search for: granstream

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...

hi all, I known taht this ML is about *, but lot of you are using BT telepnones. I'm using FW 1.0.6.7 for all phone. This firmware support VLAN tagging for QoS on the Layer2. I use it to separe the PHONE network form the PC network, which are in PC connector in the BT. And I also use the TFTP server for provisioning (software, configuration). The problem is, if you set the VLAN for QOS.

I recently posted a question RE the Sipura 941 and using different ring tones, Thanks to hads I managed to use SET(_ALERT_INFO=Classic-1) to achieve this but trying this on the GXP 2000's didn't seem to do the trick?? Has anyone one had any luck on this topic? Also haven't been able to find any info on an auto-answer for the GXP 2000, again, I have succeeded in doing so with the

To all: (newbie) I have setup a BT 100 phone and mostly everthing is working pretty good except for the message button. I have place value in the appropiate field in the web configuration but nothing seems to work. When I press the button the speakerphone led goes on but the phone does nothing else (no dialtone, no sip request to *). Does anyone have this buttton working? I would like to

I think I have something misconfigured regarding voicemails. They work great, I have this setup: Sip.conf [ext1] Context=phones Mailbox=201 Voicemail.conf [home] 201,password,name,email@mail Voicemail delivery and all works great but when I check sip extension ext1 (analog phone using a Granstream ATA 286), the stutter tone signaling message waiting does not work. Anything wrong with contexts or something? Thx Guys

...able in iax.conf ( no g729 license... yet ), if I hangup one of the two calls, latency goes down to almost none. The strange thing is that one of my friends tryed calling me with one grndstream and one XLite and that combination does not seem to increase latency, but if he calls me with two granstream he has the same results. I also try calling 2 different * boxes, and the results are the same, continuos increasing latency. Has someone look at similar results ? Thank's.

...garrett.smith@b2llc.com B2 Technologies 454 Sonwil Drive Buffalo, NY 14225 (716) 250-3408 Direct (716) 630-1548 Fax (716) 903-9495 Cell AOL IM: B2sales Specializing in New and Used equipment from vendors including Cisco Systems, Juniper, Adtran, Dialogic, Lucent, Nortel, Sipura, Granstream, Snom, Mediatrix, Carrier Access, Digium, Zultys, IPDialog and more. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041123/8831fec2/attachment.htm

Hi all,

...three types of phones connecting to asterisk via SIP. We have Polycom, Grandstream and Cisco 7940 IP phones. When we ring polycom to grandstream or grandstream to polycom then both phones can send and receive voice fine and all is well. When we dial any combination of Cisco and either Polycom, or Granstream the Cisco, no voice is being sent but the Cisco can receive voice from the remote phone fine. When we dial Cisco to Cisco it all works fine. I am at a loss to figure this out and any help pointing me in the right direction would be appreciated. We are running an old Asterisk server with version 1...

Hi list, someone has successfully change different ringtone from dialpan with asterisk with this model Granstream? for example: exten => 0,1,Playback(pls-wait-connect-call) same=> n,SIPAddHeader(Alert-Info:;info=ring3) same=> n,Dial(SIP/310&SIP/318,30,t) can not get it to work any idea o tips? regardss -- rickygm http://gnuforever.homelinux.com

...f the nortels do. Always remember that it would cost the companies a ton to make every system totally closed.... That being said, the entry price for IP phones or ADSI phones can be much lower than you think. Find a good consultant in your area, get an ATA, a TDM card, and an Aastra/SNOM/Polycom/Granstream to play with. You can order the Aastra phones from your local electrical supply company (the place with a long counter and lots of electricians drinking coffee ordering their parts.). Andrew On 10/31/06, Ken Williams <ken@intermountainelectronics.com> wrote: > > > > I knew I sh...

...risk-users@lists.digium.com Subject: Re: [Asterisk-Users] BugetTone Bug Showstopper, On Wed, 28 Jul 2004 23:31:06 -0400, Seth Remington <sremington@saberlogic.com> wrote: > On Wed, 2004-07-28 at 21:00, James Gardiner wrote: > > How do I get Asterisk to recognise the # key from the granstream phone for > > doing transfers? > Make sure the Grandstream is configured to send DTMF via SIP INFO > instead of in-audio. > > -Seth Also, don't forget to disable the "#-key as redial" feature. _______________________________________________ Asterisk-Users mailing l...

On the Granstream 102 box that I have in front of me, there is a "feature list" on the side. One of the features has grabbed my attention: " - optional voice encryption (model 102D)" Now, digging through Grandstream's site, I see that it's not offered quite yet. However, sending ma...

We're using Grandstream GXP-2000 with programmed buttons to the first 5 parking lot extensions. When a call is parked, whichever parking lot extension it's parked on lights up red. We've never used the "announce" part and I'm wondering if there's an option I can't seem to find to disable the announce so the transfer happens faster. Thanks for any help, Ken

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,

Ok, With everything restore on rtp.c, still I have no audio however the call is not destroyed immediately as before. I'm going to put a second Granstream box, and findout if between two boxes this happen too. I cannot believe that we cannot do 2 g726 on the same box at one time. Carlos -----Original Message----- From: Carlos Alperin [mailto:calperin@senecacom.net] Sent: Wednesday, December 06, 2006 11:16 AM To: 'asterisk-users@lists.digium.c...

...rsday, March 12, 2015 2:42 PM To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Subject: [asterisk-users] GXP 1405 and asterisk Hi list, someone has successfully change different ringtone from dialpan with asterisk with this model Granstream? for example: exten => 0,1,Playback(pls-wait-connect-call) same=> n,SIPAddHeader(Alert-Info:;info=ring3) same=> n,Dial(SIP/310&SIP/318,30,t) can not get it to work any idea o tips? regardss -- rickygm http://gnuforever.homelinux.com -- __________________________________________...

We have a Dlink DVG-1120M and were surprised that it was able to handle 2 simultaneous conversations to 2 seperate phones using only 1 MAC address and 1 IP address. So we asked ourselves..why can't other 1 MAC/1IP devices do this as well? I have a Grandstream 486 that has 1IP and 1MAC. But I don't see anywhere in sip.conf to add a second line to a device. Is this possible? Can this only

Is anybody familiar with these IP phones AT-723 or AT-323 I think it is made by this company: http://www.atcom.com.cn/at723E.html -- #Joseph

Hello, I have a problem with a Grandstream being behind a symmetric nat. The box which does the nat is a german "Fritz Box". This one does nat for the internal network. In the internal network is a Granstream BudgeTone 100. The nat router has a dial-up connection, so ip changes on every dial-in. |------------| |------------| |--------| |Grandstream |------------|Fritz Box |------------|Asterisk| |------------| internal |(nat router)| Internet |--------| netwo...