similar to: disconnect with mute

Hi, I'm struggling to set up a plan to allow international dialing from my US location. As I understand it, an international call can have 9 to 15 digits including country code. The problem is that the call always goes through after I've entered the 9th digit. My service provider is BroadVoice, my phones are Grandstream GXP-200's. DTMF is set on the phone to be via SIP INFO and

I've got a problem with DTMF, again. My asterisk box is connected with the outside world (PSTN) via a sip proxy. The problem is that for some reason, I need to use rfc2833 for signaling digits to the gateway and inband to accept digits from outside (eg. when someone dials one of our DIDs). It's possible to do this? I've ever tried splitting 'peer' and 'user' part in

Grandstream GXP-2000 with firmware 1.1.4.18 (beta) fixes an issue where the phone did not send rtp keepalives when on mute (resulting in disconnect from tech support hold and concalls) A side effect seems to be that Asterisk pops the following warning on the console... Jul 26 14:06:35 WARNING[31654]: rtp.c:463 ast_rtp_read: RTP Read too short Grandstream say they are not sure what it is but

Hello, Has anybody been able to get the Mute button work on grandstream? it simply does nothing. Only Hold is avalable, which is not that good. Thanks -- Anton Yurchenko<phila@dg.net.ua> Digital Generation

Hello I'm about to order a GrandStream GXP-2000 and a Linksys SPA-921 I'd like to have some user feedback about how those phones perform, and whether their LCD screen displays both the caller ID name and number (The GrandStream BT-100 only displays numbers, which isn't very helpful). Thank you.

Hi, I'm provisioning an office with limited cabling. I'm looking for a desk based wifi phone. Most of the ones I've seen are handsets. Any ideas? Thanks, WILL -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051007/1d2cc49a/attachment.htm

Hi, Any suggestions to below problem? Thanks -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jamie Heckford Sent: 17 October 2006 21:48 To: asterisk-users@lists.digium.com Subject: [asterisk-users] FW: Why does One-Touch record mute/disconnect callif not dialed quick enough? Hi List, Have an odd problem

Hi People, We use the Grandstream GXP-2000 phones, firmware 1.1.1.14, Asterisk PBX, Slackware Linux 10.2, loaded on a Intel(R) Pentium(R) 4 CPU 2.66GHz Box... The issues that we are experiencing involves our Telephone Operator's/Receptionist whom answer multiple incoming calls... As an example.., when they answer line 1 and Line 2 starts to ring they would ask the person on line 1 to

Hello, I have some trouble with the BLF indicator. I have two phones that use the same hint: 13 => hint,1,SIP/phone13&SIP/phone13-wlan This works great from the asterisk side, but it seems the status change is too quick for the attached Grandstream-phones. When I ring the extension the hint changes to "Ringing". The Grandstream blinks. Great. Now, when someone picks up one of

I started out a few years ago with some SPA-841 sets, because the Grandstream 2000 I thought I wanted was perpetually delayed. The GS had more call appearances, and I didn't want just the 4 max that the SPA offered. As it turns out, with the greater flexibility of VOIP, I don't need 'dedicated' CAs the way I needed them on ISDN previously, so 4 is actually adequate. Along the line,

Hi List, Have an odd problem with the one-touch record on asterisk 1.2.11. All works ok, however one of our users today discovered if he is a bit slow hitting the 1 key after he presses *, the call seems to stay connected but its almost like it is muted. Haven't figured out the delay yet but it seems to be if the 1 is not pressed within 1-2 secs this occurs. Any suggestions? I tried

Hi When I make a call with the Grandstream GXP-2000 through Asterisk (and SER) to landline using VSP, after I hang up the call the other party are still connected for another 30-40 seconds. I've notice that the SIP BYE is sent to Asterisk, but Asterisk sends no SIP BYE on to VSP. When I use the SPA-941 the call terminates on the other right away soon as I hang up. I have updated the

Hi I have a grandstream gxp-2000 and trying it on an asterisk 1.6. When I call internally between extensions I can hear the other person in the gxp2000, but when I call externally from the gxp I can't hear the person on the other end, but he can hear me. How do you configure the grandstream 2000 to work on asterisk 1.6? Regards /ralf ________________________________________________ Ralf

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I've got Asterisk SVN-trunk-r9059 currently running on Fedora Core 4 w/ 2 eyebeam softphones and 2 Grandstream GXP-2000. At my desk I've got the grandstream and the GXP-2000 I would like to ring both. Using macro-exten-vm and dialparties.agi Macro(exten-vm,200,200-202) the caller is sent to the unavailable voicemail but if I use

I'd recommend any of the following, which are all in your price range Snom 300 Polycom IP430 Polycom IP501 Aastra 9112i Linksys SPA-922 Grandstream GXP-2000 Cory Andrews ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Wednesday, November 01, 2006 11:17 AM To: Asterisk

Hi, I have Grandstream GXP-2000 connected to Asterisk, and Asterisk has trunk to VSP for PSTN calls. When ever I place local PSTN call, the landline doesn't hang up right away (40 sec), when I hang up the GXP-2000. The GXP-2000 seems to have problems making international calls as well. Where it hangs up soon as the other party picks up. I have used different IP phones, VSP's and etc.

The vast majority of what I've done with Asterisk has been with the Grandstream GXP-2000's. These phones work great for us for everything *except* speaker quality is quite poor and appears to be half-duplex. So now that we've bought and are using 40 GXP-2000's we're doing some testing on other phones. I've bought a Polycom 301 & 501 as well as a Linksys SPA942. While

Anyone used the new Grandstream GXP-21XX series phones. We have been testing these phones and like what we see. We are looking for a greater cross section of testing before we roll them to production. Any feed back would be appreciated. We are talking with Grandstream engineering and they are looking for feed back as well. Any input is appreciated. Thanks Bryant -------------- next part

Hi, We have a page on the wiki just for phone reviews, but I think it needs a bit of format change. Instead of individual reviews for each phone, I think each person should review all phones they have worked with and list the phones they have had access to and rank them in relation to each other. Also each review should have a date so the reader can see how fresh the data is to current.

On my Grandstream GXP 2010 I have the possibility for 6 channels and thus 6 different accounts... Line 1 I define an account that registers directly to an online Asterisk-server, somewhere in a datacentre. Line 2 I define an account that registers to the local Asterisk-server (NSLU2 unslung) When I activate both accounts, only the first account (to the Asterisk-server on the internet) registers.