similar to: Will there be any support for iLBC in IAXClients soon?

Hello All, I have a number of X-Lite users in countries where the incumbent Telco will do anything to block VOIP traffic. For some reason neither the X-Lite broadband or dialup clients would register with my server unless we configure them to use the Xten Xtunnels demo server. Once the Client has registered the call quality is great! The problem is the Xtunnels does not support other IP Hard

Any help on this would be great I have 2 TDM400P's, 2 asterisk servers (running on powerful boxes with FC1 and * v CVS 1.0.02), and 4 analogue PSTN lines from BT and whatever I do, I cannot get rid of this damn local echo. Ive tried setting the echoTraining, echoCancel (in phone.conf and Zapata.conf) , echocancelwhenbridged to every possible combination , Ive even tried running the fxotune

Hello All When I open up iaxcomm, it registers fine with the asterisk server. If I call into it, iaxcomm will ring; however if I leave iaxcomm sitting idle for awhile (I haven't figured out exactly how long) it seems to miss calls. I can see the calls coming in on the asterisk server but they never ring through on iaxcomm. If I close it and reopen it, it takes calls again fine. I thought I

Hi All, I try to compile the last version of iaxclient with the new build system, i also compiled speex 1.2b2 . When the makefile try to compile the sample code testcall I have the following mistales: gcc -I/d/Lib/iaxclient-2.0.1/portaudio-1.9/src/common -I/d/Lib/iaxclient-2.0.1/speex-1.2/include -o testcall.exe testcall.o ../../lib/.libs/libiaxclient.a

Hi, I had downlaoded iaxclient-2.0.2 and complie project *\iaxclient-2.0.2\contrib\win\vs2005* ** It gives many83 fatal and file missing error of file missing Error 1 fatal error C1083: Cannot open include file: 'portaudio.h': No such file or directory d:\mohit\asterisk\iaxclient-2.0.2\iaxclient-2.0.2\lib\portmixer\px_win_wmme\px_win_wmme.c 40 Error 2 fatal error C1083: Cannot open

About 2 weeks ago, I made the first announcement to the list about the first IAXCLIENT cross-platform builds. What we had then, was a command-line application which could make a single outgoing call. We've since done quite a bit of work on the clients, and the client now has: 1) A working cross-platform GUI 2) Volume level controls, and re-tuned audio processing 3) Multiple call apparances

asterisk 1.4.35 dahdi 2.3.0.1+2.3.0 one span on a 4port T1 card Got complaints this morning that outbound and inbound calls were "scratchy" and I made a few test calls. It kind of sounds like the gain is too high somewhere, and the audio is overdriven. Is this a problem at the carrier? I'm trying to call them now, but it's Sunday morning in the sticks, and my chances of

Hi, it took me 2 days to get my asterisk box running, so now I completed and I am disappointed of the sound quality. When I call other people their voices sound somewhat scratchy. First I thought it might be a codec problem, but I also recognized it during the ring tone or even the DISA connect tone. Sometimes it is better quality and sometimes more scratchy. Where might be the problem? I am

Hi, I'm experiencing a few (but meaningful) cases of audio distortion (or bad quality). I can't say yet how often this happens. Please listen to the following sound file: http://213.96.91.201/temp/distorted_audio_1.wav This was recorded by Asterisk while the local SIP caller was dialing out a SIP trunk (so the problem is on my side, definitely, and it doesn't seem to be related to

Hello, I need to demux video files created with ffmpeg2theora 0.28+svn18147 I have ogminfo v1.5 and getting this error (ogminfo.c) OGG stream 1 is of an unknown type (bad header?) (ogminfo.c) OGG stream 2 is of an unknown type (bad header?) (ogminfo.c) (a1/serial 1092562206) Vorbis audio (channels 2 rate 48000) Is possible, that ogminfo is older then ffmpeg2theora ? Or what can be the reason?

David Brazier wrote: > Hi > > The latest SVN trunk for speex has changed the SpeexPreprocessState to > an opaque structure, for jolly good software engineering reasons. > However, the Analogue AGC (AAGC) feature of iaxclient (in audio_enode.c) > relies on some members of this. It uses speech_prob to detect when > there is enough speech to consider AAGC and then loudness2 to

Asterisk-people, Some of you may have heard that we were working on a simple, cross-platform IAX client library called "iaxclient". We've pretty much been "on vacation" with the project for a while, but recently have made some progress, and now have the library working across platforms, and a simple test client called "testcall", up and running on 3

Hi, sometimes a ms-office file (ms word 2k, ms excel 2k) will be destroyed, when the client writes it to the server. The strange thing is, that samba writes parts of its log-files at the top and bottom of the file. Here are the first 2 lines of the destroyed file: [2002/09/04 17:03:05, 2] smbd/open.c:open_file(216) tb opened file xxx/xx/xx/xxxx/xx xx/filename_TB.doc read=Yes write=Yes

On Thu, 08 Apr 2004 10:14:09 -0400, Steve Kann wrote: >Gary wrote: > >>I have noticed lack of codec negotiation with calls thru a registrated >>asterisk box. >> >>No seen problems with outbound calls, (though I haven't specifically >>tried it), but the problem exists inbound. >> >>Easiest method for testing this was ring in via a sip client set

e.g. can we make it throw warnings etc. rsa-sha2-256 and rsa-sha2-512 are fine, they use PSS. On Sun, Jan 20, 2019 at 1:55 AM Yegor Ievlev <koops1997 at gmail.com> wrote: > > Also can we do anything with ssh-rsa? It uses both SHA-1 and > deprecated PKCS#1 padding. If it's used to sign certificates, there's > no additional protection of SHA-2 hashing before SHA-1

We have a DHCP/PXE server in a build environment, which is separate from our Spacewalk v1.5 server. We direct builds to Satellite or Spacewalk based as needed. It contains initrd & vmlinuz files for each version/arch we currently deploy for both RHEL & CEL. I'd like to keep the storage footprint to a minimum, if possible. Keeping a big directory of ISOs to loop-mount for kickstart

Hi, I have an audio quality problem regarding IAX2. I have 2 Asterisk servers interconnected via 2 LAN trunks at 1Gbps (no nat, no firewall). One trunk is SIP and the other IAX2. Normally, I use IAX2 but have noticed easily reproducible audio quality problems (voice in/out is OK but there's a "third" noise overlapping with a "scratchy sound" as if it were some kind of

Jean-Marc Valin wrote: >> As you can tell, the AAGC integration with speex was really a classic >> hack. Instead of re-creating the hack, what's probably best here is to >> integrate AAGC back into speex, and have a proper API. >> > > Agreed here. If you can come up with a clean patch to add that feature, > it's something I'd like to see in

Hello All, I have been looking for a commercial version of an IAX2 Softphone for Windows but the ones I have came across (i.e. Iaxcomm, Iaxphone, Diax) do not seem to have an updated version since April 2004 in some cases. We looked at Firefly but we sent emails to Virbiage/Freshtel with questions and could never get a response from them. Has anyone got any recommendations for commercial

Same problem here: if call come over ISDN PRI and it is for a SIP phone that equals to strong echo situation, at the SIP end. Interestingly this doesn't happen on all calls but it does on 95% of them. Asterisk load at that moment is insignificant - 1 to 2 calls. I have tried with all possible echo cancellers in zconfig.h, with and without MMX, and with and without CFLAGS+=-march=i686 in