similar to: Returning different SIP Hangup Cause

While not strictly Asterisk related this issue could certainly affect some of you: http://blog.krisk.org/2013/02/packets-of-death.html -- Kristian Kielhofner

Hi, I would like to use Asterisk as VoiceMail system ... the only issue I have is with DTMF recognition. Which mode should I force into sip.conf ( general, only for peer ? ) so that the Voicemail application is understanding password from users ... inband : works, but has some glitch ... not always good ... don't know why. rfc2833 : doesn't seem to work .. info : said to be not working

Hello everyone, I've had the time to examine some specific serial/parallel forking scenarios with Asterisk lately. Looking at chan_sip it appears that anytime Asterisk wants to tear down a call before it's brought up, it sends a 603 Declined: } else { /* Incoming call, not up */ const char *res;

Does anyone have a good list of links to the various presentations at Astricon, specifically one including a link to the performance analysis by those guys from Belgium? I would love to get a closer look at their graphs because it was impossible to read them, and I was pretty close to the front! -- Kristian Kielhofner

Dear, is any way to change , the size of voice packets? I want to increase the quality by decreasing the size of each packets, because of bandwidth failure. ? thanks in advance Mani -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081110/c1b2ed9d/attachment.htm

Hey, I am here at Astricon and about to go down to registration. Is there any interest in pictures if I take my digital camera? I am sure that someone is already doing this. (Probably someone official). I would take pictures of each day and upload them to my website if anyone is interested. Let me know! -- Kristian Kielhofner

Hello everyone, I just ordered one of these: http://www.marvell.com/products/embedded_processors/developer/kirkwood/sheevaplug.jsp Just over $110 with shipping but they are expecting the price to come down quite a bit: - 1.2Ghz ARM5 - 512MB RAM - Multiple flash storage options - Gigabit ethernet - USB 2.0 - 5 watt power usage They probably won't be shipping until late March but I

Hello everyone, As I promised at eTel last week, I have finished up work on my "Asterisk Native Sounds" project. Here's a little diddy from astlinux.org: ----------------------------------- Asterisk Native Sounds are a collection of audio prompts for Asterisk. They will improve quality, reduce CPU usage, reduce latency, and (in some cases) eliminate the need for G729

Hello everyone, I was just looking over some logs, and it appears that there have been less than 3,000 downloads for my native Asterisk sounds packages (all formats combined). What gives ;)? In my humble opinion, EVERYONE (unless you have your own in a different voice/language) that uses Asterisk should be using these prompts. How about a direct link this time:

Hi, I am new to asterisk. I have downloaded Asterisk and Zaptel from the cvs root. I am installing them on Mepis with linux-2.6.7 Whenever I try to do "make" in the zaptel directory, I get the following errors. make -C /lib/modules/`uname -r` /build SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/linux-2.6.7` make[1]: *** No rule to make target

I have NFAS setup on several quad port T1 cards (Sangoma). It mostly works well with the exception that calls coming in on channels 48,72, and 96 have no audio. I tried removing these channels from zapata.conf with hopes that the channels would not come up or be used. Now I get "Ring requested on unconfigured channel". How can I busyout these these channels so that incoming

Hello everyone, Even though a lot of it was a bit last minute, several of us from the commnunity made it to Baltimore to help Digium with their booth at ISPCon. It was a great time. Gregory Boehnlein, Brian Capouch, Christian Savinovich, Kristian Kielhofner (me), and John Todd (not pictured) were there (as well as others), and some pictures were taken (the up close ones of me were very

Does anybody know why Asterisk does not support inband DTMF for G.729? Our SIP carrier use inband dtmf for G.729. This causes problem for us to use it for our Asterisk IVR system. Any suggestion to solve this problem? Gary -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070622/43308a1f/attachment.htm

Happy New Year in advance by a few ticks for the northern hemisphere. Here's the first topic and guest for 2009: In any voice path there are several potential sources of quality problems, ranging from echo to voice dropouts and everything in between. With VoIP systems the potential for quality problems increases dramatically, often times making it very difficult to identify the source of

Hello, I am just getting started with res_config and ODBC. I have MySQL all setup and am filling it with my data. Everything seems very straight forward. One thing catches me so far: 1) How are register lines in sip.conf and iax.conf represented? i.e. register=> username:password@fwd.pulver.com/700 insert into ast_config (filename,category,var_name,var_val)

I am trying to create a %100 g729 (with no transcoding) system (using a Soekris, of course). I am running AstLinux with the native sounds, g729 is the only codec allowed, %100 SIP (g729 only allow=) - I think I am covering all of my bases. I have only "format=g729" in voicemail.conf. On an incoming call to a mailbox, everything goes well until recording the message. When the

Hi Kristian, Anywhere I can read about this Soekris/AstLinux project? ... Regards, Hans -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Kristian Kielhofner Sent: Thursday, February 24, 2005 6:02 AM To: jim@vanmeggelen.ca; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users]

Abel, In have the same issue when I have burned the image to an 800MB CF Disk. All it displays is GRUB CLI in a continuous stream. Seshu -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of abel Sent: Monday, June 06, 2005 2:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject:

Hello all, This might be one for Digium, but I would like to see some type of Wiki that people would have to wade through before they would get the information on how to subscribe to the list. This wiki should cover most of the basic stuff that gets asked over and over again just to help reduce the amount of repetition that most of you have probably noticed takes place here. I understand

I've paid them, tried to provision numbers, e-mailed support, instant messaged support, and got nowhere. I highly recommend everyone stays away from this provider.