similar to: Outgoing phone calls "muffled"

"sip show channels" shows some info about active sip channels, the current codec included. What does it say? jg" jg, sip show channels reports the Format as being ulaw for 17 active calls. Holds - no Peer User/ANR Call ID Format Hold Last Message Expiry Peer xxxxxxxxxx kbrown xxxxxxxx (ulaw) No Rx:

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

If you do not put a context in the beginning of the sip.conf file, the default is, ta da, default in extensions.conf. Putting a context=testof idea in sip.conf got things moving: sip.conf [general] port=5060 bindaddr=0.0.0.0 ;10.8.0.34 *context=testofidea* srvlookup=yes disallow=all ;read somewhere you have to disallow everything first allow=ulaw allow=alaw allow=gsm dtmfmode=rfc2833 ;;

Hi, DSCP match in /tcrules/ doesn''t work with CS6 and CS7, it provides an error "invalid value" for string and hexa values. It seems that it comes from /Chain.pm/, in the function /do_dscp/: fatal_error( "Invalid DSCP ($dscp)" ) unless defined $value && $value < 0x2f && ! ( $value & 1 ); I dont understand why "$value < 0x2f", but

I have a new APC Smart-UPS SUA2200RM2U. I've had no success with the manufacturer's PowerChute software and smart signaling, so I have decided to try NUT. The smartups driver can't make contact through the serial port. Since this is rack-mounted, it comes with a 940-1524 serial cable. As far as I can tell, this is supposed to work with smart signaling. Here's my ups.conf:

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,

Greetings, I am porting the openssh-portable 6.8 release to the HP NonStop (NSE) platform. Prior versions were no real problem, with minor tweeks. However, with the inclusion of regress/netcat.c, which depends on arpa/telnet.h, we have an issue. Unfortunately, the platform does not have this file, nor anything like it - telnet is done rather differently. We do have a version of netcat (0.7.1

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

I can use DSMARK to mark on the Egress side. Is there a way to mark/change the DSCP value of an incoming packet on the ingress side? Thanks. Jon Flechsenhaar Boeing WNW Team Network Services (714)-762-1231 202-E7

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