similar to: switching trunks based on quality

Does anyone out there have a recommendation for tools that will monitor the quality of VoIP systems? I am looking for jitter and MOS monitoring. I have a custom Nagios plugin that is alerting me if the jitter jumps out of a 20ms but I am looking for a little more detail. I would not be against writing something in Perl for Nagios to do but I don't really know where to start on measuring jitter

I am having an issue with doing FTP auto provisioning of Polycom 501's when they are behind a NAT. If I put the phone on the same subnet as the provision server it loads the configs and changes fine but as soon as I put in behind a NAT it comes up with cannot contact boot server. I have tried behind and replicated this behind a PIX 501, a Linksys SOHO router and a Motorolla SOHO router.

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Just emailing the list to see if anyone out there has used Vitelity? If so what has been your experience with service, support etc? Thanks Curt

Thanks Curt, that's "too cool for school", any idea on when this is coming to the MS SBS platform? I use SBS for myself at home and would love that level of functionality included. Does Asterisk therefore handoff voicemail storage etc to Exchange for this level of integration? Cheers, Dean ________________________________ From: Curt Shaffer [mailto:cshaffer at

I put iptables on my asterisk box and an odd thing occurs. I allow 5060 and 10000-20000. As soon as I start iptables and make a call it literally takes 60-90 seconds before the call even starts to ring. As soon as I shut iptables off, the call goes through immediately again. Its quite odd. The call does eventually go through and talks fine but it takes sooo long to connect. Anyone have some

I have been working with a couple companies who are interested in integrating Cisco VoIP (mostly call manager express) but utilizing Asterisk for AA, VM, failover trunks etc. I have found some materials and guidance out there but I think a list and/or wiki for general asterisk integration with other vendors would be great and feel that it is enough off topic that it deserves its own space. Just

I wanted to see if there was anything reasonable in price out there yet that performed an RF to IP bridge via asterisk. What I mean by this is callers from PSTN can be patched to a UHF/VHF radio and vis-?-vis. I know there is an option available for the Avaya systems but it?s a little out of the price range I?m looking for (~$200/channel). Has anyone out there found a stable way to do this?

I have decided to move on from Asterisk@Home and start compiling asterisk myself now. I got a dedicated box and put my X100P in it. I installed the server version of CentOS 4.2 (2.6.9-22.EL kernel) barebones. The box is a dell GX270 workstation with 1GB of RAM. I got a fresh copy of O'Reilly's Asterisk the future of technology and begun. I downloaded the zaptel-1.2.4.tar.gz, libpri-1.0.9,

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 am trying to perform a "rollover" when the primary number is busy. This is coming from a POTS line. Apparently I need call waiting on the POTS line as I get immediate busy from the FXS if I don't have it. So my question is this. I have an Aastra 480I CT. The call forward when busy here seems pretty straight forward. Choose the mode as busy enter the extension in the forward number

I am using asterisk version 1.2.6 with Zaptel version 1.2.5. I have a POTs line coming into a Digium TDM01B. It appears to not be getting CID at all. If I hook up a POTS phone to the line CID comes through fine. Inbound and outbound calls work fine but there is just no CID on inbound for this channel.The incoming route for the channel is Zaptel Channel 0. No DID or CID settings applied. My IP

I am having the same issue as below. Has this issue been solved or does anyone know an answer? This error recently began and we have multiple phones out of commission. PLEASE HELP!! http://lists.digium.com/pipermail/asterisk-users/2006-August/162841.html How did you find out about 468*??? It's sure as poop not documented in the Polycom Admin Guide anywhere. -----Original Message-----

Hi, Some odd behaviour here. A friend and I were talking tonight, and it seems we have both seen the same problem. We are both using aastra phones (I am using 9113is). We have a connection to and from providers via SIP and IAX. When I place a call on the local hold of the phone, and then pick them back up I can hear them, but they can not hear me. However, if I park the call,

The TDM01B is 4 port capable but has only 1 FXO module. I'm running asterisk 1.2.7 and zaptel 1.2.5. I cannot seem to get the TDM01B working. When I do the zttool it shows 4/1/0. I can dial out from a POTS phone up to the point that the cable plugs into the card. Here is my /etc/zaptel.conf loadzone=us fxsks=1 and here is my /etc/Zapata.conf [channels] language=en #include

I have been struggling over central provisioning for quite some time. I have eagerly watched each post with like problems but have yet to find a reliable answer. I have a Polycom 501 and I am trying to provision from an FTP server, and just to take any routing out of the issue it is on the same subnet. I am running the 2.0.1.0291 firmware and 3.2.2 bootrom. I set the IP info on the phone and

Sorry if this shows twice but it appears my first message was quarantined because of my digital signature. All, I have been tasked with setting up video conferencing utilizing asterisk. One of the requirements is a softset that has video capabilities. Eyebeam looks promising but I was just wondering if anyone out there knows of any freeware with comparable features of Eyebeam that they

Just curious... Is there anyone out there who has given this outfit money and actually received any service from them?

I am experiencing a situation where I am getting intermittent choppy audio. Here is the network layout: Termination provider -> IAX2 over the Internet -> 20Mb fiber connection -> router -> Asterisk My ATA connection goes into the router between the fiber and the Asterisk server on another interface here is the layout from me to Asterisk: Sipura ATA (SPA1001 running

Does anyone know of a really lightweight web interface that allows users to log in and modify attributes of their extension only? Thanks Curt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061030/892a67b2/attachment.htm