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Howdy, How can you tell if RTP traffic has been reinvited/is bypassing an * server? Sincerely, Brent A. Torrenga brent.torrenga@torrenga.com Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 +1 219 836 8918 x325 Voice +1 219 836 1138 Facsimile www.torrenga.com

Seeking recommendations on handsets for use with Asterisk. I've been looking at the Aastra 480i CT because of its cordless handset and also the new Linksys SPA-942. Anyone using either one of these with comments on them? Any other thoughts on good reasonably priced handsets? This is for just a couple of people who work from home offices and will be connecting to an Asterisk server hosted

...n. (Closes issue ASTERISK-20492. Reported by Jeremiah Gowdy) * --- Improve Code Readability And Fix Setting natdetected Flag (Closes issue ASTERISK-20724. Reported by Michael L. Young) * --- Fix extension matching with the '-' char. (Closes issue ASTERISK-19205. Reported by Philippe Lindheimer, Birger "WIMPy" Harzenetter) * --- Fix call files when astspooldir is relative. (Closes issue ASTERISK-20593. Reported by James Le Cuirot) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.20.0 Th...

...n. (Closes issue ASTERISK-20492. Reported by Jeremiah Gowdy) * --- Improve Code Readability And Fix Setting natdetected Flag (Closes issue ASTERISK-20724. Reported by Michael L. Young) * --- Fix extension matching with the '-' char. (Closes issue ASTERISK-19205. Reported by Philippe Lindheimer, Birger "WIMPy" Harzenetter) * --- Fix call files when astspooldir is relative. (Closes issue ASTERISK-20593. Reported by James Le Cuirot) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.12.0 Tha...

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...ions Severity Minor Exploits Known No Reported On 05 September 2014 Reported By Philippe Lindheimer Posted On 18 September 2014 Last Updated On September 18, 2014 Advisory Contact Matt Jordan <mjordan AT digium DOT com> CVE Name...

...ions Severity Minor Exploits Known No Reported On 05 September 2014 Reported By Philippe Lindheimer Posted On 18 September 2014 Last Updated On September 18, 2014 Advisory Contact Matt Jordan <mjordan AT digium DOT com> CVE Name...

...http://lists.digium.com/pipermail/asterisk-dev/2011-August/050626.html * Significant fixes and improvements to parking lots. (Closes issues ASTERISK-17183, ASTERISK-17870, ASTERISK-17430, ASTERISK-17452, ASTERISK-17452, ASTERISK-15792. Reported by: David Cabrejos, Remi Quezada, Philippe Lindheimer, David Woolley, Mat Murdock. Patched by: rmudgett) * Numerous issues have been reported for deadlocks that are caused by a blocking read in res_timing_timerfd on a file descriptor that will never be written to. A change to Asterisk adds some checks to make sure that the timerfd is both...

...http://lists.digium.com/pipermail/asterisk-dev/2011-August/050626.html * Significant fixes and improvements to parking lots. (Closes issues ASTERISK-17183, ASTERISK-17870, ASTERISK-17430, ASTERISK-17452, ASTERISK-17452, ASTERISK-15792. Reported by: David Cabrejos, Remi Quezada, Philippe Lindheimer, David Woolley, Mat Murdock. Patched by: rmudgett) * Numerous issues have been reported for deadlocks that are caused by a blocking read in res_timing_timerfd on a file descriptor that will never be written to. A change to Asterisk adds some checks to make sure that the timerfd is both...

do you have selinux enabled? It should not be. p p.s. - if it comes to re-installing, you can backup all your settings with the freepbx backup utility and then restore so that you don't have to re-enter everything. From: "Lachek Butalek" <lachek@gmail.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Date:

Hi, I'm trying to edit an AMP-derived dialplan: the macro "dial" uses the AGI script "dialparties.agi" to find the extension to call. I'd like to drop this script: does anyone can explain me what is its main job? Thanks -- Domenico Viggiani

Hi Folks, I want to setup a follow me routine so that asterisk can call me on the multiple numbers. I tried some of the samples at voip-info but there is a problem with those examples. I dont have coverage in my home area and my cell phone answering machine picks up the phone right away so my home phone never rings. I also want the caller to be able to leave a voicemail and the cell phone

actually it sounds like a permission issue. You said you are doing it as root, but what is asterisk running as. I've found it is very sensitive, even to ownership. Make sure the owner:group is set to what Asterisk is running as before copying. Then, I've never had problems copying vs. moving - although I could imagine it might create problems in a race condition case. p From:

Kevin, since voicemail doesn't support saving in g729 format (as far as I have seen last time I looked into the code), it would need to transcode the g729 to wav or something else at this point to save the voicemail. Isn't that why it is failing when it hits the voicemail system? (sure sounds like what the error is complaining about). philippe alex.robar@gmail.com From:

For the DID's the easiest way for you to trasmit the incoming DID is to create custom extensions for the external numbers that access the external trunk directly. (e.g. they should NOT go to Loca/xxxx. or it will not retain the orignal CID in freepbx- which is effectively how it is being sent when you put xxxxx# in the ring list). As far as why it is only ringing one of your external numbers,

From: "Time Bandit" <timebandit001@gmail.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Date: Tue, 7 Nov 2006 08:53:51 -0500 Subject: Re: [asterisk-users] Follow Me problems > Today we appear to have discovered our first bug. We have an extension > setup to "followme" by ringing that extension

Hi, I've setup polycom https provisioning with an apache/linux server. However the log files aren't saved because there is nothing to process the http PUTS polycom uses. Does anyone have a secure solution they are using in this scenario so the phone log files can be saved? philippe --------------------------------- Cheap Talk? Check out Yahoo! Messenger's low PC-to-Phone call

Take a look at freepbx 2.2 beta. We have made both ringgroups and follow-me have a call confirmation option. When used, the ringgroup/follow-me extensions that are outside lines (like your cell phone) must confirm they want the call (press 1 to accept, 2 to decline). All the while the caller hears ringing (or MoH if chosen). If no answer, they are sent on to where ever else you want them to go

The Polycom's need to have their registration time lowered. Set it to 60 seconds which will re-register every 30 seconds. The polycom doesn't have any sort of 'keep alive' feature to keep the NAT holes open. There is information on the wiki fruther describing this and how to set it up if you don't know where to look. p From: "Von L." <methodvon@gmail.com> To: