[Freeswitch-users] Attention: behavior change with r15025

/feature, especially for anthm and bkw. Kind regards, Tamas ps.: issue was related with FSCORE-454. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

[Freeswitch-users] stange segfaults with log

mistakes with building? Missed "make clean", or someting? Thanks in advance, Tamas ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBS

[Freeswitch-users] timeout while waiting for CF_BRIDGED flag

think it is a bug? Thanks in advance, Tamas ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

Re: [Freeswitch-users] Failed DTMF Sanity check

http://filebin.ca/ (up to 50MB) Tamas Michael Collins írta: > Put it out on a webserver where one of the devs can grab it with a > browser. > -MC > > On Tue, Jul 7, 2009 at 11:48 AM, Max Bridgewater > mailto:max.bridgewa...@gmail.com>> wrote: > > Thanks! What i

Re: [Freeswitch-users] Record_session cutting wav files

Hello, please try out FS >= r14143 as there were some fixes around call recording and media bugs. Please let us know the results. Regards, Tamas Mariusz Kołodziejczyk írta: > Hello > > I saw a strange behavior when i'm using record_session for outbound > call. Recorde

Re: [Freeswitch-users] Linking problems with mod_portaudio.so module

Did you make bootstrap.sh and configure before compilation? We have no issues on Ubuntu 9.04 (libtool 2.2.6a). Regards, Tamas Jason White írta: > I can report that this problem (the failure of mod_portaudio.so to be linked > properly) still persists as of revision 13970. > > T

Re: [Freeswitch-users] Ways of Integrating Sphinx...

Hi, maybe things in speex should be worth to look for too. Just my 2 cents... Regards, Tamas ps: It seems, VAD+DTX in mod_speex does not work. Brian West írta: > VAD isn't really high on my list right now. > > /b > > On May 4, 2009, at 12:46 PM, mszla...@aol.

Re: [Freeswitch-users] What would happen if a selected rtp port is used?

Hi, there is a similar JIRA ticket for auto port selection. Probably it is not exactly what you want but lets take a look. http://jira.freeswitch.org/browse/SFSIP-54 Regards, Tamas seven írta: > Hi, > > I'm running more than one FS on a server, just want to know, how does

[Freeswitch-users] loopback-b channels stay alive

have the loopback a and b leg as well as the sofia chan." but it doesn't work, because originate api doesn't let us originate inside a session. So we still using it. Thanks in advance, Tamas ___ Freeswitch-users mailing list

Re: [Freeswitch-users] Mod_limit stuck when hitting limit value

r12581 did fix the case when limit is called more times, right? Thanks, Tamas rod írta: > Hello, > > I'm now running r12590, the pbm was still there, but this was because of > a broken dialplan. > > I'm using this for exceeded limit: > > >

Re: [Freeswitch-users] Mod_limit stuck when hitting limit value

r12581 did fix the case when limit is called more times, right? Thanks, Tamas rod írta: > Hello, > > I'm now running r12590, the pbm was still there, but this was because of > a broken dialplan. > > I'm using this for exceeded limit: > > >

Re: [Freeswitch-users] Mod_limit stuck when hitting limit value

>> extension before reaching the limit extension. >> What was odd was that in ngrep I saw FS sending a 503, I will >> investigate on this. >> >> I will rerun long term test and let you know if all is ok, so I have >> to >> do for a previous mail related to

Re: [Freeswitch-users] openzap with A102

Hello, please show your configs (openzap.conf, openzap.conf.xml). Which protocol do you intend to use? (EuroISDN, etc.) Regards, Tamas Baskar írta: > *Hi Michael Collins, > > When i load mod_openzap i can able to get this output > **Successfully Loaded [mod_openzap**] wi

Re: [Freeswitch-users] Mod_limit stuck when hitting limit value

Hello, we had the same problem. we couldn't test r12581 for sure yet, but we will. This fix is only for the limit (db version), right? Would be limit_hash a better choice to increase performance anyway? Rod, do yoo have maybe experiences with limit_hash with your sipp? Thanks in advance,

[Freeswitch-users] Questions about speex

width requirements and Speex seems to have better MOS values over 10kbps (or 6kbps with VBR). Thanks in advance, Tamas ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-

Re: [Freeswitch-users] mod_shout delay in trunk

Hello, could this option be used to lower I/O load - to rather write more bytes at once rather than one by one - on file recording (record_session)? Regards, Tamas Anthony Minessale írta: > Thanks, > > We appreciate the positive feedback! > > if you revert the change I sugge

[Freeswitch-users] fifo problem

with the out method. and if this uuid is in the top of the fifo then pop it else don't pop anybody. it seems to be a hack anyway Thanks in advance, Tamas ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.

