similar to: Setting "User Info" in extensions.conf? (ZyXEL P2000W)

(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, 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 all I have just got a P2000w and experience several problems. Hopefully there is someone out there that has got it working. I saw it on Cebit and the person demonstrating it there told me that it was connected to an Asterisk server on the stand -so it should work. Problem 1: it does not register correctly It get lots of messages like this: Jun 26 19:45:19 NOTICE[1107585968]: chan_sip.c:5630

Hello, Sometimes, when a call comes in from the PSTN through our VoIP gateway, the information that is sent to our web page that logs calls includes the original CID name instead of the one that is we expect to be rewritten on the fly using Asterisk's LookupCIDName: ================= ;extensions.conf [internal] exten => group,1,LookupCIDName exten =>

I am using Sipura 841 phones and Asterisk CVS-v1-0-06/14/05. Whenever a call comes in with blank CIDName or CIDNum the phone reports the respective variable as "asterisk". I can manually set the variables to whatever I want: CIDName (alpha-numeric) & CIDNum (Numeric). But if I try to make them blank, or null, or maybe throw some alpha characters into CIDNum, they get reported

Hello, I found serveral discussions about the Zyxel ePhone Prestige P2000W and the WiSip from Pulver Innovations on this mailings list but still have some questions: 1) are there other affordable wireless SIP Phones on the market? I haven't seen or found anything else till now ... 2) is p2000w and wisip the same hardware?? so could I use firmware from both companies regardless of what

Hello, In case there are other users of the AsteriskWin32 port... I haven't really used the AGI feature of Asterisk to run an application from extensions.conf. *Win32 supports Perl, which I don't know. Apparently, it's also possible to write AGI applications as EXE's (there's a eagi-test.exe file installed by default). => When a call comes in, I'd like an AGI

Hi! I'm trying to set up the possibility for users to call my Asterisk from the net. The Asterisk is behind a Intertex IX66 in which a have set "Static domains" so it forward all calls for my hostname and external IP to the * box. when somenone calls at lars@<external-ip> it all works, but if they call lars@<hostnmame> it wont work. At my Asterisk console I get the

Hello everybody. Is there a way how to setup asterisk to notify caller's phone? Example: I have some numbers and names in asterisk database ( cidname, cidnum), and I want to display the name of person on my phone ( which has no addressbook, but can display chars ) which I am calling to be sure that I have dialed the right number. Thank you for any answer. Andrej

Hi! Then I phone other phones with my Zyxel 2000W (WI-FI) it just hang up when I answer the phone I am ringing. It works fine if I call the 2000W from other phones. I have tried many sip settings. I use this now: [205] type=friend username=205 secret=passwd205 callerid="Zyxel" <205> host=dynamic context=local nat=yes canreinvite=no disallow=all allow=g729

Hello When a call comes in, I'd like to fork a Python script that broadcasts a message so that users see the CID name + number pop up on their computer screen, and simultaneously ring their phones. The following script doesn't work as planned: It waits until the script ends before moving on to the next step, which is Dial(): =========== exten =>

Hello folks! I have a little problem when I'm trying to set up a small VPN between two hosts. The problem is that when I try to sepcify an IP-adress in /etc/tinc/netname/hosts/hostfile using Adress = xxx.xxx.xxx.xxx syslog says; Invalid variable name on line 2 while reading config file /etc/tinc/boppen/hosts/melc (Line 2 is the Adress = xxx.xxx.xxx.xxx line, and melc is the host I'm

Greetings, 2 nights ago I upgraded one of my remote servers to the latest CVS Stable, Asterisk CVS-v1-0-02/07/05-16:13:48, and my inbound and outbound caller ID stopped working. At first I thought it was my carrier, so I've been yelling at them for the past day or so. I then upgraded my core network (about 8 asterisk servers) to the latest stable as of 3am last night, Asterisk

Hello all, I recently updated * to 1.0.x from a CVS version downloaded in July, at the request of BroadVoice. Every since the upgrade, the email notifications that a voicemail has been left only contains CIDName and not CIDNum. Here's an example: "Just wanted to let you know you were just left a 0:06 long message (number 3) in mailbox 1 from PECK JASON , on Tuesday, November 16,

Hi, This problem may be related to a configuration problem but I believe it is a bug in the FOP since restarting the FOP server clears the problem. Here is the scenario: Using AgentCallBackLogin and have four agents logged in a call is made to one of the agents directly from an internal phone. Okay so far. Call is hung up and the same extension is used to call another agent okay again, no

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 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

Has anyone experiences this please: - We were running a number of ZyXEL P662HW-61 routers at our sites and all traffic was being sent over IP-SEC VPN's between devices. When we moved to a new architecture, we got rid of the VPN links so that the SIP traffic was running directly through the routers. Each site uses Snom 360 devices with the latest firmware (v4). Ever since we did this, the

Hello list, We run Asterisk CVS-HEAD-06/02/04-11:25:18 built by root@Gate01 on a i686 running Linux. All works fine except Audio is lost 10minutes into the call. This happens for every call PSTN-SIP, SIP-PSTN, SIP-SIP Example of one call setup using Snom200 and Grandstream 486: -- Executing Macro("SIP/xxxx1251-d638", "CFW|xxxx1251|SIP/xxxx1253") in new stack -- Executing

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 :