similar to: You ASKED for an Asterisk book, you GOT an Asterisk book!

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

Anyone in the Toronto area interested in getting together to share notes and swap war stories? -- Jim Van Meggelen jim@vanmeggelen.ca -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.6.9 - Release Date: 06/01/2005

Folks, In light of all the troubles people report when running more than one TDM400 card in a system, I wouldn't mind hearing what your solution of choice would be when having to connect 5 or more analog telco circuits to an Asterisk. I'll try and compile the answers together and get them into the Wiki, as I figure this could be useful knowledge for the community. TIA, Jim. -- Jim

Hello, We have finished our tests of the new Digium firmware on the quad T1 cards(TE405P/TE410P). Overall it is a big improvement over the version 1 firmware. Here's the review: http://astguiclient.blogspot.com/2005/09/digium-405p-v2-review.html MATT--- -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello On a Sipura SPA-841 handset (and also at other end) you hear a sea wave like sound - it gets louder then softer and continually repeats. I don't remember hearing this when using other handsets. But what is this effect? How can I reduce it? Angus

Dear All Can you please let us know how we can modify our Asterisk "extensions.conf" file so it interprets the subscriber dialed digits in one-by-one digit manner . At its current configuration , it interprets them in an whole packet . I mean , say the subscriber dials as "665 0000" so we need Asterisk to send it to the peer switch as 6,6,5,0,0,0,0 but not as one

Hi everyone, I have a question concerning DNS SRV lookups. The situation is like this: - one central Asterisk server - many domains with SRV records, let's say we have bar.com and doe.com Now the question is: if the SRV lookup is done for foo@bar.com the call is mapped to foo@myasterisk.mydomain.net. Is that correct? If so, I have a problem: if somebody calls foo@bar.com, Asterisk

Greetings Asterisk Users, I'm happy to announce that Russell Bryant and Leif Madsen have volunteered to host the next Asterisk Tech Tips webinar, next Thursday April 21 at noon central time. Russell and Leif are project leaders and have collaborated on two Asterisk books: Asterisk: The Definitive Guide and Asterisk Cookbook , both published by O'Reilly & Associates. Asterisk: The

Hello: My Asterisk book is on its way, so please bear with me. Based on what I have read and my actual Asterisk experiences, I am not too clear on the context-extension relationship. I am not sure if some of the error messages (Not Found) are a result of a bug or a feature. My experience so far is limited to sip.conf and extensions.conf, as I don't have a hardware board yet. First: It

Hi List! any body use www.simpletelecom.com? I subscribe to www.simpletelecom.com for A-Z termination and paid US$15.00 and US$70.00 via credit card in two days, but my account has US$15.00 only. I checked my credit card from the bank and they said me the payment already paid to merchant. I've lost US$70.00 :( so anyone here has experience with them? are they a SCAM? Thanks! </Madhawa>

Hi This is the output from show dialplan dial-sipmnf-sippt-pstn [ Context 'dial-sipmnf-sippt-pstn' created by 'pbx_config' ] 's' => 1. Verbose(1,Dialing ${ARG1} on mnf pt pstn) [pbx_config] 2. Dial(SIP/${ARG1}@${SIPMNF},${ARG2},${OUTBDIAL}) [pbx_config] 3. Set(GLOBAL(FOUNDME)=${DIALSTATUS}) [pbx_config]

Hi there, I?ve been looking to do an outbound dialer for systems alerting, etc. and have in large part followed the recipe here: http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out That and the associated pages at voip-info give a basic set of recipes for callfiles, but nowhere there or in my copy of the O?Reilly book by Meggelen, Madsen, & Smith can I find a detailed

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

Before I play around with this again in 1.2.1, regexten is still essentially broken, correct? The misconception seems to be that it allows you to execute a command upon registration from a SIP UA. Even the O'Reilly TFOT book erroneously states that this is what it is for. After reading the developer discussion though, it definitely seems to be broken. Is it fixed yet? Doug.

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:

Hi, I have what I think is a relatively advanced question. Any help is appreciated, even if it's not a complete answer. I am using Asterisk in mostly realtime fashion, specifically SIP registrations are in a MySQL table. This works fine (mostly). I also set a few variables in the setvar column, like this: callerid_internal=test <710>;did=5555551234 Again, this works

I would like to have the ability to have Asterisk announce the temperature -- not using TTS -- within the dialplan. For a non-Asterisk project, I have a cron job that periodically pulls down an XML file from weather.com containing local weather data (TWC's user agreement requires that data be cached locally). Using sed, I also create a text file that contains only the numeric value of the

I am running * 1.0.3 for some reason when I start * is does not appear to be parsing my zapata.conf file. I do not see any errors * just does not seem to know to look for zapata.conf. I am unable to use my FXO card to make calls or receive calls. I have been able to configure SIP to work correctly. Any help would be greatly appreciated, I spent most of last night searching for an answer.

> 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

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