Re: [asterisk-users] Fail2ban integration issues with Asterisk 1.4.21 under Debian Lenny

On Mon, 30 Aug 2010, J. Oquendo wrote: > Gordon Henderson wrote: >> On Mon, 30 Aug 2010, J. Oquendo wrote: >> >> I also posted a very effective iptables script some weeks ago if you care >> to search the archives. It works and is extremely effective in blocking >> these types of attacks - however,

[asterisk-users] Cisco 9971

Hi, I am having a weird issue with a Cisco 9971 phone. I managed to get most of it working, including the side car, however one of the issues is that there seems to be some sort of side tone / beep occurring roughly every 13 seconds or so, as if the phone is activated with call waiting. However

Re: [asterisk-users] Fail2ban integration issues with Asterisk 1.4.21 under Debian Lenny

On 10-08-30 01:53 PM, J. Oquendo wrote: > Gordon Henderson wrote: > >> On Mon, 30 Aug 2010, J. Oquendo wrote: >> >> >> >> I also posted a very effective iptables script some weeks ago if you care >> to search the archives. It works and is extremely effective in blocking >> these types of atta

Re: [asterisk-users] Fail2ban integration issues with Asterisk 1.4.21 under Debian Lenny

Gordon Henderson wrote: > On Mon, 30 Aug 2010, J. Oquendo wrote: > > > > I also posted a very effective iptables script some weeks ago if you care > to search the archives. It works and is extremely effective in blocking > these types of attacks - however, it will not stop a broken sipvicious

[asterisk-users] Voicemail prompts fuzzy and quiet

Strange issue that I can't figure out and I am hoping someone may have some ideas. Two Asterisk boxes running 1.2.34 (yeah I know it is old, but it runs like a top and I am not going to mess with it). *B rsyncs config from *A. *A dies. I bring up *B and it all works fine, except for one issue.

Re: [asterisk-users] TELUS British Columbia PRI Settings

On 8/30/2010 2:59 PM, jeremy.hellst...@synovate.com wrote: > The specific error message is as follows. > _ > Changing signalling on channel 24 from Unused to Hardware assisted > D-channel > DAHDI_CHANCONFIG failed on channel 24: Invalid ar

Re: [asterisk-users] Fail2ban integration issues with Asterisk 1.4.21 under Debian Lenny

On Mon, 30 Aug 2010, J. Oquendo wrote: > How about a little cron script without having to install anything? You > could run it off the hour: > > rightnow=`date "+%Y-%m-%d %k"` > > grep $rightnow /var/log/asterisk/messages |\ > awk '/No matching peer/' | sed's:'\''::g' |\ > uniq | awk '{print "ipta

Re: [asterisk-users] TELUS British Columbia PRI Settings

The specific error message is as follows. _ Changing signalling on channel 24 from Unused to Hardware assisted D-channel DAHDI_CHANCONFIG failed on channel 24: Invalid argument (22) Did you forget that FXS interfaces are configured with FX

Re: [asterisk-users] help with dialplan

On Mon, Aug 30, 2010 at 2:48 PM, Todd Reese wrote: > Thanks for pointing out the misspelling.  I've corrected that and still no > luck. > Create a new debug log with your recent changes, re-attach it the list. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com |

Re: [asterisk-users] help with dialplan

Thanks for pointing out the misspelling. I've corrected that and still no luck. On 8/30/2010 2:33 PM, Alex Bell wrote: possibly check you spelling: [from-interal] -> [dialout1] include => from-internal ?? On Mon, Aug 30, 2010 at 10:14 AM, Todd Reese > wrote:

Re: [asterisk-users] TELUS British Columbia PRI Settings

On 8/30/2010 11:42 AM, jeremy.hellst...@synovate.com wrote: > I tried those as you said, deleting my failed attempt. I've found that > using hardhdlc=24 generates an error and reminds me that FXO uses FXS > signalling and vice versa when running dadhi_restart, which seems to > indicate that it is

Re: [asterisk-users] help with dialplan

I actually found that one and corrected it. I have replaced the context with the from-internal, remote, and dialout1. Each has produced the same results of a 404 error. On 8/30/2010 2:10 PM, Paul Belanger wrote: > On Mon, Aug 30, 2010 at 11:42 AM, Todd Reese wrote: >> Here's a debug fo

