similar to: Outgoing phone calls muffled

Hello, Several people report that outgoing phone calls to our clients sound muffled, like they are talking underwater. Reported for both the Snom 870, and the polycom ip650. Incoming calls sound ok. Could this be a codec problem? My dialplan looks like: [general] port = 5060 bindaddr = 0.0.0.0 srvlookup = no tos_sip = cs7 tos_audio = ef registertimeout = 1 relaxdtmf = yes context =

All: RKG needs an asterisk consultant to help us track down issues we are having with our system. Mainly dropouts and dropped calls. If you have experience in troubleshooting these issues, please contact me at email attached to this messages. Regards, Eddie -- Eddie H. Mikell Senior Systems Engineer RKG Office: 434.970.1010 x 124 Email: emikell at rimmkaufman.com --

All, I'm looking for sip phones that support something other than openvpn. There are a lot of vendors in China (mainly Alibaba) that sell l2tp VPN phones. Are there any American vendors that support l2tp? Thanks, -- Eddie H. Mikell Senior Systems Engineer RKG Office: 434.970.1010 x 124 Email: emikell at rimmkaufman.com -- <http://www.rimmkaufman.com>

All, The users in our organization are well, quite frankly, sick of phone service that is being provided. The choppy phone calls, and drop outs are detrimental to our sales force. I've tried about everything I can think of. Moved the asterisk server from VM machine to dedicated machine More than enough bandwidth Setting 802.1p = 7 Set Dedicated voice traffic 35% of bandwidth. Not sure

Hi All, Have completely moved off the old ESI system, and things have been going pretty good with the new server. I have one issue, which has been reported by several of our customers. I've tested it, and it does indeed seem to be a problem. When the customer is asked to dial in the first three letters of the person they are trying to reach, they will be routed to the wrong extension.

All: Yesterday I discovered something interesting. I dialed 1800ANCESTRY from the asterisk system I am testing and got the number doesn't exist message. I then dialed the same number from our old system and it went through. I realized that the "Y" in ancestry made the number too long, and went back to my dialplan. How do I ignore numbers that are too long? Obviously,

Hi folks, I''ve been trying to work out this situation for a few days now without much success and dug all over the archives for similar issues, none of which seem to relate exactly. I installed Debian Etch and the Xen (3.03) kernels / binaries on a Dell server. It works fantastic with all the default configurations with one strange flaw - as soon as the /etc/xen/scripts/network-bridge

All: My company has an existing ESI IVX E-class system with 45 phones. I can add one more card, to expand it another 6 phones, but it's $8000, and then the system will have to be replaced. I have the Asterisk server up and running, with 2 sip lines from the local phone service. (Thanks to you guys, it is working great!). I'm pretty sure this is the way the company will move, and

We have a location that is having a really odd issue. We have a sangoma POTs card. We are running software echo cancellation with the card (through asterisk) to try to eliminate some major echoing problems. I've turned on both EC and echotrain, which seemed to have gotten rid of the echo for the most part. However, we are now running into an issue where the outside caller hears a star wars

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

Hello, To start with, thanks a lot for making such a great voice codec available! Having recently upgrading to speex rc1, It occurred to us that there seems to have been a regression in the quality of encoding since version beta1. We are compressing some 22khz wave files in wb mode with maximum quality / complexity in VBR, and the result was really great with speex beta1. With rc1 (or beta3),

I am using Audacity to record some voice prompts. The .wav files I'm producing are of stellar quality. However, once I turn them into .gsm, they sound buzzy and muffled. I know that some of this comes with the territory, but I wonder if there is anyone out there who does this routinely, and who can advise me as to the MO I could use that results in the highest quality in the resulting

The suppressMessages and suppressWarnings functions currently suppress all the message or warnings that are generated by the input expression. The ability to suppress only specific messages or warnings is sometimes useful, particularly for cases like file import where there are lots of things that can go wrong. Suppressing only messages that match a regular expression has rightly been rejected

Hi All, I've a single 1.2.17 Asterisk system. Gradwell here in the UK is used for PSTN calls via IAX2. Our 'net link is a dedicated 2Mb fibre connection (of which we have ever used 50% max bandwidth). We've no E1/T1 links, everything is IP based. My boss complains that many of the calls he holds with others has a bad quality. He also says that its not just him. My iax.conf file

I've just done some rudimentary testing to see how Vorbis degrades at absurdly low bitrates without downsampling. In summary, don't hope for anything decent below -q 0 for now. I tried oggenc -b <bitrate> -M <bitrate> for the below and a few in between: 24k - spectral energy "floor" captured decently, but many pure-tone blips (think old computer movie sound effects)

On 15/12/09 10:37, Blaise Potard wrote: > Having recently upgrading to speex rc1, It occurred to us that there > seems to have been a regression in the quality of encoding since > version beta1. Just curious, did you identify where exactly the regression occurred? > We are compressing some 22khz wave files in wb mode with maximum > quality / complexity in VBR, and the result was

Conditions have classes and the condition system is designed around the idea that classes would be used for this sort of thing. That is already how tryCatch and withCallingHandlers discriminate the conditions to handle. Designing and implementing a condition class hierarchy to support this is indeed the hard/tedious part. Best, luke On Thu, 10 Sep 2015, Richard Cotton wrote: > The

Hi Jean-Marc, You make that sound like its just a matter of meeting some arbitrary spec. Let's be more specific..... If you use narrow band voice down to deep bass frequencies: - 16 bit linear audio sounds good - alaw or ulaw sounds muddy - low bit rate codecs, like speex or G.729, sound awful. I assume QinBin only listened to some uncompressed audio in his evaluation.

Folks! We have purchased G729 and have been testing the codec on mUltiple Gateways. Here is what we have found. Here is the config I have used: ------------------------------- Asterisk Server On Dual Pentium Xeons with 6GB of RAM, running on Fedora Core 2 User1 is in USA on Broadband Cable User2 is in India on 64Kbps ISDN Line User1 using SIPURA SPA 2000 user2 using Xten professsional(X-pro)

Hello Jean-Marc, and thanks a lot for your quick answer! 2009/12/16 Jean-Marc Valin <jean-marc.valin at usherbrooke.ca>: > On 15/12/09 10:37, Blaise Potard wrote: >> >> Having recently upgrading to speex rc1, It occurred to us that there >> seems to have been a regression in the quality of encoding since >> version beta1. > > Just curious, did you identify