similar to: Grandstream blind transfer issue

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

-----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

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 Sorry to be a bother again but seems like I just cant get away from the problems. This time my problem is that *sometimes* a user cant transfer a call from one extension to another, I have narrowed down the problem to it only happening to calls from outside the internal system. The wierd thing about the problem is that it comes and goes one moment the user can transfer, and the next

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

http://grandstream.com/BETATEST/HT488_496_386/

Hello Users, We have Thirdlane Multi tenant PBX system in production. Asterisk version is 1.6.2.15. Attendant transfer is working, but blind transfer is not working with Grandstream (gxp-2000) phone. We have read from google that it is a bug in Asterisk 1.6.2.15. We saw the below links: <http://www.odesk.com/leaving_odesk.php?ref=http%253A%252F%252Fwww.freepbx.o

OK, now this should be really simple, but I am a bit of a newbie so please bear with me. I have an *@Home box setup with TDM04B and two POTS lines. On the SIP side, I have GXP2000 phones. Most things seem to work, but the users cannot figure out how to transfer incoming calls from one extension to another. Now I am not sure that I have things setup correctly, but is there something

hi .. I have an asterisk system with three TDM100P (single port FXO) cards and 10 Grandstream 100 phones connected to it .. 1st question: when i phone out or receive a call from one of the SIP phones onto the PSTN, there is a LOT of local echo in the handset .. the PSTN end of the call does not here this echo, but it's VERY annoying on the SIP end of things .. the echo seems to be about 0.3

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

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

Hi, Does anybody know how to properly execute a transfer (without using the |Tt option) from a GS100? Scenario: 1. I call from X-PRO (ext 1100) to Grandstream (1101). 2. Grandstream answers. Call is established. 3. Press [TRANSFER] on the Grandstream. X-PRO caller is put on hold. Grandstream gets dial tone. 4. Grandstream dials 1103 (the extension of another GS100). 5. Grandstream hangs

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

I've searched for a few days now without finding an answer. The release notes for version 1.0.5.16 of the Grandstream firmware says it supports attended transfer using replace but the docs haven't been updated so I can't work out how to enable it, or whether it should Just Work. I'm currently using the # attended transfer patch for * but would like to get back to using the

The solution to the problems with the Grandstream 1.0.4.39 firmware is to use inband (in-audio) DTMF. Neither the RFC2833 nor INFO seem to work. However, this presents another problem. When I'm using g729 to place a call, I get the warning "Unable to process inband DTMF" because inband is not supposed to work with g729 (although it does seem to work when I've tried it so far).

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.

well .. good news :) i've just put in txgain=1.0 rxgain=1.0 in my zapata.conf and upgraded the Grandstream Budgettones i'm using to version 81 of the software and all seems fine .. there is still an echo but after the first couple of seconds of call it vanishes, as the echocancelling kicks in .. so far my client is happy :) now .. i have one slight problem left .. although most of my

Hello, I am building an asterisk PBX with some stuff to make a usable VOIP / PSTN Gateway. I use the following devices: SIP Phones from GrandStream for VOIP side OpenLine4 from voicetronix for PSTN Side I am building things step by step with some priorities. I have now a working system able to place and receive calls from/to pstn. Before attempting to bring other functions (like voice

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

Hi guys, I'm trying to get blind transfer to work and automatically transfer call to another number on key sequence press. Extensions.conf_snippet [from-pstn] exten => _0399377744,1,Set(__DYNAMIC_FEATURES=blindxfer) exten => _0399377744,n,Set(__GOTO_ON_BLINDXFR=to-pstn ^0388924326^1) exten => _0399377744,n,dial(SIP/0399377704 at c5400-02,T) [to-pstn] Exten =>