similar to: Difference between Snom 190 & Elmeg 290?

Hello, I just bought a Snom-320 from ATAComm. I plugged it into my LAN, registered it with *, etc. All my other SIP gear is Sipura and works fine, both on the LAN and over the Internet. The new Snom seems like it can't process the audio from the handset mic. A steady tone is garbled, even on the LAN. I've contacted ATAComm, Snom and the company representing Snom in the US. So

Most people seem quite positive about Snom phones, I cannot share this opinion. The displays are dying quite often, and firmware is buggy. I have tried every firmware from 4.5 up to 5.x and 6.04 but keep having problems with phones locking up or rebooting during an ongoing conversation. REALLY annoying for a phone that is advertised / targeted as a business class phone

Hi, Is there a simply way to allow dialout from ISDN modem to outside number through Asterisk? I've got an server with an Asterisk and the following cards: 1. TE110 -- to telco 2. TE400P with one FXS to analog phone 3. Two HFC-S based cards in NT mode I'd like to connect ISDN modem to one HFC-S card and allow dialout through TE100 to some external number (being more precisely, to some

Hi All! I am new to asterisk and have a simple question: I was able to install and configure it as I wanted. But when I try to configure a default extension ('s' extension) for incoming sip calls it doesn't work. I just want that when someone calls my ip it get's connected to some default extension. The reason I ask is that I plan to replace a conventional pbx with asterisk.

Hi, Snom phones firmware 5.0 is now out. Try it if you like: http://www.snom.com/wiki/index.php/Main_Page. Regards, --------------------------------------------------------------------- Usman Tahir snom technology AG www.snom.com --------------------------------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL:

Would it be possible to use the programmable led+keys on the Snom phones to signal that there is an incoming call that is ringing a call group or pickup group? We use this on our existing PBX if for example the accounting dept. is out for lunch but nobody can hear their phones. This way you can see an incoming call (and we hate voicemail) :) Thanks!

Hi! I'm trying to setup a Wildcard TE110P with a PRI in The Netherlands. I get a Red Alarm on the line. Is there any way of debugging this? I've tried some configs that should work but without success. Is there any way of telling if the cabling is correct or what else the problem could be? Thanks!! Remco

I installed bristuff0.20-RC3 (which includes * 1.0.3 stable) It works fine until I disconnect the phone jack for the ISDN line. Even after plug it back in asterisk still reports that it could not create a zap channel when I try to dial out and the line gives an engaged tone when I try to dial. Re-starting asterisk doesn't solve this, I have to stop asterisk, unload the modules, reload

I have multiple locations running * where all the phone are on their own lan and all the data is on a separate lan. The problem is they are sharing the same dsl connection. The locations are IAX2 trunked together, but it only takes one data down/up load to just kill the voice. What I am looking for is a small switch with QoS that I can stick in ahead of the dsl modem. Plug in one connection from

Hi list! I'm sure the topic has been discussed but I could not find what I was looking for. What would be the best / easiest VPN software solution. I would like to install vpn software on the * server for roadwarriors to connect to with laptops running windows. Ideally the vpn solution will not require any additional software on the client side but will use IPSEC. (Ofcourse call quality

Hi list! Sorry for this slightly off-topic message but does anybody know if the standard for ISDN BRI is the same in Spain as it is in the rest of Europe (or the Netherlands). Will a standard HFC-S card work?

Hi list! I have some strange problems with Asterisk 1.0.5-BRIstuffed-0.2.0-RC5. Very regularly asterisk seems to lose connectivity with the ISDN line. If you try to call in you get the information tone that the number is not in use. Outbound calls do stil work however. Unloading the modules and reloading them and start/stop asterisk will solve the problem. Another problem that occurs

Is it possible to rewrite caller id's? I would like to have sip phones appear by their local cid (like Henk <208>) but when they call out using the PRI I would like their full DID (MSN) to appear (like 0031201234567) I could ofcourse set callerid to the main phonenumber but surely there must be a better solution? Thanks!! Remco

Hello, I have Elmeg IP290 phone and have problems with VM. I don't know how to configure this IP phone, that it could call to *97@192.168.0.1 if I pressed "VMail" button. Now if I press buttom "VMail" , ip phone dials: sip:asterisk@192.168.0.1 (192.168.0.1 - Asterisk IP). So I don't understand , from where it takes "asterisk", cause I have never write

Hi list! I have some sip phones and Sipura ATA 2000's. However after dialling a number I need to dial a # to control a device. When I dial # Asterisk kicks in and puts the call on hold. How can I change this? Thx!! Remco

I tried to get callerid working the normal way but the cid is never passed to the phone. It doesn't work untill I set SetCIDNum(0${PRI_NETWORK_CID}) in extensions.conf which I found in the wiki: http://www.voip-info.org/tiki-print.php?page=Asterisk+zaphfc Is this intended behaviour, or still a bug? It does work but it only shows one zero even though I have nationalprefix = 0

Hi list! I'm using Asterisk 1.2.7.1. with FreePBX 2.0.1 on a CentOS 3.7 box. On the * box I also have a samba share where our CRM app can dump call files and a cron script is moving the call files every second to the asterisk directory. Everything goes really quickly, the call file is placed on the samba share and very quickly moved to the asterisk dir, so far so good. But then the call

Hi list! I'm currently using a HFC-S card for my ISDN BRI line with bristuff. The instability is driving me crazy however. I'm having continuous problems where inbound calls will not work after some time of operation (the number then appears as not in use to the caller) or also outbound calls do not work. The solution is to unload the modules, stop asterisk, re-load the modules and

Hi list! I am installing an * box that will be installed on a site with KPN BRI ISDN in The Netherlands. I am using bristuff fron Junghanns. Does anybody know the correct settings for this? I will not have internet access there which makes it harder to google around on location. switchtype = euroisdn is pretty obvious but what about these settings: signalling = bri_cpe_ptmp ; p2p TE mode

Hi list! I'm still trying to figure out about the groups in asterisk. If I understand correctly, if you assign a certain group number and you assign the same call group number to a sip device the device will reing even though you did not specifically specify it in extension.conf? How will this work for ISDN BRI/PRI? I don't want some extensions to get all calls from the BRI/PRI, just