similar to: Polycom autoprovision behind a NAT

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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

What is everyone out there doing in an all IP termination environment to change trunks when quality drops to a certain provider automatically? Thanks Curt

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

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 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?

Dear List members, After succesfully installing the * on a couple of systems, and putting them on test, I observed that there is an intermittent call drop on PSTN line. The systems are - Dell Optiplex P3/500MHz/128MB - Built-in ethernet - 1 X100P (Motorolla chip) card on PCI - 10G HDD etc. - Asterisk April 17 CVS. - 2 Mediatrix FXS ATA (4 phones) - 2 Grandstream phones. - sip.conf, zaptel.comnf

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

Hi there, We have plans to install an office (not call center) with the following setup: 200 linksys 942 phones (sip + g711) on a LAN a server with a dual port E1 sangoma and a remora card with 4 fxo modules. So far when we want to setup a linksys phone, we need to use the http interface of each phone, disable/enable a lot of things and plug it into the network. this is not the best scenario for

Hi, I want to setup vsftp on centos 4.5. 2 types of ftp. they are active and pasive. What is the default type ftp type o CentOS ? Is is PASIVE ftp? in /etc/vsftpd/vsftpd.conf, I found below line. What should I do for below line? # Make sure PORT transfer connections originate from port 20 (ftp-data). connect_from_port_20=YES Should I COMMENT it out? I need a very secure ftp service. YOUR

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 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 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 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 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

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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 all I have a few Polycom 331's but after following allot of advice I can't get them to provision from a dhcp boot server. We have a sonicwall router in place. I can press setup and set the FTP boot server to my * box. From there th phones boot fine. But I cannot get them to autoprovision. I have tried dhcp option 66=ipaddres. 66=FTP://PlcmSpIp:PlcmSpIp at 192.168.1.1/ u ahve also

I've just started deploying these (well the T28P model) after years of Snom issues and they look pretty good (although the documentation is execrable; if you thought the Snom stuff was obtuse Yealink have got them knocked into a cocked hat!). Anyway, for provisioning I use HTTP with a DHCP entry like:- # # Yealink Phones # group { #

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