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http://www.voip-info.org/tiki-index.php?page=Asterisk%20bounty%20Meet%20 Me%20video%20conferencing I posted this bounty for $US2,000 some months ago. Basically I needed the ability for 4 or 5 of us to conference on a weekly basis which is why I was happy to offer this bounty, however I have only had 2 people make brief inquiries and no one has really offered any substantial indication they

...hannel to another) doesn't work for us. Txfax called with the 'caller' parameter issues CED, while the receiving side needs CNG in order to switch to fax extension with rxfax. 2. This is probably the reason why J2 and our UC don't recognize incoming fax. Thank you. Alex Zarubin Webley Systems -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040330/5735e68b/attachment.htm

Hello, The v1-0_stable cvs release doesn't include the recent change ('poll' instead of 'select') in channel.c. Will it end up there any time soon, or we need to use cvs head to pick up this change? Thank you. Alex Zarubin Webley Systems -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040513/be61bfc0/attachment.htm

...esource busy chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or resource busy chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or resource busy chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or resource busy Thank you. Alex Zarubin Webley Systems, Inc. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030731/3c7f4565/attachment.htm

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Is there a way to play adpcm, 6KHz in asterisk? If yes, where can we get this codec? Thank you. Alex Zarubin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030916/b8be2453/attachment.htm

...ten => 99,8,Playback,/tmp/voxfile (put your own extension). Pcm recording is OK, playback is OK. Adpcm recording is noticeably worse. Adpcm playback is very bad/unusable. Is it just us (all our servers with T400P and TE410P), or it's a common adpcm codec problem? Thank you. Alex Zarubin Webley Systems, Inc. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030918/17258caf/attachment.htm

...orm uses SERVICE commands for this purpose to put B-channels into 'out-of-service'/'maintenance' or 'in-service' state. What is (would be) the asterisk solution? The current libpri doesn't send any SERVICE commands, just acks the incoming ones. Thank you. Alex Zarubin Webley Systems, Inc. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031201/2454f69a/attachment.htm

...ding txgain and rxgain to the call structure[s] and setting gains from there (instead of taking global values from config files) looks like an OK solution. As far as speed - the easy way would be using sox 'speed' effect. Any comments/suggestions are appreciated. Thank you. Alex Zarubin Webley Systems, Inc. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040120/ed1d58f3/attachment.htm

...achines in the office. We do have problems when we try to send faxes to services supporting fax, i.e. J2 or our UC platform. The receiving side doesn't recognize fax. To send a fax we drop into /var/spool/asterisk/outgoing: Channel: Zap/g1/<fax number> MaxRetries: 0 WaitTime: 20 Context: webley_txfax Extension: txfax_ext Priority: 1 SetVar: TXFAX_NAME=<fax name> and extensions.conf contains: [webley_txfax] exten => txfax_ext,1,txfax(${TXFAX_NAME}|caller) exten => txfax_ext,2,Hangup Two questions: 1. Sounds like txfax sends just one CNG tone. Can we have a parameter making t...

.... Can we ask to explain how buffer underrun is implemented? We could not figure it out by looking at file.c and zaptel with ZAPTEL_OPTIMIZATIONS. 2. Which asterisk parameters/configuration vars could help? Any suggestions on Linux configuration (IO related or any other)? Thank you. Alex Zarubin Webley Systems

Hi We are having a significant (> 1 sec) delay in a multi-asterisk conference, with IAX2 legs connecting meetme on different boxes. All the other legs are PSTN (TE410P). The example configuration Slave box 1 meetme <--- IAX2 ---> Master box meetme <--- IAX2 ---> Slave box 2 meetme The delay is between Slave box 1 and Slave box 2 The primary suspect is our iax configuration

Brock Tellier <btellier@WEBLEY.COM> writes: > Virtually any program using the GNOME libraries is vulnerable to a > buffer overflow attack. The attack comes in the form: > > /path/to/gnome/prog --enable-sound --espeaker=$80bytebuffer > The following exploit should work against any GNOME program, though I...

Hi all, After a long time having no time, I have finally done some fresh work on my software fax machine. I have replaced the original carrier tracking with something more robust. I have also added 4800, and 2400 bits per second modes, and cleaned up a few bugs in areas like superfine mode operation. I apologise for this update taking so long. At ftp://ftp.opencall.org/pub/spandsp you will

Hi all- Over the past week or two, I've been trying out asterisk under Fedora 1 Linux (RedHat). In my setup (single and dual Xeon motherboards), I have so far had a very good experience in terms of performance. In doing E1 load testing, I've found that Fedora handles heavy load much better than RedHat9, probably because of its better use of the multi-threading capabilities of the Xeon.

I have been trying to find information and possible issues with running a cluster of dovecot server with different versions without any luck. We have an existing cluster with dovecot v1.1.10 and tried migrating to a new cluster with dovecot v2.0.12. We ran into some issues when we moved traffic to the new cluster and have been unable to reproduce it in our own testing. We may need to put real

Dialplan stopped working after I did cvs update for zaptel-zapata-libpri-asterisk and 'make clean', 'make', 'make install' for the above components on 04-15-03. All the config files are the same as before. Both PRI and SIP calls I am making forward calls to 's' in the default context. Fields 'from' and 'to' look normal. My attempts to fix it by

gtk12-config --version 1.2.10 glib12-config --version 1.2.10 AZ -----Original Message----- From: Tilghman Lesher [mailto:tilghman@mail.jeffandtilghman.com] Sent: Wednesday, May 21, 2003 2:27 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] gastman segmentation fault when pressing 'enter' in a command win dow On Wednesday 21 May 2003 01:22 pm, Alex Zarubin wrote:

If there are multiple NICs in the box, how do we specify the local IP address to be used for RTP? Anything in rtp.conf ? Thank you. Alex Zarubin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030609/d151f190/attachment.htm