similar to: standalone PRI-to-SIP converter

Over a year ago I saw a discussion about a standalone device which converted a T1 in/out to SIP in/out (over 10/100 LAN). Anyone recall what this device is? (I'm looking for a standalone device - not a PCI card). Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070907/7060b23e/attachment.htm

Hi, I've read this thread in this list history http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.user/261151/match=t38modem http://sourceforge.net/tracker/?func=detail&aid=3337581&group_id=152230&atid=783657 Has anyone been successful when integrating latest version of Asterisk (10 or 1.8, for instance) with t38modem ? My target setup is: fax ---<PSTN>-- SPA3102

Checkout Beronet ISDN cards and Berofix Gateway (appliance or pci card) Personally for my last installation I chose Berofix card which is rock solid and reliable, yet easily configurable. With berofix you don't need telephony drivers on the host system, the isdn card is detected as a NIC and all configuration is done using a web interface. Then you configure FreePBX a new trunk to use the

I'm running * 1.4 and can successfully restart asterisk from the command line with: /usr/sbin/asterisk -r -x "restart gracefully" However, I have a cron job that tries to restart asterisk and gets this error: No such command 'restart gracefully' (type 'help restart gracefully' for other possible commands) Can anyone think of why this is happening? Thanks

I see a lot of attempts by hackers to call 00972595301123? or 011972595115207? or variations but that same 972595 is often present. Can someone break down that dial string with an explanation? The 011 look like an overseas call (from Americas), while the 972595XXXXXX is unclear... -------------- next part -------------- An HTML attachment was scrubbed... URL:

We've deoplyed a number of pure VoIP wireless (wifi & proprietary) phones, but not dect. Is there a simple overview of integrating DECT phones with Asterisk somewhere? I assume the DECT basestation has a multi-account SIP VoIP interface, and the handsets are just plain old dect? Can you push configuration info to individual phones? (Are they individually addressible / configurable

I'm looking for a small formfactor mobo for an install that needs to handle 25 phone sets (no transcoding). I found a new dual atom 1.66GHz mobo - anyone know what kinds of call volume that will handle? MD

For high-availability Asterisk when using PRI Lines, you might also want to check out our product (FSV-4PFS). It's available at www.failsafevoip.com Bill Also to add to the last post does this device have hardware echo cancelation? if it does it could be a great replacement, if not may not be what I'd really want to use. Thanks, tom I've been looking at the RedFone foneBRIDGE2

We're looking at a large wifi phone deployment, and we're looking for wifi phones that: 1. Are SIP compliant (Asterisk friendly) 2. Provision capable (ideally TFTP of own MAC address) 3. Industrial quality (no cheap plastic stuff). 4. Well documented (and none of the "only telco's get documentation" crap) Does anyone have a suggestion? Thanks, MD -------------- next

I found some great pieces of script on the internet that I've combined to allow Asterisk to send voicemails as an MP3 file, and encode the sender name and number as well as message number as tags into the MP3 file. I even include a "cover art" image which has our company logo and PBX symbol in it. Mobile phone users love it, and Android phones can now play the attachments (without

Hi i want change my asterisk server. Actually, Asterisk work's on a IBM Server with a internal digium E1 card. For High availability, i don't want now use "internal E1" card. In my new asterisk systems, i have two server and two E1 not in the same site. I am search a hardware gateway, if possible in 1U Rack with 2/4 or 8 E1 capacity with echo cancellation. I want that this

I have a new asterisk system with a T1 card. It appears that running "ztcfg -vv " is required in order for asterisk to start properly. Is this correct? Are people adding this command to the asterisk startup script? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

We're creating a SIP gateway for a client that will take one leg of a call in via SIP, and out the other side via H.323. To minimize load on the gateway, we would like to have the RTP stream bypass the gatewayy altogether (directrtp/reinvite). Is this possible with these to protocols? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is there a recent survey of that Linux distro and version people are using for the Asterisk installations? I recall seeing a pie chart over a year ago (I think on a wiki but I can't find it again)....also hoping for something more current. I suspect RH5 and RH6 are most popular...but I'm looking for facts -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm connecting a T1 PCI card to a Nortel Option 61 switch T1 card. My Sangoma A102D shipped with 2 T1 cables - which I assume are straight through. Do I need to make crossover cables for this scenario? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071026/9cea5e74/attachment.htm

Has anyone written an app that monitors SIP/IAX registration attempts? A couple of clients are being flooded with SIP registrations (but the source IP changes every few hours so IPtables won't do).. I would think that any attempt to reg 5 times with a bad password should cause a 5 minute timeout until reg is considered again. Has anyone written such an app? The name app_hackblock is my

Roman: I created two bash scripts called Mail2Fax and Fax2Mail for use with the asterisk sever. They leverage the app_txfax and app_rxfax scripts, along with ast_fax. They make using these apps a lot easier, including being able to mail to fax@domain.ca for outgoing faxes and then extracting phone numbers from the subject line! (Makes it easy to use with Sendmail without complex rules /

Is there a simple way to get call volume from a particular trunk within the dialplan (for conditional branching)? I suspect we will have to build an AGI script but I'm hoping something new in Asterisk 13 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150627/6774c750/attachment.html>

I happened to be in the cli tonight as some (208.122.57.58) initiated a simple attack - just trying to make long distance calls from outside context. Although harmless, this went on for several minutes as the idiot just used up my bandwidth with SIP messages. Here's and example: [2011-12-28 22:53:42] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension

I have an asterisk server with a dynamic public IP address. Once the IP changes, remote clients suddenly have one-way audio again. I can resolve the problem with a restart, but am thinking have adding a cron command which does this every night. Will a "reload" cause asterisk to respect the new IP address specified in sip.conf? Or do I have to restart? Thanks, MD --------------