similar to: Incoming phone line rollover / hunt?

Hello, It's been noticed that mailing list monthly rollovers didn't happen because the machine was down when the cronjob to do the rollover would normally have fired. I will trigger them by hand now... upshot is that I'm not bothering to refile the first five days of May in the correct slot. I sincerely hope that no one cares :-) Monty --- >8 ---- List archives:

Hello, It's been noticed that mailing list monthly rollovers didn't happen because the machine was down when the cronjob to do the rollover would normally have fired. I will trigger them by hand now... upshot is that I'm not bothering to refile the first five days of May in the correct slot. I sincerely hope that no one cares :-) Monty --- >8 ---- List archives:

Hello, It's been noticed that mailing list monthly rollovers didn't happen because the machine was down when the cronjob to do the rollover would normally have fired. I will trigger them by hand now... upshot is that I'm not bothering to refile the first five days of May in the correct slot. I sincerely hope that no one cares :-) Monty --- >8 ---- List archives:

Hello, It's been noticed that mailing list monthly rollovers didn't happen because the machine was down when the cronjob to do the rollover would normally have fired. I will trigger them by hand now... upshot is that I'm not bothering to refile the first five days of May in the correct slot. I sincerely hope that no one cares :-) Monty --- >8 ---- List archives:

Hello, It's been noticed that mailing list monthly rollovers didn't happen because the machine was down when the cronjob to do the rollover would normally have fired. I will trigger them by hand now... upshot is that I'm not bothering to refile the first five days of May in the correct slot. I sincerely hope that no one cares :-) Monty --- >8 ---- List archives:

I was hoping that rolling over to the next trunk would be simple, but it doesn't appear to be so, especially for a newbie. So I'm looking for a simple way where if I get a busy on the first outgoing trunk, I can do something to get connected to the next one. Perhaps something like the big boys do and dial 9 first? I'm guessing a custom dial plan might do that but I haven't

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

> 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

I have about 5 incoming USA SIP lines, but my provider does not have any sort of roll-over or huntgroup feature. Does anybody have an idea on how I can create a general number that will ring to the next available, non-busy SIP line that I have? Is there a provider out there that would do this? Any suggestions would be greatly welcome. Thank you. -------------- next part --------------

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

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, 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

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

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

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

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