similar to: no ring in ear

Hello, I am implementing a small call center with 1 to 4 agents. For some reason I don't understand, if an agent is busy, and it is his/her turn in the queue round, asterisk gives an "all destinations are busy" message and hangs up the call. Agents are SIP lines registered with an audiocodes MP108FXS which registers each line independently. Ringing strategy is RoundRobin (most of

Anyone can share me with Music Onhold Configuration sample? Thanks in advance for your help, Kang

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

Alright, this may seem like something relatively easy to do but I must be missing something or had a neuron misfire. I am trying to get The Status lights on my Snom200 hardphones to display the status of each one of my PSTN lines in my Asterisk server. Current Config: 3 X100P cards Asterisk CVS-02/25/04-18:06:52 5 Snom200 phones I am currently using the following macro to dial out

Hi all, -------- I have a Asterisk at a public Network (official IP address). In the local network I have isntalled a Snom 200 IP phone and in my home network (behind NAT) a Snom 100 device. I can dial the Snom200 device from my home location without any problems but the Snom200 can not dial me. It always gets a "we do not rely". I tried to forward the SIP Port (5060) UDP via UPnP

Hello list, i am using: asterisk CVS-10/13/03-11:54:33 chan_capi-0.3.0 ATA-186 V2.16.1.ms over MGCP Situation: ISDN calls ATA ISDN speaks with ATA ATA-Phone presses Flash and speaks to another one (SIP/snom200) ATA-Phone hangs up ISDN talks to SIP/snom200 snom200 hangs up The incoming extension of ATA keeps busy for a time (20 sec?), even its not off-hook anymore! Any ideas? -- Swapping

Hi, in the Snom FAQ I found the following information: After staring up, the phone tries the URL given in the "Setting URL" of the phone. ... BTW this setting can also be set via DHCP. .... option tftp-server-name "http://192.168.0.9/snom200{mac}.htm" The documents used: FAQ-04-06-14-sf.pdf "Setting up DHCP for snom phones" FAQ-04-03-24-sf.pdf "How can I

Hello. Has someone been able to make work faxes over sip, i have one mp108 fxo and one mp108 fxs, my setup is : telco analog line -----> mp108fxo -----> Asterisk ------> mp108fxs -------> fax machine 1) Asterisk detects the tone from the sending fax ( i am receiving ) but looks for extension 'ff' not 'fax', ( at least that's what * complaint, invalid

Hello, I'd like to be able to see if a channel is use and handle the call differently if it is. The best I can find is the command ChanIsAvail(). The problem is, I have an snom200 phone which does call waiting, so even if it is engaged in a call, a second channel is still available on it. I would like to be able to differentiate between these two cases: no calls engages, or calls

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040106/3efdeb9f/attachment.htm -------------- next part -------------- Hello Though my SNOM 200 phone receive a voice mail but it doesn't show MWI on the LCD panel, Instead it keeps displaying "DND SW-REG,Call-log". Though I can access my voicemail using exten

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

Sorry I forgot the subject in the last post. When my snom200 receives an inbound SIP external sip call, it somehow rejects the call and with a busy tone. The debug shows the following error: channel.c:1142 ast_read: Exception flag set on 'SIP/sipphone-7796', but no exception handler what does this mean and how can I debug it further?? Thanks JC -------------- next

Hello, After receiving and finally being able to configure my new Polycom IP 600 phone Here are my initial experiences: GOOD STUFF: - The sound is wonderful from both handset and speaker (711U) - The large multi-shade LCD screen is beautiful (for a phone) - The phone has an integrated Ethernet switch so you can connect a PC to it as a passthru - The AC adapter was included (small thing, but

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

On recent releases of the snom200 firmware, the MWI indicator will turn on, but won't turn off when the message has been checked. It works on firmware 2.03o, but not in 2.04g or newer. I filed a bug report with snom, but they're claiming it is an asterisk issue and that it should have been resolved. They suggested that I ask on the list. "Anyway, Asterisk had a bug where it

Hi, Have posted to this list a couple of times and have always received great responses and help. I have a basic * system setup Using 3 X100P cards with 6 Snom200 IP phones. It was a bit of a struggle getting everything up and running but have been pretty happy with the flexibility and ease of *. My major problem is one that has been discussed on this list many times before. The echo

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:

This is troubling. Shouldn't your hubs/routers autosense the 10MBPS? ---------- Original Message ---------------------------------- From: WipeOut <wipe_out@lycos.co.uk> Reply-To: asterisk-users@lists.digium.com Date: Wed, 15 Oct 2003 07:53:13 +0100 >Steven J. Sobol wrote: > >>On Wed, 15 Oct 2003, Jon Pounder wrote: >> >> >>Nothing works. Call transfer

Has anyone had any experience with wireless LANs and Asterisk? We have and here are my impressions. We configured an Asterisk in the office as a precaution to see how it would work for our own retail customers. Our office is open space, about 800 sq ft. (20x40 area). We use Snom200 and Grandstream SIP phones. Using the latest Linksys wireless access point (WAP54g) and 3 wireless bridges