similar to: ZyXEL Prestige 2000W - cant make a call?

Hi, I have not been successful in working out how to dial a # within a phone number. EG: exten => _12345,1,Dial(Zap/1/0868563823#,5,t) or exten => _08XXXXXXXX,1,Dial(Zap/1/${EXTEN}#) I'm trying to append a # character so that I can use a cellsocket (mobile phone to pots adapter) connected to an x100p. I think that asterisk is simply ignoring the # character. The docs on

Hi everyone, I decided to have a look at SIP & NAT again and I've been at it for a [quite a] few hours but typically nothing is working for me. Actually I'm not sure if SIP and NAT can ever work but some emails on this list do suggest that someone has got it working, once, maybe. I'm experimenting with a ZyXEL 2000W [WiFi Sip phone] which supports "Outbound Proxy",

Hi everyone, This is off topic and is for GS technical support really but it seems that there are a lot of Budge Tone 100/101/102 users out there. I've got a Budge Tone-100 (101 - without the extra 10base ethernet connetion?) here. I changed the configuration through its web based interface and I clicked the reboot link. But then something went wrong and ever since then it doesn't

Would anyone know why Voicemail in * doesn't get the DTML keypresses from my Cisco 7940 running SIP (POS3-07-3-00) ? Is it something to do with "dtmf_avt_payload: 101" setting in SIPDefault.cnf in the tftp server? Thanks for any help! Derek -- Derek Conniffe Rivertower Ltd DDI: (Local Ireland) 01 201 0146 (International) +353 1 201 0146 Mobile: (Local Ireland) 086 856 3823

Hi everyone, I'd say this question has come up and been answered before but I haven't been able to find it. I have a Cisco 7940 that I've upgraded to SIP firmware (currently P0S-3-06-3-00 - for some reason there was a failure when trying to upgrade to V7 so I left it at V6). The problem I'm having is that when I connect to voicemail the DTMF key presses dont seem to work

Hi Derek, I'm not sure your recording will match with my needs. I wanna be able to do this myself with our currency here. Can you just tell me what to use and how to use it ?? Thanks. Daniel. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Derek Conniffe Sent: lundi 7 f?vrier 2005 11:59 To: Asterisk

Hi Ray, I was wondering if the "qualify" option is used [in sip.conf] to keep a connection (from the SIP phone inside the firewall to the Asterisk server outside the firewall) open then would the firewall not allow two way communication without incoming port mapping/NAT (providing that the SIP phone started "talking" first)? I'm not sure about that - I'm being

Hi everyone, Sorry for forwarding and top-posting this email again but its as if my TDM40b has keeled over yesterday. After a few hours last night and swapping the card to another asterisk server (with exactly the same result) I needed to have the FXS ports working ASAP this morning so I have repaced the functionality of the TDM40b with some Grandstream handytones which I already had in

Hi Everyone, I've been using Asterisk now for a few months for my small office (which is mostly just me while other guys are always on the road so we rely heavily on telephones) - I'm very excited with Asterisk as it can do everything I've ever wanted to do with a PBX. I'm having a problem with an S100U USB --> Telephone interface. I haven't actually made it work yet

Has anyone worked out a way to transfer the Calling Party's Category codes to Asterisk through E1 / T1 connections? I know this is normally available on SS7 interconnects but is it also available to asterisk on the ISDN signalling channels? (I kind of doubt that it is......) Thanks, Derek -- Derek Conniffe Rivertower Ltd DDI: (Local Ireland) 01 201 0146 (International) +353 1 201

Hi all, I have a 7940 running the latest SIP firmware (V7 - thanks Doug Lytle for the tip on the V7 firmware upgrade!). Its almost working perfectly - I can make calls either though my local PSTN or over VOIP but for some reason if I dial my voicemail (which is mapped fine to the VM button on the telephone) it doesn't detect my DTML keypresses so when I press 1 for new messages it just

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

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

If you see a wj0011 version of firmware for Zyxel Prestige 2000W floating around (I found it in a German forum), KEEP AWAY. It completely trashed the wireless networking in my phone. -- ========================================== Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600 Fax: +613 99401650 FWD: 512237

Does anybody have any experience with the ZyXEL Prestige 2000W? I am having problems with the line tear down when I call another extension. If nobody picks up at the other end when I hangup the 2000W, the other extension continues to ring. Is there any way to hangup a SIP call if there is no more traffic? Asterisk seems to think that there is still a connection open. This is pretty annoying

To all: (newbie) I have setup a BT 100 phone and mostly everthing is working pretty good except for the message button. I have place value in the appropiate field in the web configuration but nothing seems to work. When I press the button the speakerphone led goes on but the phone does nothing else (no dialtone, no sip request to *). Does anyone have this buttton working? I would like to

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Anyone have any experience with either of these, I 'd appreciate some feedback? Plus it seems pretty easy to steal a connection with this. Zyxel Prestige 2000W WiSIP thanks, - -- Steve "They that would give up essential liberty for temporary safety deserve neither liberty nor safety." Benjamin

Hi Everyone, I've got a weird problem with both Firefly & iaxLite (both IAX softphones). They don't seem to stop ringing when an incoming call is make to them. If the call is answered the conversation starts both ways but the ringing sound still keeps going and the softphones keep displaying that a call is coming in (but they do not display that the call is answered). I read

Hi all, I've had a strange problem this morning and I know someone who has reported exactly this problem to me too last week: - I've setup a new server running Asterisk 1.2.4. Currently there is no Zaptel hardware install (but there will be soon). This server is behind a NAT router on an DSL line. The remote IAX server on the Internet (which handles the call termination / origin)

Hi everyone, I've just upgraded from Asterisk 1.0.X to 1.2.5 (and the matching latest libpri & zaptel from www.asterisk.org). All compiled fine but now I've got a weird problem with my EuroISDN lines connected to a Digium quad E1 card: - Asterisk suggests its answering calls and playing voice prompts to the incoming call but, in fact, its not really - the caller hears a couple