similar to: Sound issue with Asterisk

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Thank you and sorry...There is something going wrong with the system I only sent one mail... _____ From: Kerry Garrison [mailto:kerryg@techdatapros.com] Sent: Wednesday, May 11, 2005 5:14 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [SPAM] - RE: [Asterisk-Users] Grandstream-Budge tone - Email found in subject This is usualy a problem with either

I have an asterisk Redhat 9 box running 4 hardphone extensions. Inter-extension calls are crystal clear. However when I dial out through my iconnect account I get a lot of jitter. At first I thought it was my nat gateway but after I programmed on of the hardphones (budge tone 100) for direct dial to iconnect I have clear voice transmission. I have no way of explaining this. My asterisk sip.conf

Hello, I tried to configure a cisco 7905 IP phone using the skinny channel but I had not much luck. The relevant portion of skinny.conf is: [cisco1] device=SEP000F3487F8E3 callerid="Alex" <123-456-789> mailbox=500 callwaiting=1 transfer=1 context=default threewaycalling=1 line => 500 ; Dial(Skinny/500@cisco1) I set up the tftp server, and prepared the following

Hello all, I need a virtual firewall/router solution. I''m thinking of a netscreen 1000 but I want to know if it can be done in Linux. Here is my idea: 1 Linux box 2 GigE interfaces 1 interface setup with a public IP address ($PUBIP) 1 interface setup with 802.1q VLAN trunking with 100 vlans assigned ($VLAN1-$VLAN100) a /25 subnet routed to $PUBIP from my core routers All $VLAN

Hi everyone, This is off topic and is for GS technical support really but it seems that there are a lot of Budge Tone 100/101/102 users out there. I've got a Budge Tone-100 (101 - without the extra 10base ethernet connetion?) here. I changed the configuration through its web based interface and I clicked the reboot link. But then something went wrong and ever since then it doesn't

HI , I installed asterisk in fedora core 3 machine perfectly. and i have 10 units of GrandStream IP phone ( Budge Tone-100 ) . I wanted to know how can i use it as extentions in my LAN ? Asterisk PBX alredy there. I didn't try to do any configurations of any files . What are the configurations has to be made with asterisk ? Thanx in advance, Luke. Send instant messages

Hi All; I just need to know the openion about Grandstream phone, actually I tried Budge Tone 201 and I chocked that there is a noise in the handset (zzzzzzzzzzzzzzzzzzzzzzzzzzz) always, but in the speaker the sound is good and no noise. Anyone has idea about Grandstream, and if they have a lot of problems and such noise in handset? Or my luck was bad that this phone is defected? Regards Bilal

Hi; Have two grandstream Budge tone...Connected them to the network and able to make call to/from them. But when the coming call answered, I can not hear any voice and also my voice is not heart... I am able to hear voice only if I pressed the hold button and take the call again....This problem also Occurs in calls from x-lite to cisco7940... Does anybody has any idea or documentation

We have 3 Grandstream Budge Tone 100 phones which are being very fluid on incoming calls. They are set up as extensions 2501, 2518, and 2536. When calling out to another phone, they always identify themselves correctly. But sometimes they will respond to the wrong incoming calls. (By respond, I mean that the phone rings and if someone picks up the receiver, the call then goes thru.) For

Hi, I am looking for hardware sip phone with very good sound quality. Can anyone recommend ? I use to have Grandstream Budge-Tone 100 but I feel that the sound is not very satisfactory and volume too soft Regards ASLAY

Hi, I search How To transfer call between my SIP phone. I have an PSTN line (X100P) and 10 grandstream budge tone phone. For example I want : - Reveive an external call and send it to SIP/phone1. At this point no problem. - After my receptionnist want transfert extern call at SIP/phone2... I don't known how to properly transfert call.... Thanks

I finally got my Asterisk all setup and everything seems to be working except for menu interaction between my Grandstream Budge Tone 100 and my Asterisk. I have the SIP phone setup to properly connect when pressing the 'Message' button and that's working perfectly. When the menu starts, it says press 1 to read your messages, but pressing 1 (or any number) fails to send. Does anyone

I'm not sure if this is possible, but I was hoping to find an address book that runs on Windows XP that will allow me to select a phone number and send that to my Asterisk. The Asterisk system would make the call and connect the call to a SIP phone (Grandstream Budge Tone-100). Is there anything out there that can do that? Thanks, Dave -------------- next part -------------- An HTML

I have a quick question. I am using Asterisk 1.4. I have a user that has changed phones (grandstream budge tone 200 to a polycom 330). I have changed the sip.conf and extensions.conf. I have also unplugged the old phone and plugged in the new phone. I get the ext showing on the phone, but when I do a sip show peer 5000 the old ip address and phone show up. I did a sip reload and a dialplan reload.

Hello: Hoping someone might know how to resolve this (probably an easy one). I have one Asterisk PBX with a single NIC and an FXO card with PSTN line attached, and one IP phone (Budge Tone 100) on the LAN. Via the phone I get no dial tone, and dialing 9, <number> doesn't allow me to dial out. I also need the phone to ring when the asterisk PBX is called. I have modestly tweaked the

Hi, I'm using the current version of Ubuntu, and the Wine version the system down loaded. I'd like to use my photo application "Fixfoto" but for some reason I can't get it to work. I can use some apps e. g. XnView, or IrfanView without any problems but for some reason I can't get Fixfoto to work with Wine. I've installed Fixfoto using Wine but when I click on the icon

Hi I've been using Asterisk for a while now with the TDM400 and it seems to be working fine. I'm using version 1.2.2 and I've struck a problem when I added a Budge Tone 100 SIP phone to the network. The phone rings when calls come in and I can make calls but in all cases (internal or external calls) the other party cannot hear me even though I can hear them. I'm sure I've

Hi I am using asterisk 1.4 on RHEL4 I am sending this mail to the mailing list , to enquire wheter any one had faced simillar problem which I am facing now I am facing a problem which is not able to solve or understand , the problem is that I cannot log into the mailbox from a VoIP hardware phone , while I am able to login to the mail box using soft-phone for the same users

Le 26/04/2020 ? 08:31, Gordon Messmer a ?crit?: > Are you sure the input lacked a space character, and that this isn't just a > font rendering bug? 100 % positive. I tried it on a different sandbox server without X installed, but only with xorg-x11-xauth. >From my workstation running OpenSUSE Leap 15.1, I connected to my admin console using the following command: $ ssh -X