there is a symmetric nat and router does not allow a static NAT. By sniffing traffic I can see the port is using new but in case it change, how can automate the process of advertising the correct port? Cheers! -- Forwarded message

Re: [SR-Users] Different rtpproxy for different media type

Hi, Yes thats possible as I think you can do a search in SDP body for "video" if found then select the rtpproxy instance else select other one. Regards, Sammy On Jan 14, 2016 06:39, "Koray Vatansever" wrote: > Hi everybody, > > Is there a way to use different

Re: [SR-Users] Fwd: Kamailio and NAT

Ahh, I thought Asterisk is in the public internet, but actually you connect to a provider (vendor), which seems to run Kamailio as well. Using information from 100 trying is too late, as the INVITE was already sent... so one more question before trying to propose a solution. Do you have to

Re: [SR-Users] Different rtpproxy for different media type

I think I couldn't explain myself clearly. I want to see the following media lines in resulting SDP: m=audio 10076 RTP/AVP 97 98 c=IN IP4 rtpproxy1.example.com a=rtpmap:97 speex/8000 a=fmtp:97 vbr=on a=rtpmap:98 telephone-event/8000 m=video 20007 RTP/AVP 96 c=IN IP4 rtpproxy2.example.com

Re: [SR-Users] Sipcapture on SPARC

Hello, this looks like an issue on the SQL query syntax. What version of mysql server do you run there? Cheers, Daniel On 14/01/16 22:38, Spencer Thomason wrote: > Hello, > I’m attempting to deploy a homer sip capture server on SPARC and I’m seeing > some strange date related errors: > ERROR:

[SR-Users] Rewriting header of BYE message to carrier

Hi, We're running a system with Kamailio running in front of Asterisk just handling registrations and forwarding everything else to Asterisk. But we're having an issue during hangup on incoming calls. If the initiator hangs up, the call completes successfully. But if one of our phones hangs up,

[SR-Users] Sipcapture on SPARC

Hello, I’m attempting to deploy a homer sip capture server on SPARC and I’m seeing some strange date related errors: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Incorrect datetime value: '2052-06-12 16:21:06' for column 'date' at row 1 (1292) Is there some

[SR-Users] IMS terminating dialog does not re-run through all iFCs after calling t_newtran()

Hi, My CSCF environment has an application server acting as a B2BUA. It's originating a new session, and creating a new terminating session back to the S-CSCF (two separate sessions per-call). When I perform a re-INVITE with a diversion to transfer the call to another URI it is all good until

Re: [SR-Users] Different rtpproxy for different media type

Do you want to change/update the SDP once the call is established ? shouldn't it just work like this: if(has_totag() && is_method("INVITE")) { if(search_body("video") { set_rtpproxy_set("1"); unforce_rtpproxy(); set_rtpproxy_set("2"); offer_rtpproxy("$avp(myflags)");

Re: [SR-Users] Fwd: Kamailio and NAT

Do you control the Asterisk? If yes, depending on Asterisk capabilities of building replies, you may be able to do some automation to detect the external port. Cheers, Daniel On Thu, Jan 14, 2016 at 3:47 PM, Nelson Migliaro wrote: > There is not a public Kamailio, only

Re: [SR-Users] Fwd: Kamailio and NAT

Is the kamailio behind nat communicating with another kamailio on a public IP? Cheers, DAniel On Thu, Jan 14, 2016 at 1:33 PM, Nelson Migliaro wrote: > Thank you Daniel for your answer, > > As you mention, there is a symmetric nat and router does not allow a > static

Re: [SR-Users] Different rtpproxy for different media type

Hi Sammy, I'm not sure it will work. Assume the following scenario: Kamailio receives INVITE with audio only SDP and selects rtpproxy-1. After a while video is enabled with REINVITE. Now SDP has video and kamailio selects rtpproxy-2 according to your solution. In this case, most probably video

[SR-Users] IPTel URI forwarding to PBXes not working

I am stumped as my DID provider can't route via SIP URI to PBXes but they can to IPTel. I can call my own PBX, and from my own PBX to the PBXes URI without issue. When I route the DID via URI to IPTel and set voicebox it works and routes to the voicebox. When I add a forward for the PBXes URI