Re: [OpenSIPS-Users] number of opensips children

Hi Brian, the average load (which is very small in your case) is not really relevant. Maybe you have a spike in traffic - like more than 8 requests received in the same time. So, my advice is : 1) check on the network level if you receive bulk messages at a moment 2) maybe the processing you do

Re: [OpenSIPS-Users] What is protocol/port mismatch?

Hi Brian, The backtrace from gdb seams really corrupted, but the pstack output is a bit consistent. What is strange is that dlg_validate_dialog () seams to call fm_realloc() which is not in the script. So, to eliminate any possibility of a mem corruption (the crash was in the mem manager), I s

Re: [OpenSIPS-Users] t_relay() not relaying payload

Hi Thamer, actually I see a 408 TIMEOUT for the second re-INVITE - but I see not outgoing re-INVITE in the second step (only the inbound message) Looking at the 2 re-INVITEs, I would say you have a routing issue as the first re-INVITE has as RURI a public IP (sip:1...@uac.ip.address.here:5060)

Re: [OpenSIPS-Users] priority differences between carrierroute and drouting

Hi Gabriel, The selection of the Gateways (from the list for a prefix) is a bit more complex in Drouting as you have 3 ways of choosing (including random, group based, etc) - see the "sort" param http://www.opensips.org/html/docs/modules/devel/drouting.html#id272114. But indeed, maybe a new mo

Re: [OpenSIPS-Users] Starting OpenXCAP without any logs

Hi, El 03/02/10 15:49, CheeWii escribió: > Thanks a lot ! > Looking forward to your reply kindly~ : ) > Last night I setup a machine with the same package versions as yours and I easily reproduced the issue: OpenXCAP rums OK in foreground, but it fails to start in background. The issue is rela

Re: [OpenSIPS-Users] priorities with opensips

Well, this might be possible using a simple external script. If you put all the calls through the dialog module, you have an external MI command to terminate a dialog (from the proxy). So what you need is a script to query (via MI - dlg_list command) what are the active calls and to terminate (v

Re: [OpenSIPS-Users] TLS call failed

Hi Steven, For the NOKIA N97, could you post the entire log (debug 4) for the INVITE part (covering the receiving of the INVITE also) ? Regards, Bogdan doolin wu wrote: > Hello, > > I'm trying use TLS feature of OpenSIPS-1.5-tls. TLS was > configured and server run successfully. > I tried to

Re: [OpenSIPS-Users] RTP listener

Hi Yannick , check http://lists.opensips.org/pipermail/users/2010-January/010515.html Regards, Bogdan Yannick LE COENT wrote: > > Hello all, > > > > I would to listen and record RTP streams in real-time. > > > > RTP proxy seems to be able to record RTP streams in pcap format. > > > > Is th

Re: [OpenSIPS-Users] Need help Nathelper + rtpproxy

Hi, do not pass both I and E flags !! As you call force_rtp_proxy() twice (once for request and once for reply), you need to pass once the I and next one the E flags, depending which interface you want RTPP to use for the request or reply. Regards, Bogdan Zilla1000 wrote: > I am having probl

Re: [OpenSIPS-Users] opensips 1.6.1 crashes on NOTIFY?

Hi Alex, updating from 1.6 SVN branch, are these crashes still happening ? Regards, Bogdan Alexander wrote: > Any ideas? I'm afraid to touch something and make things worse :) > > 22 декабря 2009 г. 14:23 пользователь Alexander > написал: > > What about this one: >

Re: [OpenSIPS-Users] priorities with opensips

Thank you Bogdan! this is really a useful hint! -- View this message in context: http://n2.nabble.com/priorities-with-opensips-tp4506066p4512524.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists

users@lists.opensips.org

On 28.01.2010 11:07, Indiver wrote: > I forgot to mention that files are not storing by callee or caller number. > Moreover it is taking its own unique caller id. How to over come this in > order to modify the recording file name as callee<->caller and time stamp > format. File names are created u

users@lists.opensips.org

On 29.01.2010 11:29, Bogdan-Andrei Iancu wrote: > I doubt you can change that as RTPproxy is not decoding the RTP stream - > as the name says, the tool is only RTP aware, so cannot interpret the > content. But I guess you can google for some other audio tools to help > mixing the 2 streams. Check

[OpenSIPS-Users] Res: RTP listener

Hi Yannick There is an open source software http://oreka.sourceforge.net/ capable not only to listen, but to record and manage the calls. It requires a separate machine connected to a SPAN/MONITOR port of the network switch. I have tested both OREKA and the RTP Proxy feature, it works, you will

[OpenSIPS-Users] BYE - 404 not here

Hello everyone, i have a problem when a call is hangup by the callee, i think i probably have some general routing logic Problem and i cant find any way to solve it. caller --> asterisk (62.66.66.67) --> opensips(62.66.66.66) (+rtpproxy on the same machine) --> pstngw (213.20.11.11)

Re: [OpenSIPS-Users] BYE - 404 not here

On 04.02.2010 12:56, Max Mühlbronner wrote: > But if the call is established and the callee hangs up, the BYE is not > received by the original calling side so it stays connected. > My opensips knowledge is still very basic, so please excuse if it is > some dumb routing mistake made by me. I suppo

Re: [OpenSIPS-Users] One way audio - media port changed (opensips / mediaproxy)

