similar to: Re: Wine & GSM 6.10 codec

Dear all, I'm planning to buy some IP phones with GSM audio codec support in order to use with an Asterisk SIP server I have implemented and nowsuccessfully running with softphones like Eyebeam and Twinkle. A vendor offer to me the SNOM 300 IP phone, that support GSM 6.10 audio codec. I've looking for GSM 6.10 codec in the web but there is no helpful information. Just I enter the

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*I have a Ham Radio program, named CQ100, it works fine using WINDOZE, but when I installed the same program on my linux system everything works except there is no-audio I'am using Ubuntu 7.10 linux... The author told me that windoze uses GSM 6.10 codec, plus ACM audio compression manager, these are built-in... So by anychance does anyone know of a program that one can get to use on a

I have around 20-30 softphones behind NAT .. My sip.conf has nat=yes and they all are able to register and make calls with no problem . My voip carrier supports gsm as well as ilbc .. Server takes calls from sip phones , does call recording in between and forwards to voip carrier . My problem is that half of my softphones use ilbc and rest use gsm and my provider supports both gsm as well as

Dear all, I have Asterisk 1.4 with SIP. I have a voicemail implemented with GSM sound files. The problem is I have IP phones Utopix HyperPhone 202 which support only G.729a/u and G.723.1 high/low, but not GSM. If I choose G.729A the "pass-throu" calls among users are OK, but Asterisk can't transcode GSM to G.729A in voicemail calls. My company doesn'y want to pay for a G.729

Hi, I'm using Windows Messenger 5.1 (It supports SIP) with Windows XP to connect to another SIP user using asterisk in the middle. The codec selected is gsm and I have a problem because the sound sent by my machine reach the end point all wrong, it seems just noise... If I use uLaw (or gsm from other machine with othe UA) everything works fine. I think the audio codec used

Hi all, I'm trying to run Ventrilo (voice chat) on WINE. I have the latest version of wine. I installed correctly Ventrilo. Now I have to setup it to use the GSM 6.10 codec. I read on-line that I only need to add "MSACM.msgsm610=msgsm32.acm" to the system.ini file, and add the file in the /system directory. I did it, but when I choose the codec GSM I get an error: "Unable to

Larry Shields wrote: > *I have a Ham Radio program, named CQ100, it works fine using WINDOZE, > but when I installed the same program on my linux system everything > works except there is no-audio I'am using Ubuntu 7.10 linux... > I wanted to install it on unix system freebsd, but too few compitable on that systems. That system only for netware )

Dear all, I have implemented an Asterisk SIP server for a WAN VPN over Internet. We have users distributed along all my country (Argentina) that register to my Asterisk in order to talk among them. I'll plan to have voice and voicemail with GSM codec, because we can't afford the payment for the G.729 licenses (it's an administrative problem of our company, not an echonomical problem).

Dear all, sorry for the Off-Topic but I have an Astreisk 14 voip server connected to Twinkle and X-Lite clients. I have to use the GSM codec for all of my clients, and it was set up in the sip.conf specifically in "allow=gsm" line. Twinkle has GSM codec built in, but when I open X-Lite audio codecs settings I can't see the GSM codec, being that the official web site and the PDF

All, Has anybody come across a problem with the MS-GSM codec on Asterisk? We have a client for Windows XP that uses the MS-GSM codec. While the codec seems to work when talking directly to Asterisk, when 2 of these clients talk to each other through Asterisk the sound is all mangled on both ends. I assume MS-GSM is the same as used by Windows Messenger and so my problem may be related to

I've a question about the bandwidth consumed by IAX2/GSM. According to the wiki page, the GSM codec should run about 13 kilo-bits/sec for a voice encoding. However, watching gkrellm when I initiate a call to Digium, it looks like the channel is taking a consistent 5-6 kilo-bytes/sec. That's a lot more bandwidth than it should take. Is there perhaps a setting I have wrong somethere in

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 list! I only can make successful calls if I disable gsm with "disallow=gsm". As soon as I allow gsm the following appears at the console. There are much much more Lines with "File dsp.c, Line 1198" but I cut them for a better survey : --------- Log Start ------------- Asterisk Ready. WARNING[5126]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on

I am configuring a test Asterisk server (1.8.9.2) to practice setting a single codec globally, to avoid transcoding as much as possible. Since all of my recordings are in gsm format, I am trying to make the SIP clients use gsm everywhere. I am using Ekiga on Fedora 16 x86_64 for my tests. [root at elx2 asterisk]# cat /etc/asterisk/sip_general_additional.conf

Hello, are the gsm codecs in Cisco as53xx or 36xx (gsm, gsmfr) copatible with Asterisk's gsm codec? Thanks in advance, Thomas

Hello All, I would like to only use the gsm codec for all the calls, is it possible I want to use minimum possible bandwidth as we have most of calls over Internet. Regards, Sanjay Rajdev

Hello All, I'm trying to get my SNOM 200 to work using the GSM codec. The problem is it wont' pass dtmf digits. I have it dtmf outband on, inband off. In my sip.conf I have dtmfmode=rfc2833. Whenever I press a digit on the snom phone I get "invalid gsm data" so I think it's a configuration problem in the snom, anyone have a working configuration using gsm? Thanks -Matt

Hi, I am not sure if I am doing something wrong, but I can't get MeetMe to work with GSM codec (Asterisk 1.6.1 SVN r190371). My config files below: ---- sip.conf: ---- [general] context=common canreinvite=no bindport=5060 bindaddr=78.105.1.127 disallow=all allow=alaw allow=gsm rtptimeout=600 rtpholdtimeout=3600 rtpkeepalive=30 nat=no jbenable=yes tcpenable=no realm=dev-sip.wima.co.uk

Currently I am creating .wav files and then converting them via SOX to .au file format, then running them through a gsm codec convertor which all works fine except that it sounds like the recording was made with a sock in my mouth !! Could someone in * land help me to get a good sound quality with gsm format. Thanks in advance. -------------- next part -------------- An HTML attachment