similar to: GXP2000 and hint LED's

Hi, I am sorry to be asking this but the wiki is down and has been for a couple of days and I need to get this working before Monday to get my live system setup. Trying to get the Snom 190's and soon to arrive 3com 3102's to use the function keys and for the life of me I can't work it out from the conversations on the archive what I am going exactly wrong here? The snom 190 with

Hi Everyone, Merry Christmas :-).... My Asterisk Box doesn't have a sound card, it is running Asterisk 1.02 Zaptel 1.02 Libpri 1.02 Mpg123 0.59r All compiled from source with kernel 2.6.9-1.6 on Fedora Core 2 Any help would be very much appreciated..... The error I am getting is -- Executing WaitMusicOnHold("SIP/snom-james-849d", "30") in new stack Dec 12 00:27:29

Charles S. Antrim wrote: > I am using a card that has an fxo and fxs module. I am no where near an expert but I have my sip phone working through my pstn line and this is my config. /etc/asterisk/sip.conf [general] port = 5060 bindaddr = 192.168.69.1 context = sip disallow = gsm allow = alaw disallow = ulaw nat=disable srvlookup=no localnet=192.168.69.0/255.255.255.0 subscribecontext =

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

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

hi, i'm a bit puzzled because i do not get include and hint to work with the new realtime enginge (cvs-head from 2004-12-09). other things (sipfriends and "normal" extensions) work perfect with the realtime engine. the entries in the static extensions.conf file i used before where: exten => 183,hint,SIP/snom220 exten => 183,1,Macro(stdexten,443,SIP/snom220,183) exten =>

I've tried to use a Cisco power injector to supply power over Ethernet to a GXP2000 phone without success. Although when I plugged these phone to a PoE capable Cisco Switch it worked without a problem! Knowing that all these three equipments implement IEEE 802.3af protocol, why doesn't it work with the Cisco power injector? Anyone also had this problem before? Thanks, Ricardo Carvalho.

I am trying to set up the BLF on a GXP2000. Currently what I have is extensions.conf: [globals] polycom430=SIP/101 [internal] exten => 101,1,Macro(voicemail,${polycom430}) [macro-voicemail] exten => s,1,Dial(${ARG1},10,tT) exten => s,2,VoiceMail(u${MACRO_EXTEN}@default ) exten => s,102,VoiceMail(b${MACRO_EXTEN}@default) [ext-local-custom] exten => 101,hint,${polycom430}

I am looking to setup paging using the auto answer feature on the Grandstream GXP2000. I am thinking I will follow the method as described here: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page I will setup the 4th account on the phone to auto answer. Does anyone else have a method that works better? I also looked at the allpage AGI written on Voip-Info. But it seems

We are doing a medium sized office in NYC with 80 phones. The customer originally requested Polycom 601 phones. The COO also authorized us to purchase 2 Grandstream GXP2000 phones for the mail room. We find these phones much easier to configure and work with asterisk . They support BLF & intercom right out of the box. They can also be centrally managed and provisioned. They also sound great

Scenario, I got some very good help earlier from Joseph getting me up and started but I have a couple of small problems still. Setup: FC2 Asterisk 1.0 Zaptel 1.0, TMP400P, FXS Port 1 FXO Port 4 Analog dialout line and Analog handset plugged in. Problems: 1. Incoming calls work and the phone rings and can be answered no problems, (although I wouldn't mind being able to adjust the ring but

I have 2 sip accounts setup - 200 and 202. If I do sip show peers I get: sip show peers Name/username Host Dyn Nat ACL Mask Port Status 202/202 192.168.0.6 D 255.255.255.255 5060 Unmonitored 201/201 (Unspecified) D 255.255.255.255 5060 Unmonitored 200/200 192.168.0.3 D 255.255.255.255 5060

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

1. We just dial the extension directly and have speed dials setup for the first 6 parked positions. We don't use *8 at all. 2. Change the config on the phones under Account to "Send DTMF via RTP (RFC2833)" -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Noc Phibee Sent: Thursday, February 08,

Hi i use a lot of Grandstream GXP2000 with BLF How to set up on the same key BLF blinking call interception? So that someone is able to take a call that is destinated to another user phone Thanks bye

Hi i have two problems with my Grandstream GXP2000 : 1- When i wan pickup a call, that's don't work's (*8EXTEN) and when i test whit Softphone, i have a error too, he say me *8201@MyContext not found .. in features.conf, i have: [general] parkext => 700 parkpos => 701-720 context => parkedcalls

Hi all gusy, i have a big problem with gxp2000 and asterisk 1.0.9 The phones after a few go in "busy" state, if you call it get the busy tone but the phone can male any type of call. This is my sip.conf [502] language = it username = 502 secret = <password> host = dynamic type = friend context = local canreinvite = yes dtmfmode = info callgroup = 1 pickupgroup = 1 callerid = 502

Sorry to post such a newb set of questions but I have been hammering about trying to get Asterisk running on FC2 machine reading everything available (I think that is what stuffed me, shouldn't have read it all :-) ). Config FC2 running Asterisk 1.0.0, with the h323 compiled in and installed correctly. Amazingly enough I have everything compiled correctly and installed. I am running a

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