similar to: ECHO Tutorial

We are having trouble with an installation that is getting a lot of echo on some calls. The installation is all SIP phones and they have a VoIP provider. When we call through the voip provider and into another of their customers (voip throughout) there is no echo problem. If we call in their landline, through the TDM400's FXO to one of the SIP phones, there is no echo problem. Sometimes

Hi, Can anyone tell me what should be included in h323.conf to get asterisk to talk to a Cisco 2600 gateway? Any statement examples for extensions.conf would also be appreciated. Thanks. Will chan_h323 use the Cisco as a gateway anyway? Regards, Steven Thomas

I have read different posts over the months wondering who Vonage uses for their VoIP technologies. I stumbled across this article (although it's from 2002, I think) that suggests strongly that they use Cisco. There is no telling what they might use in conjunction with this but this should clear some of the conjecture.

I have a client with 20 GXP-2000s. Everything seems to be working fine. However, after a couple of weeks of use, the client is having a hard time adjusting to the new IP based phone systems and only misses one feature from their old Lucent system. That is, they had 8 analog lines before and all their old Lucent phones showed a button for each line. So, it was easy for anyone to say,

Hi All! First, let me introduce myself, as this is my first post to the list (I've been lurking for quite some time now). My name is Matt Hardeman, and I work for a software development firm in Birmingham, AL. We are interested in the Asterisk PBX and it's various configurations first as an internal solution for our occasionally bizarre telephony needs, and eventually are interested

Hi, I can't upgrade an old 7960 from SCCP 3.0 to SIP 7.5. Could you help ? From http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080094584.shtml#sccptosip2, I got the following: 1. Copy the desired binary image from Cisco.com to the root directory of the TFTP server. 2. Specify the image in the configuration file image parameter for the protocol

Is there any tool that can do LCR for Asterisk but also take into account MOS scores? Is it possible to automatically generate MOS scores on random "calls" so as to keep an updated database on a per provider, per destination, per time-of-day score? Hopefully, with that information we can create a better LCR module or script? Thanks, Daniel

Hi! Can anybody help me with convert Cisco 7940 CallManager Phone to a SIP Phone? I have continious error in tftp log: connect from 192.168.1.111 Mar 18 12:12:30 AKrasavin utftpd[10081]: peer requests OS79XX.TXT, conversion octet Mar 18 12:12:30 AKrasavin utftpd[10081]: unterminated option value in init packet Mar 18 15:12:30 AKrasavin xinetd[10068]: START: tftp pid=10080 from=192.168.1.111 Mar

There's an excellent tutorial on Cisco's web page at http://www.cisco.com/en/US/tech/tk652/tk701/technologies_white_paper09186a00800d6b68.shtml It will tell you just about everything you wanted to know about echo and more :) The short answer to your question, however, is that echo is comprised of two components: volume and delay. Increase either one and the problem gets worse. In the

Hi, We are currently trying to setup Asterisk with iBasis. One question/problem we have is that Ibasis has told us to send the INVITEs to one IP address and all media to a different IP address. How can we do that in Asterisk? Thanks

Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to

Does FreePBX support virtualization of its services? For example, can I use it to provide virtual PBX to different clients under the same instance of FreePBX? Or is it more geared to single office-type installation? Thanks, Daniel

We are looking for an offline billing solution. We have a couple of particular requirements: 1) Since it's offline, we need to be able to import the CDR. 2) A way to support account credits based on referrals. Meaning, that if a member refers a new account, that member would get a free month of service, or similar type credits. 3) Generate invoices in either HTML or PDF format so they can be

I'm using a few GXP-2000 with firmware 1.0.2.13 and everything seems to be working fine. However, there are a couple of issues I'd like to know if are possible: 1) Even though the phone has 4 line appearances, if I am speaking on a line, the phone can no longer receive phone calls. I can manually select another line and make calls, but when Asterisk tries to send a call to it, I

Is there any documentation you guys can point us to in order to learn more about the proper use of the Local channel? We don't currently use it. However, while evaluating other people's billing and management systems for Asterisk, we noticed they make extensive use of it. Thanks, Daniel

Hi all, I just bought the 7970G phone. It's a beautiful phone. In trying to make it work with Asterisk, I've read many posts on the net. However, all of them make reference to having to install the SIP firmware on the phone. Where can I get it? Thanks

I just upgraded my 7970G to the SIP firmware. What I'd like to do is have the 8 line buttons be able to make outbound calls using the same account (for practical purposes, same caller-ID). Since the phone is going to have a single public DID, when a call comes in, it should ring on the first available line. So, if I'm on line 1 and a call comes in, it should ring on line 2. How can this

Hi, I'm trying to have a SER machine send calls to an Asterisk server working as an IVR. I was able to do this part just fine. Also, when the caller makes certain options in the IVR, the call is then transferred to an extension via SER. This part is also just fine. However, I'm trying to get Asterisk out of the media path once the caller has made a selection in the IVR. Can

I try to use meetme app after reading manual i compile and install zaptel with ztdummy when i make lsmod i have ztdummy 2532 0 (unused) wcusb 20064 0 (unused) zaptel 179168 4 [ztdummy wcusb] usb-uhci 26348 0 [ztdummy] usbcore 51616 0 [wcusb usb-uhci] after it i recompile asterisk and after it i have

I was just wondering if the original audio is always the best audio. I'm sure every compression format including vorbis is based on trying to make the output sound the closest to the listener to the input. I was wondering if there is any possibility that there would be a way of modifying the huffman tables or something in some way to make the output sound better than the original? ---