similar to: Sound Quality

Hi All, Thanks to the people that responded to my question. The suggestion of using SJphone instead of the Xten phone resolved the problem :-) I now have another sound related problem! I've installed mpg123 and ensured that the binary is copied to /usr/bin as well as the default install location. mpg123 plays mp3s back perfectly via the desktop speakers, but when I playback an mp3 via the

I've searched the archives a bit and have not really come up with a good answer to my queries. I have * running on a RH9 box behind a LinkSys NAT box. I can talk with iConnectHere outbound just fine. I am trying to configure an inbound Xten softphone from outside. I have that user set as NAT in sip.conf (seems to help), but I still cannot establish a full session. I think the problem comes

Dear Guys, I?ve test Xten Lite softphone to connect to my Asterisk Box but it registers all the three lines at the same time and if I try to dial an extension it tries to reach 3 Ext. at the same time, can somebody haved this trouble? and how can I fix it. Also, I ?ll like to have the Xten LITE or PRO Softphone (Lite is free and PRO about $50.00 USD) it can hanle 3 lines (lite) and 6 lines

Hi all, I'm running a stable Asterisk on a HP DL380G2 1.4Ghz 0,5GB RAM equipped with 1x TE410P and 2xJunghanns QuadBRI running in NT-mode. Connected to the BRI-Ports are 12 Fax-Modems (Elsa MicroLink ISDN/TL V.34) which are only operating in dial out analog mode to deliver fax messages. After a while of running fine (50-200 dial out connections) on some S0 spans the following message occurs

Ok I have a question. I have Xten-lite working with our Asterisk system and I am able to make and get calls. But the main problem is the sound is very choppy and sometimes it cuts off words. I have tested it with ulaw and alaw as well as GSM. They all do the same. ulaw seems to work better. I also have an ATA-186 which works great without this problem. Here is my Sip.conf settings.

Hello, Is there an utility for asterisk for codec conversion? I tried google but i haven' got anything. I am trying to initiate a call with G711 codec to asterisk and i would like asterisk to call a gateway with an g729 codec, therefore making a codec conversion from g711 to g729. I know chan_oh323 does it by specifying the OUT_CODEC variable, but chan_h323 does not. And i was wondering is

Hi, I have a deltathree account and I can place calls but I can't receive calls. I use Grandstram sip phones. When I call my deltathree phone # the voicemail is answer :(( I need some help and solutions from the guys who allready are using deltathree. I search on Internet and I try all types of configurations... :( This is my configurations files: - sip.conf - [general] port = 5060

I've got the XTEN Lite soft phone mostly working with * but it's dropping out like a very bad cell phone call. The GSM codec is worst (unusable), G711u and G711a are best but not good enough to use. I don't think it's a lack of bandwidth. What tuning options or approaches should I be investigating to make this work. Also, what's the best soft phone(s) for Windows XP?

pstn -> sip gw -> * -> C7960 When I dial into * via the pstn, I hear the ivr menu just fine (good quality). I press 3000 (valid extn), and I begin to hear ringing however the ring back is very very choppy. I answer the C7960, and speech is clear in both directions. Place the C7960 extn on hold, and the MOH is very choppy. Checking 'sip show channels' indicates both the sip gw

Hi folks, well the easiest soft phone I have found (from this list) is the x-ten. great little piece of software. BUT, on a windows machine I need to plug in a USB device and find the software to drive it which can actually use a normal handset (similar to the s100p) but can be used to actually generate/answer the calls and actually rings when a call is received. Any ideas ?? Gary .

Chris Lattner wrote: > On Dec 21, 2007, at 1:08 PM, Richard Pennington wrote: > >> I'm a little further along now. I've started to put together a simple >> driver for Elsa and LLVM that I'm calling "ellsif" (cute name, I think >> it works). >> >> The file being compiled is a "printf" function. Here are timing >> results

I'm a little further along now. I've started to put together a simple driver for Elsa and LLVM that I'm calling "ellsif" (cute name, I think it works). The file being compiled is a "printf" function. Here are timing results for optimized and unoptimized runs: [~/elsa/ellsif] dev% ./ellsif -v test/ofmt.i -time-actions Adding test/ofmt.i as a preprocessed C file

On 12/21/07, Richard Pennington <rich at pennware.com> wrote: > I'm a little further along now. I've started to put together a simple > driver for Elsa and LLVM that I'm calling "ellsif" (cute name, I think > it works). Er. Hm. Can you explain the name? The problem with names like "ellsif" is that it sounds like "else if". I like the

On Dec 21, 2007, at 1:08 PM, Richard Pennington wrote: > I'm a little further along now. I've started to put together a simple > driver for Elsa and LLVM that I'm calling "ellsif" (cute name, I think > it works). > > The file being compiled is a "printf" function. Here are timing > results > for optimized and unoptimized runs: Cool, this is

Any users of the VocTel VOIP service? (Canadian) How have you found the quality (Choppy / smooth audio)? Any problems registering? (I have been unable to register for hours) After reading about the collapse of a big USA VOIP provider, I'm curious Thanks, OCG -------------- next part -------------- An HTML attachment was scrubbed... URL:

> I've build gcc many times over the years for different target processors > and was never able to get my head around it internally. It is incredibly > complex. I also didn't like the fact that I had to have N copies of gcc > to support N processors. Scott McPeak is rather familiar with the internals of gcc and edg and says elsa is far simpler. > I became interested in

Daniel Wilkerson wrote: >> I've build gcc many times over the years for different target processors >> and was never able to get my head around it internally. It is incredibly >> complex. I also didn't like the fact that I had to have N copies of gcc >> to support N processors. > > Scott McPeak is rather familiar with the internals of gcc and edg and >

I'm running Asterisk 1.2.13 on a Solaris 10 X86 box behind an IPCop firewall on a 5Mbps down/512 up cable connection. I'm having sound quality problems when users call in for voicemail and with music on hold. The sound is choppy and muffled while souding pretty good for calls inside the network. I'd appreciate some pointers as to where to start looking to improve things. I've

Does anyone know how to maximize music on hold quality on calls inbound from PSTN? I know that it is common to have choppy and static sounding music on hold when connecting via PSTN but how can that be minimized? I assume that the bitrates, type of music, etc can minimize the effects. Does anyone have any experience in this area? Do you know where I should look for more information?

Hi Ted there was a drift from the initial topic. worth a new thread! 2014-07-27 9:13 GMT+02:00 Ted Mittelstaedt <tedm at mittelstaedt.us>: > On 7/26/2014 12:18 PM, Arnaud Quette wrote: > >> Hi Eric, >> >> sorry for the lag, summer time... >> >> I'm first seconding Charles comments >> >> 2014-07-09 12:31 GMT+02:00 Eric S. Raymond <esr