similar to: Need a good extensions.conf sm bus config w/polycom phones

Hello- We've got a patched-together extensions.conf that's barely working for us, and we need to get real about using Asterisk. We've got a couple of remote workers with Polycom IP-601 phones, and a single asterisk server, using a couple of incoming DIDs from teliax and sixtel. Does anyone have a good extensions.conf that they'd be willing to share, that provides a

I posted a week ago to see if anyone could share a good, or great, dialplan for a small business who wants to make good use of some polycom 601s. I received a pointer to some the loligo dialplan, which is nice, but doesn't help me with the polycoms. And nothing else, except a number of "please share if you find one" requests. Is nobody willing to share? Thanks- Tom

I am considering using iax.cc (sixtel) and wondering if anyone had opinions, good or bad. Are there outages with any regularity? How responsive are tech support? How is packet loss? I am particularly interested in termination to the UK, but will accept any comments people have. Thanks -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US

Hello, I'm looking for US DID and US50/CA 800# Providers. I found voiceconduits.com 8 month ago, there interface looks good, but there are still not live, I believe they won't be any time soon. I found sixtel, but order take eternities, they probably won't get my orders right any soon. So i'm looking for a good provider for DIDs and 800# from the US and CA, who offer online

Hi everyone, I have been dealing with a certain issue with a particular customer site for months now. The problem occurs when there is an error with caller id as shown in the following: WARNING[16036]: chan_zap.c:6309 ss_thread: CallerID returned with error on channel 'Zap/3-1' When this happens, it appears that the call still goes through as I can see the caller still navigating

Hi Everyone, I have a DID number with livevoip, but I have been experiencing two problems that I can't seem to resolve. I am not sure if they are in any way related. I have other DIDs with iax sixtel but I do not have that problem. Livevoip seem to think that the problem might be with my configuration. Can someone help me figure out this problem please. 1) When an incoming call to my

I know the Polycoms work with NAT, but you have to specify the public IP. Is there anyway for it to discover the external IP automatically? I like the phones (been playing with a 301) but for some of our clients who have a dynamic IP (and no hope of getting a static ie cable or residential DSL) I'd be afraid to use them since you have to specify the IP. What about the Cisco phones?

I was just checking out the dids for all of my fail over providers and noticed that neither DID that I have with SixTel work. Both pause for a long long time The local number gives a recording: 'The number you have dialed is not in service or is assigned in a different area code. Please check your number and dial again'. The 800 number just rings busy. Anyone else having this issue or

Aloha, Does anyone have a good source for Polycom SoundPoint? IP 600/500/300 phones? Everyone sells Cisco 79XX. Aloha, Matt

Whoa, talk about flying under the radar. I got a few DIDs *months* ago from Sixtel (or iax.cc). Initial respone was great, but then it seemed that the only tech-support person had fled the country. No responses, bad responses, poor call-quality. I had a few $s left in the balance and kind-a just forgot about them. Not worth 10 minutes of my time to get $10 of my money back, quite frankly.

I am trying to do the following: 1. Call comes in to my * box over IAX (VP Connect DID) 2. Check to see if call should be forwarded to my cell 3. Forward the call to my cell phone and take * out of the media path. I am able to do all of the above except * is not able to natively bridge the call. I am using sixtel and for the call forward portion, but the calls don't connect before sixtel

Hi All, Has anyone gotten the polycoms or the linksys phones to accept oprtion 66 on the dhcp request for the address of the tftp config server? We have the dhcp server issuing the proper IP of the tftp server, but the phones just sit there and never try to contact the tftp server for their configs. We can see the proper option going from the dhcp to the phones with ethereal trace. Thanks JR

hello list, I subscribe to Sixtel.net for a DID just to see how it worked. they say DID active inmediatly , but after 4 days , I have no DID , I tried to call tech support office and it seems they all ways are in hollidays, a support ticket no good luck, I call back service also not good luck. so anyone here has experience with them? are they a SCAM? fortunally only put there $10 bucks but

Hi, I've ordered a few IP501s from PC Connection, basically since we have an account with them. I like the phones for what they do, and now would like establish a relationship with a reseller that can give us maintenance and access to the most current firmware. What are some good resellers out there? Regards, --- Gavin Adams VP of Technology Promisant (USA) Inc. Email: me@gavinadams.org

Anyone using IAX.cc / Sixtel? Would love to hear experiences good or bad. Thanks! -- Start Your Own ISP! http://www.YourOwnISP.com

Hello, a user outside the office regularly gets a call from ext. 101 but that extension does not exist in my extensions.conf. when the user pickup the phone no one answers. Any Idea how to fix this issue? that user uses Polycom SP 450, Thanks in advance, Motty -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, We have updated the Wiki page for Polycom phones: http://www.voip-info.org/tiki-index.php?page=Polycom+Phones We posted several configuration specs as well as a link to an admin guide for the phone. We also posted a link on there to two firmware versions for download. The official Asterisk-Polycom support website should be up and live sometime in January. If anyone has anything to add

Stay away from Alliance Systems. We ordered $15k worth of Polycom's over a month ago and we're still waiting. Our account rep's communication with us on what the delay has been, has been terrible. Doug. -----Original Message----- From: Gavin Adams [mailto:me@gavinadams.org] Sent: Friday, January 27, 2006 8:26 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion'

I'm having some trouble with Polycom Soundpoint phones. I have had good luck deploying them on a local network, but now I've tried putting some in place which access their * server across the network. The * server is on a public IP and the polycoms are behind a NAT on a cable modem broadband connection. Every so often I get: May 27 16:12:08 NOTICE[29728]: Peer 'Polycom1' is now

I am using Asterisk CVS from last week and have been using Realtime SIP for a couple weeks now without any problems. Yesterday I decided to turn on Realtime IAX but I am having problems dialing to my long distance providers like Voicepulse, Sixtel or Nufone. I get the following: -- Executing Dial("SIP/2001-3761", "IAX2/password@voicepulse/19566680301") in new stack