Re: [Freeswitch-users] record session in fifo

m. so these are handled the same way as a "normal" manual call. otherwise I guess we have to duplicate this logic. so we didn't find elegant solution for this neither thus I asked maybe there is. Regards, Tamas On Thu, Jan 29, 2009 at 5:44 PM, Anthony Minessale < anthony.min

Re: [Freeswitch-users] record session in fifo

in advance, Tamas Here is the callflow again: 1, originate a loopback channel via event socket 2, loopback-b channel is hunting the dialplan, wich decide routing, caller_id, the need for recordings and so forth, and bridge a sofia call 3. the record_session is running on the sofia ch

Re: [Freeswitch-users] record session in fifo

/resume functions into api function, what I should turn on and off on demand? Anyway, I quess this is a bit extreme circumstance, and it isn't so important to us now. Thanks, Tamas Tamas Cseke írta: Hello, Thank you your help. I tested with r11489, but moh is still recorded in fifo. I

Re: [Freeswitch-users] record session in fifo

Hello, Thank you your help. I tested with r11489, but moh is still recorded in fifo. I quess you I should test the CF_PAUSE_BUGS in r11466. But I didn't find where you check this flag. Is it maybe possible you forget to commit something? Thanks, Tamas I didn't find where y

[Freeswitch-users] record session in fifo

with &fifo out application 7 sofia channel is bridged to the consumer 8 loopback channels die after transfers everything is recorded into one file. but the problem here is again the unwanted recording in the fifo while the caller is waiting Could you please advise me any solutio

Re: [Freeswitch-users] Channel variable 'call_timeout'.

caller leg. > > > > On Fri, Dec 5, 2008 at 10:34 AM, Tamas Cseke wrote: > > >> Hello, >> >> I have the same problem, >> >> I don't understand the difference between >> >> progress_timeout >> originate_timeout >> call_timeo

[Freeswitch-users] loopback+intercept+bind_meta_app

e and there is not really a and b legs. Might not. Is it possible to use dtmf bindings with intercept? If it should work, then maybe it is an issue with loopback? Thanks, Tamas ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org

Re: [Freeswitch-users] DNS RV faiover

want to use the default port. Regards, Tamas Anthony Minessale írta: > it might not. > > try putting the value in register-proxy as well > sip:host.tld > > > On Wed, Dec 17, 2008 at 8:49 AM, Tamas Cseke wrote: > > >> Hello, >> >> We'd l

[Freeswitch-users] DNS RV faiover

use SRV records. Is there any differences between the 2 method? Thanks any help, Tamas ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswi

Re: [Freeswitch-users] Jitter + Packet Loss

Hello, I've added bounty for RTCP already: http://wiki.freeswitch.org/wiki/Bounty#RFC_3611_-_RTP_Control_Protocol_Extended_Reports_.28RTCP_XR.29_support I know that the requester wants stats for RTP but maybe we could make a joint bounty ;) Regards, Tamas Jonathan Palley írta: &g

Re: [Freeswitch-users] Channel variable 'call_timeout'.

I mean timeout for an answer. which variable would control this? I quess it was call_timeout previosly. Please explain me this timeout variables Thanks, Tamas Michael Collins írta: > FYI, it is on the channel variables page but it's in a crazy place under > "unknown functi

[Freeswitch-users] voicemail disk quota poll

Hello, I would like to use a disk quota in users' voicemail (http://jira.freeswitch.org/browse/MODAPP-173) We haven't decided yet what would be the better prompts to play to the caller when the mailbox is full. Please advice some messages! Thanks in adva

Re: [Freeswitch-users] strftime(%u)

Hello, I had the same issue, this is ignored in switch_apr.c:switch_strftime. I don't know why, but %w is almost the same so I didn't deal with it.. Regards, Tamas henkoegema írta: > Brian West-3 wrote: > >> what platform are you on? >> >> On Nov 22,

