similar to: Fwd: Complimentary Subscription to VoIP Industry Publication

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Hello everybody. I have a problem with an integration between an Asterisk (1.4.24.1) on FreeBSD 7.0 and a Shoretel 7.5 server. To make a very long story short, when someone behind asterisk call an extension behing shoretel everything work as expected. When someone behing the shoretel server call someone behind asterisk the first 10 seconds of the call seems ok but then the line is dropped

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

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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

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

Hi Steve; Currently with Dahdi, for the below configuration that we were using it in the rc.local, how it will be? I think there should be something new to be used instead of ztcfg, what it is? And what about other lines? They need to be changed? touch /var/lock/subsys/local /sbin/modprobe wctdm /sbin/ztcfg -vv /usr/sbin/fxotune -s /usr/sbin/safe_asterisk Regards Bilal --- On Thu, 5/1/08,

Hi Steve; Currently with Dahdi, for the below configuration that we were using it in the rc.local, how it will be? I think there should be something new to be used instead of ztcfg, what it is? And what about other lines? They need to be changed? touch /var/lock/subsys/local /sbin/modprobe wctdm /sbin/ztcfg -vv /usr/sbin/fxotune -s /usr/sbin/safe_asterisk Regards Bilal --- On Thu, 5/1/08,

ast_waitfordigit that accepts milliseconds as input. Douglas Garstang wrote: > Admittedly I have not used the ExternalIVR app. Is it any good? > > I'm not sure I agree that Asterisk is GOOD for building IVR's. Sure, > it can do it, but boy it is UGLY. There's also the fact that you can't > call Backgound() in a macro, which forces you to use Read() which >

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

Hi, I am new user on asterisk (for that matter linux) and i have lot of embedded programming experience. We have a new project from our client, to design a box that takes the telelphone line as input and route the line to the respective user with different ring tones. The box should be programmed by the users with buttons. Features. 1. I should be able to store some .wav files for different

I see a variety of DECT 6 phones available CHEAP at costco. Is there a way to convert these to SIP? I recall someone talking about a Siemens devices that works with all DECT phones, making them SIP (I think).... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081205/fdd14af9/attachment.htm

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

Folks, First of all, this email is sent to -users and -biz, but please follow- up to the -biz list only. I have set the reply-to, but I fear mailman will strip it off ... Please don't flame me for posting to -users, I'm just not sure who lives on -biz as the (signal/noise | net.kook | troll) factor has been pretty bad on that list lately (poor Rehan!!) ;-) Telephony Depot

Hello NUT-UPSUSERS, I would just like to try and help other and indicate that I was able to use the development trunk and get the Tripp Lite OMNI1000LCD UPS (USB) working with the newhidups drivers on Gentoo Linux. The one that is currently available at many local Costco stores and online. In case others need to replicate, I've attempted to document my steps over here.

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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