similar to: X-Lite <-> Snom200

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

Look like SNOM have rolled back the firmware version of the 200's from 1.16w to 1.16q.. Anyone know why? -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

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

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 wonder how IAX compares to SIP bandwidth-wise? I've tried both over overseas IP connection, and somehow SIP seemed to work better. Peter

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

Make sure you are using G.711a, G.711u or GSM codecs.. I have not been able to get iLBC to work and someone the other days couuld not get SPX working.. You will need to enable/disable the codecs in X-Lite.. If you also want to control the codecs that * uses then put the following in the general section of your sip.conf disallow=all allow=alaw allow=ulaw allow=gsm Hope that helps.. > Hi,

Hi, I know this is slightly off topic but I figured the knowlege here is probably the best on the subject.. I want to setup remote offices with 4 to 6 SIP phones (SNOM 200) using ADSL and the internet to connect to the Asterisk box.. These phone will be behind an ADSL router using NAT... I don't want to setup another Asterisk system in each office so IAX is not an option.. I could use

Hi, I have just got my SNOM 200 to start doing some real testing with *.. I am trying to use the GSM codec but the quality is really bad, Is that normal? does anyone actually use GSM?? Also are there any 'gotcha's' that I need to look out for so I don't spend hours trying to get somthing working that really doesn't work anyway.. Thanks.. later.. --

Hi, I have VoiceMailMain on extention 1001 so it would be nice to get that sent to the phone instead of asterisk@[IP Address] when there is a message waiting.. Is it possible to change the extention that is sent to the phone when MWI is lit up?? Thanks.. -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by

I am going to try and use a passive AVM fritz BRI card for my * setup.. Here is the thing.. I need to order my BRI from BT.. The service that looks to be the one to use is what they call ISDN 2e becasue this has the option to setup hunt groups across multiple ISDN2e lines so I could add another line later to get 4 channels.. According to the BT website in order to use the hunt grouping across

*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.

Hi, Does the G.729 module support adding more licences??

Hi, I have been trying to figure out why the S100U is not performing very well on RH9.. Here is my thinking..( may be totally wide of the mark but here goes anyway) I remember reading somwhere that the sound system used by RH has changed... Does the S100U not depend on the sound subsystem?? So what I think is that the sound subsystem in RH9 and the S100U are not happy working together.. Does

Hi, Is Asterisk (or can it be set up as) a SIP proxy? Thanks -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

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

Hi, I installed s fresh system yesterday and it seems that zttool did not install!! ztcfg is there.. Anyone else had this problem or is it just me? -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Hi.. How do you accept user input while waiting or playing moh? My Dialplan is as follows.. ring,ring,.. Hello thanks for calling blah blah... Please enter the extention number blah blah... WaitMusicOnHold(10) If no input pass call to operator.. The problem is that the user has to input the extension while they are being told what to do.. any input during Wait or WaitMusicOnHold is ignored...

Hi, Has anyone worked out a way to use Asterisk in a Hot Desk environment?? I have not been able to think of a way for the user to have control over which IP phone will ring when that users extension is dialed without the user needing to reconfigure the phone.. Something like this would be cool.. User dials *8555 (or similar) and is prompted to enter their extension and then password, after