similar to: Understanding codecs and transcoder

Dear Mailinglist-User currently we`re working with an IP-PBX, based on Asterisk, with SIP, H.323 and ISDN-Capabilities. SIP and ISDN works fine, but H.323 not. In our first test, we started to connect Asterisk to an Cisco IOS-Gatekeeper with the "chan_oh323" (version 0.6.5). We successfully tested in/egress calls without any problems. But when we started to connect our Asterisk

Hi all, I am using chan_h323 from Asterisk CVS to interconnect with GNUGK v2.2.2. Then I made call from a H323 EP, thru GNUGK, to SIP EP on Asterisk. However, I only heard ringing when the call was answered on SIP side. Below is the debug from chan_h323. Any help is welcome. Thanks. *CLI> == New H.323 Connection created. -- Setting up Call -- Call token:

Dear friends, I found the thread on this subject this summer but wonder whether it has been taken any further. I have an important medical problem where X is computed from a three independent and complicated measurements (exchangeable sodium and potassium and total body water - i.e. X = (Nae+Ke)/TBW ) and Y is serum sodium concentration (all data in Edelman, JCI 1958). I have the individual

Hello, I'm trying to use the GSM codec with Nufone H323 but it's not working. Does somebody has some idea? Have I missed something? Thanks!! Celso Fassoni Some additional info: (I'm using CVS-HEAD - downloaded today) monkey:~# cat /etc/asterisk/h323.conf [general] port = 1720 bindaddr = 192.168.0.100 ; this SHALL contain a single, valid IP address for this machine

Hello everybody, I?ve been trying to solve a problem for several weeks now but it really beats me. There are several hard phones connected to an Innovaphone 3000 VoIP gateway. On the other side I have a SIP softphone connected to Asterisk. The problem I have is that on incoming calls (hardphones to softphone) I only have outgoing audio (from soft to hardphone); everything is OK when I call the

Hi, In voicemail.conf: ; Supported values: ; 'filename' filename of a soundfile (single ticks around the filename ; required) ; ${VAR} variable substitution ; A or a Day of week (Saturday, Sunday, ...) ; B or b or h Month name (January, February, ...) ; d or e numeric day of month (first, second, ..., thirty-first) ; Y Year ; I or l

Hello All: Does the Asterisk support to insert an off the board transcoder for a call? Thanks, Charles ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping -------------- next part -------------- An

hi folks. i have several remote sites with total of 200 sip phones connect to our Asterisk server. i want to minimize bandwidth usage and thinking about getting a Digium TC400B transcoder card. what are your experience with it? how's the quality? also if there are 120 active channels in used. will the 121 person able to make calls? will it support more channels if i put 2 cards in the system?

Hey Guys, I've been working on Icecast 2.3.1 on my test server (Virtual Server-win2k3) And I am not able to get either stream transcoder v3 or v1.25/6 (whatever the current dos one is) to work on it. Also, there is a problem with the people from the game secondlife, they also cannot connect to the server to listen to it. Has something changed in the latest configuration? I've gone

Hi All, Should anyone be looking for one, here's a DirectShow-based command-line transcoder for Theora: http://www.eddiana.com/?totheora.html It can do basic video transcoding, as well as scale video, watermark video with a graphic, grab frame stills, capture from video sources, and perform a couple color transformations. Ed

Dear all I have Asterisk which is connected with avaya through E1 back 2 back now i have on asterisk side G.711 codec and Avaya also useing G.711 codec everything fine. I need G.729 on my asterisk side. can i have lots of SIP phone on my lan and issue is i have 2 to 3 building so problem is LAN is congested thats why i need G.729 Now testing perpose i have download

What I am asking for is something to take an incoming SIP INVITE, change the codecs listed in the SDP, forward the (new) INVITE to a media gateway, perform the reverse codec handling for the 200 OK and perform RTP transcoding on the resulting 2 legs of the call. -How can asterisk do that ! -do any one know a distribution contain asterisk have solution like that ? Regards -----Original

Hi All, I have recently installed a TC400B card into a system and am trying to get it to work. As far as I ca tell from the docco on Digiums website, there is no config as such unless you want to enable / disable only 1 codec, otherwise by default it runs as 92 channels of either. I have tried asterisk 1.4.9, 1.4.10 and 1.4.10.1 along with zaptel 1.4.4 and addons 1.4.2. The zaptel modules

I'm running Samba 3.0.7 on SuSE 9.1 with OpenLDAP for auth. I've been mostly following along Terpstra's Book "Samba 3 By Example", as lots of other people here have been. The three big problems I'm having are: 1) After a Windows XP user logs out for the first time, upon next login they get an error message saying "Windows cannot log you in because your profile

Hello, I'm running Asterisk 1.0.1 (the same was with 0.9, 1.0). When it receives inbound H.323 call it makes connection and uses local 127.0.0.1 address to send audio stream: remoteIpAddress: 127.0.0.1 When making outbound calls from Asterisk it makes correct connection to send audio stream. Is it a bug in h.323? Is there some more settings to make in .conf files? See detailed debug below:

2008/12/9 Olivier <oza-4h07 at myamail.com> > > > 2008/12/9 Tilghman Lesher <tilghman at mail.jeffandtilghman.com> > > On Tuesday 09 December 2008 09:14:11 Olivier wrote: >> > Hi, >> > >> > In voicemail.conf: >> > ; Supported values: >> > ; 'filename' filename of a soundfile (single ticks around the >>

Dears Any one knows a standalone voip transcoder software name,not an ip pbx. What I want is to transcode the incoming sip calls from g711 to g723 or ilbc or g729 ..... and forward it to a media gateway .. Regards Khaled chehab ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by

I'm having tons of trouble getting the fromstring to work in voicemail.conf. I've tried both voicemail and voicemail2 but the emails still seem to be coming from asterisk pbx. Has anyone had any luck with this? ================= Here's my voicemail.conf: ; ; Voicemail Configuration ; [general] ; Default formats for writing Voicemail ;format=g723sf|wav49|wav format=wav49|gsm|wav ;

Hi, all! when my h323 phone dial in Asterisk system, i can hear nothing. and the following is the log slice i picked from /var/log/asterisk/full. ps: i am using red hat AS5 kernel 2.6.18-53.el5,Asterisk-1.4.24.1, pwlib_v1_11_0, openh323_v1_19_0_1. Best Regards! 81948 [May 6 10:07:34] VERBOSE[11579] logger.c: -- Remote UNIX connection 81949 [May 6 10:07:51] VERBOSE[29627] logger.c:

Greetings, I read from archive that OCFS is for kernel 2.6. which means RHAS4.0 is supported. However, I can't find the download for this version. Is the support official and where can I download it. Thanks / regards, Ivan Wong Database Administrator e2Open Inc. (www.e2open.com) Suite 34.03, Level 34, Menara Citibank, 156, Jalan Ampang, 50450 Kuala Lumpur, Malaysia DID: +603 2776 6397