similar to: best SIP-softphone?

Hi all, This is my first question to this list, so please be gentle... Last week I installed a X100P FXO-card. And with not much tweaking I had it running fine, there is only one problem right now and it is the soundquality. When making a call sound is always perfect, both for the calling party and the called party. The problem is recieving phonecalls. The soundquality differs a lot then. The

I built ffmpeg2theora from SVN, using libogg 1.1.4, libvorbis 1.2.3 and Thusnelda from SVN. Running ffmpeg2theora --help reports that ffmpeg2theora accepts a range of -2 to 10 for the --audioquality option and a bitrate of 32 to 500 for the --audiobitrate option, but it doesn't appear to actually accept any of the low end numbers. It appears that the lowest acceptable value for

I did not install soxmix in my linux box. If you having issues with mixmonitor, you can put both legs of the call into a conference and record the conference -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Matt Roth Sent: Thursday, April 13, 2006 1:20 PM To: Asterisk Users Mailing List - Non-Commercial

I just check the source code, Monitor uses ast_writestream and it eventurally goes down to au_write, g723_write, etc. They don't commit to the disk. So, in effect, if you have a lot of ram, the audio should stay in ram until it gets swap out or the file is closed. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On

hi everybody, just tried to install my cisco 7960 - but without much success :-( I want to set it up as a sip phone - but I can not setup the phone's IP address... after plugging it in it says "Configuring IP" - I unlocked it and entered the "Network Configuration". I can see the edit-buttons but when I trie to press then it says "That key is not active here"

I've this error How i can find the problem? Apr 23 12:24:43 WARNING[131081]: PRI: received TEI check request for TEI = 89 Apr 23 12:24:47 WARNING[131081]: PRI: received TEI check request for TEI = 89 Apr 23 12:24:48 WARNING[131081]: PRI: !! Got a UA, but i'm in state 1 Apr 23 12:24:53 WARNING[131081]: PRI: received TEI check request for TEI = 89 Apr 23 12:25:02 WARNING[131081]: PRI:

I use xlite , but it isn't support video when it is free. who used better softphone ? Thank u. Best Regards Zhao Zigang ??? Alcatel Shanghai Bell Co., LTD *:388,NingQiao Rd.,Shanghai 201206 *:086-21-50554550-7762 *:Zigang.Zhao@alcatel-sbell.com.cn

Hi, I am facing a (for me) strange problem. When placing a SIP-Call I normally get connected and the dialplan is executed. The Call-Flow is controlled by a PHP-Agi-Script. The script answers the call (via AGI-Command) and a simple voicefile is played. SOMETIMES(!) I get disconnected immediately after the Answer - without any reason. This happens about all fifth call. Later on you will find

Hi, When an audiopacket is received too late I could interpolate this frame. The problem is that I don't know if it is a true bufferloss or just the last audiopacket of a talkspurt. Now my question is if it's harmfull for the audioquality that at the end of the talkspurt one frame is interpolated? Or would this be almost inpossible to hear since the last audiopacket in the talkspurt

I have a client that want to try the softphone with USB handsets route to see if hardphones will even be needed. I always push for hardphones (Polycom) so I am not sure about softphones or USB handsets. This is going to be for a 300+ seat call center onsite and many offsite, I plan on using OpenVPN for the offsite machines. Any advice on softphones, handsets, or practical experience with

Hi, the attached patch (against current SVN) adds three new options to ffmpeg2theora: --keyint (set keyframe interval) --smoothness (set the theora_info::sharpness encoding parameter) and --noautosync (disable the new frame dropping/duplicating code). It also fixes a bug with the processing of --cropright that sometimes segfaulted. The sync adjustment code is now tuned to be more reliable. BTW

Just passing this bit of feedback on! :) jack. ----- Forwarded message from Thorsten Titze <thorsten.titze@hangar-eleven.de> ----- Delivered-To: jack@64.139.18.57 Delivered-To: jack@icecast.org Delivered-To: jack@xiph.org Date: Sun, 07 Oct 2001 12:42:19 +0200 To: feedback@vorbis.com From: Thorsten Titze <thorsten.titze@hangar-eleven.de> Subject: Ogg Vorbis RC2 library for Dreamcast

more dreamcast news ----- Forwarded message from Thorsten Titze <thorsten.titze@hangar-eleven.de> ----- To: feedback@vorbis.com From: Thorsten Titze <thorsten.titze@hangar-eleven.de> Subject: Ogg/Vorbis for Dreamcast Organization: http://www.hangar-eleven.de Hello, just wanted to let you know that my sources of the Dreamcast Ogg/Vorbis Player library have now been included into

This is fixed in 2.2.27 with ddc96f7 lib-storage: Fixed error handling in list=children checking Aki On 23.01.2017 12:00, Aki Tuomi wrote: > Thank you for your report. > > Aki > > On 23.01.2017 11:56, Thorsten Hater wrote: >> OK, I found the problem in my config. If I use an default namespace with an >> empty >> name, instead of "inbox" it works as

Hi List; I need to configure a softphone to be client and use it with Asterisk, which is the recommended one? Is it iax2? Regards Bilal ____________________________________________________________________________________ Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games.

As workaround, you can either use VOLATILEDIR setting for maildir, or you can set maildir_stat_dirs=yes in config. Aki On 24.01.2018 14:45, Aki Tuomi wrote: > > Ah seems you have found a bug, apologies. =) > > We'll look into it. > > Aki > > > On 24.01.2018 13:34, Thorsten Hater wrote: >> Well, I have to scrub most data from the output, but it is what I

Hi everybody, I will receive my CISCO 7960G tomorrow. I've ordered it as a "global spare" without any callmanager licence. Now I don't know if I can get firmware-updates so could please someone send me the SIP-firmware? Is the default firmware the "skinny" one? Wich would be better to use with asterisk? Thank you very much

OK, I found the problem in my config. If I use an default namespace with an empty name, instead of "inbox" it works as expected. Here the log for this case Starting program: /usr/local/libexec/dovecot/imap -u **** imap(****): Debug: auth input: **** home=**** uid=48 gid=48 quota_rule=*:bytes=1000M imap(****): Debug: Added userdb setting: plugin/quota_rule=*:bytes=1000M Debug: Effective

Hi, I did added the default location and stripped down my config to a very basic level, dropping all plugins and database queries, see below. The segfault still appears in the same location. As I have build from source, I wonder whether you can reproduce the problem? Thorsten $ doveconf -n # 2.2.26.0 (23d1de6): /usr/local/etc/dovecot/dovecot.conf # Pigeonhole version 0.4.16 (1dc4c73) # OS:

Sooner or later everyone needs to install a package that is not included on the official installation CDs/DVD. Until now I just used rpmseek.com and Rpmfind.Net but of course this is far from optimal for security and compatibility to reasons. So what is the recommended way to get and install "non official" packages that are not part of the repository?! Thorsten