search for: bt200

I have just upgraded my Polycom 501's from 1.6.2.0041 to 2.1.0.2708 to get the microbrowser. Almost everything is fine except when receiving calls from a BT200 (1.1.14 and earlier) the Polycom rings but when answered, drops out and the BT200 gets a busy tone. I have many PAP2T's and SPA3000's etc and they all cal call the Polycom without problem. Does anyone know what could be going on ? Thanks. -- Kevin Withnall http://kevin.withnall.com/ IL...

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm testing with a Grandstream BT200 telephone and, according to I read, it has a LED that blinks if for that extension messages were left. In "Voice Mail UserID", under the "ACCOUNT" tab, I put *100 that is the extension in which my Asterisk answer the voicemail service and if then I press MESSAGE button, the tel...

Hi All, I have working asterisk version 1.4.24. I have a blind transfer issue with grandstream bt200. I have updated the latest firmware to the phone. The phone is sending the *refer* to asterisk but asterisk is not able to connect with the another call that i have checked in sip debug. I am using transfer button of the grandstream phone. Can anybody provide help for this issue? Thanks in advance!...

Is there an ATA type device out there that has low level audio out for connecting speakers? My asterisk server is in one building, I wish to have speakers in another building and connect them up to a low level audio device that I can call into and speak. Can I connect speakers into the FXS or FXO of a grandstream HT 503? Does that work? I'd prefer not to have a PC setting there (sound

Hi All, I have setup asterisk 1.4.22; so far everything good. Except, I am still searching for voIP phones. Which grandstream phone should I buy, this is going to be for small office for testing purposes. I am on a budget, hoping to find someone here who has some used to sell or point me in the direction of a seller. I am in the US. thanks, Mike

If anyone can help im trying to get my jabra bt200 or bt250 headset working with chan_bluetooth. They seem to pair ok but they will not come out of "Negotiating" state. I get this on first start of *: [HS] jabra > AT^SPTT=? [HS] jabra < ERROR If anyone can be of help please advise, im pulling my hair out on this one. T...

Hi I have a server with the last version of asterisk branches, zaptel branches, 2 Digium Card with TDM800P 16 HPEC channels (Echo Cancelation), 40 Grandstream BT200 and 10 Grandstream GXP2000. zapata.conf echocancel=64 rxgain=0 txgain=0 when i place a call o receive a call, I finish a sentence i hear a ssssssss, AND when the both side talks at the same time i have choppy audio. Any help i appreciate. Thanks Ruben

...bout the problem is that it comes and goes one moment the user can transfer, and the next call he can't. I am running: * Asterisk 1.4.17 * Zaptel 1.4.7.1 * Libpri 1.4.3 Using the following phones and firmware * Grandstream GXP2000 (with ext pad) : 1.1.4.14 * Grandstream BT200 : 1.1.4.18 I have set up the phones to log debug logs to a syslog server, I am still trying to figure out what exactly the log says. Is it an * problem, or Grandstream problem Does anyone know if I am able to see the keysequence the user types into the phone (just in case it might even be a us...

Hi folks, Am new to asterisk pbx systems. I am trying to figure out what to do, I'll list and folks feel free to give feedback and advice. MAIN purpose for usage: 1.exposure to setup an asterisk box 2.get home phone service via VOIP/internet connection. tasks so far ------------------ 1. setup and install asterisk (1.4.x) --> DONE -currently configuring sip.conf

...100 Trying Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK41bcdf63;rport From: "Steffen" <sip:116 at 192.168.150.151>;tag=as085e9ac6 To: <sip:119 at 192.168.150.11:5060> Call-ID: 17eb890c14a50dfa541efbb517835bb4 at 192.168.150.151 CSeq: 102 INVITE User-Agent: Grandstream BT200 1.1.1.14 Content-Length: 0 --- (8 headers 0 lines) --- asterix*CLI> <-- SIP read from 192.168.150.11:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK41bcdf63;rport From: "Steffen" <sip:116 at 192.168.150.151>;tag=as085e9ac6 To: <sip:119 at 19...

...aw, alaw, GSM, etc), the result is always the same. Tried to change the RTP port but the result is the same. The grandstream IPhone is behind a NAT, However there are others IP Phones working fine behind NAT (from other locations, outside my network) Changed the IP Phone from Grandstream GXP2000 to BT200, got the same probem The error in the IP PHONE display is "481" The X-lite Softphone works fine in the same network. Any help please ? Regards Claudemir F. Martins -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asteri...

Hi Guys: I'm using Asterisk with Addpac 2620 as gateway, internally I'm using Grandstream BT200, unfortunately when I called to external phones (PSTN), and I have to choose some extensions, the Phone don't dial the extensions, I believe that DTMF relay in ADDPAC is not working well. I'm using RFC 2833 and ALaw for SIP Channel (Between ASterisk and ADDPAC). Someone have any experiencie...

Hello, I've enabled BT-200's SYSLOG logging, and I get some message whose meaning is obscure to me. In particular, in a day I got the "Deletion of invalid timer" message almost ten times from one phone, which has some call problems. Can someone point me to a resource on BT200 error codes? Thanks, -- Andrea Spadaccini Multimedia Technologies Institute s.r.l.

These is my scenario. Asterisk 1.4.16 Zaptel 1.4.8 Grandstream BT200 Grandstream GXP2020 Grandstream GXP2000 For some reason the end user ask to configurate son direct access like *01,*02,*03 thru *78. After they began to use these direct access, I cant place a 3 way CONFERENCE. I remove the direct access, but i dont know if one of them block the CONFERNCE....

Some light relief .... SNOM say "Please note that you will not be able to reach us by phone." http://www.theregister.co.uk/2008/03/13/dont_call_us/ regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com

Hi All; I just need to know the openion about Grandstream phone, actually I tried Budge Tone 201 and I chocked that there is a noise in the handset (zzzzzzzzzzzzzzzzzzzzzzzzzzz) always, but in the speaker the sound is good and no noise. Anyone has idea about Grandstream, and if they have a lot of problems and such noise in handset? Or my luck was bad that this phone is defected? Regards Bilal

...1 Hi all! In the network of my house I was testing the security with my Asterisk installation. The first test that I'm doing is an man in the middle attack. In this scenary, the attacker is a virtual machine that it tries to see the SIP traffic between a PC with a softphone and a Grandstream BT200 telephone. But it draws attention to me between the PC with softphone and the telephone I see traffic ARP or ICMP that could make to try between the equipment but does not see RTP. Is there some special consideration that it must to observe? I am doing it to the capture with: # tcpdump -i eth0 -n...

I'm preparing to deploy a small number of Grandstream BT101's and GXP2000's to a remote location (which I won't have access to). I'd like to have them pull a config file from my server - I'm almost there... The phones are looking for the config file on my webserver which is good. I need to generate that file however. I see a tool on the GS website to generate

Can anyone who has used both comment on the pros and cons ? Need to buy about 30 of these, for a small company with limited IT support. Julian ______________________________________________________________________ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email