similar to: Using Zyxel Analog Telephone adapter with a GSM gateway

Hi, I have recently started a job as a system administrator, and as part of my responsibilities I have to look after an asterisk system. Quite impressed with it, but have one or two niggling issues. One of the last things my predecessor here did, was install a VoiceBlue mobile gateway unit, and though it seems to work ok, nearl 20 seconds pass from dialling a number to the call connecting, which

I'm looking at setting up a VoIP GSM gateway to connect to my asterisk box. What experience have people on this list have with GSM gateway hardware. I have been looking at the 2N voiceblue products. Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061028/87014bd9/attachment.htm

Hello All * users. I have been looking for a way to allow GSM termination through Asterisk to occur. I am not a Telekom fundi and I have set up IAX2/IAX/SIP on asterisk with the ZAP channels via the Digium TDM 400P. I am unable to find any place that can tell me the cost of the VoiceBlue with a currency to I can calculate the cost of buying one. Alternativly - or just out of interist - I

Hi All, I am looking for a GSM VoIP gateway for use with Asterisk. I have come across VoiceBlue by 2N but it's price is beyond my reach. Are there any other alternatives out there? I've scanned across the mail achieves for an answer to this without much success, if the question has already been answered kindly point me to the resource. Allan.

Just a warring for everybody regarding this company. I emailed 2n.cz to buy 2 gateways to integrate 3g network and GSM with my asterisk box. They emailed me back to contact the local dealer for them. So I did and I bought the 3G analog gateway (bought from the local dealer) and BLUEVoice gateway(I bought it from a dealer in Serbia when I was there for a meeting) too. Having problems setting them

Because the VoiceBlue is only 4 channels and I am supporting 100 cell users, I needed a way to overflow calls to the PRI of all 4 channels are full. Unfortunately, there seems to be no built-in mechanism to determine if the gateway is full, so this script parses the output of "asterisk -rx sip show channels XXX.XXX.XXX.XXX" to determine the number of channels currently in use. Hope this

Hello, I am looking for a gsm gateway that is SIP based i.e no need of FXO/FXS analogue connection. I searched the email archives and found messages from 2008 but not sure how accurate these are. What do you use and how well it works ? The only sensible one I found is one made by portech and one that is made by Eurodesign. The one from portech is like a trunk while the one from eurodesign

Hi all, Simple question: Is anyone successfully using an ATEUS VoiceBlue SIP <-> GSM box with your Asterisk setup? They look pretty nifty: http://export.2n.cz/index.phtml?l1=products&object_id=138 Thanks! .......jurgen -- thinger@gmail.com is jurgen's gmail address. Check out http://jurgen.ca/ for more yummy goodness.

Hi, Has anyone tried the ZyXEL Prestige 2000W with * ? Is it worth the money ? Best regards Matthias -- _;\_ Matthias Cramer / mc322-ripe System & Network Manager /_. \ Dolphins Network Systems AG Phone +41-1-847'45'45 |/ -\ .) Libernstrasse 24 Fax +41-1-847'45'49 -'^`- \; CH-8112 Otelfingen

I have a zyxel wifi phone and it works just great registering and callin on * but when it comes to menus it just not working (buttons dont change menus) . Anybody else have this problem? Need help. thanks. -------------- next part -------------- A non-text attachment was scrubbed... Name: Jim Van Eeckhoutte.vcf Type: text/x-vcard Size: 136 bytes Desc: not available Url :

I've just seen this post: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg41132.html and it took me back to play again with my dust collecting 2000W. Does anybody got DTMF to work? My sip.conf looks like this: [400] type=friend context=from-sip username=400 secret=verysecret disallow=all allow=g729 dtmfmode=rfc2833 host=dynamic nat=yes qualify=300 canreinvite=no My phone is

(I posted this note on http://www.voip-info.org/wiki-ZyXEL+P2000W+configuration too) I tried to put together comments that were asked for on the P2000W. These configs seem to work fine for a ZyXEL P2000W, thanks to Giles Scott for getting me started with it. DTMF keys work fine and are "read" correctly by Asterisk. It's important that you upgrade your phone to a modern version

Hi Everyone, After a fair amount of faffing ive managed to get the 2000w working with asterisk for IP -> PSTN calls (i.e. get the phone to make and receive calls over our BT line). The final solution is to set up outgoing VoIP calls but I now know that without a SIP aware router I can think again! (damn you iptables!) In the mean time I'm trying to figure out why I can't get the

Hi, I'm trying to get a Zyxel P2000W (reportedly also sold as WiSIP by Pulver) to work with an asterisk box. The phone connects nicely to an external VoIP company (sipgate.de reportedly using asterisk themselves) but there is a strange problem with my asterisk: - Incoming calls via ISDN (chan_capi) to the Zyxel work perfectly. - Outgoing calls (capi/sip/iax) via asterisk are dialling out

Hi! I'm having a ZyXEL P2000W that I'm using together with my Asterisk box (CVS from some week ago). When I get a call directly to the *-box (lars@hostaname.domain.se) I see on the console that Asterisk get the calling users name as CIDName and his SIP-address as CIDNum - but at the P2000W i only get "asterisk" as CLIP? And I can see in the P2000W that "asterisk" is

Hi, Does anyone have suggestions hooking Zyxel Prestige 2002 to Asterisk? I have tested Zyxel Prestige with both supported codecs. Call with G.711 sounds very choppy and cracking. Almost can't understand a word. Today I installed G.729 support into Asterisk but unbearable voice quality remains. It's a little bit better though. I have tested that Zyxel ATA with some commercial SIP

Hi All, Today I got a couple of ZyXEL Prestige 2000W WiFi phones. I'm having a problem making SIP calls although I can receive calls just fine. When I try to make a call the phone makes some sound (like "bup bup bup bup bup bup beep beep") and then I just hear hissing background noise (not too loud - like comfort noise). I upgraded to the latest firmware on the phone - Wj.00.10

Hi Has anybody tried to use the Zyxel Prestige 2000W_v2 with Asterisk? I have the latest firmware and can now make outbound calls. So it looks like Asterisk does accept the configuration in sip.conf. However, I cannot receive any calls. Also, I keep getting this message on the console: chan_sip.c:7733 handle_request: Registration from '<sip:51@10.99.1.133;user=phone>' failed for

Hi, I am having an odd problem with one of our Zyxel P-2000W v2 wireless SIP phones. The phone can no longer receive calls. DND is not turned on and the phone has the exact same configuration as the other 2 phones (they each have unique extensions and such but all other settings are the same.) What can I do to debug this in the CLI. Nothing seems to shed any light on the problem. I am using

Hi, I know we've talked about this phone to death. I have pretty good voice quality, with and without WEP enabled, using the G729a codec and DLink & Netgear access points. I am facing one obstacle that is driving me insane. Does anyone have the call hold feature working? If you do... how did you make it work? The instructions say to press the left button to place the call on hold,