similar to: Asterisk@Home Ver 1.2 Whats new?

Hi everyone, Is there a simple way of answering a different ringing extension from a sip phone using AAH? I have absolutely zero technical know-how when it comes to modifying conf files etc. Still working on figuring it all out. ;) That brings me to my second question... where the hell does one find an extensive manual of sorts that explains all conf files and what the strings all mean etc?

Some people don't know how to conduct themselves on usenet and mailing lists. Despite the mailing list guidelines being crystal clear about no html posting and not top-posting we have people still doing so. People still use text-readers to access the mailing list so you could be denying access to these people by flouting the mailing list guidelines. This is an example of top-posting

Is it me... or is it voipjet? This week I've been trying various providers, just can't seem to get voipjet to work. I signed up with voipjet but so far can't get it to work inbound or out bound. I always get 'all circuits busy'. May 12 22:27:05 VERBOSE[2442]: -- Executing [1;36;40mDial[0;37;40m("[1;35;40mSIP/101-ad89[0;37;40m",

I know this has been gone over before but It seems that most of the traffic to this list is the same 10 questions being asked over and over again. What if someone (with some programing skill) wrote a script so that when someone posted to the list, it would search the wiki and google and respond to them. If they where not happy with the response, then they could forward there message back

I've been going nuts lately trying to get rid of an annoying echo problem that makes my asterisk server unusable when clients try to call me. Here's the breakdown of the issue - Hoping that someone can throw me a clue: My setup is as such: Single AMD Athon machine with X100P clone card and voip through multiple providers . * Inbound calls through the X100P that do not bridge to

I was just checking out the dids for all of my fail over providers and noticed that neither DID that I have with SixTel work. Both pause for a long long time The local number gives a recording: 'The number you have dialed is not in service or is assigned in a different area code. Please check your number and dial again'. The 800 number just rings busy. Anyone else having this issue or

Anyone use livevoip? opinions? -- JD Austin Twin Geckos Technology Services LLC email: jd@twingeckos.com http://www.twingeckos.com phone/fax: 480.422.1250

The registry's are stored in DB. Just export your database with 'database show' Schedule it with cron to run every 5 minutes or so. You can do that with -rx command line switch for asterisk. Send the file across to other node and pipe it through awk/perl/cut or whatever you like and import it when you bring the other node up. You will have to stop and start asterisk I think. I

Hi all, when I type this: ruby script/plugin install svn://projects.jkraemer.net/acts_as_ferret/tags/stable/acts_as_ferret from my rails project the plugin is not getting installed, its simply going back to my rails application. I don''t find the plugin in the vendor/plugin directory I am using Instant Rails for your information. Could anyone please let me know whats going wrong...

First encounter with * Just downloaded & installed aah-2.7 Started up AMP, but i can not find any reference towards isdn. I presume there has to be some configuration done for my Eicon-Diva-pro. Does aah actually support isdn-bri? On the mail-archive i found some references, but these are rather old ( they speak about the coming release of aah-2.1) aah-handbook (version 1.6) doesn't

Dearfolks-- I have been told by an experienced R programmer and teacher whom I trust that it is easier to understand R code if you read it aloud, as the language that it is. However, she was clear that reading it aloud was not simply reading the marks on the screen: you read A.df[5,] as "the fifth row of A.df" (or "the fifth row of data frame A"), not as "A dot df left

Hi all I have installed AAH 2.2 in my P4 PC following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp and made as per the guide says and downloaded SJ Phone, and registered user and when i try to dial the 19197543700 i get message that, all circuits are busy now, please try your call later and when i see in the console i get this mesage any help Called easycall/19197543700

Hi all, We use AAH to run our office telecoms registered with two Sipgate accounts. Today, Sipgate appeared to have had problems with their server with oneway audio on every call. In order to cause the Sipgate message service to pick up in stead of our AAH box, I simply unplugged the network cable. We now have problems where AAH does not seem to access the network. I plugged the network cable

For a while, I''ve been getting that HMT is replacing HABTM. It appears that HMT can do all of what HABTM can do and more. The question is: Should I stop using HABTM? Let''s take a simple case: A case has many categories For a given category, there are certain valid statuses Category has_and_belongs_to_many :statuses Status has_and_belongs_to_many :categories Question: Is

Hello, does anyone know if there is a tool in existence that can monitor samba shares like smbstatus but is much more advanced (and runs in console) , I mean a tool that has menues, some search options, and things sorted out nice. A Samba realtime monitor would be the best thing with lots of details and lots of options. I tried Swat, but that aint really a solution to watch who is connected

I'm trying to set up extensions and have them forward to my cell phone or other phones I have and include them in call groups. I tried *72480204xxxx and *7298480204xxxx, I get the recording that unconditional forwarding is set to that number.. but when I call that extension I just get silence and eventually it hangs up. Someone throw me a clue stick? JD

I'm still relatively a novice with asterisk and am having issues with echo. The calling party that calls a PSTN number doesnt hear the echo, but the answered side via sip or forwarded to another PSTN number over voip hears excessive echo that makes it difficult to communicate. I've been playing with the zapata.conf settings for echocancel, echotraining, rxgain, txgain, etc and am

Hi Does anyone know the clear advantages over using asterisk rather than asterisk@home. Is the home version limited in anyway etc? Many thanks in Advance Scott

Hey all, is it safe to run the asterisk-update.sh script that comes with AAH to upgrade only the asterisk binaries? Doug has chimed in a few times saying 'upgrade' when I post problems, but Aah makes this really painful. I'm using AAH 2.0 & am fighting a number of 'bugs' that only seem to be manifesting in my installation. Can I safely upgrade just asterisk and not any of

I'm still a learning but I purchased a dph-140s to test with AAH 1.5. I think this is a rebadged ACT P104SLD which others seem to have working with *. It seems to be configured and registered similarly to the softphones I've been using just fine, but it does not receive or send audio (it will send audio to vm), or perform the loopback test. It seems to signal and receive calls fine. The