search for: goncharovsky

Hello, I am trying to configure new opus codec in asterisk 14, but unable to find any examples of codecs.conf settings for this codec. All I am trying to do - setup peer with using opus in narrow band mode (8kHz sampling rate). Does anybody know how to configure chan_opus? -- Regards, Igor Goncharovsky Unistim Dev: http://unistim.igorg.ru -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20161025/bc549364/attachment.html>

Dear, is any way to change , the size of voice packets? I want to increase the quality by decreasing the size of each packets, because of bandwidth failure. ? thanks in advance Mani -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081110/c1b2ed9d/attachment.htm

Hi, Is anyone aware of a SIP hardphone with Busy Lamp Fields supporting 2 colors (or more) ? This could be very useful to support extended presence, for instance. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090519/0b8f1b62/attachment.htm

...in a path more than one directory deep - results in no config parsing on module reload (Reported by tootai) * ASTERISK-23827 - autoservice thread doesn't exit at shutdown (Reported by Corey Farrell) * ASTERISK-23814 - No call started after peer dialed (Reported by Igor Goncharovsky) * ASTERISK-23673 - Security: DOS by consuming the number of allowed HTTP connections. (Reported by Richard Mudgett) * ASTERISK-23246 - DEBUG messages in sdp_crypto.c display despite a DEBUG level of zero (Reported by Rusty Newton) * ASTERISK-23766 - [patch] Specify timeout for datab...

...in a path more than one directory deep - results in no config parsing on module reload (Reported by tootai) * ASTERISK-23827 - autoservice thread doesn't exit at shutdown (Reported by Corey Farrell) * ASTERISK-23814 - No call started after peer dialed (Reported by Igor Goncharovsky) * ASTERISK-23673 - Security: DOS by consuming the number of allowed HTTP connections. (Reported by Richard Mudgett) * ASTERISK-23246 - DEBUG messages in sdp_crypto.c display despite a DEBUG level of zero (Reported by Rusty Newton) * ASTERISK-23766 - [patch] Specify timeout for datab...

...TLS retransmission (Reported by NITESH BANSAL) * ASTERISK-23564 - [patch]TLS/SRTP status of channel not currently available in a CLI command (Reported by Patrick Laimbock) * ASTERISK-23754 - [patch] Use var/lib directory for log file configured in asterisk.conf (Reported by Igor Goncharovsky) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.10.0 Thank you for your continued support of Asterisk!

Hi: i am surveying ip phones for our company. we will use them with asterisk. we have office in taiwan, hong kong,singapore and china. cisco and polycom are too expensive for us. we try several china brand ip phones. they are all cheap and some of them have good quality. but most of them won't offer future firmware support, which we think it's important for ip phones.

...t been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * Fix an issue with Music on Hold classes losing files in playlist when realtime is used. (Closes issue ASTERISK-17875. Reported by David Cunningham. Patched by Igor Goncharovsky) * Resolve a potential crash in chan_sip when utilizing auth= and performing a 'sip reload' from the console. (Closes issue ASTERISK-17939. Reported by wdoekes. Patched by Richard Mudgett) * Address some improper sql statements in res_odbc that would cause an update to fail on...

Ok, I think I asked this previously but don't remember seeing an answer... Yes, you can "tickle" an SPA94x or 962 and have it fetch a config from a TFTP server... But is there no way to simply "push" a couple of lines of XML config to it directly via an HTTP POST (sans TFTP server)? Thanks, -Philip

What's the difference between the TE121 and TE122. I read the description on Digium's site and it isn't clear to me. Best regards, -- Guilherme Loch G?es Visite nossa loja virtual: http://www.shopvoip.com.br Not?cias e F?rum sobre VoIP com software livre: http://www.asteriskexperts.com.br -------------- next part -------------- An HTML attachment was scrubbed... URL:

I was pointed to the following: http://asteriskforum.ru/viewtopic.php?t=1761 It is in Russian, which I don't speak, but it references an Asterisk patch. Is this patch in 1.4.17? Is it scheduled to be in 1.4.18 (or whatever ships after 1.4.17?) Anyone work with this?

