similar to: process_sdp: Multiple audio streams are not supported

hi, i.m. newbie in asterisk. asterisk 1.0.3 is my current version. i like to store usernames and passwords in a sql database. i like to log failed authentification-passwords, to create a blacklist for securityreasons. i thingk a sql-database is a good way to log these actions. i don.t find debugging-options to output invalid login-passwords. Ok, i have made the following: debian is my OS.

Hi! I have benn playing a little with quesues tonight and I found out if there are at least one member-extension free the announcement with p'the place in the queue wont be played to the person who called in. Is this possible to change so the announcement will be played even if there are free member-extensions? I think that would be nice (well it's not how ACD-groups usually works but

hi evrey one, i'm in some kind interesting in developping some asterisk programme like doing a small programme including some of these services that do a telephone operator. but abviously i need to know about programming in asterisk in thos to files i think :) (extensions.conf and in sip.conf files) so i'm asking if someone can give me a puch,i will be very glad thanks in advance

Asterisk is sending a 481 in response to an INVITE for reasons I do not understand. Here is the INVITE: INVITE sip:8009499014 at X.YYY.32.3:5060;transport=udp SIP/2.0 Record-Route: <sip:X.YYY.32.10;lr=on;ftag=247898> To: <sip:8009499014 at X.YYY.32.10 :5060>;tag=ac86f72d2bfe10395b2e62e01c70bf66.0f65 From: "Scott Thompson" <sip:7166359474 at X.YYY.32.10>;tag=247898

Is anybody using Mitel phones? It appears that when you page a Mitel phone using asterisk's MeetMe, the paged phone will hang up the call its on to take the page. Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110621/764a6fa9/attachment.htm>

Hi, I'm using asterisk 1.2.1. Is there anybody out there who knows what this warning means? *WARNING: chan_sip.c:3470 process_sdp: Unknown SDP media type in offer: image 5004 udptl t38* Google does not help at all. TIA Giorgio Incantalupo

Anyone else have issues with T.38 where the call drops after T.38 is attempted to be negotiated, with a message like the below? WARNING[31046]: chan_sip.c:4978 process_sdp: Unable to lookup host in c= line, 'IN IP4 100101'

I'm experienced in statistics, but I am a first-time R user. I would like to use R for correspondence analysis. I have installed R (Mac OSX). I have used the package installer to install the CA package. I have run the following line with no errors to read in the data for a table: NonLuxury <- read.table("/Users/myUserName/Desktop/nonLuxury.data.txt") The R online help

Hi all, I have a package that is specific to a task I was repetitively using a few years ago. I now needed to run it again with new data. However I am told it was built with an older version or R and will not work. How can I tweak the package so it will run on 11.1? It was a one-off product and has not been maintained. Is there a way to "unpackage" it and repackage it to work? I

Hallo! Wenn du das hier lesen und verstehen kannst, interessiert dich vielleicht auch dieser Link zur neuen deutschen Ruby On Rails-Usergroup: http://rubyonrails-ug.de/ + Weblog + Mailingliste Viele Grüße ~Tobias PS: Danke an die Gründer!

Hi All, I'm trying to upgrade asterisk server to 1.8.x from my asterisk 1.6, But when making A Call from SIP Client, I got cli Warning ... and no call has been made. My Sip Client is using lib java peers client http://peers.sourceforge.net/ with standard codec PCMU/PCMA [Jan 28 23:03:32] WARNING[1654]: chan_sip.c:8942 process_sdp: Unsupported SDP media type in offer: audio 0 RTP/AVP 0 8

Is there a way for asterisk's voicemail to send an email (including voicemail attachment) to multiple email addresses? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110408/8908db5f/attachment.htm>

I'm wondering if there is a simply way to perform a voicemail callback feature using AGI. For instance, a caller leaves a voicemail, the voicemail will then call the owner of the voicemailbox determined by a database look up. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Does anyone have any application examples for Confbridge in Asterisk 10? I'm just looking for simple ad-hoc functionality similar to meetme in 1.8. Thank you in advance.

A friend asked me for help to auto-dial and play a prerecorded message for a political campaign. I've briefly googled auto dialer scripts but haven't seen one that really stands out. Are there any free or cheap auto dial solutions that you nice folks recommend? Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm monitoring Asterisk with Nagios. Nagios constantly alerts because of too many zombie processes. I eventually had to disable the notification for the alert but why does Asterisk create so many zombie processes, I've see more than 30 at times and it generally stays in the 20s... just seems unusual and wondering if it's harmful, thanks in advance. -------------- next part

Hi! I heard many things over ogg, but all what i can see is shit. O.K. The oggdrop.exe play with me, have this tool a version number? I can't see it. Every other tool don't run under my system (Win98; Athlon1,5GHz; 768MB RAM). For example every version of oggenc produce the error: "OGGENC.EXE kann nicht ausgef?hrt werden"! There are a german mailing list? An how can i register

Dear list members, I'm interested in showing that within-group statistical dependence is negligible, so I can use ordinary linear models without including random effects. However, I can find no mention of testing a model with vs. without random effects in either Venable & Ripley (2002) or Pinheiro and Bates (2000). Our in-house statisticians are not familiar with this, either,

Hello there i have installed samba on my SGI IRIX box and its up and running. I have a few win xp machines. wanna make the printer availabe on my IRIX as a network printer. I tried adding the printer thru add printer wizard in windows it gets added but it shows the status as "status:Access denied, unable to connect" how should I resolve this problem ? I am using HP laserjet 6MP plz

Is anyone using asterisk with fail2ban? I have it working except it takes way more break-in attempts than what is set in "maxretry" in jail.conf For example, I get an email saying: "The IP 199.204.45.19 has just been banned by Fail2Ban after 181 attempts against ASTERISK." when "maxretry = 5" in jail.conf Perhaps someone else is experiencing this or has resolved it,