similar to: SIP on TCP

Here's the latest batch of words to get shipped out to Allison Smith. Please submit reasonably small changes to me by tomorrow 10:00 AM Eastern time, and I'll add them. As usual, donations to what will be a ~$110 USD expense would be appreciated, as I am paying for this round out of my pocket. Please send to paypal address "jtodd@loligo.com". I did not include all

I really should be doing something better on this beautiful weekend, but I'm trying to save myself some time for later projects by documenting some things that have been particularly troublesome in the past. That being said... I've written up a configuration guide for the Cisco ATA-186, which describes some of the features that are possible to set in the ATA and specifically what

Hi, There is any simple way to have external access to the Asterisk database? I want to be able to edit a specific family only, which is a phonebook. Thanks, Dan

Hello All, There have been some reports of person(s) checking out the latest version of Asterisk using CVS, and having serious difficulties compiling and installing it. I thought the person(s) who are concerned about having these problems might like to know that I checked out Asterisk Version: Asterisk CVS-09/14/03-08:48:21 compiled and installed it with no problems of any kind. Everything I

http://www.nero.com/us/631911127302064.html Have you all seen this? Its a SIP softphone put out by the people that do the CD burning software Nero... Check it out it works with * Dave

Hi, I've had a search through the archives and didn't find much. Is anyone using the Monitor application? I have it working but there is a really big drawback. The files are always called the same thing, which means if I make 2 calls one after the other the first recording is lost. I half expected Monitor to use something like ZAP-2-1-<yyyymmddhhmmss>-in/out.wav for it's

Can asterisk act as a SIP registrar or location server? I would like to be able for a user agent(client) to register with whatever client they are using as "username@domain-name.com". Rather than the entry/username/password that is setup in the sip.conf file. That way a user could log into any SIP enable client and their calls would follow them around. I have read the sip.conf man pages

The soundfiles I submitted earlier today have been cleaned up, and added to the Digium CVS server in a more formal manner. Also, some of the really bad formatting in my .txt description file has been rectified. All of the sounds on my website are now on the Digium site, and I will be submitting future changes via patches to Digium for additional sounds. Ideas welcome for more text; I may

Sipura does include STUN as an option. It has for quite some time. We are using it with all of our Sipuras behind NAT'd gateways and it works great! Try upgrading to the latest Sipura firmware rev. Darren Nay > -----Original Message----- > From: John Todd [mailto:jtodd@loligo.com] > Sent: Saturday, May 22, 2004 1:57 PM > To: asterisk-users@lists.digium.com > Subject:

http://www.loligo.com/asterisk/misc/Presentations/Asterisk-features-20040106.xls I had been asked a while ago to put together a short Excel spreadsheet listing many of the "common" features of Asterisk as compared to a typical PBX. Many PBX vendors supply an exhaustive list of their features, and I figured I'd take as many of the unique features as others had offered, and put

I'm wondering if there is any command-line interface available for working with values stored in astdb. Of course, I can run "asterisk -rx "database show" " or other commands like that, but I was hoping for a local command that would allow manipulation or output in some other form. Is astdb in a standard db format? JT

This is a fairly large list of hardware phones, features, and URLs, with some vendors I've never heard of before on it. I suspect that any phone that doesn't explicitly say "SIP" is an H.323 phone. http://www.tmcnet.com/it/0403/0403sg.htm#sidebar2 JT

I have several phones running Cisco's 6.0 SIP software release at this time. Two of the phones have not shown any abnormal behaviors, but one of them has an unsettling propensity to lock up after several hours, where the softkey labels disappear and the phone stops registering, requiring the standard *-6-settings reboot sequence. Otherwise, the phone seems to work OK except for a slight

This is not specifically Asterisk-related, but I think that sometimes a "pre-emptive" clue is a good thing. Viking Electronics (http://www.vikingelectronics.com/) has some neat widgets that attach to phone lines, which I'm sure many of the people on this list would find at least somewhat useful in conjunction with their Asterisk systems. Many of the widgets can be replaced with

Here's a clip of comments lifted from a Cisco bug list. This will be perhaps useful to those of you who have just purchased a Cisco phone off eBay. JT ------------- (1) Short problem description: Documentation on how to load SIP image on phone with skinny software (2) Longer problem description (what happens): If the phone is loaded with the Cisco Skinny code, then there is a small

Hello. Well I finally rx'd my dev kit (new batch of TDM's apparently. I'm on Mandrake 9.1 There were no hardware install instructions, it would have been nice to know whether the 4-way power connector was to be used or was for some other future or expansion purpose. It came with a floppy disc, no label and it wasn't even write protected. The only readme file was

John, I spoke with Level(3) last week regarding SIP termination. They quoted $0.01/minute, with an 11 Million Minute / Month minimum. Ugh! -dg -------------------------------------------------------------- Darnell Gadberry President binaryMedia darnell AT binmedia DOT com ------------ Date: Fri, 19 Dec 2003 21:12:22 -0500 To: asterisk-users@lists.digium.com From: John Todd

does anyone have a sip working through a cisco pix firewall? i can get the phone to register and the call to be negotiated, but as soon as the call is answered there is no sound and the call ends immediately. im sure this is due to the RTP negotiation being rejected by the pix. any helpful ways around this? right now my only solution is to put a small box outside the firewall and IAX the

I've spent some time on the Wiki adding documentation on all asterisk applications from the cli 'show application xxxx' commands. I've also added some cross references and pointers. http://www.voip-info.org/tiki-index.php?page=Asterisk If you find this useful, please go there and help us build a reference database. A Wiki is a wonderful collaboration and documentation tool -

Hello All, I am a newbie looking to learn about Asterisk. I'm new to IVR and all that goes with it. I would like to know if it is possible to grab the number of an incoming call, have Asterisk, or third party software return the call with an automated voice message allowing the original placer of the call to select another person to call (eg "Select 1 for Bob") then have