search for: abrahamsson

...a thread on reviewboard regarding this (https://reviewboard.asterisk.org/r/654/). It was Paul Belangers work trying to make the installation process take care of this. But the conclusion seem to be that the parts concerning this was postponed. So, did it make it in some other way? BR, Torbj?rn Abrahamsson -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111019/5a4e5150/attachment.htm>

...ng, and replaying, the voicemail works, and we receive both an h264 and an wav-file. What I now wonder is how to convert these into one file playable by a (standard) media player. I have not found any real good leads by google:ing, but of course I may have missed it Any pointers? BR, Torbj?rn Abrahamsson -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101214/6899049e/attachment.htm

Hello, I have written an AGI script for asterisk that uses google translate for text to speech synthesis. It supports a variety of different languages, local caching for the voice data and wideband audio. The voice in most languages is female and the quality of the synthesized speech is very high. More info about the script can be found here: http://zaf.github.com/asterisk-googletts/ the first

...is discussed for the 1.4 branch, and the result is that the problem should be fixed. But this is still a problem in 1.2 branch. Will this be corrected in a new release, or is this not considered a security fix and hence ignored? Actually isn't this a fix for a security fix... BR, Torbj?rn Abrahamsson

..., but I would like it to be dynamic, ie not having to update the friend definition every time a different domain is used. I understand that I would need to use outbound proxy in the client to prevent it from dialing the domain directly. Is this possible? Any alternatives? BR, Torbj?rn Abrahamsson -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140218/bbb463b8/attachment.html>

...------ From: Matthew Jordan <mjordan at digium.com> Date: 01/29/2015 10:41 AM (GMT-05:00) To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] JITTERBUFFER function On Thu, Jan 29, 2015 at 4:56 AM, Torbjorn Abrahamsson <torbjorn.abrahamsson at gmail.com> wrote: > Hello! > > > > I am going to use the JITTERBUFFER function in a SIP (and local channels) > only setup, but have some questions of how to use it: > > > > 1. Do I need to activate jbenable in sip.conf? Or is it eno...

...ow should I invoke this to make the buffer belong to channel B? Maybe using b option to Dial? So that when a JB-enabled device (B) calls out one just calls JITTERBUFFER from the normal dialplan flow, and if there is a call to the device (B) one need to use b option? Sound correct? BR, Torbj?rn Abrahamsson -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150129/f474aff8/attachment.html>

On Thu, Jan 29, 2015 at 4:56 AM, Torbjorn Abrahamsson <torbjorn.abrahamsson at gmail.com> wrote: > Hello! > > > > I am going to use the JITTERBUFFER function in a SIP (and local channels) > only setup, but have some questions of how to use it: > > > > 1. Do I need to activate jbenable in sip.conf? Or is it eno...

...re server? mail_max_userip_connections works well when the client is connection to the mail store without proxy, but when using proxies the POP/IMAP server will register the remote IP (rip) as the proxy server's IP address - thus a low limit will be reached quite easily. Thanks -- Martin Abrahamsson

...found the examples you mention. But they are only examples of how to start the buffer on the calling channel. Maybe adding what you just told me here to that page would make it easier for others to grasp as well? Anyway, thank you very much. Very appreciated! Keep up the good work. BR, Torbj?rn Abrahamsson

...s up after a longer interval. As it is hanging up after 5 seconds it is not such a long step to start thinking about 'qxdw(5)'... I'm not totally sure how to interpret what the w option does. What is a marked user? And what happens when the timeout occurs? Any ideas? BR, Torbj?rn Abrahamsson

Hi all, I am trying to send an extra header in SIP INVITE Message , i.e (email="me at me.com") but when I check the Message at the target that header is not there So I is Askterisk altering the Message and Is there away to include extra headers for SIP INVITE Message? Thank u