Re: [asterisk-users] help with dialplan

possibly check you spelling: [from-interal] -> [dialout1] include => from-internal ?? On Mon, Aug 30, 2010 at 10:14 AM, Todd Reese wrote: > Hi all, > > I've been have problems with getting this system on line and would like > to acquire some help with the extensions.conf. > > My current probl

Re: [asterisk-users] help with dialplan

On Mon, Aug 30, 2010 at 11:42 AM, Todd Reese wrote: >  Here's a debug for extension 150 > In the future, simply attach your debug log to your email. Here is your problem: [Aug 30 11:34:55] NOTICE[2079] chan_sip.c: Call from '150' to extension '6789542133' rejected because extension not found in

[asterisk-users] Wifi + SIP + Asterisk

In my old office, for conference purpose , gotomeeting was used. also for the lecture delivery, same gobomeeting was used, most the time , we need to listen voice only. also, we use to share desktop screen. But as far as I know SIP is the standard for video telephony. SIP can handle video +Audio. n

Re: [asterisk-users] help with dialplan

Unfortunately, that didn't work. The phone is still giving me a 404 error. I have my own system that is 1.6.2.7 with Grandstream phones that works fine. Using it as a guide, I built this server for a client which also has Grandstream phones. Last week, it dialed out fine. Since the weeke

Re: [asterisk-users] Fail2ban integration issues with Asterisk 1.4.21 under Debian Lenny

Gordon Henderson wrote: > > > > So.. Get a copy of the sipvicious code from http://blog.sipvicious.org/ > > (or directly from http://code.google.com/p/sipvicious/ ) and learn how to > > use svcrash.py as that's the only thing that's going to ultimately stop a > > long-term attack on your site.

Re: [asterisk-users] Maximum Wait Time queue option

Thanks Warren for your help On Mon, Aug 30, 2010 at 9:21 PM, Warren Selby wrote: > On Mon, Aug 30, 2010 at 10:31 AM, Tino wrote: > >> Hello, >> >> Is there any option to set the maximum number of seconds a caller can wait >> in a queue before being pulled out ? >> >> Thanks >> >> > > In the Que

Re: [asterisk-users] Maximum Wait Time queue option

On Mon, Aug 30, 2010 at 10:31 AM, Tino wrote: > Hello, > > Is there any option to set the maximum number of seconds a caller can wait > in a queue before being pulled out ? > > Thanks > > In the Queue() command itself there is a timeout parameter. From your asterisk box, try running: asterisk

Re: [asterisk-users] TELUS British Columbia PRI Settings

I tried those as you said, deleting my failed attempt. I've found that using hardhdlc=24 generates an error and reminds me that FXO uses FXS signalling and vice versa when running dadhi_restart, which seems to indicate that it is the wrong variable name. I also notice that if I change that variab

Re: [asterisk-users] TELUS British Columbia PRI Settings

It is a half turned up PRI, so 1-12 should be correct? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: August 28, 2010 12:49 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-use

Re: [asterisk-users] help with dialplan

Todd Your context must be set to where you want your extension to start each time it dials out. Without getting into your dialplan code too much try changing the context to point to dialout1 context=dialout1 If dialout1 is working you should be able to dial. The best way to handle this is to

Re: [asterisk-users] help with dialplan

Here's a debug for extension 150 [Aug 30 11:34:53] VERBOSE[2099] config.c: == Parsing '/etc/asterisk/logger.conf': [Aug 30 11:34:53] DEBUG[2099] config.c: Parsing /etc/asterisk/logger.conf [Aug 30 11:34:53] VERBOSE[2099] config.c: == Found [Aug 30 11:34:53] VERBOSE[2099] logger.c: Aster

Re: [asterisk-users] Play a number of files to a caller

On 08/30/2010 01:58 AM, Julian Lyndon-Smith wrote: > ControlPlayback can't be used with ExternalIVR, can it ? > > We use ControlPlayback in our current dialplan, what I am wanting (in > concept) is to have a meetme/conference room where one of the parties > is a caller, and the other party a file

