similar to: PCM Voice Quality Issue on CVS Version

Hi, Do the g729 codec licenses for Asterisk work on a SIP environment (only SIP UAs running g729 + Asterisk)? I would like to buy a couple for a SIP test lab but I have not found any documentation on wether it works for SIP UAs or not. The Digium page only mentions: "The G.729 codec works with all Digium cards." Can somebody tell me please? Thanks, Ricardo Villa

I've just had the most appalling performance from * ever. Dialling: Cisco 7960 => asterisk => IAX produces sound drop outs so extreme that the call is useless. I noted this in an earlier post. Dialling: Cisco ATA186 => asterisk => IAX is fine. Frankly, I think this is such a bad problem that it should be sorted in advance of any of the new features that seem to be

Hello, We are using asterisk 0.4.0 on debian sid with Cisco 7960 and ATA186 phones. All sip entries have: callgroup=1 pickupgroup=1 However I am unable to remotely pickup a ringing phone using *8#. I get fast busy tone. Is there some flag to add in extensions.conf ? Thanks in advance,

Checking the earlier mails, it stated that the phones were $75 (100) & $85 (102) ref :- http://lists.digium.com/pipermail/asterisk-users/2003-June/013483.html Well, I just called Ovislink/dgtimes and was quoted $90 & $100 and the person said there was no price change. Anyone on this list actually bought them at the $75 & $85 rate ??? Regards...Martin -- Too much is just enough.

Hello, I have had several instances over the last month of Asterisk freezing, sometimes after 12 hours, sometimes after 8 days. The common elements are that: - all Zap channels lock[hangups don't register and no new calls in or out] - no new in/outbound calls can be made on Zap or SIP channels - people who are still connected to calls can continue to talk - in the CLI interface, you can

Hi everyone, I am interested in estimating this type of random effects panel: y_it = x'_it * beta + u_it + e_it u_it = rho * u_it-1 + d_it rho belongs to (-1, 1) where: u and e are independent and normally zero-mean distributed. d is also independently normally zero-mean distributed. So, I want random effects for group i to be correlated in t, following an AR(1) process. I am

Today class we are going to be talking about the wonderful line of GrandStream products. Or should I say BarbieTone phones? Who else is having MAJOR issues with the grandstream products? How many times have you been told upgrade upgrade upgrade? How many of you have paperweights, granted the phone is light as a feather and couldn't weight papers down in the first place? How about that

Symptom: Phone after about 15mins will stop functioning Problem: DHCP lease renewed but default route dropped Fix: Assign a static ip and problem is resolved. Upgrade to new firmware once it is released It turn's out that these phones have a few issue in 1.0.3.81 firmware. The phone may stop transmitting packets if configured with DHCP, if DHCP is being provided by certain devices. Netopia

Upon testing the g.729 licenses in our lab we found the following issues in a plain SIP environment: 1. When g.729 licenses are used in leaving a Voicemail, they do not get released upon clearing the channel. 2. If the voicemail.conf has options for 2 format types (ie. wav49 and WAV), it consumes 2 g.729 licenses. I have reported this to the bugtracker. maui*CLI> show version Asterisk

Hi, My name is Joseph Villa, I'm new to the message boards and I'm also new to Samba. I just got an e-mail back on our Nessus scans.. Here are the 2 that are relivant.. 1.) The remote host has accessible LOGS$ share. ScriptLogic creates this share to store the logs, but does not properly set the permissions on it. As a result, anyone can use it to read the remote logs. Solution:

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

Hi, I configured asterisk on redhat linux 9 box. I installed two different ip softphones (SJPHONE and X-PRO) and got them registered with asterisk. The call from one phone to another does get routed via asterisk, but there is one problem coming up. As soon as call is accepted by the end user , it is automatically disconnected with the error "cannot align media streams". If I enable SIP

Hi, I'm away at a conference in Amsterdam. My home is in Cambridge in the UK. On a whim, I tossed an ATA186 and a phone into my bags before leaving home. I was able to plug my ATA186 into a LAN here at the conference and was connected to my home Asterisk in a few seconds. Total time from unzipping my bag to talking to home no more than 15 seconds. OK, so the kit could be more portable,

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

Ahh Mr Carbuyer ... you should have _specified_ you wanted tires with that new car We can still help you though, it will just be an extra $$ above the price we quoted you I understand the concept. I see it in many industries until a company comes along that cares about it's customers I still think that digium is the best buy (for the small scale stuff that I'm interested in anyway) ...

Hello Everyone - Well, I think I'm getting closer with the asterisk connection. This is my setup and I keep getting this error below in ,my /var/log/asterisk/messages file. I have opened 5060 port on the firewall box. I would this is Warning which I can ignore! But I see the connetcion coming but NO DIAL TONE on mt ata186 box sitting in my 192.168.200.x site! I'm using ATA186(cisco

Did Linksys really use Asterisk for the SPA9000 software? -- Andres Technical Support http://www.telesip.net

Hello, I am interested in estimating this type of random effects panel: y_it = x'_it * beta + u_it + e_it u_it = rho * u_it-1 + d_it rho belongs to (-1, 1) where: u and e are independently normally zero-mean distributed. d is also independently normally zero-mean distributed. So, I want random effects for group i to be correlated in t, following an AR(1) process. Any idea of how

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

I've configured Asterisk 0.7.2 to work together with Cisco ATA186 (SIP,G.711. RFC2833) and OpenPhone (H.323, G.711). But there is an issue while calling from ATA186 to OpenPhone via Astrisk - when I press any key on analogue phone connected to ATA, Asterisk shows following message: -- Executing Dial("SIP/519-3781", "OH323/62.213.36.100|20|Tt") in new stack --