search for: bt100

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...

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 - Asterisk is Asterisk asking...

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...

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... URL: http://lists.digium.com/pipermail/...

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...

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...

...lMain ("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 settings and have set both (asterisk and BT100) to info. This has not helped. My phones register with SER (port 5060) and use Asterisk for voicemail (port 5064). My configs are below along with my BT100 settings: ;Grandstream BT100 SIP Server: x.x.x.x:5060 SIP User ID: 2092 Authenticate ID: 20...

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 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: F...

...nyone have any idea as to how I would solve this? Hope someone can shed light on this, Many thanks, Aisling. -----Original Message----- From: Aisling [mailto:ashling.odriscoll@cit.ie] Sent: 07 September 2005 13:54 To: 'asterisk-users@lists.digium.com' Subject: Eeven Stranger - Asterisk BT100 Password Issue 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&...

...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 decided to plug out my BT100 and left it plugged out for a few hours. When I plugged it back in and dialed 9999 for voicemail, bizarrely I could hear the voicemail main menu and was prompted for a password. When I entered the password, I was able to listen to the messages...This is what appeared on the Asterisk console --Ex...

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 experienc...

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 - and a very mer...

...ut 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 Server: time.nist.gov Symptoms: DHCP works fine for me. All config like IP, Mask, DNS, router etc. LED: 1900-01-01 SIP server: With 192.168.100.180 work fine with asterisk but with central.mydomain.com not work....

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...

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=40...

...92.168.70.26:5060 for 3600 Dec 24 12:16:35 NOTICE[15776]: chan_sip.c:7742 handle_request: Registration from '<sip:52221@192.168.70.2>' failed for '192.168.70.26' -- SIP Seeding '52221' at 52221@192.168.70.26:5060 for 3600 -- Saved useragent "Grandstream BT100 1.0.5.20" for peer 52221 -- SIP Seeding '52221' at 52221@192.168.70.26:5060 for 3600 from debug /var/log/messages Dec 24 12:16:34 DEBUG[15776]: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = '52221' Dec 24 12:16:34 DEBUG[15776]: MySQL RealTime: Ev...

We are having many successes with Asterisk and starting to get the hang of it. But, I am still having problems getting my Budgetone BT100 (firmware 1.0.4.50) to work fully. I can receive calls, but cannot make them. We have the latest version of Asterisk, Fedora Core 3, Digium TDM400P with one FXO and one FXS card configured and working well. We have a PSTN line going into the Digium card, standard phone going out. Also, we have...

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=dynam...

Hi Guys, Sorry about writing to that list but could not find better place. I have Grandstream BT-100 phone, btw, was working great with Asterisk. I have upgraded the phone, and during upgrade something went wrong. Right now when I power the phone I can only see some garbage on the LCD display. does not react on any buttons, pings,..... Maybe somebody has any idea if it is fixable or I can just

Hi, after upgrading to Asterisk 1.2.8 from 1.2.7.1 I got a problem with Grandstream BT100 after making an attended transfer (FLASH + NUMBER + SEND + WAIT ANSWER + TRANSFER). After the transfer, the display clears all the info except the clock, there is no dial tone, the WEB admin stops working. Incoming calls make the display light turn on but there is no ring and no callerid on the...