similar to: Asterisk sales materials

This is probably a good time to point out that there is a good litmus test for all Voip providers. PRIOR to purchasing anything, send them an email and request the sales information. Ask about their servers or their policies or anything you can think of. How they respond will tell you a lot. If it takes forever, you can tell that they are either really busy, really indifferent, or something in

----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Saturday, September 13, 2003 7:55 PM Subject: Asterisk-Users digest, Vol 1 #1279 - 16 msgs > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

Slightly off topic, but maybe some can suggest something off list... Trying to upgrade a 7960 that was running skinny. I've got sip v3.2.0 installed and running, and am able to place calls via *, etc. However, when upgrading to v4.4.0 I can never get to the point of being able to place a call (eg, no dialtone, etc). I can ping the phone, look at the Network Config, etc, but I can't

Hi Rich, I had to change all my nat=yes to nat=route in the sip.conf. nat=yes seems to be ignored in today's CVS. Walter > > Message: 5 > Date: Fri, 27 Aug 2004 08:45:19 -0600 > From: Rich Adamson <radamson@routers.com> > Subject: Re: [Asterisk-Users] sip change? > To: Asterisk Users Mailing List - Non-Commercial Discussion >

Hot Pennies.net Special Situation Alert Watch This Stock Trade!!! STOCK DETAILS Symbol: IPFS Exchange: OTCBB Shares Outstanding: 13,379,000 Current Price: $1.10 Industry: Wireless Manufacturing and Technology COMPANY DETAILS Integrated Performance Systems Inc 17300 N. Dallas Parkway, Ste. 2040 Dallas, Texas 75248 Phone: 972/381-1212 Web:

Hi, I am new to asterisk , i am getting the following error,& the /etc/zaptel.conf file entry is as follows defaultzone=us loadzone=us span=1,1,0,esf,b8zs,yellow bchan=1-23 dchan=24 Parsing '/etc/asterisk/zapata.conf': Found Jul 1 18:33:35 WARNING[16384]: chan_zap.c:664 zt_open: Unable to specify channel 1: No such device or address Jul 1 18:33:35 ERROR[16384]: chan_zap.c:5296

Anyone interfaced * to the Nortel option 61c pbx via T1's, pri, etc? Need to begin planning the implementation, purchase cards, etc. Any recommended approaches, configuration problems, etc? Off-list is fine if you'd like. Rich radamson @ routers . com

I'm about to download Festival source, apply the astrisk diff's, and initiate basic testing. Thoughts are to download v1.4.3 (latest per the fesitval website. If anyone has an existing how-to, install notes, tips, or any suggestions I'd greatly appreciate it. Direct email is fine if you'd rather not post them. Thanks, Rich radamson@routers.com

you can always do a "restart when convenient" within asterisk, and it will do it's thing when all lines are clear.... -----Original Message----- From: Jonathan Moore [mailto:moorejon@usd465.com] Sent: Thursday, January 08, 2004 12:31 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Re: 911 and lawsuits and redundancy Is there a way to reload a module from the

Interesting story. I was following up any wild goose chase i could to find a resolution including cleaning dust out of the machine. At 4 pm I plugged the line in and called for the sake of luck, and it was back to working. In the mean time, thanks for the troubleshooting advice. I will keep this in my log book so I have the steps for next time something goes awry. ------------------------------

Rich Adamson <radamson@routers.com> wrote: > >> I have been investigating on this list and found that faxing is not >> reliable between Zaptel >> cards and that Digium does not support nor >> recommend fax over the TDMXXB interfaces. >> >> Is this true ? Will fax not be reliable enough for my customer ? > > True... the TDM card (or its drivers)

Extreme N00b, I am getting the error message "a target does not exist" when running the make install inside the zap directory, probably pretty common, possibly a package I didn't install, just need some insight on it. The same occurs with the libpri and asterisk. -----Original Message----- From: asterisk-users-bounces@lists.digium.com

Hello , I learn handbook-draft.but I think I don't understand asterisk. where I can find some learning book about asterisk? thank u. B.R. John. -----????----- ???: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]?? asterisk-users-request@lists.digium.com ????: 2004?12?24? 7:51 ???: asterisk-users@lists.digium.com ??: Asterisk-Users Digest, Vol 5,

Abhishek, In reverse order 3/ yes it is freeware, though some of the termination boards are available for sale from www.digium.com 2/ yes you can interface to Cisco handsets running SIP. 1/ Does it have a gui interface - the short answer is no. The longer answer is depending on what you mean, if you mean programming - then no though a number of people have developed sql interfaces. If you

I've been looking around for a small external sip fxo gateway, sending emails to possible vendors, etc, and can not seem to come up with anything that fits. Suggestions anyone? (No channel bank & T1 card suggestions, please. I've also just completed an eval of the Mediatrix 1204 which does not support the requirements.) The market between two fxo pstn lines (pair of x100p's) and

Does anyone know whether the linux machine running * have to have a sound card on it in order for musiconhold to work for sip phones? I've tried about everything (including tons of google searching) to get it to work, and nothing. When a call is placed on hold between two C7960's, the CLI indicates: -- Executing Dial("SIP/3002-c418", "SIP/3000|20") in new stack

Kind of off topic but I know CVS is the "prefered" way of upgrading, however are there such things as "stable" CVS upgrades? It seems a lot of the CVS's have a lot of devel bugs in this that I would be scared to put even near production. Just IMHO. :-) Matt -----Original Message----- From: Rich Adamson [mailto:radamson@routers.com] Sent: Friday, August 27, 2004 9:15 AM

Rich Adamson <radamson@routers.com> wrote on 2005-03-23 09:08: >> >> I have a x100p card and it doesn't detect a hangup from the calling >> party when going in voicemail(). My PSTN provider is sending open >> loop disconnect (voltage decrease for a given moment of time). >> Actually Progress Detection is HIGHLY EXPERIMENTAL so it should not >>

I would have to agree, an IAD locking up is bad either way you look at it. Even if you're there to reboot it on demand, it takes nearly 5 minutes to come back up. What kind of servers are they? What kind of phones? In all honesty, none of our IAD's ever lock up. And ones that did were defective and replaced. -----Original Message----- From: Rich Adamson [mailto:radamson@routers.com]

There is an option that you can add to your dhcp server option 150 IIRC. -----Original Message----- From: "Rich Adamson" <radamson@routers.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: 2/27/06 4:37 PM > Has anyone done any integration work with Covad's hosted solution ? I > am considering