similar to: Grandstream Quality Survey.... :P

I have upgraded my grandstream phone from firmware 1.0.3.78 to 10.0.4.30 and now I am having problems with early dial. On the older firmware earlydial worked fine with my asterisk server, but now as soon as I have dialed the number I get a congested tone, and the number 4 flashes up on the LCD screen. Has anyone had this problem, and if so, how do I fix it? -------------- next part

I thought I'd post my experiences for the benefit of anyone else who may be at the point I was when I first started with asterisk. I have 2 incoming analog lines (north eastern U.S., Verizon) where one is set to ring if the first is busy. I bought a bare-bones system from abs-pc with the following components: POWER SUPPLY 450W ALLIED ATX450P4 R(41) MB NFORCE2 A7N8X DELUXE ASUS RTL(Standard)

Is there anyway to turn off the call waiting beep in the grandstream and/or cisco ata186? I have a dial statement in my extensions.conf that rings 5 phones at the same time by combining them with the & in the dial statement. i.e.) exten => blah,blah,Dial(SIP/GS1&SIP/GS2&SIP/GS3&SIP/ata186a&SIP/ata186b,25,t) If one of those lines is being used, then the user gets a really

Hi, I've tried the latest CVS Zaptel and Asterisk after following the Echo fix threads. But echo is the same if not worst. Has anyone managed to alleviate their echo from these latest changes? --------------------------------- ALL-NEW Yahoo! Messenger - sooooo many all-new ways to express yourself -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm having a horrible experience getting a Cisco ATA-186 to work with *. I can make calls from the ATA with no problems. However, incoming calls make the ATA ring once, and then the call is disconnected. I have no problems with my Sipura 2000 or my Grandstream phones. I am running 2.16.1 sip code on the ATA 186. Neither * nor the ATA is behind a NAT. They are both on public IP addresses

I have an asterisk server running on Redhat 8.0 with a Digium TDM400P w/4 FXO modules (TDM04P) There are 2 lines going into the Digium card. One line is a Vonage digital line, and the other line is a Comcast voice line. I have a SIP Grandstream 100 phone connected to the Asterisk server. I also have IAX configured with FWD. The problem is that on occasionally, after talking for about 20

Hello! Is there a phone compatibility list anywhere? I know Asterisk is supposed to be compatible with IP phones that support SIP, h.232, and IAX, but a list of phone known to be supported would be a nice addition to the documentation. I need a buisiness phone, and the Cisco is _EXPENSIVE_ I like the 3com 1102, but it is NBX phone, and I can find no documentation of this phone working

Hello, today I found this security report regarding Asterisk SIP Security. http://www.securiteam.com/securitynews/5LP0720B5G.html Maybe It could help somebody who isn't using a newer than 15th of August cvs version. Best regards Lubo

Hi all,

Is it safe to assume that a fresh CVS build will not have the SIP translation problem described? Regards, Christopher --__--__-- Message: 11 Date: Mon, 29 Sep 2003 12:45:40 -0700 To: asterisk-users@lists.digium.com From: "Ernest W. Lessenger" <ernest@oacys.com> Subject: Re: [Asterisk-Users] Is somthing broken? Reply-To: asterisk-users@lists.digium.com At 12:33 PM 9/29/2003, you

Hello, after upgrading my asterisk from cvs few days ago I discovered that all my anddefault asterisk voice messages/prompts was played wrong - they was played so slowly ... and the IVR system wasn't usable at all. Today I upgraded everything: a fresh cvs copy, made make install and make samples - but after that the problem with IVR was the same: ALL default asterisk sounds in gsm format

hi, can some one tell me a good IP phone (not software, but a "real" phone :) that work well with asterisk? how mutch does it cost a good IP phone? i made a VoIP network for my company, but now we are using a client for PC phone... i'd like to buy a IP phone, can someone tell me witch model i should buy? thanks, Angelo

Hi I have a brand new ATA186 with the following firmware: Version: v2.15.ms ata186 (Build 020919a) I have been through the archives about how to configure it, but my colorful configuration web page does not have the same fields that people say I need to adjust. Even the examples on Cisco's web site don;t match. For example, I don't have the GtkOrProxy field, which is an important

I don't mean to be a troll in any way shape or form. I was on IRC last night and I observed the following convo. below. What do you guys make of it ? [02:14] <bkw__> Let me tell you how chidlish digium and Mark Spencer is. I walk into a restaurant with them all here at Astricon wearing my sangoma shirt and he asked me to leave. [02:15] <Dovid> u serious ? [02:15] *** mog

Hello every one, I have got a H323 gatekeeper for testing. The informations are something like this: account code: test01 gk ip address:192.168.10.12 I don't know how to set it in the h323.conf or oh323.conf, I have tried it for almost one day but I always got the error. Help me please. Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL:

If you have been paying attention, you already know this, but this weekend I have spent time ironing out the various details with my chan_skinny code that has been out there, if you knew where to look. I believe I now have all basic features operational and am going to be working on getting the class 5 (hold, transfers, call waiting and caller*id, etc) operational in the comming week(s).

Hi, I am a beginner to Asterisk. Can anybody clear my following doubts regarding the configuration needed? 1) What is the ideal system configuratin required?(like processer, RAM, h/d space etc) 2) How many connections it can handle at a time? 3) How many Virtual PBXs it can handle? 4) Whether Postgres or Mysql is best suited? 5) How many IVR's it can handle simultaneously? 6) How many

Guess who's here to do an Asterisk demo this week without the power supply for his SPA-841. I have an ATA186 with me. Both phones use a 5v supply. Does anyone know whether the supplies are interchangeable? Thanks in advance; sorry for the noise. B.

-----BEGIN PGP SIGNED MESSAGE----- Greetings, I've just about got Asterisk up and running and am wondering the following. Currently, I subscribe to both Vonage and Broadvoice and as such, I've got a Sipura and Cisco ATA186. Although I'm sure this is expressly prohibited somewhere in my service agreements, can I reprogram these devices to access my own asterisk server rather than

Hi All, I'm looking at getting some Cisco VoIP hardware to play with in combination with a Asterisk server. I've heard that there is beta software available to do SIP on the 7905G. So, I'm thinking of either getting a 7905G or a ATA186. My dillema is, which one to buy? I can get both for about the same price, has anyone had any experience with using a 7905G with Asterisk? On