[asterisk-users] Maximum Wait Time queue option

Hello, Is there any option to set the maximum number of seconds a caller can wait in a queue before being pulled out ? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us fo

Re: [asterisk-users] Play a number of files to a caller

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith Sent: Monday, August 30, 2010 2:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Play a number

Re: [asterisk-users] help with dialplan

Here is the sip.conf portion for extension 150 [150] deny=0.0.0.0/0.0.0.0 type=friend secret=1234567890 qualify=yes port=5060 pickupgroup= permit=0.0.0.0/0.0.0.0 nat=yes host=dynamic dtmfmode=rfc2833 dial=SIP/150 context=from-trunk canreinvite=no callgroup= callerid=device <150> accountcode= cal

Re: [asterisk-users] help with dialplan

Todd How do you have the context in the phones sip configs set? Bryant From: "Todd Reese" trees...@gmail.com Hi all, I've been have problems with getting this system on line and would like to acquire some help with the extensions.conf. My current problem is that the phones won't dialout.on t

Re: [asterisk-users] help with dialplan

On Mon, Aug 30, 2010 at 10:14 AM, Todd Reese wrote: > I've been have problems with getting this system on line and would like > to acquire some help with the extensions.conf. > Post a debug log of the problem: http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt -- Pa

Re: [asterisk-users] SIP Debug Messages

On Mon, Aug 30, 2010 at 3:19 AM, Positively Optimistic wrote: > Is it possible to send sip messages (debug) to a file or syslog server > without having them present in the console? > http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt Specifically, *CLI> sip set debu

[asterisk-users] help with dialplan

Hi all, I've been have problems with getting this system on line and would like to acquire some help with the extensions.conf. My current problem is that the phones won't dialout.on the VOIP lines listed as dialout1, dialout2, dialout3. This version of asterisk is 1.6.2.11. Below is the ext

Re: [asterisk-users] Fail2ban integration issues with Asterisk 1.4.21 under Debian Lenny

On Mon, 30 Aug 2010, Nikhil Nair wrote: > Hi, > > I've recently had a fairly prolonged SIP registration attack, 18 hours in > this case and often with 200 attempts per second, and suspect I've had a > number of these in the past. Almost everyone has - read the fine archives, then google for sipvi

Re: [asterisk-users] Web-meetme

- "Flavio Miranda" wrote: > > Hi all, > > I am trying to set up Web-meetme in Asterisk 1.6. After some attemps, I am > receiving the message: DB Error: connect failed > What could be ? It's very likely the connection failed to your database... Check your database name and credent

[asterisk-users] Fail2ban integration issues with Asterisk 1.4.21 under Debian Lenny

Hi, I've recently had a fairly prolonged SIP registration attack, 18 hours in this case and often with 200 attempts per second, and suspect I've had a number of these in the past. The main symptom I noticed previously was, because Asterisk was responding to each registration request it receive

Re: [asterisk-users] Could MeetMe invite someone to the conference?

Thank you ! Do you konw how to realtime billing for MeetMe conference? 2010/8/30 Paul Belanger : > On Sun, Aug 29, 2010 at 10:52 PM, Zhang Shukun wrote: >> but i want to know if i can invite some one to the conference when i >> already in the conference? >> > http://www.voip-info.org/wiki/view/A

[asterisk-users] How to Billing for MeetMe Conference?

hi,Dear all as you know, MeetMe has cdr for each attendant. but the fee is always paid by the moderator. not by each one. and the the members in the conference are dynamic changed. in this scenario, how to billing for MeetMe conference? and i want to hungup all the calls when the account f

[asterisk-users] SIP Debug Messages

Is it possible to send sip messages (debug) to a file or syslog server without having them present in the console? If so, does anyone know what kind of performance hit this would create. Instance has approx 800 sip peers. -- _

Re: [asterisk-users] Play a number of files to a caller

Hi Tilghman , thanks for the help. ControlPlayback can't be used with ExternalIVR, can it ? We use ControlPlayback in our current dialplan, what I am wanting (in concept) is to have a meetme/conference room where one of the parties is a caller, and the other party a file to be controlplaybacked b