similar to: Little help with Conference

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

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 and old firmware versions at this

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 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:

Hello, I have a problem with a grandstream IP Phone. The SIP autentication is OK, but when try to call someone I get the message --> WARNING[14281] chan_sip.c: Error in codec string '=audio 5004 RTP/SAVP 3' I tried to change the CODECs (ulaw, alaw, GSM, etc), the result is always the same. Tried to change the RTP port but the result is the same. The grandstream IPhone is behind a

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

Under Advanced make sure this is set: Call join on Xfer (2 calls): to off ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tommaso Calosi Sent: Thursday, May 04, 2006 4:02 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Unwanted conference with snom320 and asterisk

I have 13 Snom 320 with asterisk 1.07 bristuffed. The problem is that sometimes on random basis, when one customer is placed on hold and another call arrives, the customers are put in conference with each other. This look very strange to me, but I've disabled the confernce button on the snom phones to prevent the human errors, but it still occurs. Investigating I've discovered that a

Hi, I'm using Grandstream IP phone GXP2000, with Asterisk 1.4.15 I'm using g729 codec and want to use only this codec for the calls. My normal calls are going fine. But issue is coming when I'm using the conference from the Line1 and Line2 Option. When I'm initiating the conference at that time, IP phone is sending the G711ulaw for the conference call, while in my phone I've

I have just upgraded my Polycom 501's from 1.6.2.0041 to 2.1.0.2708 to get the microbrowser. Almost everything is fine except when receiving calls from a BT200 (1.1.14 and earlier) the Polycom rings but when answered, drops out and the BT200 gets a busy tone. I have many PAP2T's and SPA3000's etc and they all cal call the Polycom without problem. Does anyone know what could be going

Hi I just want to know if anyone have problems with server DELL 1600, Like: Hangup Call. Thanks Ruben

Hi All- I have an asterisk server and GXP2000. If I want to send a call to someone else (external), I can transfer the call where I can announce it, and then send it over. But what I'd like is to start a 3-way conference, and then drop out. But if I do a conference button on the phone, and then drop out, the other two are not left to finish their conversation (the call is ended).

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 internal calls. No NAT no DNS name (only IPS configured). I configured

Thx John !! Hmm I found now on voip-info.org a lot of Beta releases which should fix my problems... Kind of strange whats going on with Grandstream devices and their firmware ... If you install the latest "official" release you can expect a few troubles with Asterisk 1.4.11 (one way audio --> randomly, dropped calls). So you have to install the BETAS whether you want or not...

-----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

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

Hello, We have several clients with GXP2000 in their network and behind NAT. We have one particular client that has several GXP2000 behind a Linksys RV082 VPN Firewall/Router which is doing NAT services. According to SIP packet inspection, it detects it's a symmetric NAT. The problem we have is that even though we have configured Asterisk AND the GXP2000 to register every 60 minutes, they

If anyone can help im trying to get my jabra bt200 or bt250 headset working with chan_bluetooth. They seem to pair ok but they will not come out of "Negotiating" state. I get this on first start of *: [HS] jabra > AT^SPTT=? [HS] jabra < ERROR If anyone can be of help please advise, im pulling my hair out on this one. Thanks Jason Price NOTES: JABRA BT200/250

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

hi, we are experiecing the following probem, if anybody have come across such a problem or a solution to this please let us know. our set up is, an Asterisk server equipped with, 4 port station interface card ,single port fxo card and several soft sip phones we have found problems with the following scenarios, outside caller (calling through fxo interface) <------------------------------>