similar to: Broadsoft - Asterisk interop

Ottawa, Canada ? February 15, 2006 - PIKA Technologies Inc. today announced that they have integrated PIKA?s high-density analog computer plug-in boards with the open source Asterisk PBX, with the introduction of PIKA Connect for Asterisk. PIKA Connect for Asterisk is a software layer, available free of charge and distributed under the GNU Public License (GPL), which allows interoperability

I am looking for a sample sip configuration of a SIP provider that runs Broadsoft VoIP switch. My sip provider is Conecta from Brasil, that only give me a SIP IP address to configure my asterisk box, when I call them for support or authentication data to load on my sip.conf, they say me that I don?t need such data, so, anyone knows how I would configure my Asterisk box against a Broadsoft peer?

Hello, I've seen several references in IP phones manuals to Broadsoft's BLF-List URI feature (also referred to as List-Oriented BLF). With this mechanism, a server is able to update the BLFs an IP Phone is supervising without asking the IP phone to reboot, as for a reason I don't know, most phones send BLF-related SUBSCRIBEs during boot time. Is this feature compatible with Asterisk

asterisk-users at lists.digium.com, you are invited to Title: VoIP Users Conference Time: Fri Mar 7 12:00 - 13:00 (Timezone: Paris) Where: http://voipusersconference.org Calendar: Description: Every week we try to get guests with ideas, products and services you haven't had time to check out to come and talk about what they're doing. Tomorrow, Pika Technologies will be with us.

Every week we try to get guests with ideas, products and services you haven't had time to check out to come and talk about what they're doing. Tomorrow, Pika Technologies will be with us. Friday, March 7that 12:00 PM (Eastern US) 9AM PST, 5PM GMT *** Call (724) 444-7444 or SIP:123 at 66.212.134.192 *** After the call connects, enter the conf: 22622# and your_PIN# (or 1# if you

Grasping at straws here...is anyone using a Dell PowerConnect 2224 24-port unmanaged 10/100 switch in a deployment? I have two separate asterisk installations with bad one-way audio where the only common elements left are the Dell switches and Polycom IP-500 phones. Two different ITSPs - one location uses IAX2 to the outside world, the other SIP. Two different bandwidth providers, one SDSL 1.5

Hi, all-- What do you make of this? Here's my call log--looks like there are a lot of calls going in and out of the server that are not real incoming or outgoing calls. Does anybody have any clue what is happening? 2006-11-14 16:41:00 Local/8183... 8183461773 "8183461773" <8183461773> 8183461773 NO ANSWER 1 47. 2006-11-14 16:40:59 IAX2/Voice... 8183461773

Hello I don't know much about game consoles, and I was wondering if someone had successfully ported Linux and Asterisk to the current hardware, ie. Nintendo Wii, Sony PS3, or Microsoft XBox360? Thank you.

Hi all I'm trying to get asterisk 10 spandsp get faxes from 012 in israel (they use broadsoft switches) using T.38 more reliable and would like to know if anyone knows of any changes I could make or ask them to make. As it stands now I get much more reliability receiving faxes with iaxmodem with no T.38 which is funny than when having asterisk receive using T.38 and if I try using gateway

Contacted Scott Willard at Polycom this morning, he has since been reassigned to other duties within the organization. Mr. Willard's tone seemed optimistic, and he referrred me to Roger Austin, Regional Channel Manager for Voice. Roger's reply to my inquiry is as follows: "Cory, We appreciate your interest in Polycom VoIP Phones. Polycom deploys our VoIP phones with our VoIP

Hi list, I'm in central-europe and signed yesterday a broadvoice account. My Asterisk box is CVS 2005-08-25. Problem I face is: "Failed to authenticate on REGISTER to 'phonenumber@sip.broadvoice.com' (Tries 2)" then "Registration for 'phonenumber@sip.broadvoice.com' timed out" and finaly "Giving up forever to register

Greetings to all. We have developed an application for asterisk to accept creditcards using the Authorize.net gateway. I am looking at reverse phone lookup services in order to get the *full* address of the caller (not just state,city). This is a very low volume service as you can imagine so i would prefer something with low or no minimums and a pay-as-you-go scheme. An API would be a plus,

I'm still not getting my outbound to work. I've seen two patches relevant to broadvoice for chan_sip.c which apparently have already been added to CVS. I'm dropping all outgoing calls after ~30 secs. Asterisk doesn't seem to know they're gone though. I called my cell w/ broadvoice and turned on sip debug AFTer the call had physically dropped: *CLI> sip show registry

Hi, I'm using gotoiftime on asterisk, but it seems&nbsp; there is a difference between the asterisk time and the system time. could it be because i adjusted the system timezone on my linux? do asterisk not detect the change of timezone on the system? How can I fix this prob? Regards, nhadie -------------- next part -------------- An HTML attachment was scrubbed... URL:

Anybody know a reliable fax solution for 1.4.30 branch? I am using PikaFax on another server and works very well (about 3000 faxes a week), but it appears they no longer offer their product to open source asterisk, only for there "WARP" appliance. NOT really looking to migrate from 1.4.x to 1.6.x -------------- next part -------------- An HTML attachment was

Gang, I recall several months ago that there was a company that was giving away a free 1-port T1 card, with some specific conditions. Do any of you recall who that was? My Google searches are coming up empty and now I'm wondering if I was hallucinating... Thanks, MC -------------- next part -------------- An HTML attachment was scrubbed... URL:

On 29 Jan 2015, at 11:07, Administrator TOOTAI wrote: > Le 28/01/2015 22:03, Steven McCann a ?crit : >> Hello, > > Hi > >> >> I'm investigating a situation where there was a hundreds of minutes >> of >> calls from an internal SIP extension to an 855 number in Cambodia, >> resulting in a crazy ($25,000+) bill from the phone company. I'm

----- Forwarded message from Stelios Zimeras <szimeras at irisa.fr> ----- Date: Wed, 20 Oct 1999 11:35:52 +0200 From: Stelios Zimeras <szimeras at irisa.fr> Organization: IRISA, Campus de Beaulieu, 35042 Rennes Cedex, FRANCE X-Mailer: Mozilla 4.7 [en] (X11; I; SunOS 5.5 sun4m) X-Accept-Language: en To: rgentlem at stat.auckland.ac.nz Subject: R installation problem Dear Sir, I try

Hello For those SOHO customers (ie. at most, a couple of POTS/ISDN connections and simultaneous SIP calls) who'd rather not use a big, noisy PC to run Asterisk, I'd like to offer an alternative that has the following features: - not old hardware sold on eBay, ie. it must be up-to-date hardware sold by a company currently in business - compact, silent - has room for a 2.5" hard-disk,

I have a curious problem connecting with OpenSSH (5.1p1 on FreeBSD, 6.0-beta on OpenBSD) to some managed switches running "IPSSH-6.6.0". When I connect to the switch with the SSH2 protocol, I can't use the switch's CLI, because I can't terminate a command line. I press <Enter> and nothing happens. Same for ^M (CR) and ^J (NL). When I connect to the switch with SSH1