similar to: Creating a SIP gateway for use behind NAT

I'm hoping someone can help me out with this. I am basically just trying to separate my outbound and inbound calls into separate contexts instead of having everything in a single context. Any help would be appreciated. Perhaps I've missed something really obvious.... Here is the network layout: <remote> <--TDM400P--> <nattedbox> <--IAX--> <liveipbox>

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 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

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

Hello all, I would like to document some things I am doing with asterisk, but would prefer to do this from a web interface. I am unfamiliar with any software that allows you to create online documentation from a web interface. Ideally I will be able to create documentation online from a browser, which then when saved, is immediately ready to be read online. Perhaps I can setup different authors

Hi i have buy a Digium TDM400P with 1 fxo modules TDM01B for connect my asterisk to a french analog line. In my zaptel.conf, i have: loadzone=fr defaultzone=fr fxols=3 when i load the module, i have in logs: Nov 24 06:13:40 gw kernel: Freed a Wildcard Nov 24 06:13:40 gw kernel: Zapata Telephony Interface Unloaded Nov 24 06:13:40 gw zaptel: Removing zaptel module: succeeded Nov 24

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 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

hi all, I don't know how to setup asterix to work as PBX. If I want just basic configuration with 2 SIP phones (snom, ata), what all I have to write in the configuration files, or respectively in the configuration of ata and snom ? If there is any good documention available, send me URL too. All (ata, snom) are behind firewall (nat) and astrix is on the public IP, but I can move for

Hi, Is it possible to use asterisk with Free World Dialup (FWD) ? Did someone manage to make it work? how? Best, -P -- __________________________________________________________ Sign-up for your own personalized E-mail at Mail.com http://www.mail.com/?sr=signup CareerBuilder.com has over 400,000 jobs. Be smarter about your job search http://corp.mail.com/careers

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I have a medium sized business that is interested in implementing * as their PBX system. They currently have a Panasonic system with Panasonic handsets that they are going to replace Asterisk with, as the current system is maxed out, and they don't even have voicemail capabilities. I have been considering using an Adtran Atlas 550 with FXO and

Hi List, I am sorry that I may bring the old question to the community. My question is 1. How can we determine if asterisk is working normally or not ? what kind watchdog process do we have at this moment ? 2. In case the running asterisk is mulfucntion, is there any available way to auto restart asterisk ?? Please advise if you could. Thanks.

Hi, I'm evaluating what could keep me from upgrading production systems to 1.6.2. As 1.6.2 seems pretty close to 1.6.1 but I gave 1.6.2 a try but I met an issue with BLF-pickup which kept me from going further. Have you met other issues I should include include in my checklist ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk Documentation Project, in conjunction with O'Reilly Media are pleased to announce the official release of Asterisk: The Future of Telephony on Friday, October 14, 2005 at AstriCon 2005 in Anaheim, CA. In the true spirit of Open Source, the authors and O'Reilly Media have published the book under the open, Creative Commons

Dear All, I want to play music in my AGI while i am searching for a field in DB. Actually during some processes in AGI i need to play music . Thanks in advanced. Regards, Mehdi

Forgive me if this is a FAQ question but I didnt see anything on the website of forum spelling out the difference between 1.4.x and 1.6.x Obviously 1.6.x is in development. Is it stable enough for production use? What are the new features being implemented in 1.6.x? Will Cepstral work with 1.6.x? Thanks, Mike -------------- next part -------------- An HTML attachment was scrubbed... URL:

Can someone show me how to set up DUNDi, I will be using it to connect 14 asterisk servers internally. I don't want to use it on the external world. If anyone has any examples of connecting 2 or 3 (if their is a difference) machines in a DUNDi co-operation that would be helpful. Johnathan Falk Network Administrator Clinton Community Schools

Hi all, I think I've run out of options in terms of what I know about this. I have created a user called asteriskuser and granted all privileges to the asteriskcdrdb database. Then I created the table via the cdr_mysql.txt file. I have edited the cdr_mysql.conf file to reflect this, and added load => cdr_addon_mysql.so after compiling it from the latest CVS. If I check the

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