similar to: Exception flag set - snom200

Hi Steve, I am having this problem in which RxFax is still holding the file after receiving a complete fax. Somehow the zap channel is still active but on the fax client it was sent successfully. If you call the line it is still busy. Changed from phase 3 to 4 >>> MCF: 8c HDLC underflow in state 8 Changed from phase 4 to 3 Slow carrier up <<< DCN: fb DCN with final frame tag

Hi, We try ti hide the caller id at calls trought E1 in EuroISDN (Spain) using restrictcid=yes and doesn?t work. What can I do, thaks Pedro -----Mensaje original----- De: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]En nombre de asterisk-users-request@lists.digium.com Enviado el: mi?rcoles, 31 de marzo de 2004 12:00 Para: asterisk-users@lists.digium.com

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

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

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

Try this one. Took me a while too. http://www.snom.com/download/share/snom200-2.05c-SIP.bin > -----Original Message----- > From: M3 Freak [mailto:m3freak@rogers.com] > Sent: Monday, May 17, 2004 11:42 AM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Snom200 Firmware: I only see 2.04g > > Hello all, > > I've noticed several messages about the

Hello. I have two snom200 ip phones and 1 mp108fxs (audiocodes 8 fxs) and i dont get a ring in the caller phone when I dial from a snom200 to the other snom200 or the mp108fxs, I made a debug with ethereal, and I can see a "Ringing" packet being return from the called snom200 or mp108fxs to the asterisk box, but it is not being re-transmitted to the caller snom200. Altough

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 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

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

Hello all, I've noticed several messages about the latest firmware on Snom's site, 2.05b, and today I see that another update is listed, 2.05c. However, when I go to the download page (http://www.snom.com/support_dl_en.php), the latest firmware version available for the Snom200 is 2.04g. Are the newest firmware releases not yet available, or am I doing something stupid? Thanks, Kanwar

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

This is a serious problem! I have brought up this issue in four previous attempts to get some feedback. I find it hard to believe that no one else is having this same problem. Apr 11 13:27:36 NOTICE[4446]: channel.c:1906 ast_read: Dropping incompatible voice frame on Local/103@sip-00f3,2 of format alaw since our native format has changed to slin Apr 11 13:27:36 NOTICE[4446]: channel.c:1906

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

Hi everybody i have a problem trying to connect an incomming phone call from pstn to my (soft phone) iaxcomm, the phone rings but when i try to answer the call, asterisk sends a message like this. Jun 1 19:33:17 NOTICE[5013528]: channel.c:1223 ast_read: Dropping incompatible voice frame on IAX2[192.168.222.99:4569]/16 of format GSM since our native format has changed to ALAW i'm working

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

hi,all i have just set up asterisk 1.6.0.7 rc1 with spandsp 0.0.5 pre4 to ReceiveFAX, link to a E1 (DE410P) using dahdi this can receive the fax from E1 successfully, but i see many error message in the log like this: [Mar 16 09:24:38] ERROR[23540] channel.c: ast_read() called with no recorded file descriptor. when i receive a 5 pages fax, i will see this error message over 200 lines..... it

All, Im having a problem with ReceiveFax where its generating a ton of these messages the entire time the receivefax app is running receiving my fax. [Apr 7 22:16:06] ERROR[26918]: channel.c:2520 __ast_read: ast_read() called with no recorded file descriptor. Im running on Centos 5.2 with all patches. asterisk-1.6.0.9 asterisk-addons-1.6.0.1 dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2

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