search for: madsen

...getting when someone trys to call be via FWD (all outbound calling works fine, and inbound has worked before as long as I have everything in the same context, but I am trying to seperate the contexts like in my previous dialplan): Executing Dial("SIP/fwd.pulver.com2-312e", "IAX2/madsen:password@nattedbox/55555@from-sip") in new stack -- Called madsen:password@nattedbox/55555@from-sip -- Hungup 'IAX2[207.61.247.201:4569]/1' == No one is available to answer at this time And here are the config files: *** NAT'd Box *** extensions.conf ---------------...

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:

...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 In my configuration I have internal SIP clients registering from 192.168.0.0/28 and my * address is at 192.168.0.100. Using the host address of the * box as the inside_net variable the audio from 192.168.0.0/28 was sent to the outsid...

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 pu...

...entation and submit it for inclusion. A section to submit documentation edits would be nice, as well as maybe a history timeline or something like that? Just some thoughts. If you know of something like this, please let me know. In the meantime, I'll be googling some more. Thanks, -- Leif Madsen - Telecommunications Technology Sheridan College - Trafalgar Campus ICQ: 3445119 FWD: 18924

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

Hi: Which was released for free download under a Creative Commons license for "The Future of Telephony, by Leif Madsen, Jared Smith, and Jim Van Meggelen". Regards. --------------------------------- Sick sense of humor? Visit Yahoo! TV's Comedy with an Edge to see what's on, when. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail...

...B? (yes, the database is started :)) I've verified that I can login to mysql with the username and password, and it's fine. Then I do a select * from asteriskcdrdb.cdr; to see if anything has been written, and sure enough, just Empty. Any and all help greatly appreciated! Thanks, Leif Madsen.

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, 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

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

...or channel bank is capable of? I just need someone to give me a brief explanation how it usually works, and if someone was implementing an Asterisk system, how they would go about providing a call rollover (single advertised phone number, but allow multiple incoming calls) Thanks in advance, Leif Madsen. -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.2 (Cygwin) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQE/bmJ86gq3eQ0gpNURApYRAKDViZhTUrygxcM3yqlPkdifK4jpuwCfeKVU wympd2pcbcUW1LA4HDeRLzY= =DLug -----END PGP SIGNATURE-----

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:

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

...ot;, "525867.000000", "525868.000000", "525869.000000", "525870.000000", "525951.000000", "525952.000000", "525954.000000", "525962.000000", "525964.000000", "525965.000000" )) /Jesper Runge Madsen Aalborg Universitet Denmark

-----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 the Make script uses should .version file not exist. It runs well for me. - -- Steve "They that would give up essential l...

.../ patch -p0 < /path/to/patch Nothing should fail. cd /usr/src/asterisk/ make cp /usr/src/asterisk/channels/chan_sip.so /usr/lib/asterisk/modules/ Restart your Asterisk and try it. If you want to call a NAT'd Asterisk box, my Free World Dialup number is 18924. Currently online. -- Leif Madsen <leif@hacklocalhost.com> http://www.hacklocalhost.com

...txgain + Immediate? (yes/no) + CallerID String + Name + Number + Enable mailbox indication? + Mailbox number(s) to be associated with this channel + Context Would you like to setup registration for FWD, IAXtel, SIPPhone or iptel? + -- +------------------------------------------+ |Leif Madsen - http://www.hacklocalhost.com| +------------------------------------------+ | @| leif at hacklocalhost dot com | | SMS| sms at hacklocalhost dot com | | FWD| 18924 IAX| 1700-363-0761 | |iptel| 8972-1969 sipph| 1-747-386-1618 | +-----------------------------------------...

...ions such as this? Are you charging on a per hour basis? A per line? Are you charging to activate certain features such as voicemail, call parking, call queueing? I'm trying to get an idea of how a standard Asterisk installation might be charged to a business. Thanks in advance, - -- Leif Madsen - Telecommunications Technology Sheridan College - Trafalgar Campus @: leif.madsen@sheridanc.on.ca ICQ: 3445119 FWD: 18924 IAX: 1-700-674-5480 -----BEGIN PGP SIGNATURE----- Version: PGPfreeware 6.5.8 for non-commercial use <http://www.pgp.com> iQA/AwUBP2DBheoKt3kNIKTVEQJfaQCfe1wqiKxU...

...ox 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 - http://www.hacklocalhost.com| +------------------------------------------+ | @| leif at hacklocalhost dot com | | SMS| sms at hacklocalhost dot com | | FWD| 18924 IAX| 1700-363-0761 | |iptel| 8972-1969 sipph| 1-747-386-1618 | +-----------------------------------------...