Re: [Freeswitch-users] apr_md5 windows build problem

no, freeswitch.2008.sln or whatever is it Michael Jerris írta: > did you try 2008 with the freeswitch.sln file? > > > On Oct 30, 2008, at 9:10 AM, Tamas Cseke wrote: > > >> Hi, >> >> The answer is pretty simply for me: I have only 2008 express. >> But

Re: [Freeswitch-users] apr_md5 windows build problem

Hi, The answer is pretty simply for me: I have only 2008 express. But I tried 2008 express too as I said earlier and got the same. Maybe it is not an issue with 2008 professional. Best regards, Tamas Michael Jerris írta: > It just hit me, this is 2005, not 2008, thats why. Can we tak

Re: [Freeswitch-users] apr_md5 windows build problem

Hello, I still have problem after a fresh checkout. I tried with MSVC++ 2008 express and I got the same errors too. Tamas Michael Jerris írta: > This should work on a fresh checkout. > > Mike > > On Oct 27, 2008, at 10:23 AM, Tamas Cseke wrote: > > >> Hello,

[Freeswitch-users] apr_md5 windows build problem

d, 0 up-to-date, 0 skipped == Thank you, Tamas ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-

Re: [Freeswitch-users] callerids

I guess Jonas Gauffin írta: > Hello > > How do I specify callerid in the dialplan? > > I tried with: > > > > > The gateway still receives the callerid that I've specified in the user > directory. > > Thanks, > Jonas > > >

[Freeswitch-users] api originate, then bridge to a uuid?

9c46-11dd-b3ea-09becf1e00a9 ${uuid})}) My problem is in this originate I can specify only app, so I try eval and then bridge, but eval don't execute it just eval. Is it possible to bridge a certain uuid to the originated call? Thank you a

Re: [Freeswitch-users] Load test - performance not even matching Asterisk

Hello, Could you please check this out: http://jira.freeswitch.org/browse/FSCORE-197 and test the performance with the attached patch? It would be nice to make sure that it doesn't change the performance negatively. Just if you have time and are you interested in :) Thank you, Tamas Jon

Re: [Freeswitch-users] Turn on CN packet generation

Hello, to answer for my own question: Tamas írta: > Hello, > > Is there a way to let FS send RTP CN packets by itself? > No, FS just forwards the CN packets (or absorbs). > What I mean is for example when I turn on VAD for an outbound call and > allow CN (suppress_cng=f

[Freeswitch-users] Turn on CN packet generation

Hello, Is there a way to let FS send RTP CN packets by itself? What I mean is for example when I turn on VAD for an outbound call and allow CN (suppress_cng=false) than lets FS sends out CN packets even when party A is not sending them. http://lists.freeswitch.org/mailman/listinfo/fre

[Freeswitch-users] openbts

Hi, Have you seen this? http://openbts.sourceforge.net/background.html I guess, FreeSWITCH would be better for this ;) Regards, Tamas ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman

[Freeswitch-users] CNG

on't really know how should I manage this case. In sofia profile I have suppression commented out: and I tried before bridge: and Any idea? Thanks in advance, Tamas ___ Freeswitch-users mailing list

Re: [Freeswitch-users] tricky bridge

Many thanks! That works :) Regards, Tamas Brian West írta: > This is in the default config that does exactly that: > > > > > > > > > > > > > > &

[Freeswitch-users] tricky bridge

in advance, Tamas ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

[Freeswitch-users] typo

Brian, I've got a little spelling fix for you :) Index: src/switch_core_session.c === --- src/switch_core_session.c (revision 9385) +++ src/switch_core_session.c (working copy) @@ -1047,7 +1047,7 @@ switch_log_prin

[Freeswitch-users] multiple sip domains

I could change the to host, authorization would work...How could I do this? Thanks any help, Tamas ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http:

Re: [Freeswitch-users] T.38 pass-through

I have 2 sip profiles, one for handling incoming request from cantata and another for sending out requests to CW. In both I've set: and in dialplan I have: I'll try to catch somebody for further investigation. Thanks, Tamas Anthony Minessale írta:

