similar to: Local Outbound Calls on PRI

I've got a client who would love to be able to leave an asterisk server running sompelace, and achieve telephone connectivity using an IP phone from within his Motorhome in his words "I want to be able to work from a mountaintop in Glacier National Park" I've done some initial testing, and a SNOM200 SIP phone comes close enough to working that I have not ruled out the idea as

I've got a 4 port T1 card in my Asterisk box with a PRI from Qwest as my PSTN interface. I'm experiencing random dropped calls on the various SIP devices I have tested. Network connectivity to the SIP devices looks ok, and I have tried a variety of the devices including all of the following. Grandstream 286 Grandstresm 486 Sipura SPA 1000 Mediatrix 2102 Some example lines from my logs

I've got automon up and recording calls on demand from information I found in the list archives, however instead of ending up with one monolithic file, I've got a -in and -out version of the files in my monitor directory? Anyone have suggestions how I could end up with a monolithic file that does what I want?

http://www.boingboing.net/2006/06/05/play_zork_by_phone.html Let me preface this idea with one comment: I don't have the time to do this - I don't even have time to eat these days. But someone out there has the cycles to do this... and it would be very cool. OK, so now Zork is attached to Asterisk, but using the less-than-clear Festival engine. There are beta tests of the LumenVox

Pardon my ignorance, but I was hoping someone could clear up something for me. - For a few POTS lines, digium has a single port card for that, or a T1 card to a channel bank. - For 10 or more lines, digium has a T1 or E1 card for that too based on PRI channels - For 100's to 1000's of lines, I suspect a soft-switch is in order??? A traditional phone company will sell: - POTS lines for

Does anyone know if the HandyTone 486 has the option to turn the two ethernet ports into either a switch/hub, or does it have to do NAT ? Thank you, Steve Maroney

Anyone know who that good looking female is thats on the Digium.com website ? Ok, my Real question is I noticed that Digium has relesed a new T1 card with an echo canceller. I also noticed that its supports E&M Circuits. Im I have very little knowledge on T1 circuits and traditional PBX's so what Im asking is can I use Digiums T1 card to connect to another PBX via a tie line ? Or does

Scott's SpanDSP FAQ mentions that the senders' TSID (20 bytes sent by sending fax machine) is made available, but doesn't mention where or how. Is there a variable or something that's set? I didn't see it in http://www.voip-info.org/tiki-index.php?page=Asterisk+variables Thanks, Woody

I need to connect up to sixteen phones per building, I can use a cheap hub, but POE would be useful. Is there a cheap POE hub available? Everything I have seen has been expensive. Chris Mason

What is the deal with "CSR's"? TWTelecom is telling me that I can't port a number to their service without a Customer Service Record. Apparently this is easy with Verizon, and not so easy with some other companies. Basically I'm at a brick wall with a couple of ports because TWTelecom is telling me I HAVE to get a CSR and certain other providers (Time Warner Cable for

Hey guys, I was wondering what POE solutions are being used ? Ive done some searching on google and found that PowerDsine seems to be good brand. Any comments,suggestions, and experiences on POE hubs other POE products would be greatly appreciated. Thank you, Steve Maroney

Hi all, I am currently facing an issue with Dovevot version 2.2.29.1 where subfolders in a public namespace are only visible to the 2nd folder level: namespace Public -> folder "1st" -> folder -> "2nd" -> folder "3rd" .... When one user creates a subfolder below the 2nd level other users do not see that folder and are unable to subscribe to this

Im trying to get IAX to work between my * and FWD. I activated my iax2 account on iax.fwdnet.net and I get the output: "Registered to '65.39.205.121', who sees us as 68.14.203.254:4569" when I start asterisk. I tried used the Call Me tool on fwdnet.net but I dont get any calls even though the Call Me tool says everything looks ok. Can someone call my FWD number and just leave

Im looking for an Intercom solution thats interoperable wit Asterisk. Ive read several posts about people using the 2nd lines on some SIP phones w/speaker phone. Unfortunatley I dont that is going to cut it in a large warehouse enviroment. Does anyone have a solution that uses a "loudspeaker" ? Thank you, Steve Maroney

Hello all, I am using dovecot for some time now but always had very basic configurations. I recently have configured a new, additional namespace in order to move archived mails (Thunderbird Archives) to another storage system. namespace { type = private separator = . prefix = Archives. location = maildir:/var/vmail/archive/%Ln/Maildir subscriptions = yes list = children inbox = no

On 04.05.2017 10:27, Andreas Oster wrote: > > > Mit freundlichen Gr??en > > Andreas Oster > NOVA Elektroanlagen GmbH > Carl-Zeiss-Str. 3 > > D-76275 Ettlingen > > Tel.: +49 (7243) 5490 22 > FAX: +49 (7243) 5490 54 > aoster at novanetwork.de > http://www.novanetwork.de > > Gesch?ftsf?hrer: J?rg Amann, Claudia Blasi > Registergericht: Mannheim, HRB

Hey guys, Im interested in hearing about servers (and thier hardware specs) that successfully run both asterisk and samba for an office of maybe about 12 extensions (SIP) and about 12 workstations. Im hopeing to not only replace a traditional PBXs with Asterisk/Linux but to provide a solution to needs such as a file serving, email serving, etc. Ive read the Success stories form voip-info.org but

Hey guys, Im about to sign up for VoicePulse Connect. Of course, I plan on using my asterisk server to "register =>" with the service. I would rather sign up with VoicePulse via SIP instead of VoicePulse Connect. My asterisk box is behind another Linux box serving as my nat/firewall. Does anybody think I will have a problem ? Should I stick to IAX and VoicePulse Connect or can I use

I have a PRI link up and running between Asterisk and a Nortel Norstar MICS v4.1 . I'm having a problem getting the textual Caller Name across the link from Ast to Ns, however numeric Caller ID arrives and displays fine. From Ns to Ast both elements come through fine. I'm forcing dummy values for testing using: exten => s,1,SetCIDName(Test) exten => s,2,SetCallerID(1234561234)

Hey guys, How much disk space is used by Asterisk to store voice mail for about 10 - 20 users/mailboxes Thank you, Steve Maroney