similar to: Budge Tones pick up wrong calls

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

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; 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

I'm following the instruction from Jo?o Amaro from the page http://lists.digium.com/pipermail/asterisk-users/2005-February/090752.html Everything went fine until I run the 'make' command under asterisk-oh323-0.6.5. I got the error message chan_oh323.c:5220: too many arguments to function `ast_channel_register' I have attached the error message. I'm running asterisk CVS

https://bugzilla.mindrot.org/show_bug.cgi?id=2518 Bug ID: 2518 Summary: docs: remove redundant lists of algorithms Product: Portable OpenSSH Version: 7.1p1 Hardware: All OS: All Status: NEW Severity: enhancement Priority: P5 Component: Documentation Assignee: unassigned-bugs

Dear All It is a little hard to give a good small example of my question,so I will show the full data on the bottom and the attachment.Maybe some one could tell me an appropriate way to show it.I'm sorry for the inconvenience. Q:How to generate a 53*53 diagonal matrix by my data Some problems confused me are that: 1.Since it is a diagonal matrix,I have tried to transform col1 and col2 to

https://bugzilla.mindrot.org/show_bug.cgi?id=2501 Bug ID: 2501 Summary: VerifyHostKeyDNS & StrictHostKeyChecking Product: Portable OpenSSH Version: 7.1p1 Hardware: All OS: All Status: NEW Severity: normal Priority: P5 Component: ssh Assignee: unassigned-bugs at mindrot.org

https://bugzilla.mindrot.org/show_bug.cgi?id=2536 Bug ID: 2536 Summary: X11 forwarding slows down client Product: Portable OpenSSH Version: 7.1p1 Hardware: Other OS: Linux Status: NEW Severity: enhancement Priority: P5 Component: ssh Assignee: unassigned-bugs at mindrot.org

A new problem has cropped up with compiling vignettes for my package BayesFactor. I'm not sure when it started, but I can tell you it didn't occur on R 2.15.3, and it does on 3.0.0 RC (2013-03-31 r62463) (session info is at the bottom of this message). I have defined methods for which.min and which.max for a class (I've defined both S3 and S4 methods for the class

Greetings, Perhaps someone can offer some advice on how to get this working. I will detail my process and whats been done. on the XP side of the house (client side) 1. XP_signorseal registry hack applied, value at 0 now. 2. following values set to disabled: domain member: Digitally encrypt secure channel data (always) domain member: Digitally encrypt secure channel (when possible) domain member:

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

Hey Steve and everyone, I looked at the configuration, and unless I am missing something I don't think they are configured # ztcfg -vv Zaptel Configuration ====================== Channel map: 0 channels configured. In the zapata.conf file, it is the sample version, but I didn't notice anything in there that related to what you said. Or is it in a different file or location? I am

Greetings! I loose all my metadata after concatenating corpora. This is an example of what happens: > meta(corpus.1) MetaID cid fid selfirst selend fname 1 0 1 11 2169 2518 WCPD-2001-01-29-Pg217.scrb 2 0 1 14 9189 9702 WCPD-2003-01-13-Pg39.scrb 3 0 1 14 2109 2577 WCPD-2003-01-13-Pg39.scrb .... .... 17 0

Dear all, I have encountered a strange problem with read.table(). When I try to read a tab delimited file I get an error message for line 260 not being equal to 14 (see below). Using count.fields() suggests that a number of lines have length not equal to 14, but not 260. Looking at the actual file, however, I cannot see anything wrong with any lines. They all seem to have length 14, there

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

[Dec 31 10:39:45] WARNING[17884]: pbx.c:2518 __ast_pbx_run: Invalid extension '11', but no rule 'i' in context ...[snip]... When testing IVR and pressing "1" from my Grandstream SIP-phone, the above message is printed on the Asterisk CLI. How come Asterisk receives my "1" as "11" ?? Settings in my SIP-phone are : Send DTFM : via RTP(rfc2833) &

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

Hi, I am using asterisk-1.4.15, My sip configs is like [2501] type=friend username=2501 secret=2501 canreinvite=no host=dynamic dtmfmode=rfc2833 context = sip disallow=all allow=ulaw incominglimit=1 nat=1 queue.conf is like [gen-enq] joinempty = yes musiconhold = default strategy = rrmemory servicelevel = 60 timeout = 60 retry = 5 wrapuptime=5 announce-frequency = 90 announce-holdtime = yes