similar to: Snom 200 phones not working.

I am trying to get the following phones for testing. Is there a distributor in the US that is able to sell me these Sip phone and ATA adapters? I can not afford the Cisco phones there too hard to configure and too expensive! 1 - Sipura SPA-2000 2 - Grandstream Sip phone BT-102 1 - Grandstream HT-286 1 - Snom 105 Sip phone. I have called and emailed chagres but they have not reply. Nor

Ok here is another problem I have run into. I have a Sipura 2000 and I have been able to configure line 1 with only one small problem. But I can't get the line 2 working with asterisk. Here are samples of my sip.conf and extensions.conf. If I disable line 1 I can then get line 2 working. Is there a sample configuration for the Sipura to get both ports working with Asterisk. Sip.conf

Hello I am still having problems with IAXTELL and FWD configuration. I get the following when I dial 17009965342 which is set as an example to dial to FWD people. 1+700+99+ 5 digit number. I have placed XXXXX where my passwords are. CLI> Executing Dial("Zap/14-1", "IAX/abatista:xxxxxx@iaxtel.com/917009965342@iaxtel") in new stack -- Calling using options

I just got 2 IpDialog phones for use with my Asterisk system. I have been able to get the phones to just dial local extensions but it is not able to register with my system correctly. I would like to know if someone has set these phones up before and how they did it! Is there any examples for use with Asterisk? They seem simple enough to config with there web interface. Thanks

I have about 5 snom 200 phones working fine with everything. Voicemail, Transfers and all. Except I can't seem to use them to pickup parked calls nor place a call on park. I also have sipura-2000 with analog phones that are able to pickup parked calls and to park them. Most of them are on firmware 2.04g I have upgraded one to 2.05c for testing but this did not fix the problem. I get no error

I would like to prevent * from dialing 900 and 976 numbers. I setup the following settings in extensions.conf. But this does not seem to work! I don't know what I am doing wrong please help! exten => 1900XXXXXXX,1,Congestion exten => XXX976XXXX,1,Congestion exten => XXX976XXXX,1,Congestion exten => 1XXX976XXXX,1,Congestion exten => 91900XXXXXXX,1,Congestion exten =>

I would like to know if anyone has come up with a script for 911 dialing rules that put correct information on our locations. We have our office in 3 different building one being our production & shipping dock. It is almost 2 blocks away. We are connected with Ethernet Wireless between the buildings and have Sip phones setup in the other 2 locations. All the phones are working just fine.

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

I have a problem that sometimes lines will go into what I call never never land. The Asterisk system will put a line with <Zombi> on it when you type show channels it will make the analog phone line dead. And on the CLI it says: astsvr*CLI>Zap/1-2<ZOMBIE>(macro-twoline-exten,s,1)Up Dial Zap/1-2|20|r I have tried to release it with soft hangup Zap/1 & also soft hangup

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 a new user question. Sorry I know most of you are Linux experts I am not! I am just getting my feet wet with this. And I am sorry to ask this stupid question. I was following an installation post from Wiki that said when using RH 9 you need to make sure that you have the following installed first and you should check them with the following command. Are there any other items I need to

With a recent update to Asterisk CVS, and versions 2.02r and t of the Snom 200 firmware, I'm getting the Snom phones stuck reporting "Busy.": -- Got SIP response 486 "Busy Here" back from 10.12.34.248 -- SIP/3064-b07d is busy They're on-hook, not doing anything. They are registered fine. I can pick them up and call out over Zap or IAX fine. I just

I tried to connect my * to IAXtel, but i always get this errors. chan_iax2.c:5849 socket_read: Registration of 'mnetwork' rejected: Registration Refused On dial a iax number i get: chan_iax2.c:5526 socket_read: Call rejected by 69.73.19.178: No authority found chan_iax2.c:5528 socket_read: Immediately destroying 3, having received reject chan_iax2.c:2411 iax2_hangup: We're hanging

Dear List, I'm having a blinking MWI light on the snom 320 even when there's no message waiting in Asterisk. We've managed to make the voicemail button work using fromdomain=192.168.0.1 in sip.conf vmexten=2500 (our VoicemailMain application extension in extensions.conf). We also added notifymimetype=application/simple-message-summary also in sip.conf to allow SIP simple MWI

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:

Hi All, I have been busy with this problem for a while now, but I can't find any solution. First I thought this was a problem with the phones, but all my phones have this problem. (2 SNOM 200, 2 GS 102, 2 Cisco 7960). I tried all firmware versions I could find for the phones. First, my situation: - No NAT, No Firewall, same subnet - Codec configuration: In general: disallow=all

I feel this is in error some place. If I call a sip device that is not registered or not connected at the time. Asterisk will send that call to voicemail to busy not unavailable. Is there a way to correct this? Ariel Batista Kasi International - Computer Networking Ph: 305-574-6721 Fx: 305-574-0212 -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, Great job on the Asterisk@home project. Looks great this new version is really nicer looking. But I have a few questions. 1) For the new web access http://localIP/maint how and where do I change the password. 2) Since I don't use the Amp section for setup the .conf files I use my own. How do I get the asterisk server running status up. I have it running and works but shows up as not

I have a question on SIP devices that are setup and working but you change the login name and contents to them why does asterisk need to be shut down and restarted for them to work? I have reloaded extensions and done a reload command. But the updated sip phones do not work until I shut down and restart asterisk. Is there any other way to update them without restarting the system? Since the

I have a Cisco 7960 & IpDialogs that I am not able to use the transfer button on it. What happens is that it puts the call on hold and then it gives you a dial tone. You can dial but it will not transfer the call. What we are trying to do is transfer to extension 700 for parking so another person can pick up the line. We can not use the # key to do this due to we have several IVR's