similar to: app_hackblock to prevent SIP/IAX reg trolling

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

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

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:

I'm looking for recommendations for standalond PRI to SIP converters. (Needs to be outside the asterisk box - so a PCIe card won't do) I've used redfone but this project doesn't need the redundancy features... Thanks!

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:

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

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

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 --------------

I have a simple asterisk install (1.4.18), and want to use call parking. I can successfully park a call (I see on the CLI that the call is parked to 701). Everything is pretty default. However, I can't pickup a call from another phone. When I dial 701 from a phone, asterisk can't find that extensions and notifies the person picking up that the extension doesn't exist. (It appears

I'm working on a script that needs to determine the extension (eg: 123) of the phone that initiated the call, or CALLERID number if an externall caller. Is there a simple way to do this? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090724/b663db29/attachment.htm

I can't find this in the wiki/email history..but I'm sure it's based asked before. The port range define in rtp.conf - is that for connections initiated by asterisk? Or the port range asterisk listens on? Or both? Thanks! MD -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is there a prebuild module/dialplan which gives me a nice interface to recording messages? Assuming I can't use the voicemail command, I need to offer users a way to record, playback, erase, rerecord, etc. I can probably do it through dialplan but it feels like I'm reinventing the wheel. Thanks, MD