similar to: SNOM200 firmware roll back!!

Hi, I have just been playing with the latest X-Lite.. It works fine with Asterisk.. As for codecs I tested G.711a/u, GSM and iLBC... iLBC is the only one that didn't work.. not sure why.. But the bigger problem is that when I call another extension that is using a Snom200 the call connects but there is no audio in either direction.. I have tried G.711a/u and GSM and while X-Lite shows that

incominglimit is already implemented for SIP. Just specify under the endpoint how many incoming connections are allowed. For example, [cisco] type=friend username=cisco secret=blah nat=yes ; This phone may be natted host=dynamic canreinvite=no ; Cisco poops on reinvite sometimes qualify=200 ; Qualify peer is no more than 200ms away

My system seems to be generating a call on its own... Unfortuately I can't give much more information.. I have an X100P and an S100U.. My Modem and the X100P share a common line.. When I am on the internet (which is most of the day) * just sits there and does nothing (apart from when I am testing ideas for the dial plan), but at night when I am sleeping and the modem is not connected then

Does anyone know if this unit works with Asterisk? Thx. B.

*This message was transferred with a trial version of CommuniGate(tm) Pro* Hey all, I've just done a quick (but functional) web front end for searching the CDRs in a MySQL database. Anyone interested in trying it out? I'm wondering what to add to it next. So far you can seach using source, destination, CLI, channel and date ranges. It also displays ALL fields in the database table.

Hey all, ? I'm trying to get an E400P and 2 X100Ps working together in the one box and don't seem to be having much luck. ? I can get the two different types of boards working separately, but not together.? I've made calls on both the X100P and have seen sync and correct signalling on the E400P.? But when I try to enable to configs together I get the following: ? modprobe wcfxo

Hello, I'm new user of asterisk. Can anybody pls tell me how to use asterisk or any detail how to link???? i installed Asterisk-0.4.0 on i810 onboard sound card with Redhat 7.1. when i type "asterisk -vvvc" i get *CLI> prompt Prakash Get Your Private, Free E-mail from Indiatimes at http://email.indiatimes.com Buy The Best In BOOKS at

Hi Anybody have a example or use this modem as FXO, acording to intel this modem is a real full duplex modem, so i have one question why i can we use this modem as fxo My idea is simple when you recive a call in the modem you can transfer this call to any other place using for example a IAX2 or Openh323 protocol This is posible? Thanks

hi, i have a external usrobotics modem, i want to use it with asterisk to interact with the pstn, what i have to do? thanks, -- santiago jos? ruano rinc?n administraci?n servidores y servicios de internet red de datos universidad del cauca -----BEGIN PGP MESSAGE----- Version: GnuPG v1.0.6 (GNU/Linux) Comment: For info see http://www.gnupg.org

On Sun, Aug 17, 2003 at 03:44:21AM -0500, Gnophone Support wrote: > Hello, and thank you for registering at gnophone.com. Your login > information is listed below: > > Username: miernik > Password: ******* > IAX Phone Number: 17002916107 > > Please login as soon as possible to > http://x.linux-support.net/directory/ to complete the

Hi all, I'm setting up my first * install and have it peering with another * machine using IAX across the internet which provides our pstn gateway. So far I have the IAX "friend" set up correctly but when I make a test call from an external phone, I get: WARNING[5126]: File chan_iax.c, Line 648 (get_timelen): Don't know how to calculate timelen on 8 packets I have set up a

Can anyone provide me with a step by step on how to set up Windows Messenger on a Windows XP Pro box as a SIP client with asterisk? I'm interested in doing various tests of my asterisk server from the Windows perspective of the world. In the alternative if someone could provide information on another Windows based fully functional easy to configure iax or SIP client that would suffice as

Hi, I have more or less decided to do with the SNOM phones for the next stage of testing with Asterisk becasue they seem to be the best value for money and have support for the GSM codec and easy upgrades.. But now I have to decied wheather to get the 100 or 200 and if its the 200 then I need to have some justification for the extra cost.. Can someone who knows these 2 phones tell me what the

I live in Tennessee, USA, and work 1000km away in Texas. Thanks to the wonders of broadband I never leave home (well, not for WORK, that is. :) I'm setting up an Asterisk system whereby I'll have an extension in Texas, so clients can reach me at a local telephone number. We have a VPN set up already (OpenVPN, which I highly recommend to anyone needing such a thing.) It does

where can I get the 2.05 firmware all i see is the 2.04 firmwares :-) also anyone got a fix for the horrible speaker phone on the 200's

Hey all, I was playing around with IAXTEL last nite and have outgoing calls working a treat. I'm sure I woke a few people up in the US with my annoying test calls. :) Anywayz, incoming calls are a different matter. I have a NAT firewall my * box is sitting behind and the server 'appears' to have registered correctly with IAXTEL. Thing is, when I try and call my 1700 number

Hi All, When the Caller hangup at the voice menu, the wildcard X100P didn't disconnect the calls properly and it just keep looping at the voice menu and timeout and loop again, are there any methods can fix the problems? Please help! Thanks, Randal

> > No, this is not the case currently with any of the Cisco SIP software > loads that I am aware of. If you find this to be incorrect, please > let the list know. Cisco has not deployed much of the featureset in > their SCCP phones (such as paging/intercom) into the SIP phones due > to lack of standards/interest/political capital. > > JT Ok, after further

I have just started to play with callgroups and pickupgroups.. I updates my * from CVS this morning (about 15 mins ago).. I have placed callgroup=1 and pickupgroup=1 into each of my 3 phone configurations in sip.conf.. I place a call from phoneA to phoneB, then I go to phoneC and dial *8# , the call does not get picked up by phoneC and continues to ring on phoneB.. Have I not configured

A while ago SIP transfer via the # key on a call to a cell phone via iconnect was working. I updated to the current CVS tonight and now that functionality is gone. Any ideas as to how to enable it again? Thanks in advance -russ