Re: [Freeswitch-users] T.38 pass-through

I've missed one fact: FS runs in OpenVZ guest. I don't know whether it is important or not t38 reinvite seems to be ok. Regards, Tamas Michael Jerrie írta: > This should work fine, just as well ad bypass_media except it can also > handle nat. What exactly is not

[Freeswitch-users] T.38 pass-through

so something might be wrong with the proxy mode. Cantata and FS box are on the same switch so this is not a network issue. No NAT involved in the whole scenario. Could anybody give an advice? Thanks in advance, Tamas ps: FreeSwitch Version 1.0.trunk (9233M) Linux mt-sw1 2.6.18-12-fza-amd6

Re: [Freeswitch-users] mod_portaudio send3rtppacket/60msinsteadof1packet/20ms

hings. Csaba can make further tests on Tuesday. Regards, Tamas Michael Collins írta: > > Tony has a valid point here. Is there any way for you to try on > different hardware? Portaudio, in the grand scheme of things, isn’t > very high on the food chain when it comes to FS developme

Re: [Freeswitch-users] eavesdrop dtmf commands

spy: sip>proxy --->sip -FS --->eavesdrop the portaudio channel. 1., portaudio is hear the spy, but the original call is active yet 2. make other half deaf (sip) and portaudio talk with the spy 3. everyone can hear, and talk with eachother. Thanks, Tamas ps: next week I

[Freeswitch-users] eavesdrop dtmf commands

talk with the other half during spy, could you tell me if is it possible? Thanks any help, Tamas ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http

Re: [Freeswitch-users] Freeswitch win32 error (Procedure entry point inet_pton could not be located in the dynamic link library WS2_32.dll)

I am using Platform SDK 2008, and you? Regards, Tamas > Hi, > > Over the days, my interest in FreeSwitch is overwhelmingly increasing. I > love this software. I am sure that one day i will be competent enough to > contribute to this project. > > I get this error when i compil

Re: [Freeswitch-users] Scheduled Hangup Issue

solution for this? Thanks, Tamas On 4/11/08, Anthony Minessale <[EMAIL PROTECTED]> wrote: > > I found a bug where some code to optimize for escaped strings not needing > expansion skipped the expansion which is not ok cos we need to strip the \ > in the result > > Update to

Re: [Freeswitch-users] Scheduled Hangup Issue

Is it any other solution for setting a global variable with the value of the bridge_uuid on answer? Thanks any help, Tamas > oops i meant \$ > > On Thu, Apr 10, 2008 at 4:24 PM, Anthony Minessale < > [EMAIL PROTECTED]> wrote: > > >> try escaping the $ with /$ &g

Re: [Freeswitch-users] windows build problems

be on uninstall something where left. I don't know. I installed 2008 express, and problem didn't appear already. Regards, Tamas On 4/10/08, Michael Jerris <[EMAIL PROTECTED]> wrote: > > Whoops. Missed that. Fixed the switch_ivr_play_say issue and another > build prob on msvc 20

Re: [Freeswitch-users] Scheduled Hangup Issue

;t set yet. Could you advise a solution for this please, if there is? Something like: Thanks, Tamas On 4/2/08, Anthony Minessale <[EMAIL PROTECTED]> wrote: > > > I added support in the latest trunk to set the variable > execute_on_answer to an application of your choice > &

Re: [Freeswitch-users] windows build problems

Hi, As I wrote in previous mail, it was svn trunk r8070. Regards, Tamas Michael Jerris írta: > Is this with svn trunk (what svn revision?) > > Mike > > > On Apr 9, 2008, at 5:01 PM, Tamas wrote: > > >> Hello, >> >> I tried to compile r8070 on

Re: [Freeswitch-users] windows build problems

tings\jalsot\asztal\fs\libs\voipcodecs\src\oki_adpcm.c(224) : warning C4244: '-=' : conversion from 'int' to 'int16_t', possible loss of data 23>c:\documents and settings\jalsot\asztal\fs\libs\voipcodecs\src\oki_adpcm.c(230) : warning C4244: '-=' : conver

[Freeswitch-users] windows build problems

Hello, I have 2 problems: 1, libvoipcodecs C2143: syntax error : missing ')' before ' I don't know what the problem with brackets, but with the attached modification it complies. Do you have any idea, why? :) 2. freeswitchcorelib acl functions can't find inet_pt