Hmm, you are right, that wasn't the full syslog for that call. Investigating further I see that I get the following: Jan 11 22:28:07 sbc1 /usr/sbin/opensips[27792]: CRITICAL:dialog:log_next_state_dlg: bogus event 6 in state 2 for dlg 0x7f5e880692a0 [1305:480665684] with clid 'a27f94c89e3e13c41

Re: [OpenSIPS-Users] One way audio - media port changed (opensips / mediaproxy)

Hi Magnus, El 04/02/10 12:54, Magnus Burman escribió: > Hmm, you are right, that wasn't the full syslog for that call. > Investigating further I see that I get the following: > > Jan 11 22:28:07 sbc1 /usr/sbin/opensips[27792]: > CRITICAL:dialog:log_next_state_dlg: bogus event 6 in state 2 for dlg

Re: [OpenSIPS-Users] BYE - 404 not here

File attached, Thanks very much for taking a look. Regards Andrew Pogrebennyk schrieb: On 04.02.2010 12:56, Max Mühlbronner wrote: But if the call is established and the callee hangs up, the BYE is not received by the original calling side so it stays connected. My opensips knowledge is still

Re: [OpenSIPS-Users] Does Opensips Presence Support Content Indirection in PIDF ?

Hi Mani, Can you please tell me in what RFC is CID in PIDF defined? Regards, -- Anca Vamanu www.voice-system.ro mani sivaraman wrote: > Could any one please let me know if opensips support CID (content > indirection) in PIDF PUBLISH and NOTIFY ?. Is there any configuration > to be done to e

Re: [OpenSIPS-Users] BYE - 404 not here

>From my experience, this usually happen either from a configuration file error, or from the terminating UAS who sends the BYE not sending it to the RURI in the contact header from the original INVITE. Can we see the original INVITE as it hits OpenSIPs? -Brett On Thu, Feb 4, 2010 at 4:56 AM, Max

Re: [OpenSIPS-Users] BYE - 404 not here

sorry, of course, here are the invites: *Invite from asterisk --> opensips* U 62.66.66.67:5060 -> 62.66.66.66:5060 INVITE sip:123549...@62.66.66.66 SIP/2.0. Via: SIP/2.0/UDP 62.66.66.67:5060;branch=z9hG4bK16ff74c2;rport. Max-Forwards: 70. From: "49302332434343" ;tag=as1fcd8c32. To: . Contact

Re: [OpenSIPS-Users] Starting OpenXCAP without any logs

Following your suggestions,I really make openxcap running in fork mode successfully~ Thank you ,keep you posted !:) 2010/2/4 Saúl Ibarra Corretgé > Hi, > > El 03/02/10 15:49, CheeWii escribió: > > Thanks a lot ! > > Looking forward to your reply kindly~ : ) > > > > Last night I setup a machine

Re: [OpenSIPS-Users] high-availability - senario

Option 2 works fine for me. Thanks From: users-boun...@lists.opensips.org on behalf of Stanislaw Pitucha Sent: Tue 02/02/2010 3:03 PM To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] high-availability - senario On 02.02.2010 14:17, Julien Chavanton

Re: [OpenSIPS-Users] B2BUA help

Hello Anca, An ven., janv 29, 2010, Anca Vamanu schrieb: >opensipsl...@encambio.com wrote: >> An lun., janv 04, 2010, Anca Vamanu schrieb: >>> I see that you say that the prepaid scenario does not work for you. >>> What version are you testing with? >>> >> Solaris 11 x86 (nv-b91) >> OpenSIPS

Re: [OpenSIPS-Users] What is protocol/port mismatch?

Hello Bogdan, An jeu., févr 04, 2010, Bogdan-Andrei Iancu schrieb: >The backtrace from gdb seams really corrupted, but the pstack >output is a bit consistent. > Sorry about that. >What is strange is that dlg_validate_dialog () seams to call >fm_realloc() which is not in the script. > Maybe a lin

Re: [OpenSIPS-Users] Config Suggestions Request (SIP Trunks)

Mike O'Connor wrote: > I've never once been able to get sipXecs to send out a register packet, > the are very clear in the documentation that they expect a sip trunk. ie > static configuration. Have you tried using the sipXbridge module? http://sipx-wiki.calivia.com/index.php/SipXbridge_Overview_

Re: [OpenSIPS-Users] t_relay() not relaying payload

Bogdan you are correct. Thanks for your help. It was an issue with fixing the NATed contact on the first reply. On 4-Feb-10, at 4:14 AM, Bogdan-Andrei Iancu wrote: > Hi Thamer, > > actually I see a 408 TIMEOUT for the second re-INVITE - but I see not > outgoing re-INVITE in the second step (on

users@lists.opensips.org

Hi Andrew, Which proprietary tool using for mixing of RTP files. So that we contact them and try to use that. Cause that we have facing problems while using sox and rtpbreak interms of voice quality and other issues. Regards, Nehru. -- View this message in context: http://n2.nabble.com/Query-r

Re: [OpenSIPS-Users] ping gateways in lcr or drouting?

Bogdan-Andrei Iancu wrote: > None of them do support pinging to GW, but I guess it will be a nice > feature for DR.. Bogdan, Another nice feature would be to add a pseudo-variable to do_routing() to take caller's URI from. LCR module already can do that. Probably I will come up with a patch for