similar to: MP3s in /var/lib/asterisk/mohmp3 causes Asterisk crash

Hey all, I am in the middle of creating a new user wizard which will generate all the .conf's the new Asterisk user will require to get themselves up and running in Asterisk without having to touch a single configuration file. This is what I have come up with as a rough draft. It is far from complete, so I'm asking people to submit things that should be added, changed, removed

I'm just curious if I was to place my * box behind a a FW/NAT box running linux, if my SIP calls will still work. Box right now is a RH9 computer using iptables as the FW. I wouldn't mind placing my * box behind it, but I'm wondering if anyone has actually gotten NAT working with *? Thanks, -- +------------------------------------------+ |Leif Madsen -

I'm sorry this is somewhat offtopic, but I do plan to use this to help me create documentation for the * project.. so I guess it is somewhat on topic :) Anyways, I am looking for some sort of document control system. It should act somewhat like a CVS where it keeps previous versions, allows people to submit documentation, keeps track of who has what document open etc.. etc.. The

Grandstream and Global IP Sound have inked a deal in which Global IP Sound will provide its royalty free iLBC codec to Grandstream. GS will integrate this codec into the BT and HT product lines

Hello all I would like change the all announcemennt(Voicemailmain,Voicemail etc.). But I don't know how to change the these each prompt. Do we have any guide book for this? Please teach me about changing the voicemail or other prompt. Thanks

Does anyone on this list know of any Linux based apps that will work with Dialogic or Brooktrout that provides voice-mail/Autoattendant only?? It seems that Panasonic, Avaya, and Mitel all use Unix/linux based OS on their firmware for their proprietary voice mails. My wish list would be; A software that provides all of the drivers for a dialogic or brooktrout board Voice Mail Messages in WAV

In attempting to make my asterisk server the latest and the greatest tonight I attempted to upgrade my CVS. In the asterisk directory I ran the make clean, executed the CVS update -d, and after all the files completed ran make upgrade. My problem is that when I pull up the CLI the cvs version that is showing is the same date as my initial install. Does this mean that the upgrade did not go

Hello-- I've been asked an interesting question, and I'm too ignorant to answer it authoritatively (yet). Can anyone help me? Question: If I'm going to implement a somewhat small (10-80) phone system, and I have a choice of using VOIP phoneset (like SNOM or Grandstream or Cisco, etc), vs. cheap analog touch-tone phones, exactly what features will I kiss goodbye if I use the cheap

Has anyone tried installing * on a box with two eth interfaces which is acting as a NAT box? I have only one IP at this point and I would like to get * working without all of the NAT issues. My idea is to run * on my gateway (which is also running the firewall and masquerade services). All of my UAs (Grandstream + Xten X-LITE + gnophone) will be inside the NAT screen, and will connect to the *

I'm trying to get the total Linux/* installation size as small as possible. I'm wondering if anyone has looked at the installed packages list from the Redhat installation [rpm -qa] and has parsed out all packages not needed for * to run. I follow the custom install guide from Andy Powell but the installation yields 948+ Meg with 340 installed packages. I'm sure most of those packages

Thanks to ww and his patch on bug #104, I have successfully implemented Asterisk behind NAT without using STUN or anything crazy. It's quite straight forward. Until this gets tested enough and put into CVS, you will have to patch your chan_sip.c file to do this. I'm sure within the next few days this will get put merged into CVS if no one finds any problems. I tried this on chan_sip.c

> I'm pretty sure that is incorrect. The inside_net is the ip address of > the asterisk server, and the inside_mask is the subnet mask. At least > that is how I have mine setup in my sip.conf, and it works. > > inside_mask for the internal mask would make more sense to me as well :) > > -- > Leif Madsen <leif@hacklocalhost.com> > http://www.hacklocalhost.com

L.S. Situation: I am running Samba 2.2.7a as my DPC in a small business network with win98/ME clients. A big Fiery printer/scanner/copier device is also part of this network. It has its own printserver. Since Win98/ME can not print to an LPD queue directly, I have setup Samba to spool the printjobs send by the clients to the Fiery server. All is working well, but there is a small problem. This

I have the following in extensions.conf: [global] MP3ROOT=/var/lib/asterisk/mohmp3 [default] exten => 1111,1,Answer ; Answer the line exten => 1111,2,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten => 1111,3,MP3Player(${MP3ROOT}/sample-hold.mp3) The command that runs is: 14030 pts/0 S 0:00 /usr/bin/mpg123 -q -s -b 1024 --mono -r 8000

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Hi, rpcclient $> deldriver "Fiery X3e 31C-M PS v1.1 eu" Failed to remove driver Fiery X3e 31C-M PS v1.1 eu for arch [Windows 4.0] - error 0xbb9! (samba 2.2.8) Thank you -- Guy Roussin

Hi all, I would like people to email me at 'leif at hacklocalhost dot com' some example configuration files for VoIP providers which * can register with. I am going to expand upon the FWD php "wizard" I created for these other providers, but I need some examples as I don't actually use anything but IAXtel and FWD. So far sipphone and iaxtel has been mentioned. I can

Does anybody know why I can NOT hear the MusicOnHold - using SJphone on another PC in our network (normal playback is not a problem) . See the * output and the line configured in extension.conf below (also mp3player does not function) Any suggestions? *Asterisk output:* *CLI> -- Executing WaitMusicOnHold("SIP/jeroen-bf54", "30") in new stack --

I have a couple of packages (?fiery? and ?reqres') that uses a mock of a rook request for their examples. The mock is an R6 object containing a rawConnection, along with a finalizer that closes this connection when the object is removed. So far so good. I?ve recently been getting CRAN errors due to the rawConnection not being closed in examples > cleanEx() Error: connections left open:

Hello Asterisk's Users ! Is anybody knows if the default MP3 tracks provided with the lastest release of asterisk is free of rights or not ? The default tracks are : - fpm-calm-river.mp3 - fpm-sunshine.mp3 - fpm-world-mix.mp3 Regards, -- S?bastien Mortier AbsysTech Tel : +33 892 460 991 Fax : +33 320 745 005 Gsm : +33 620 792 429 Assistante : Sarah Foucart sfoucart at absystech.fr