similar to: Snom 200 stuck on "Busy"

As a follow up to my earlier posting, the problem with the Snom 200 Busy signal was the firmware! I reverted back to 1.16x and everything's OK. That made today pretty complicated, since I already had a new kernel and a new Asterisk build I was trying all at once.... Hopefully someone else will benefit... - Matt

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031221/41e5bcd1/attachment.htm -------------- next part -------------- Hello All, Whenever I try calling SNOM 200, I am getting "Everyone is Busy at this time" message though line is free. I tried with firmware image of 2.02t and 2.03b (this I received from SNOM customer support) but

I have 2 Snom 200 and would like to get them to work properly with Asterisk. With the Firmware 2.02t I am able to use the phone. But only one line configured. With there newer firmware 2.03o it will not allow me to make calls. But I can get calls on the unit. Again the 2nd line is not able to be registered. Is this an issue with Asterisk or Snom? I could use some example configuration

Hi all! I updated my snom200 to 2.02t and now MOH from * don?t works anymore... only the MOH from snom server and if i clear the MOH server field in the phone i have no MOH at all..( with the transfer button, moh plays using a extension). Someone with that problem? I downgrade to 2.01s but nothing changes. Miklos -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am using the snom 200 with Phone type snom200-SIP Version snom200-SIP 2.04g Bootloader URL http://www.snom.com/download/snom200-boot1.9.bin Firmware URL http://www.snom.com/download/share/snom200-2.04o-SIP.bin I am using asterisk stable tree. I had to disable "Challenge Response on Phone" on my snom; I could not get it to work with

Check the Redirection on the web interface of your Snom 200. If it says "When Busy" that's your problem. It should say "Never". Also make sure on Sip->Lines your line appearance says "All" or else you will have the same problem. Hope this helps :) Greg Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original

Hi All I have a snom 200 phone here which works perfectly when using the handset to playback the voicemail messages etc. However when I play back the voice using the speakerphone it sounds choppy. Anyone had this problem before? Regards Mark

Hi, I'm trying to set up my SNOM 200 with extensions, with different ringtones - but it doesn't seem to work. I've defined two extensions for it in Asterisk and in the SNOM 200 configuration. In the SNOM home>settings>SIP>Lines config page, I have set the ringer for the first extension to "ringer1", and "ringer6" for the second. In

I cannot get the transfer button to work on a Snom 190, I cannot get the # to work either. Any ideas? Regards Thorben -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041207/5ac06747/attachment.htm

Hi We are about to buy several Snom phones. Does anyone have warnings or advices against these phones ? Our finalists were Cisco, Polycom and Snom. We will be using only the SIP protocol. Thanks Patrick

Please forgive me if this is a silly question. I've been following this thread in the hope that I could put my * server and snom 200 into full-time service very soon. I need to find out how to have the lines configured so that it does not return a busy reply when only one call instances is engaged. Am I supposed to create multiple extensions on my asterisk dialplan to reflect the 5 call

Ok, not exactly an Asterisk problem, but... I picked up some SNOM 200 phones because SNOM's have been recommended for use with Asterisk and they have line buttons that can subscribe to presence. However, they don't appear to power up when connected to my Negear FS108P, which is an 802.3af Power-over-ethernet capable hub. I am pretty sure these are the SNOM 200b, in that the ethernet

Hi, I've looked through the archives and seen references to placing calls on hold on a snom 200 (any version of the firmware but we have the latest: 2.05e.) Basically, we can't place calls on hold on the snom 200! The manual talks about the Flash button (which is really the "R" button, as far as I can tell.) Pressing the R button will immediately disconnect the incoming call.

We're using firmware 7.3.30 on an installation of Snom 300 phones. Should we stick with it, or do the newer firmwares have better support for Asterisk? Thanks John

Christian, Just a quick heads up warning that something seems to be amiss on the Snom 200 running v2.03h code. The Message Waiting Indicator is blinking (due to a voicemail left in Asterisk) as its suppose to. Some time later the LED stops blinking (as though no voicemail); then later it starts blinking again. I left a voicemail to x3008 (button 2) this morning at 8:48am, and purposefully have

I'm using the SNOM 200 latest SIP beta (so that I can have the GSM codec, etc). Everything seems to be working fine, but the music on hold doesn't play when I use the HOLD button on the snom. Any suggestions? Thanks, --Ernest

Snom 200 has be set up with extended key pad. The product literature also mention multiple sip registration. How many registration can it handle? It does not seem to appear in the user manual. David Kwok

Hi list, how does one get a SNOM 200 MWI to work with * ?? When I press the MWI button it doesn't connect with voice mail on my * box. thanks

Hi folks, in order to establish backward compatibility we made an image that automatically detects if the other side does not support RFC3264. Please try it out, we would be very interested if this image is a progress! http://snom.com/download/share/snom200-2.03b-SIP.bin Thanks, CS

I decided to give the whole family IP phones for christmas, all hooked into my asterisk server, so all the nephews can have their own lines. However, one of the phones I got was the SNOM 200. That's worked fine for me on my own network, but I'm having bad luck getting it to work behind NAT talking to Asterisk. It talks to my termination/origination provider, which seems to ruthlessly