similar to: Multiple SIP Phone Calls Overlapping on the Same Phone

I am trying to create an IVR system that uses both POTS and IP phones and I have a few problems that I encountered with the IP SIP phones (Grandstream Budge Tone 102). 1. When a user hits the hook fast enough, the user can create multiple IVR connections that gives the appearance of an echo that is phased a few seconds apart. The way to reproduce this is by hitting the hook fast and furious. The

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

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, 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 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

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

Hello, all. Alas, another convoluted question. All the simple things are, well, simple so I suppose we only need to trouble the list with squirrely problems! We've noticed a call history problem when using Asterisk where the call history on the Snom phones (with which we are very pleased) reflects the number of the PBX extension used by the B2BUA to dial the end point. I assume the same

Hello, Anyone have any luck getting storeconfigs + MySQL working? I was getting some freaky uninformative errors: err: Could not call: wrong number of arguments (1 for 0) It''d be great if we could get a wiki page up, there appears to be a pretty large consensus of people who are unable to get it running. Regards, AJ

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 have a Dovecot installation on a Fedora 4. With the last update from Dovecot RC10 to Dovecot RC27, the daemon kills itself every night with this error: Mar 17 05:23:11 mail dovecot: Time just moved backwards by 6 seconds. This might cause a lot of problems, so I'll just kill m yself now. The error appears just after the night maintenance script executes a ntpdate syncronization ntpdate

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Greetings again folks, One of the things I like about Fedora Core is the fact that I can add apt as a repository tool. Has anyone done this for CentOS-3? Why was yum chosen? I prefer apt over yum because of features and was wondering if anyone had created an apt repository for CentOS. Thanks in advance for the info, I'm not trying to start an

I''m backed off Edge Rails and all the way back to 0.7.2 of rSpec to make my tests pass. The changes to rSpec are coming together fast and furious and they seem too good to pass up. Is there an approved way to upgrade the components in the absence of prebuilt stuff? Every time I''ve tried, I''ve been punished harshly by the Version Conflict Gods. Just asking for

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

https://bugzilla.netfilter.org/show_bug.cgi?id=987 Bug ID: 987 Summary: nf_conntrack_reasm.c : Silent discard of overlapping fragments is not silent Product: netfilter/iptables Version: unspecified Hardware: x86_64 OS: Debian GNU/Linux Status: NEW Severity: normal Priority:

I had meant to copy the list on this; must have hit 'Reply' instead of 'Reply All'. P Ehlers -------- Original Message -------- Subject: Re: [R] rbind with partially overlapping column names Date: Mon, 16 May 2011 11:14:11 -0600 From: Peter Ehlers <ehlers at ucalgary.ca> To: Jonathan Flowers <jonathanmflowers at gmail.com> On 2011-05-16 08:56, Jonathan Flowers wrote:

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