[asterisk-users] 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 =

Re: [asterisk-users] 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

[asterisk-users] Tired of dropouts and garbled phone calls - where to go next?

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

[asterisk-users] l2tp phones - only in China?

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

[asterisk-users] Asterisk consultant needed in Charlottesville, VA

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

[asterisk-users] using ${EXTEN} with waitexten

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

[asterisk-users] Two asterisk servers, two different service providers

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

[asterisk-users] Directory routing to wrong extension if dial tones are pressed too quick.

seconds between each keystroke, everything works ok. Are they any settings that can be adjusted for this? Thanks, Eddie Mikell -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

[asterisk-users] asterisk un-registering from provider

All: Starting switching over my phone lines. I got phone line 1 switched. Everyone working. I switched the second phone line, and it worked about an hour, then I started getting errors from the cli saying the server could not register with the providing. I restarted the system, and it

[asterisk-users] What is the voicemail u option

All: Still trying to get Grandstream to play personal greetings recorded by user - no luck. Someone mentioned the u option. What is that? Something in voicemail.conf? Eddie -- _ -- Bandwidth and Colocation Provided by

[asterisk-users] How to get asterisk to playback personal greetings using grandstream gxp-2000

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

[asterisk-users] Pattern matching - how to ignore numbers after 10 digits

...@ia.ntelos.net) ; local exten = _9XXX,1,DIAL(SIP/${EXTEN:1...@ia.ntelos.net) ; local exten = _9011XXX!,1,DIAL(SIP/${EXTEN:1...@ia.ntelos.net) ; international exten = _911,1,DIAL(SIP/${ext...@ia.ntelos.net) ; emergency Thanks! Eddie Mikell

[asterisk-users] Stress Test new system

and am thinking of adding 6 or so soft-phones to various pc's to make a total of ten outgoing calls at the same time. Any thing else that can be tested before we go live (total of 60 users)? Thanks, Eddie Mikell -- _ -- Bandwidth

[asterisk-users] More clarification on outbound sip channels.

to clarify. I have gotten a couple of users who haven't been able to call out, and wasn't sure if I wasn't rolling over the sip lines properly. Best, Eddie Mikell From: Jim Dickensondicken...@cfmc.com Subject: Re: [asterisk-users] Multiple SIP lines. To: Asterisk Users Mailing List - Non-Commercial

[asterisk-users] Multiple SIP lines.

to define multiple sip lines in either the sip.conf or the extensions.conf? Now when I dial out, I just use exten = _9.,1,DIAL(SIP/${EXTEN:1...@xx.tracfone.net). How does it know which sip channel to use? Hope that is clear. Thanks for all the help. Eddie Mikell

[asterisk-users] Bridging old system (ESI IVX E) with new Asterisk server - it is robbery!

All, Thanks for the suggestions, but the system is a plan non-sip, non-ip, non pri setup. It's pretty much a closed box setup. And the prices for the card and support are robbery - which is why we aren't going to go with another setup like that. While it has been reliable - I don't think

[asterisk-users] Bridging old system (ESI IVX E) with new Asterisk server

system, and Regis is at extensions 155 on the asterisk server, Jake can call Regis and vice versa. I've pondered on this over the week-end, but don't see an easy way to handle this. Thanks! Eddie Mikell Senior Systems Engineer The Rimm-Kaufman Group

[asterisk-users] A matter of Context

Message: 15 Date: Mon, 19 Apr 2010 17:46:46 -0400 From: Ryan Bullockrrb3...@gmail.com Subject: Re: [asterisk-users] A matter of context To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID:

[asterisk-users] I figured it out!!

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

[asterisk-users] A matter of context

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