similar to: best practice

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Has anyone ever seen an Snom320 lose settings? It's been working fine for months and then I got a call this morning saying that it was asking for country, timezone etc. I logged in remotely, and it had lost the server address, username, password, mailbox and ringtone. - -- Kind Regards, Matt Riddell Director

Quick question ... I am interested in installing Asterisk and using SIP or IAX to connect to another system. If I am not planning to install analog or digital cards to connect to another system do I need to install zaptel still? -------------- next part -------------- An HTML attachment was scrubbed... URL:

>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

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Has anyone done any integration with this? All I know so far is that it appears to 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 -

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

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all, anyone else seen RX/TXFax crashing Asterisk on latest Asterisk SVN? I've now seen it on two machines I tried to set up - one it seems because the tiff file was malformed, but the other is doing: tiff -> tx fax -> zaptel -> pstn -> ddi -> zaptel -> rx fax -> tiff The above crashes every time. If no one else has

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

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

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

(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

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:

Greetings, I am writing an AGI script that needs to check on the idle/busy status of a number of SIP peers (mostly SPA9xx phones, with a few Polycoms and Snoms thrown in for fun). Is it possible to call Asterisk functions (e.g. SIPPEER) from AGI scripts? Based on my Googling, I would guess in the negative. I have tried various permutations of Set() and Eval() without success. I have also

I have the following situation: I drop a call-file into the Asterisk spool directory and I get called back. That all works. And I have this script: #!/usr/bin/perl -w use Asterisk::AGI; my $AGI = new Asterisk::AGI; my %input = $AGI->ReadParse(); $AGI->answer(); my $i; $i = $AGI->channel_status(); $AGI->say_digits($i); $i =

Hi, >However, I believe that zaptel >= 1.4.6 or zaptel 1.2 >= 1.2.21 should >support hires timers for timing on kernel >= 2.6.22 . > >What version of Zaptel do you use? > I was using version 1.4.5.1 I just downloaded and installed version 1.4.7, configure/make/make install finished without an error, but when is used modprobe ztdummy the system said: FATAL: Error

Quick Question... I'm trying understand, if it's possible to run an agi script to obtain a user's account balance and from there asterisk would be able communicate that value back to a sip phone. Is that phone feature, or is that an asterisk feature already? Think of this as on a prepaid platform...Before every hangup, the account balance is sent to the user. Hope I'm clear