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I've been working with a telco for the past two days trying to get a PRI span up and running. This is a small-ish telco and I get the feeling they don't do this very often. Anyway, they specified a pretty standard setup: ni2 switchtype, esf framing, b8zs coding, etc. All of my b-channels are up, but we're having a heck of a time getting the d-channel to come up. He finds out that

At work, I have 4 branch offices at which I've deployed asterisk. Call termination/origination at each branch office is handled either through a frac PRI or 3rd party SIP provider. Soon, I'll be replacing the legacy PBX at our HQ with asterisk. Each branch office has between 3 and 20 employees, each with their own extension and DID, and at headquarters, we have about 70 people, again

Hello I've never built an IVR before, so I was wondering if someone could share some code from their extensions.conf that would perform some of thoses steps: 1. When a call comes in from the TDM FXO port, answer 2. If no CID, play message "No CID available. Please type the number where you wish to be called back". Loop until OK or remote party hung up 3. When CID is available,

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Hello all - I'm just starting to play around w/ asterisk, and I've run into a seemingly simple problem that has really manged to frustrate me... I'm running the latest cvs version of *, and am trying to dial in to the default extention 1000 demo using x-lite. I can dial and hear the greeting no problem, but when I try and send any DTMF tones, I don't get any response. Is there

Following the successful deployment of asterisk servers at several of our branch offices, in the near future, I'll likely be implmenting an asterisk server at our HQ. We currently have a T1 PRI terminated on a legacy PBX. I'll be doing a stepped deployment in which, via a dual T1 linecard, the asterisk server will initially pass all incoming/outgoing calls directly through to the PBX.

Hello, Beside Cisco 79x1-GE, I'm not aware of any Gigabit SIP Phone. Did I miss something ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070612/b9b701b3/attachment.htm

I've been going back and forth with my telco for several days, trying different configurations to get a new PRI to come up. The bchannels are all up and the T1 is not in alarm status. The dchannel refuses to come up however. We've tried ni2, qsig, and now dms100 for the switchtype. The telco tech I've been working with says that he's been sending "reset all channels"

...ith Cisco's CallManager software. Start with this link: http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79xx Hope that helps. Good luck! Jason Burbage ------------------------------ Message: 22 Date: Wed, 16 Jan 2008 19:11:22 -0600 From: "Erik Anderson" <erikerik at gmail.com> Subject: Re: [asterisk-users] Anyone Using a Dell PowerEdge T105 in Production To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Message-ID: <fc40260f0801161711t7cc395b6ob58b3ffa7385d490 at mail.gmail.com> Co...

...For the A200 you need an additional case slot (does not need another PCI connector) for every 4 ports over 4. The same goes for the A400 on every 12 ports over 12. Christian ------------------------------ Message: 17 Date: Mon, 22 Oct 2007 16:41:19 -0500 From: "Erik Anderson" <erikerik at gmail.com> Subject: Re: [asterisk-users] Extensions.conf for basic IVR? To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Message-ID: <fc40260f0710221441u12274367ja0b04424a1acf241 at mail.gmail.com> Content-Type: text/pl...

All - I've been wrestling with how to best structure the sip device accounts on a new asterisk server I'm deploying. All of the sip devices (currently only Linksys SPA941s) will reside on the same subnet as the server, and I have already set up a decent automatic provisioning system for the phones. When the rollout is complete, there will be about 100 SIP devices authenticating and

Shortly, I'll be purchasing a Sangoma A102. I'm wondering if I should spring for the hardware echo cancellation circuit or not. Upon initial implementation, the 2 T1 Ports will be used as a passthrough as we slowly transition off of a legacy PBX. Eventually, we'll only be using one of the ports, and will be providing VoIP service to a bunch of SIP deskphones. So - with that usage

All - I've been trying to pick out a bluetooth conference phone that I could use with a softphone along with my asterisk server. I've been looking at the TrendNet TVP-SP4BK. Have any of you used this device or any other bluetooth conference phone? How have your experiences been? Thanks! -Erik -- Erik Anderson http://andersonfam.org

All - this is probably a simple problem, but I've been pulling my hair out trying to figure out what I'm doing wrong. I'm building a *simple* IVR menu. Here it is: [main-menu] exten => s,1,Answer exten => s,2,SetMusicOnHold(default) exten => s,3,DigitTimeout(5) exten => s,4,ResponseTimeout(30) exten => s,5,Background(logic-main) exten =>

Greetings all - I'm interested in using an asterisk box to supplement and add VoIP capabilities to our legacy InterTel Axxess PBX. After searching through the list archives and through google, it seems that the best way to go about this is to connect the two systems via a T1. Is this correct? The PBX currently doesn't have any VoIP capabilities, so that's not an option for

Hello all - I'm about to deploy an asterisk server here at work. Before deploying, I'd like to do an extended load test on the system. I currently have T1 crossover cables connecting ports 1->2 and 3->4. Would there be an easy way to script generating a bunch of calls across these spans? I envision generating 23 calls over the 1->2 span and 23 over the 3->4 span. I'd

I am currently using a T1 PRI from TWTelecom for DID and outgoing calls, but I recently discovered that they're offering call termination/origination over SIP trunks in my area now. If they could deliver these SIP trunks to me over a guaranteed-QoS circuit, this would be of great interest to me. We're already using a DS3 circuit from TW for our internet uplink, so I'd imagine it

Hello - I have (finally) obtained a fair amount of success in connecting my Intertel Axxess PBX to an asterisk box via a T1 PRI. I can place calls from the Intertel side through the T1, out to an IAX2 softphone and the calls get routed correctly and all of the CID information stays intact. However, when I call from the IAX side to an extension which should route back through to the Intertel