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...address, username, password, mailbox and ringtone. - -- Kind Regards, Matt Riddell Director _______________________________________________ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHl6vCDQNt8rg0Kp4RAospAJ9DUNge64n7u3RkQWsodHgdOS/higCgwNFy VfZUUNJIgzeC4Hy5vg0f+mY= =tpnK -----END PGP SIGNATURE-----

Hi All, I am trying to use the AMD (Answering Machine Detect). But it is not sending the AMD_Status as either the Human or Machine, it hangs up in middle. can any one suggest us, what might be the problem and possible solution to it. below is the log -- Executing AMD("SIP/sip-ffe0", "") in new stack -- AMD: SIP/sip-ffe0 14082284927 (null) (Fmt: 4) Apr 23 08:00:26

>Soumya Kat wrote: > What I would like to know is how to get information such as SIP users, > number of SIP connections and traffic associated with those from asterisk > using a C Code. >Russell Bryant > There is actually no good way to do this inside of Asterisk right now. It's > certainly all possible ... it's just software ... but there is no > straightforward

...o use some non standard form of SIP. Any pointers? - -- Kind Regards, Matt Riddell Director _______________________________________________ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFH1KVoDQNt8rg0Kp4RAgSfAJ0aXGIkKi6kGAjZK8TtSV2mMj79qQCdHhAS 1jZ9sjtsTJ3O1R9J3giztw8= =Mlnt -----END PGP SIGNATURE-----

...has seen it/has a fix for it I can do a back trace. - -- Kind Regards, Matt Riddell Director _______________________________________________ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFH1yudDQNt8rg0Kp4RAtpyAKCkh6AQgHWYtW6gB8NgMnm/E2RoUgCgmvsx lVnVU5Bd8Wwk1b8GAaLyhxY= =4997 -----END PGP SIGNATURE-----

hi, all asterisk 1.4.24 , zaptel 1.4.10.1 , E1 Manager API Action : Action: Originate Channel: ZAP/G1/8888888 Callerid: 12345678 Context: callout Exten: s Priority: 1 extensions.conf [callout] exten => s,1,Answer() exten => s,n,Wait(10) exten => s,n,Hangup() when the phone 8888888 pick up , it will come to callout context, after hangup, one cdr generate, but the

We just completed a new implementation of voicemail for Asterisk. It's much cleaner than Comedian mail and can emulate several voicemail user interfaces, including Audix. It's a great replacement for Audix. All of the sounds/prompts are presently being re-recorded by a professional female voice. If you are interest in the app, let us know at nt_jnewman at yahoo.com. Justin

For some reason, I have been unable to find the answer to this online or in books... I want to have a "click-to-connect" feature on my website where the user enters their phone number and then my asterisk server calls their phone and my phone and connects the two calls to each other. All I have are: 1. A Server 2. A DSL connection 3. A Router and DSL Modem 4. A static IP What do i

Hi All, I'm stumped on this and I looking for some clues to fix this. This is a new install of Slackware 12.1 onto an IBM x330 Server. Asterisk 1.4.20.1 plays the wav files and the Cepstral_Allison Swift just fine, but when I play the gsm files the audio quite choppy. And, the files produced from the MixMonitor don't even record any audio other than noise. I have a hard drive from

If i use AMD() or the code below, now the problem is the fax machine/modem detection and answer machine detection get detected as the same. If i need to seperate the two how do i do that? For example, if i use AMD() to detect an answer machine by saying any greeting exceeding 2.5 seconds is a machine, how do i distinguish between a fax/modem and a long greeting? ---- dave cantera

...mISDN, mISDNuser all to no avail. Does anyone have any ideas? - -- Cheers, Matt Riddell Director _______________________________________________ http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com (Free Asterisk Voip Community) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHSzNrDQNt8rg0Kp4RAiw4AJ4nyTu5Bh/Cg5ij9X5WNH8VRbCevwCfcadG 6Eae/P72BPp/+FBTg7atXuI= =Sf66 -----END PGP SIGNATURE-----

Hi, I'm testing Asterisk 1.4.15 with the -g option. When it crash didn?t generate core file in the /tmp folder. What is happening?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071130/dc693449/attachment.htm

Hi All We received following error .Please help us to sort out. WARNING[3281]: frame.c:1426 speex_samples: Had error while reading wideband frames for speex samples. Regards Nirukshitha ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search.

Hi, Using Asterisk 1.4.17. I'm calling a PHP script through AGI. No matter what my script returns (0 or -1), AGISTATUS always appears to be 0 = SUCCESS. I was wanting my script to be able to return a value to the dialplan and then test AGISTATUS but it looks like I'm going down the wrong path. Any suggestions? Thanks, Brian -------------- next part -------------- An HTML attachment

I'm no longer on the DEV mailing list, but: # svn checkout http://svn.digium.com/svn/asterisk/branches/1.4 asterisk svn: URL 'http://svn.digium.com/svn/asterisk/branches/1.4' doesn't exist http://svn.digium.com/svn/asterisk/branches/ -- /Nick -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am setting up an Asterisk server to provide voice messaging in a campus setting. I am interested in how others have Asterisk set up in regards to firewalls and web interface access to minimize security risks.

(alternative title - what did I do wrong? or suggestions to make this work) Thought I'd try 1.6 beta5 (and 1.4.18 didn't want to compile vpb /usr/lib/gcc/i386-redhat-linux/4.1.2/../../../../include/c++/4.1.2/i386-redhat-linux/bits/gthr-default.h:48: error: ? does not name a type ) 1.6 did compile and almost works. 'cept it thinks the .gsm files are not played. from

I've been searching around for a while, and haven't found an answer to this question, so here goes: Does anyone know if AMI can be configured to allow requests from another client without having to authenticate first? I would like to be able to restrict it based on IP address, and not require a login. Any help is appreciated. Thanks! -- Brooks R. Bridges Telecommunications Manager

Hi, Russell's blog[1] is down and there are not much information about this any where else. Any one with more information about res_ais and how it is used? raj [1] http://www.russellbryant.net/blog/index.php/2008/06/10/asterisk-16-now-with-distributed-presence/

Hi! What are the typical ways to work around the 64 groups limit? thanks klaus