search for: gxp2020

Hi, I`m using several GXP2020 phones with newest Firmware 1.1.4.18. Asterisk Version: 1.4.11. It happens several times that users complain that the caller cannot hear the transmitted voice from the phones.... Also now it happens quite often that callers on hold beeing dropped. Environment: ISDN with chan_misdn and SIP inte...

Hi, I'm using a GXP2000 (that's sharing the same GXP2020 firmware file) with the latest 1.1.5.10 beta release. It's working since a week and seems working very well. Before I was using the 1.1.5.3 and I had no problem. 1.1.4.xx versions, instead, are not performing like that one (audio, deadlocks and other minor issues). You can find a lot of info an...

Hi, Has somebody experiences with the Grandstream GXP2020 / 2000 phones in a business graded installation (with really traffic on .... not 3 calls a day ;-) ) Of course with Asterisk PBX (1.4.1 or 1.4.11 or 1.4 in generall) Thanks! Kind Regards, Erik

...s). So you have to install the BETAS whether you want or not... That you have to use unique ports is a rumour and not SIP standard. As John said --> IP:Port must be unique . I definitely not understand why I should use random ports. Kind Regards, Erik ____________ > I`m using several GXP2020 phones with newest Firmware 1.1.4.18. I had issues with phone locking up using 1.1.4.18. I've now gone to 1.1.4.22 and have eliminated that. > Asterisk Version: 1.4.11. Me too. Still testing 1.4.13 on a non-production system. > I use on every phone the 10000 as local port and in the r...

Hi, Is anyone aware of a SIP hardphone with Busy Lamp Fields supporting 2 colors (or more) ? This could be very useful to support extended presence, for instance. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090519/0b8f1b62/attachment.htm

Hi list, I'm wondering if there's a way for multiple users to share the same voicemail box and have their BLF flashing when voicemail comes, i.e. in a home phone system where there's a general vm for everyone. I'm using couple Grandstream GXP2020. Any suggestions? Kelvin Chan | Positronics Ent. Product Development | | unit 272 604-628-9330 (direct) | 8128 128th St. 604-585-2119 at 104 (main) | Surrey, BC 604-585-3056 (fax) | Canada, V3W 1R1

...age=be [BT201] type=friend context=intern host=dynamic username=BT201 secret=testpaswoord canreinvite=no callerid=Jonas Kellens <52> qualify=yes [GXP1200] type=friend context=intern host=dynamic username=GXP1200 secret=testpaswoord canreinvite=no callerid=callerid? <51> qualify=yes [GXP2020] type=friend context=intern host=dynamic username=GXP2020 secret=testpaswoord canreinvite=no callerid=Kristof Teirlinck <50> qualify=yes Musiconhold.conf : [default] mode=files directory=/var/lib/asterisk/moh random=yes Features.conf : ; Sample Call Features (parking, transfer, etc) confi...

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. Do you know if i can...

...ome (zoiper softphone). I have internal extension 60, and I call the internal SIP-client 10 at the firm via an IAX-connection over internet. My colleague at phone 10 answers my call. I ask him to transfer me with my colleague at extension 50. He then presses "transfer" on the grandstream GXP2020 (I get music) and dials the number 50. Phone 50 rings... and I loose connection... It is absolutely sure that my colleague's phone at extension 50 was not occupied. He was just awaiting my call... This is wat the CLI says (from log-files) : [May 15 16:51:41] VERBOSE[2743] logger.c: -- Ex...

...ting the feedback of a customer that a conversation is like half duplex : when he talks, the other end of the call is no longer heard. What could be the cause of these drop-outs ? A call that is coming in from the PSTN is routed through an IVR-system to the correct internal SIP-phone (Grandstream GXP2020). Where do I start searching for this problem ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090527/9da616a4/attachment.htm

Hi, I used Grandstream (gxp2000, gxp2020) and Snom (370) SIP Phones, but with 2 extensions BLF status does not work correctly. have someone ever tested a Sip Phone with more then 60 BLF without problems? Can someone suggest me model and brand? Thanks, bye. -------------- next part -------------- An HTML attachment was scrubbed... URL: ht...

...I need to know is how to detect whether a call from a member of that department is a transfer or an original call. I've looked at the TRANSFER_CONTEXT setting, but that's only for transfers with # and the T and t flags to Dial(). But we use SIP hardphones (Linksys SPA942 & Grandstream GXP2020), which have built-in transfer functions, and we would like to continue using those for transfers, rather than building it into features.conf or dialplan... Because we prefer attended transfers, and the user experience seems more modern. So, does anyone know of a way to detect whether a call from...

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

Hello, When I tried to install chan_h323-1.0.1-module.i386 RPM i got these errors. Failed dependencies: libh323_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 libpt_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 But i found the same files in /usr/lib/libh323_linux_x86_r.so.1 /usr/lib/libpt_linux_x86_r.so.1 What to do for asterisk to detect the same

Our small company is replacing Cisco CallManager with Asterisk (because we are tired of sending them money) and I am documenting the process as I go on my blog. I am trying to make the notes as easy as possible in hopes that I can ease someone else's pain. Here is the link: http://qvlweb.blogspot.com/2009/03/asterisk-pbx-system-install-01-what-i.html Please feel free to comment on the