similar to: Asterisk behind NAT << How to do it.

> 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

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

Hi All, I have just compiled the newest version of mpg123 on a RedHat 9.0 system (mpg321 has not been installed) and I am using the newest CVS version of asterisk. Whenever I place any mp3 files in the /var/lib/asterisk/mohmp3/ directory, Asterisk crashes a horrible fiery death. If mp3s exist in that directory, then I can't even start Asterisk. If I start it without files then copy

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

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

Hi, I have a strange problem trying to update Asterik on one of my boxes. I have done the following: - delete all the old source files - download the new file using: cvs checkout zaptel zapata libpri asterisk - compiling the new source file using: make clean ; make install Everything is compiled fine, except.... when I stop and start Asterisk server, it is still the old CVS version:

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

Go to www.google.com type in your search query add this to the end of your search query: site:lists.digium.com e.g. http://www.google.com.au/search?hl=en&ie=utf-8&oe=utf-8&q=Australia+site:lists.digium.com The mailing list used to be on www.marko.net, I'm not sure if the whole archive was moved across, you might want to search with site:www.marko.net OR site:lists.digium.com

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Here's a version I modified which grabs either a development or stable verision, and does a backup before updating from CVS. It also asks for addon's and cc. Leif Madsen did the original development and Mark released it. My changes does the minimum changes to previous version, to get what I need. It does the same version checking as

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

Thank you to everyone who has offered so far! I've had formal offers from Martin List-Peterson, William Suffil, Greg Varga, Brian Cuthie and Ed Guy (hopefully I haven't forgotten someone....!) Now we just have to decide where the best spot to host it is. What do you guys think? For this week, I don't care if this is a one off. At some point I'd like to have a weekly conference,

Hi, I've been on the list for slightly under a month now and noticed; a) a fairly high amount of traffic, b) a lot of questions which come up more than once, c) the archives at lists.digium.com are not searchable. I have started development to import the mailinglist archives into a MySQL database and creating a full text search possibility on this. My questions; 1) Is this already done

Hi all, I had a chance to use some call conferences that had some very neat functionalities: - When you call you are first asked for your name - When someone joins the conference a message "<name> is now joining the conference." is played. - When someone leaves the room a message "<name> has left to conference." is played. How can I set MeetMe/Asterisk to have

Suppose: ipchains -A forward -s inside_net -d 0/0 -j MASQ -m 100 (similar setup with iptables: iptables -A PREROUTING -t nat -s inside_net -d 0/0 -j SNAT iptables -A PREROUTING -t nat -s inside_net -d 0/0 -j MARK --set_mark 100) eth0 = outside iface eth1 = inside iface now: tc filter add dev eth0 ... handle 100 fw should catch packets marked by the above rule in ipchains (iptables). Ok. When