similar to: Budge Tone-100 as a Ext in the LAN

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

Hello, I have a problem with one of three [topic] phones. The phone, which is on the LAN in the same subnet as Asterisk, keeps unregistering from the Asterisk server. Whan it is unregistered there is no way to make a phone call from it, but once it is rang by any other of the phones it registers to Asterisk again. The other two are absolutely fine. The problematic one [ecco] puts this messages

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

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

Hi Once I installed astrisk , how do we connect with SIP interface ? Can somebody guide me how to integrate SIP interface with asterisk ? I want to use Astrisk just for IVR purpose. Thank you Luke -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130511/a6d32bfe/attachment.htm>

hi users! i'm trying to Install a new version of samba (samba-2.2.8a) since tree days! i have a Suse linux 8.2 Server and in my Server run alredy samba-2.2.7 ( smbd and nmbd run in /etc/samba) I 'v installed and configured now samba in the /usr/local/src/samba-2.2.8a/ directory like this : ./configure --sysconfdir=/etc/samba --localstatedir=/var/log/samba --mandir =/usr/local/man

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 all, I have migrated with some tables in my application. After some days i need to add few more tables to database. The initial tables in database have some data. when i try to migrate the database for second time with rake db:migrate it is saying the alredy table exist <table name> and rake is aborted. i dont want to loose the old data and i want to add new tables to alredy exiested

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

Dear All, its a offtopic question but really apprecite if someone would advise n help i have been running a mil server with sendmail and have sbl-xbl.spamhaus.org as my dnsbl. i had other servers which are alredy out now that is relays.ordb.org and dsbl.org have already been out of my sendmail config. any one knows of ny other servers i could add in my sendmail config apprecite ur help

I´m looking for a way to work with compose primary keys. I alredy know that plugin http://compositekeys.rubyforge.org/ but it doesn´t have examples. Please help =] -- Horus Augustus C. C. Lima Sagarana Tech Mobile +55(85)8842.4402 Desk +55(85)3304.6530 augustus-b3GZhX+fmr6XmMXjJBpWqg@public.gmane.org http://www.sagaranatech.com --~--~---------~--~----~------------~-------~--~----~ You

Hi all, I'm writing my graduation theses : analysis VO-IP protocols , and I cannot find any documents about Cisko Skinny Client Control Protocol. I have Cisco CallManager and some IP-phone and I'm sniffing traffic between that, but I don't understand, how this protocol works. Clearly i'm looking for description of SCCP commands and explanation some basic SCCP scenarios or what

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

I'm starting out w/ a Cisco 7940, running the Sip image version 6.3. I've downloaded/installed asterisk via cvs. I've set the phone up to get its info via dhcp - the dhcp, tftp, astericks box & phone are on the same network. I've gone through and setup a test account per the instructions @ http://voip-info.org/wiki-Asterisk+phone+cisco+79xx but time I do a sip show

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