Hello all, I would like to use say.conf settings but every time i restart asterisk i have to load manualy "say load new" is there a way to do it automaticaly i use asterisk 1.4.19 Thanks

...ues.asterisk.org/view.php?id=16867> Fixed playing dialtone in some scenarious when conversation already started - Fixed dispalying on-screen information when using Redial softkey (DN number and timer displayed). - Not sending short ring in case of call forward enabled on phone -- Regards, Igor Goncharovsky Blog: http://igorg.ru -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101102/994dd055/attachment.htm

...t been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * Fix an issue with Music on Hold classes losing files in playlist when realtime is used. (Closes issue ASTERISK-17875. Reported by David Cunningham. Patched by Igor Goncharovsky) * Resolve a potential crash in chan_sip when utilizing auth= and performing a 'sip reload' from the console. (Closes issue ASTERISK-17939. Reported by wdoekes. Patched by Richard Mudgett) * Address some improper sql statements in res_odbc that would cause an update to fail on...

...by Steve Davies) * ASTERISK-23650 - Intermittent segfault in string functions (Reported by Roel van Meer) Improvements made in this release: ----------------------------------- * ASTERISK-23754 - [patch] Use var/lib directory for log file configured in asterisk.conf (Reported by Igor Goncharovsky) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.28.0 Thank you for your continued support of Asterisk!

...NG about passing an empty string is a bit over zealous (Reported by Matt Jordan) * ASTERISK-23985 - PresenceState Action response does not contain ActionID; duplicates Message Header (Reported by Matt Jordan) * ASTERISK-23814 - No call started after peer dialed (Reported by Igor Goncharovsky) * ASTERISK-24087 - [patch]chan_sip: sip_subscribe_mwi_destroy should not call sip_destroy (Reported by Corey Farrell) * ASTERISK-23818 - PBX_Lua: after asterisk startup module is loaded, but dialplan not available (Reported by Dennis Guse) * ASTERISK-18345 - [patch] sips connection...

...solved in this release: Bugs fixed in this release: ----------------------------------- * ASTERISK-23911 - URIENCODE/URIDECODE: WARNING about passing an empty string is a bit over zealous (Reported by Matt Jordan) * ASTERISK-23814 - No call started after peer dialed (Reported by Igor Goncharovsky) * ASTERISK-24087 - [patch]chan_sip: sip_subscribe_mwi_destroy should not call sip_destroy (Reported by Corey Farrell) * ASTERISK-23818 - PBX_Lua: after asterisk startup module is loaded, but dialplan not available (Reported by Dennis Guse) * ASTERISK-18345 - [patch] sips connection...

...solved in this release: Bugs fixed in this release: ----------------------------------- * ASTERISK-23911 - URIENCODE/URIDECODE: WARNING about passing an empty string is a bit over zealous (Reported by Matt Jordan) * ASTERISK-23814 - No call started after peer dialed (Reported by Igor Goncharovsky) * ASTERISK-24087 - [patch]chan_sip: sip_subscribe_mwi_destroy should not call sip_destroy (Reported by Corey Farrell) * ASTERISK-23818 - PBX_Lua: after asterisk startup module is loaded, but dialplan not available (Reported by Dennis Guse) * ASTERISK-18345 - [patch] sips connection...

...NG about passing an empty string is a bit over zealous (Reported by Matt Jordan) * ASTERISK-23985 - PresenceState Action response does not contain ActionID; duplicates Message Header (Reported by Matt Jordan) * ASTERISK-23814 - No call started after peer dialed (Reported by Igor Goncharovsky) * ASTERISK-24087 - [patch]chan_sip: sip_subscribe_mwi_destroy should not call sip_destroy (Reported by Corey Farrell) * ASTERISK-23818 - PBX_Lua: after asterisk startup module is loaded, but dialplan not available (Reported by Dennis Guse) * ASTERISK-18345 - [patch] sips connection...

...by Steve Davies) * ASTERISK-23650 - Intermittent segfault in string functions (Reported by Roel van Meer) Improvements made in this release: ----------------------------------- * ASTERISK-23754 - [patch] Use var/lib directory for log file configured in asterisk.conf (Reported by Igor Goncharovsky) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.28.0 Thank you for your continued support of Asterisk!