similar to: what is wrong with gsm files

Hello, after upgrading my asterisk from cvs few days ago I discovered that all my anddefault asterisk voice messages/prompts was played wrong - they was played so slowly ... and the IVR system wasn't usable at all. Today I upgraded everything: a fresh cvs copy, made make install and make samples - but after that the problem with IVR was the same: ALL default asterisk sounds in gsm format

Hello, today I found this security report regarding Asterisk SIP Security. http://www.securiteam.com/securitynews/5LP0720B5G.html Maybe It could help somebody who isn't using a newer than 15th of August cvs version. Best regards Lubo

Hello, There is a new asterisk vulnerability report at this address: http://www.securiteam.com/unixfocus/5HP0H1PB5S.html This is the second security report regarding asterisk for 8 days (http://www.securiteam.com/securitynews/5LP0720B5G.html) Both fixes was reported and fixed silently. My question is: Is it possible in the future such a security problems to be reported in this mailing list

Hi Everybody, I am new to Asterisk. Can anybody suggest me some link where I can find architecture level detail of this system. My aim is to find out how easy it is to port it on a new hardware (T1/E1 and POTS)? Any input is highly appreciated. Regards Arun

I'm looking for a prepaid calling card platform that: * easily scales to multiple servers with a common database for: redundancy, capacity, and performance Looking to start with capacity to handle 100 simultaneous calls and be able to easily scale to 1000+ simultaneous calls. * in addition to the normal anti-fraud measures, supports an API for easily adding new anti-fraud tests along the

Hi everyone, I'm having a little problem and was wondering whether anyone would have any ideas or pointers for me. I've been working on load-balancing asterisk and have had a pretty successful setup using LVS and IP tunneling (plus a bit of iptables nating). I am only load balancing the SIP registration while the RTP between the SIP phone and the asterisk server and between the

Today class we are going to be talking about the wonderful line of GrandStream products. Or should I say BarbieTone phones? Who else is having MAJOR issues with the grandstream products? How many times have you been told upgrade upgrade upgrade? How many of you have paperweights, granted the phone is light as a feather and couldn't weight papers down in the first place? How about that

Hello, we have an asterisk (1.4.21.2) with ZAP and mISDN channels, the mISDN are 2 incoming GSM Modems, each with 2 simcards. No, the mISDN line one and two are fine, but when I get a call on line 3 something with the time is wrong. Timeconditions fall through to off-hours even if the time of the call is clearly inside business hours, here a log excerpt: [Aug 26 11:04:36] VERBOSE[3112]

Is it possible to restrict the number of concurrent calls made to a SIP peer? Or maybe the number of concurrent calls made to a particular extension. This way I can avoid asterisk trying to make more voice calls to my remote SIP gateway then I have bandwidth to handle. /davidh

Hello every one, I have got a H323 gatekeeper for testing. The informations are something like this: account code: test01 gk ip address:192.168.10.12 I don't know how to set it in the h323.conf or oh323.conf, I have tried it for almost one day but I always got the error. Help me please. Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I am still confused about the way to use asterisk with QuickNet Phonejack. If I am not wrong, The phonejack card should be using the phone.conf as the asterisk channel. I was initially confused with the ZAP channel (The digium card), now that I have found out that Phonejack should use the Linux Telephony Devices and its configuration file is phone.conf, but the question is I do not know

I have 4 TDM T1's going in to a IVR system. The IVR messages are recorded .wav format - The system appears to crap out at about 40 calls - Would using GSM or some other format help save CPU cycles? Using 1.2, Dual Xeon and 2GB ram TIA -- Barton Fisher Innovative Communications 714-228-5400 Ext 5410 http://www.icpage.com -------------- next part -------------- A non-text attachment was

Hi, > R.version.string [1] "R version 2.0.1, 2004-11-15" I have found two possible bugs in function validObject. Details below. Cheers, Scott ================================== Scott D. Chasalow Principal Statistician Statistical Genetics and Biomarkers Bristol-Myers Squibb Company Email: scott.chasalow <AT> bms.com ================================== ------ BEGIN R

Hi, I have recently discovered the project along with the PhoneJacks produced by quicknet, they could be the answer to something I have been looking into. I would like to be able to test using a dial-in server & possibily also a Windows RAS server, however I only have 1 phone line. I was thinking that I create a setup like that illustrated below to solve the problem:- Ext 1000

Dear friends we have the following problem. We have Operating system=OpenBsd installed computer and an other one with windows 98 when we change the password via smbpasswd myname ... then we have the following problem. My sambapassword is really changed. Since when I try to communicate samba server through windows 98 installed computer I see there is no problem. (Since we can see the shared

Hi! We are in process of setting up an audio guide that will cover notable places of our capital Riga, Latvia. The target audience are tourists that dials a free phone number from a mobile handset to listen to a 3 minute introduction to historic place. All audio, 10+ languages are recorder in studio at 44KHz. The audio is stored on server in A-law 8KHz because we'll be pushing it through E1

My setup: X100P and Quicknet PhoneJack. I can't seem to properly set up a Zap channel for my X100P. Here are some of my configurations: [zaptel.conf] fxsks=1 #X100P fxoks=2 #Quicknet PhoneJack defaultzone=us loadzone=us [zapata.conf] [channels] context=local signalling=fxs_ks channel->1 ;X100P [extensions.conf] ... [local] exten=>_NXXNXXXXXX,1,Dial,Zap/1 ;I'm pretty sure the

Hi! All I would appreciate if someone could advice me on how stable is sip-h323 & h323-sip translation as well as how many calls can it handle when doing such translation.( assuming single 2.8Ghz intel processor & 1GB RAM) Regards, John -- ___________________________________________________________ Sign-up for Ads Free at Mail.com http://promo.mail.com/adsfreejump.htm

Hello all, I'am trying to use 2 gnophones on my LAN. But I can't get any sound. Here is my configuration: 1. on PC1, I have: - Asterisk compiled from CVS (CVS-05/15/03) and it runs. - Gnophone binary version from Debian: 0.2.4+cvs.20020624-3 - a sound card working ( module es1371 for /dev/dsp) - I registered it as "alice" in extensions.conf 2. on PC2, I

Dear Rusers, I have some data in .csv file like "2004-8-1" and "2004-10-1", and i need to convert them into days from the origin (January 1, 1960). I have tried the function date.mmddyyyy(), but cannot get it. Anybody can show me how to handle the date data? Thanks very much! My dataset like: time 2004-8-1 2004-10-1 2001-9-1 2002-9-1 -- With Kind Regards,