similar to: Help with Welltech 2FXO gateway, GS BT100 and Asterisk

I've been using Asterisk for a few weeks now. I have a (1) BT100 phone and a Sipura-2000 for all my analog phones. All has worked rather flawlessly, until today. I was on the BT100 phone today. During my phone conversation, the BT100 disconnected and went into a "click" mode. 2 "clicks" per second I think. Asterisk was fine, I picked up one of the analog phones,

I'm seeing all sorts of problems and it's probably more of my lack of experience than anything else. I have a BT100 running 1.0.6.7 code. When I go to the status page it says it's not registered (hmm, that's not good). I also can't get dial tone but I can dial! I'm afraid I'm lost any good pointers? I've done a sip debug and all I'm seeing for the BT100 -

Hi all I'm fairly new to Asterisk, so be nice :-) I was wondering if anyone has been able to get the Welltech K1000A USB phone working on Linux. I see audio and HID drivers loaded when it is plugged in to my Fedora Core 1 laptop, but that's about all that happens. I've searched all the usual places (FAQ, Google, etc) but not found anything helpful. Asterisk is working fine with

Does anyone know why the bt100 cannot park and pickup a parked call? attendant announces the call is parked at extension 701 but the call cannot be retrieved by any other phone. also, the bt100 constantly rings when the phone is hung up after parking. anyone experienced this? using the basic features.conf [general] parkext => 700 ; What ext. to dial to park parkpos =>

Hi, I am getting the following error when I attempt to listen to voice messages by dialing 9999 (I can hear nothing): --Executing VoiceMailMain ("SIP/2092-6918", "2092") in new stack --Playing 'vm-password' (language 'en') WARNING: app_voicemail.c:4922 vm_authentication: Unable to read password. I read in previous posts that this may be to do with the dtmf

Ok this is a simple question... What has been your experience with the WellTech 38xx series (I'm looking specifically at the 3802) VoIP gateway? I'm looking for a good (and hopefully not too expensive) VoIP/T.38 gateway for my office. Asterisk intergration is not a major factor at this time but may be later on. How well does it work? Is Echo a problem? Do the T.38 capablities

Following on from my below email, things have taken another bizarre twist.. I have continued getting the error when 2092 tries to listen to messages by dialing 9999. --Executing VoiceMailMain ("SIP/2092-6918", "2092") in new stack --Playing 'vm-password' (language 'en') WARNING: app_voicemail.c:4922 vm_authentication: Unable to read password. Then I

hi all i upgrade a bt100 phone and it can't resgister with asterisk Dec 1 13:25:49 NOTICE[1112980400]: chan_sip.c:7519 handle_request: Registration from '<sip:@172.16.4.249>' failed for '172.16.4.226' is was working with the version 1.0.5.3 some bady now what is hapening? thanks in advance Rodney

Hi All, I have tried searching everywhere but I cannot find a definitive answer as to if and how the conference button works on the BT100. Could anyone be kind enough to fill me in on some info on how to use the conferencing feature, as well as any configuration in asterisk thats needed, on this phone? Thank you, James -------------- next part -------------- An HTML attachment was scrubbed...

Hi, After a long way of problems (shipping, customs, etc) finally I got Welltech working. Here below my comments. - The documentation is poor and have errors - The web configuration is not complete. However is useful for the basic configuration parameters. The command line is necessary for modify all parameters. - The software upgrade is easy. Initially the gw came with H323, we upgrade to

Hi All, I'm sure this is something simple that I have missed somewhere. When I make a call using BT100 over IAX2 with Voipjet terminating I don't get a ringing sound whilst I'm waiting to be connected. The destination party can answer the call (they do get ringing) and conversation can take place. I don't get this problem with X-Lite softphone? Any help appreciated -

Hi all! I have the following setup: Phone lines -> traditional PBX -> Welltech 3802 -> VPN -> Asterisk -> Linksys PAP2/Welltech ATA-151 -> phone There is 2 pieces of Welltech 3802 (2 port FXO) connected to 4 (2x2) PBX extensions. Asterisk is a proxy here. Each device successfully register itself. I tried the setup above with Linksys and Welltech devices as well. I setup

I have a number of Welltech FXS devices (running the SIP code) that I'm trying to register with Asterisk. Has anyone had any success doing this? If I have the Welltech set in peer-to-peer mode, I can call to and from the Welltech, to Asterisk, just fine. However, because it's in peer-to-peer mode, I can only call phone numbers that are in the static phone directory. The phone

Hello everyone, I have a problem regarding gre tunneling, I have two linux box both of them has a private network and the linux A is connected to the internet via wireless radio and the other linux B is connected to the internet via lease line. Here is the setup of my two linux box Linux A eth0 = 203.189.x.1 (internet) eth1 = 192.168.1.1 (going to hub private network) Linux B eth0 = 205.198.x.1

hello all- has anyone had any problems with the handsets on BT100's. Just picked one up for my lab and the speakerphone works great but I am only getting one way audio (incoming) from the handset. Since the speakerphone works fine, I can't think of any config. reasons why the handset wouldn't other than a faulty handset. Any thoughts or experiences? Jason Kawakami Technical

Any feedback on this brand and in particular on doing business with WelltechUSA? I am looking to the Wellgate 3701A which is a 1FXS-1FXO arrangement. I am hoping to replace the near worthless Grandstream HT-488. This company is telling me that I need to wire $ directly into there bank account. Most unusual. Thanks for any feedback on this, Marty

No idea for this problem? Alex -----Mensaje reenviado----- From: Ing CIP Alejandro Celi Mari?tegui <alex@linux.org.pe> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Subject: [Asterisk-Users] [Fwd: newbie DNS problem with BT100] Date: Tue, 22 Mar 2005 19:42:30 -0500 (Sorry, but my english is very bad) Hi I'm newbie with

Hello, I think I might have an inkling as to where the issue may be at. For some reason when I create a new context, a directory is not created in /var/spool/asterisk/voicemail. The default context resides there but new ones are not created. Has anyone ever experienced this or does anyone have any idea as to how I would solve this? Hope someone can shed light on this, Many thanks, Aisling.

(Sorry, but my english is very bad) Hi I'm newbie with Asterisk, but i was able to install and configure Asterisk with 3 budgetone 102 and 2 Handytone 206 and works fine for me. I have a problem and i don't see answer in forums: DNS resolution: First Day: ========== In configuration menu of the BT100 I use: DHCP SIP server: central.mydomain.com or 192.168.100.180 Use DNS SRV: Yes NTP

My BT100 won't register with my Asterisk server, it always comes back with a 403. I've included my sip_additional (only one to to have the username 2201) and a portion of the sniffer trace (packets 27 & 28). This has me puzzled as I have my SPA-3K working (incoming and outgoing). On my BT100 I get no dial tone, I can't call it (asterisk says the extension is busy) but I can call