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Robert, > Is there a product to simulate a PRI trunk? (Robert > Goodyear) TDMoE emulates a T1. ;) Once the TDMoE link is up, Asterisk just sees 24-lines that appear to be a T1 instead of having to deal with all of the complexities of VoIP. This is useful, since probably 75% of the utility of VoIP is really just the fact that it can run over a network. It's al...

We recently received a bug report about a compiler issue when using GNUTLS: http://bugs.gentoo.org/show_bug.cgi?id=67628. I can confirm the same bug using 0.99.11. Help? Thanks, g2boojum -- Grant Goodyear Gentoo Developer g2boojum at gentoo.org http://www.gentoo.org/~g2boojum GPG Fingerprint: D706 9802 1663 DEF5 81B0 9573 A6DC 7152 E0F6 5B76 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: not av...

...was a PRI type of thing. Like there was a module that loaded up as a fake PRI that your Asterisk box could use to connect. Thanks, Wiley ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Robert Goodyear Sent: Friday, June 24, 2005 9:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Exposing Zap Channels on Server A to be Used ByServer B On Jun 24, 2005, at 9:07 AM, Wiley Siler wrote: Hello All, I remember there is a way to use two...

TDMoE was it. Thank you!!!! Wiley ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Robert Goodyear Sent: Friday, June 24, 2005 2:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Exposing Zap Channels on Server A to be UsedByServer B On Jun 24, 2005, at 1:31 PM, Wiley Siler wrote: From: asterisk-users-bounces@lists.digium.com [mailt...

Does anyone know if there's a service out there to -- for a fee -- inject our DID into the LEC's CLI database so a called party gets our associated name? /rg

...r with Asterisk behind another NAT is troublesome. Has anyone had any luck doing this by interfacing Asterisk to the WAN using 1:1 NAT translation to give it a public IP while still firewalled? In my instance I'm using m0n0wall, but this is a hardware-neutral question. Thanks. -- Robert Goodyear Brand Up LLC http://www.brand-up.com

...works for Word files. I have a link to ooffice for Winword.exe. But I've had no luck with acroread with: [Software\\Classes\\AcroExch.Document\\shell\\open\\command] 1017688446 @="Z:\\usr\\bin\\acroread.exe \"%1\"" TIA, Wayne --- Wayne Richards Phone: 330 796-4462 Goodyear Tech Center Fax: 330 796-3947 Department 431A e-mail: wrichards@goodyear.com PO Box 3531 Akron, OH 44309-3531 And one for the lawyers: This email may contain confidential/proprietary information and may not be copied or disseminated without the expressed written consent of the Goodyear Ti...

Sip to sip calls are fine, both local on Asterisk and over a SIP gateway, however some people who call on the PSTN line say we are very queit and vice versa, can the volume be turned up on the PSTN line? The volume buttons on the VoIP phones only turns up the others voice, so this is a fix for us, but how do I make our voices louder for the people on the PSTN line? Thanks. Paul.

Does anyone know of a way to simulate the signaling of a PRI trunk for testing/setup purposes? I realize this may be a rather naive question, but I was wondering if you could take a TE110, for example, and using a crossover cable (or not?) and some means of emulating the NI2 signaling protocol connect it to another TE110 on another machine to test and verify an installation before the telco

Hi, I have a TrixBox install with a Sangoma A200 and 4 FXO ports, there are 3 BT lines connected directly to these ports. One of the lines has BT FeatureLine Compact and this is the line I am having problems with, the other 2 lines are working perfectly, detecting CID, answering incoming calls and placing external calls via SIP devices. I am receiving a error log entry: chan_zap.c:

Is the X100P FXO PCI Card capable of detecting a fax, answering the call, and then emailing the fax content to an email address? Thanks. Paul.

> -----Original Message----- > From: Robert Goodyear [mailto:me@jrob.net] > Sent: Tuesday, March 22, 2005 1:21 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Is there a way to get inserted into an LEC's > CLIDB? > > > Does anyone know if there's a service out there to -- for...

...ecting to SIP Phone 100 the caller should hear a soundfile... wiki says nothing about an Dial-option to play a soundfile to the caller ;-( Roland Zagler -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Robert Goodyear Sent: Saturday, July 02, 2005 8:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] play message to callee before connect toincoming call On Jul 2, 2005, at 10:23 AM, Roland Zagler wrote: > Hello, > > i try to do the following: > > 1) cal...

Hi, Just playing with a couple of Budgetone 102 phones and they are pretty good for the price. The only problem i'm having at the moment is when I get a voicemail on the Asterisk box the LCD flashes. Dialing *98 goes to the VoiceMail Manager, and asks for mailbox, I enter 201, then asks for password, enter my voicemail password set in the Extensions -> webadmin, then hit the

I'm all up for reading and looking round for people in the same boat to try and solve the issue together, but there appears to not be large community yet, just the asterisk mail lists. I got Asterisk working with X-Lite great now for internal calls and also calling land line numbers etc. The two problems i'm currently having are: 1. When someone calls in the phones ring 3 times then

I'm looking to expand my bandwidth for my Asterisk PBX. Why should I choose a T1 over DSL for my asterisk server? I found someone offering T1's for $290 a month + Loops or 3 Meg for $561 a month + Loops. Is this a good deal? Thanks Bart -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi. I'm having a terrible time with call quality coming into my * box. I'm using VoicePulse over a 1.5/1.5 mbit line. Outbound calls are crystal clear on both the RX/TX sides of the conversation. Inbound calls, though, are HORRIBLY garbled on the RX side. I can barely hear the caller, but they report my quality is fine. Getting loads of garbled sounds and weird echoes. (Could just be

Hi all, I have setup a Dell PowerEdge 2550 with a Sangoma A200 card with 2xFSO and 1XFS modules. The PowerEdge specs are 1 x P3 1133MHz, 512MB RAM. Sangoma A200 has 3 analogue PSTN lines connected. This server is based in Office 1, with 5 users all with a Linksys SPA942 VoIP Handset. There is another Office (Office 2) connected to here using VPN. There are two users in Office 2 with the

...Durham", "Shreveport", "Laredo", "Yonkers", "Tacoma", "Anaconda", "Butte", "Suffolk", "Buckeye", "Augusta", "Cusseta", "Huntsville", "Boulder City", "Goodyear", "Hibbing", "Norman", "Sierra Vista", "Georgetown", "Carson City", "Chattanooga", "Lynchburg", "Columbia", "Mobile", "Athens", "Little Rock", "Yuma", &qu...

If anyone knows someone with CVS access can they please take another look at http://bugs.gentoo.org/show_bug.cgi?id=22080. It has been in bugzilla for nearly nine months. Would like to see this in portage - masked is fine - so at least everyone is working off the same file. Cheers Zach. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: