similar to: Tired of dropouts and garbled phone calls - where to go next?

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 --

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,

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

"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:

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, 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>

> In my case, I have good incoming quality and terrible quality going out. > That is, I can hear people perfectly well but they complain that my > voice drops out and is garbled regardless of who places the call. This suggests to me that you may have congestion problems in your "upstream" traffic flow. Setting QoS on the packets may not help, if whatever router you are using

All: Getting ready to put the system in production. Any suggestions on "stress testing" the system? I'd like to initiate say 10 sip phone calls to make sure the provider has the bandwidth. Can you do that in CLI? I've called 4 numbers simultaneously with the hard phones I currently have and am thinking of adding 6 or so soft-phones to various pc's to make a total of

Jim, and all: Thanks for the response. If I can repeat what you are saying: you don't have to define the multiple lines in sip.conf? For comparison, with my current esi setup, we have 10 outgoing lines. If one line is busy, then the service just rolls to the next number. This is set up with the phone service. That doesn't have to done with outgoing sip lines? Does the dialstatus

All: Some of the people who dial into to our system will press the pound key when entering an extension for the directory key. When waitexten gets that, I get an error messages as, for example 123# doesn't match any extension. I was going to use ${EXTEN} to just use the first three numbers, but I'm not sure how to use this with WaitExten. so I have exten =>

All: Still experimenting with the asterisk server for the company. My local phone company has given me two sip numbers to experiment with, say 444-456-1234 & 444-456-5678 Calling in and out works, and I've spread a couple of the phones out with some co-workers. My question is this: Do I have to define multiple sip lines in either the sip.conf or the extensions.conf? Now when I

All: I am looking to install another asterisk server in an office located in a different part of the country. I think I can configure the sip and extension conf files, so that the internal phones at the two locations can call each other. My question is this, how do I properly configure the sip file for a different provider at the new location? Can I use a different register statement for

Hi all, We have recently implemented an Asterisk system using Trixbox (asterisk v1.4.4 at the moment, yet to move to 1.4.9) but are getting pressure to switch back to our old key system unless we fix two major issues. So please help me avoid switching back! An overview: We have about 12 Linksys SPA941 SIP phones connected on a private switched network to our asterisk box which is a

All: I am using the standard voicemail in asterisk. Everything works well, except, if a users wants to record their own personal greeting, it doesn't playback. I can see the soundfile being created. I suspect it is a setting in the voicemail.conf, or an option I am over-looking on the grandstream, but if anyone can point me in the write direction, I would certainly appreciate the help.

Howdy all... When making SIP calls through my X100P from X-Lite to the PSTN I'm getting 3-5 second dropouts in both directions. I've tried ulaw and GSM, but that doesn't seem to make a difference, and the * box is on my local net. Here's my hardware: Celeron 2.4GHz, 512MB, Slackware 9.1, 2 X100P, 1 T100P. Any ideas what could be happening, or pointers as to how to shoot this

First of all, *thanks* to all the developers for their hard work on Ogg Vorbis. It's greatly appreciated. I'm having a problem with encoding in Debian GNU/Linux 3.0. The oggs contain occasional dropouts and "blips" here and there. Example files: Original (~7 MB): http://personal.inet.fi/musiikki/nebularia/test/t2title.flac Ogg Vorbis file (~800 KB):

All: I've starting building an asterisk system for our company, which has about 60 users. I am new to asterisk, so thank you for your patience. I've stripped the sip.conf and the extensions.conf down to the bare minimum: Here is my extensions.conf file [globals] [general] autofallthrough=no [default] [fromprovider] exten => YYYYYYYYYY,1,Dial(SIP/151,20) [phones] exten =>

I have time-series data from a pair of inexpensive self-logging 3-axis accelerometers (http://www.gcdataconcepts.com/xlr8r-1.html). Since I'm not sure of the vibration/shock spectrum I'm measuring, for my initial sensor characterization run the units were mounted together with the sample rate set to the maximum of 640 samples/sec. Unfortunately, at this sample rate there are significant

I'm trying to use RingCentral with wine. The ui displays, but I get an dialog box error message because of an msxml (ie) dependency, and the console produces the text listed below. I run winetricks fakeie6 and try to run, but it still complains... Should I install IE? What version of IE should I install? Thanks, -Jmt Jmt:~/.wine/drive_c/Program Files/RingCentral/RingCentral Call