similar to: Asterisk 1.4 and GUI Configuration Help

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

I just purchased a Grandstream gxp-2000, budgetone102 and a HT-386. Browsing to each device by IP address, I can get logged in using admin and I can see the advanced settings, however, if I try to change the settings and clicking the Change button, it just brings me back to ask for the password again.. I can't get into the Status page or any of the Account1-4 pages either. It just keeps

I'm communicating a softphone (SJPhone) to a Grandstream phone GXP-2000. The codec order on each one is the next: SJPhone: GSM - iLBC - PCMA - PCMU GXP2000: G729 - GSM - PCMA - PCMU (I have a G729 license, so there's no problem with transcoding G729) In my sip.conf, I've defined the following codec order: disallow=all allow=g729 allow=gsm allow=g726 allow=alaw allow=ulaw And my

I'm building a dialplan for use with a bunch of GXP2000 desk sets. During testing, we had some user issues surrounding the lack of an on-phone dialplan. Users would hit 9 and sit there waiting for a redial tone, and the GXP would time out, sending just '9' to *, which couldn't do much other than spit back a 404 or play pbx-invalid. I turned on the "early dial" option

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

I have 5 GXP 2000's with firmware 1.1.4.25 running with Asterisk 1.4.13. Is anyone else getting the following error in the asterisk console: [Oct 22 11:39:01] WARNING[7100]: rtp.c:1142 ast_rtp_read: RTP Read too short every couple of seconds when a handset is in a call? I didn't notice this happening when I was using an older GXP2000 with the same firmware (doesn't mean that it

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

Version - Asterisk SVN-trunk-r12793M (1.2.4) I have 4 Grandstream GXP 2000 phones configured. However at the moment, I have had to disable BLF, Hints, and Call Limiting due to an extremely annoying bug which seems to make the phones channels "lock" in busy after a call has been hungup. If I do a show hints after say extension 200 has hung up I get the following -= Registered

Good day , I have configured TDM410P (asterisk 1.6.x) on Cent OS 5, but dahdi take some time to hangup the call when playing the IVR..(it will send the hangup signal after finishing the IVR promt..) is there any specific setting to avoid such incidents ? iam using busycount as 3, signalling=fxs_ks ;toneduration=100 callwaiting=yes threewaycalling=yes callreturn=yes

>From the usual place, http://www.grandstream.com/BETATEST/GXP2000/ Note, there are two (and it took me a bit of a while to figure out) images to be loaded. Copy the ...a.bin's and the .bin's to your http provisioning directory, and reboot. The phone _must_ load the .bin files before it understands the ..a.bin files. After it loads the first one, the phone does lock up with the

I am looking to replace all of the old "Bell" (POTS) phones in my home and office with IP phones. As you can imagine I don't have a huge budget to work with but I want phones that will provide acceptable voice quality and durability. There are basically three categories as I see it 1. satellite phones (low cost, low function) 2. primary domestic phone (good quality, POE capable,

I just downloaded the zip file from grandstreams website to upgrade my gxp2000 firmware from 1.0.0.3 to the latest but seems there are some files missing on the zip file... Anybody been able to upgrade their firmware? My website shows this files as missing: 201.133.125.152 - - [11/May/2005:16:47:16 -0500] "GET /firmware/ring1.bin HTTP/1.0" 200 12737 "-" "Grandstream

Hi, I have several GXP2000 phones which used to work fine with Asterisk 1.2. However, after upgrading to Asterisk 1.4.21.2, whenever I initiate a call from a GXP2000, it gets dropped after 20 seconds exactly. I have "early dial" enabled on the GXP2000 and "pedantic=yes" on the server. If I disable "early dial", all works well ("early dial" or "overlap

I'm hoping people can suggest some ideas for debugging a problem that I'm having with DTMF. Unlike most of the DTMF problems reported here, it has nothing to do with Asterisk interpreting DTMF. My problem is with the synthesis of DTMF tones on outbound calls on a PRI connected to a TE412P card. I'm running * 1.4.10.1 with Zaptel 1.4.4. It is important to note that these problems

Thanks for reading this. I've been pulling hair for days trying to resolve this, and any help someone can give me would be very much appreciated. I have an Asterisk box that is basically a P4-3GHz, a Digium-recommended SuperMicro X5SSE-GM motherboard, 2GB RAM, 250GB IDE hard-drive with UDMA, a SoundBlaster Live! 24 sound card, a Digium Wildcard TE110P, and a Digium Wildcard TDM400P with 4

Try to use 2.6.10 kernel not 2.6.12.3 kernel Coz I got some problems with the kernel 2.6.12, I had the same Digium Wildcard TE110P, Good luck ________________________________ Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Robert Christian Inviato: marted? 26 luglio 2005 19.13 A: asterisk-users@lists.digium.com Oggetto:

All, I have AAH 1.0 installed using Digium TDM04B and Grandstream GXP2000 phones. All seems well other than the phones have to be reset up to 5 times per day. It is like they lose thier ip connection or maybe thier SIP connection. Has anyone else experienced this issue? I have the phones set for static IP addresses and that doesnt seem to help either. Any help would be greatly

Good evening all. I'm having this strange behavior when dialing two or more simultaneus calls via IAX to other * boxes. Sound starts to have more latency, wich increments until it's almost impossible to talk (6 or more seconds), I try this calling with two grandstreams, one grandstream one tdm410p, one tdm410p and sjphone, sjphone and one grandstream, the result are similar.

I tried 1.0.1.12, 1.0.1.13 and now 1.0.2.13 on the GXP2000. I doubt it's the firmware if it is happening on two completely different phones. I read of a bug in Asterisk that exhibited this exact behaviour that was apparently fixed in Asterisk 1.2.0 as far as I can tell. http://bugs.digium.com/view.php?id=3644 > -----Original Message----- > From: stoffell [mailto:stoffell@gmail.com]

I have 20-30 GXP2000's connected to * over a T1 line. Neither end is NAT'd and there is plenty of bandwidth available over the line. The GXP's are 1.1.5.15, which is the latest. I have a problem where the phones keep dropping off of * and I get a "failed to register" message in the log of *. Sometimes they eventually connect and sometimes, I have to reboot them to