search for: pitucha

...- doesn't do any other modification or audiostream translation - only message passing. If someone's interested -- code + short doc is available at http://www.gradwell.com/tmp/iax_proxy.tar.gz Development will continue - any opinions / comments / contributions are appreciated. Stanis?aw Pitucha Gradwell Dot Com

...rring side should be billed for it. What can I do to see the difference between the channels here? If there is an A->B call going on, I'd like to know which side did the transfer - but whichever side does it, I get back to context 'default'. Any ideas? -- Kind regards, Stanis?aw Pitucha, Gradwell Voip Engineer T: 01225 800 831 | F: 01225 800 801 | E: stan at gradwell.net | www.gradwell.com Gradwell ? Internet for Business People Phone Services | Business Broadband | Email & Website Hosting Can switching to VoIP today put some change in your pocket? Registered Address: 26 Ch...

Hi, I'm trying to access audionativeformat / other codec variables in the hangup handler of a call (with ${CHANNEL(audioreadformat)}), but I get no response. Also 'core show channel ...' doesn't list those variables. Are they always set by asterisk, or only in some scenarios? It's a simple SIP-SIP call with audio passing through asterisk, same codecs on both sides. I see that

...ions of nat and qualify for the peer that has problems - rtp comfort noise is simply not sent. After trying to make it work for a day or so, I reported it as a bug (https://issues.asterisk.org/view.php?id=15466) but maybe someone here has some ideas how to make it work? -- Kind regards, Stanis?aw Pitucha, Gradwell Voip Engineer T: 01225 800 851 | F: 01225 800 801 | E: stan at gradwell.net | www.gradwell.com Gradwell - Internet for Business People Phone Services | Business Broadband | Email & Website Hosting Can switching to VoIP today put some change in your pocket? Registered Address: 26 Ch...

[This email is either empty or too large to be displayed at this time]

...ou? Do you use any workarounds? I'm asking here, because it would be strange if that functionality was broken since 1.4.8 and noone noticed ;) Exact scenario I'm using is described in the bug: https://issues.asterisk.org/view.php?id=15833 Thanks for any help. -- Kind regards, Stanis?aw Pitucha, Gradwell Voip Engineer T: 01225 800 831 | F: 01225 800 801 | E: stan at gradwell.net | www.gradwell.com Gradwell ? Internet for Business People Phone Services | Business Broadband | Email & Website Hosting Can switching to VoIP today put some change in your pocket? Registered Address: 26 Ch...

Dear All Can you please do me favor and let me know how can I convert *.wav files into 32 bit 44 KHz ? Please be informed that I have specific sound files in *.wav format that I converted them into *.gsm format with the aid of the following command : #sox FR00003.wav FR00003.gsm It got through but the voice quality is poor . I need to convert the original *.wav sound files (their file attribute is

Hey I'm looking for an advanced scenario for sipp, that can be used for testing asterisk. Mainly I'm interested in making random calls between sipp pseudo-users. Did anyone try to do something like this? Or has anyone got an example scenario with working loops? Thanks

Hello, I've got a problem with rtp handling by siemens c450 and similar. I experience a couple seconds of silence between early media and normal call (normal call's rtp is dropped by phone). This is caused by SSRC changing (even though marker bit is set). I have all relevant patches applied - it still happens on 1.4.21.1 and every version before that. Especially

Hi, I was wondering if there's any sense in increasing audiobuffer above the minimal '2' in meetme, if every channel is already dejittered before (Local/.../nj - as described at: http://www.russellbryant.net/blog/2007/10/09/asterisk-jitterbuffer-support-for-applications/) Will it help in anything, or just increase delay? Thanks, Stan

Hi, I noticed something bad happening on our systems lately. We have lots of asterisk threads running, but most of them are completely idle - strace doesn't show anything happening there. The only thread doing work seems to do everything. I see it sending mysql queries, writing logs, sending both SIP and RTP, etc. etc. That doesn't seem right. This system uses a lot of AGI scripts, mysql