similar to: Asterisk and Shoretel integration

Hello All, Here's the problem, we have happily set up several Asterisk servers to offer commercial service in the UK, our wholesale SIP termination partner (Magrathea - use SER/CiscoGW to provide us the service on a public IP address) - till now we have used Asterisk to connect clients on private IP's with Asterisk doing the required conversion for SIP/IAX between public and private

I've been anxiously awaiting ShoreTel version 6 because of it expanded use of SIP. My plan was to upgrade out ShoreTel server at our main office to version 6, and use Asterisk in our small remote offices, and have them all be able to directly dial each other's extension. (i.e. CEO in main office can dial ext 401 and get directly to secretary at small remote office, and vice-versa)

Hi, Someone who has had experience with shoretel VoIP systems, can you please give me a run down of how Asterisk is either better or worse? I am completely unfamiliar with Shoretel systems, but someone had suggested we look into them. I said, you bring your Shoretel features, and I'll show you 10 things Asterisk does or can do that Shoretel doesn't do. I still believe that's

We have a Shortel system at out main site. We're putting Asterisk servers at several smaller remote sites. I know I'll be able to get the Asterisk servers to talk to each other via IAX, but can they talk to the Shoretel server? Basically, I'd like to be able to, from the main site with Shoretel, dial an extension, and reach that phone at a remote site, and vice-versa. Thank

I've got a ShoreTel 210 MGCP phone drops calls. My packet capture indicates that the phone may be trying to renew its registration with *, but reports Restart Method of Disconnected (frame 2), then * seems to take that as a sign that it has lost the connection and closes things down. The phone, meanwhile, seems to think it can continue the conversation until a few ICMP "port

Okay, so I'm a noob. Asterisk looks very promising, so I say "thanks" and "good job" to all who contribute. My * test box is up and running with soft phones using IAX and SIP, so now I'm on to testing hard phones. I borrowed a couple ShoreTel 210 phones from somebody who had them on hand but they only support MGCP. I see that there's an mgcp.conf in

I have a client who is using a Shoretel PBX. This PBX apparently does not send DTMF information OOB, but instead sends this inband via the B-channel. This is traversing an Asterisk box via a PRI. The user calls the IVR (1-800-CALL-DHL), receives audio, but is not able to present DTMF to engage the IVR. With some light research it appears that the DSP is not activating until the call is

Hello We recently moved to Asterisk 1.2.4 (from 1.0.x) and our 10 Uniden UIP200 have stopped working ever since. We can make a call with the UIP200 to any other extensions, but it can not receive a call. In fact the UIP200 always appears offline: It does show up in asterisk a few seconds after the UIP200 reboot: -- Saved useragent "Uniden SIP Phone p2 Ver BS4.70" for peer uip200 but

SIP user => Asterisk 1.6 server => SIP Trunk => external destination: works AMI script => Asterisk 1.6 server => SIP Trunk => external destination: Failed to authenticate on INVITE to '"asterisk" <sip:asterisk@(ipaddr)>;tag=alphanumeric' I?ve tried doing things like ?include => contextwithtrunk" in various places, googling, re-reading relevant

Hey, all. While ADA can still be downloaded, that's about all that I see. No development, no recent mention, and -- perhaps worst of all -- it appears not to work properly under 64-bit systems. So, assuming Digium's abandoned it, are there any suggestions of alternatives? Right now, I'm replacing a Shoretel system, and I'd *dearly* love to avoid the incredibly fat client they

I'm new to the asterisk world and have been playing with an asterisk server with 1 FXO card for a couple of weeks. Now I'm looking to start adding IP Desk Phones but I'm unable to come to a decision on what phones to use. I like to look of the Polycoms, but because we are not a "phone company" I can't see us getting reseller authorized for them. Shoretel has some nice

I have a client requirement that multiple phones can be dialed, however they don't want the pstn phone to pick up automatically because of voicemail etc, nothing can be changed on the phones, how can I handle this requirement, by the way no zap channels are involved, all the pstn phones are behing another sip gateway.

Today I had a problem where the internet connection is unstable so calls are getting dropped all over the place. The one thing I do not understand is that at least 30 phones on the internal network went to "No Service". Since they are on the same network segment and on the same subnet I do not see why the Internet connection sould affect them. The asterisk server is behing NAT and we

Hi all, Anyone got a link or the file to Mr. Guy's notes from his speech at Astricon.. I looked for them or relevant info he showed ad FWD's webpage but have not been able to find them. He had a working FWD registered thru Asterisk behing a NAT and a couple of samples of their new scripts, talking weblog (from Mr. Pulver) and other cool new features. If anyone got these please send em

Hi List, I have a 2Mbps SDSL link which gets saturated during peak time because about I have about 3 E1 worth of g729 traffic going thru. So I'm planning to use IAX2 trunking to reduce bandwith requirement and squeeze out each and every bit of this (expensive) bandwith. I've set up two boxes (debian etch), one in a remote data center (which has plenty of bandwith) and one behing

> That makes it fragile, and that’s why I disagree with your “90% done” assessment. > What if the service behing the hook is down for a few days? In the long-term view, a pretty trivial catch-up script ought to be able to synthesize a sane history after any amount of down-time. People could even run it locally for their bisecting needs if it was that important to them. In the short term, I

So is this Digium taking over Pulver's' void? Same color scheme and fonts. Thanks, Steve ---------- Forwarded message ---------- From: Digium <webannounce at email.digium.com> Date: Thu, Jun 12, 2008 at 10:34 AM Subject: Complimentary Subscription to VoIP Industry Publication To: stotaro at totarotechnologies.com If you are having trouble reading this email, read the online

What dbPowerAmp does is encapsulate the wav file into a FLAC container without actually compressing it. I guess it can be a bit arguable if you can technically call it a FLAC file :) It's like using "Store" mode in RAR, for instance. The idea behing this "uncompressed FLAC" is to give the FLAC tagging abilities to people who don't want to compress their files. But

Hi everyone, I am fairly new to the idea of VoIP, although I've been reading about it off and on for the last few years. Now it is starting to look mature enough to consider implementing it, but there is one thing that I haven't been able to get a clear answer on... With Vonage, you are using the Vonage network - it is their responsibility to route your call to the endpoint, which is

Hello, I started using clangd to get better static analysis and code refactoring tooling with the R sources (using eglot-mode in Emacs, it just works once you've generated a `compile_commands.json` file with `bear make all`). I noticed that the static analyser can't understand several header files because these are not self-contained. So I went through all .h files and inserted the