similar to: BT100 can't register

I'm afraid I'm quite confused by what I've found on the Wiki. I have the following dial plan that works: exten => 2201,1,Dial(sip/2201@gs1.uucp,20,) exten => 2201,2,Voicemail(u2201) exten => 2201,3,Hangup exten => 2201,102,voicemail(b2201) exten => 2201,104,hangup When the phone is in use it goes to voice mail as busy. When not picked up, as

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 -

I've got a strange situation with musiconhold. It works if I dial my extension 6000: >From extensions.conf: exten => 6000,1,Answer exten => 6000,2,MusicOnHold() Debug output if I call 6000: -- Executing Answer("SIP/gs1-b6ee", "") in new stack -- Executing MusicOnHold("SIP/gs1-b6ee", "") in new stack -- Started music on hold,

Hi All, I am a newbie to asterisk, and here is my first problem, where I do not know any further. I have to grandstream BT100 connected to asterisk. Working fine, for calling to each other, and to call via a IAX-Link to the outside. If I try to call the initial demo from the samples.extensions.conf I have nothing to hear. The CLI fine reports: -- Executing

Hello, I've found GlusterFS to be an interesting project. Not so much experience of it (although from similar usecases with DRBD+NFS setups) so I setup some testcase to try out failover and recovery. For this I have a setup with two glusterfs servers (each is a VM) and one client (also a VM). I'm using GlusterFS 3.4 btw. The servers manages a gluster volume created as: gluster volume

Dear list, I have a data frame like: > log2.ratios[1:3,1:4] Clone a1 a2 a3 1 GS1-232B23 -0.0207500 0.17553833 0.21939333 2 RP11-82D16 -0.1896667 0.02645167 -0.03112333 3 RP11-62M23 -0.1761700 0.08214500 -0.04877000 how to make it to look like: > log2.ratios[1:3,1:4] a1 a2

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,

Hi All, I'm trying to configure 2 GS BT100 connected to asterisk and Welltech 2 ports FXO gateway. I configure WellTech 2ports FXO and GS BT100, both GS BT100 can call each other without any problem but when I tried to call a local extensions connected to my Welltech FXO gateway, I couldn't hear any voice on both ends. I would like to ask if anyone has ever encountered this kind of

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

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

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...

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 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 -

(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

> Dear list, > I have a data frame like: > > > log2.ratios[1:3,1:4] > ID a1 a2 a3 > 1 GS1-232B23 -0.0207500 0.17553833 0.21939333 > 2 RP11-82D16 -0.1896667 0.02645167 -0.03112333 > 3 RP11-62M23 -0.1761700 0.08214500 -0.04877000 > 4 RP11-62M23 0.2761700 -0.15214500 -0.05877000 > the 3rd and

Hello, Has anyone gotten a Grandstream BT100 to work with Level 3's 3Tone? I've been able to get my extension to interface with it, but there is no sound and the call on the GS side terminates prematurely. Here is the relavent portion of the SIP.CONF [4007] ; Budgetone BT100 type=friend insecure=yes context=test-budget username=4007 fromuser=4007 callerid=4007 host=dynamic nat=yes

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

hi all I can not register my new granstream bt100 phone with asterisk, i have old of they working perfectly but they have an older firmware(1.0.5.3). any bady now where i can read about this or now what i have to do??? My sip.conf: [10] type=friend context=unr username=10 callerid=10 usecallerid=yes hidecallerid=no canreinvite=yes host=dynamic dtmfmode=info nat=no mailbox=10 callgroup=1

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

Hi I have defined my SIP phones like this ... [Sip1] username=gs1 callerid= "Full name" <1001> etc etc Now, when I do this in a given extension exten => nnnn,1,NoOp(${CALLERIDNUM}) then I get "<gs1>" as callerid and not "<1001>" as defined with callerid= Sure, I could set the usernames to their respective extensions, but I don't want