[asterisk-users] NAT issues
Hello, this isn't an Asterisk specific problem but I don't know who else to ask for help. This is my setup, it oftens finds double NAT situations: [Asterisk box] - [Firewall IPCop] -INTERNET- [Random Router] - [Softphone] In certain situations, when two or more client softphones use the port 5060 at the same time and try to register, the UDP translation state of the port fails to assure the connection and drops both phones. If I change the client ports to random ones, they register, they can make calls and everything. It just happens if there is port clashing. I am not sure how to tackle this situation as enforcing random ports to the softphones is not viable for the setup. Is this a problem with the IPCop Firewall? I tried flushing the conntrack tables yet this situation kept happening. It gets to the point that nobody can use the 5060 port after a while (when everyone is trying to register). Thank you, Perssy Llamosas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ztdummy problems
Hello list, I have found some strange problem with the ztdummy timing, maybe you have already have this problem before, I would appreciate some hints here or maybe I need to file a bug. First of all, some background: I decided to upgrade my testing machine to the current version of Asterisk (1.4.13 to 1.4.19.1). I have tried this before and the result was the same, I am guessing that something must have changed in the zaptel code from 1.4.7 upwards since this process always fails in the same timing issue. Installing: Everything went ok until I found this issue, Playback never actually playing anything, hanging there forever. Unloading the zaptel modules makes Playback work as it should, it's not the expected setup. Loading ztdummy again and trying zttest -v it just hangs there forever without outputting anything. Trying compiling ztdummy without RTC since that worked fine in a Xen virtual machine, no success. This particular computer works fine under Asterisk 1.4.13 and zaptel 1.4.6 lspci shows that there is an usb device: 00:1f.2 USB Controller: Intel Corporation 82801BA/BAM USB Controller #1 (rev 01) lsusb shows there is one device. Bus 001 Device 001: ID : So what am I doing wrong? The computer has a Centos Linux 5 with the minimal install, not even base was selected. It's an IBM NetVista pIII 1GHz. Help is appreciated. Perssy Llamosas. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What replaces Macro() now? And how do you do the equivalent?
Hello, I believe that using GoSub combined with Set replaces macro perfectly. There is even a Return application. PLL. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk reltime mode with Postgresql
Hello, If you activate debug you will see that you get those warnings because Asterisk is trying to check users that only exist in the sip.conf file. PLL. Original Message Subject: [asterisk-users] Asterisk reltime mode with Postgresql From: Andrew Nowrot [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: 17/02/2008 07:16 a.m. Hi I am having problems with Asterisk 1.4.18 and realtime architecture. I use Postgresql-8.3 as the database. Everything works OK; all sip phones (their configs are in the database) are able to register to the server and I can make calls (dialplan is in the database), but each time Asterisk reads the information from the database it shows me this on the console: [Feb 17 12:32:50] WARNING[620]: res_config_pgsql.c:207 realtime_pgsql: Postgresql RealTime: Could not find any rows in table extensions_conf. [Feb 17 12:32:50] WARNING[604]: res_config_pgsql.c:207 realtime_pgsql: Postgresql RealTime: Could not find any rows in table sip_conf. [Feb 17 12:32:50] WARNING[604]: res_config_pgsql.c:207 realtime_pgsql: Postgresql RealTime: Could not find any rows in table sip_conf. [Feb 17 12:32:50] WARNING[9219]: res_config_pgsql.c:207 realtime_pgsql: Postgresql RealTime: Could not find any rows in table extensions_conf. [Feb 17 12:32:50] WARNING[9219]: res_config_pgsql.c:207 realtime_pgsql: Postgresql RealTime: Could not find any rows in table extensions_conf. [Feb 17 12:32:51] WARNING[9219]: res_config_pgsql.c:207 realtime_pgsql: Postgresql RealTime: Could not find any rows in table extensions_conf. [Feb 17 12:32:51] WARNING[9219]: res_config_pgsql.c:207 realtime_pgsql: Postgresql RealTime: Could not find any rows in table extensions_conf. Like I said before everything seems to work fine, but why I get this warnings. Is it a bug or I configure my server in a wrong way? Cheers Andrew ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] centos 5 vs OpenSuse 10.3
I doubt it. hxxp://boycottnovell.com/2007/10/02/opensuse-103-release/ Original Message Subject: Re:[asterisk-users] centos 5 vs OpenSuse 10.3 From: Per Jessen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: 18/10/2007 11:25 a.m. Julian Lyndon-Smith wrote: Apart from religious grounds (!), is there any pros or cons why I should choose one over the other for a new install of asterisk ? I doubt it. A distro is a distro. /Per Jessen, Zürich We use only openSUSE. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] centos 5 vs OpenSuse 10.3
I doubt it. boycottnovell.com/2007/10/02/opensuse-103-release/ Original Message Subject: Re:[asterisk-users] centos 5 vs OpenSuse 10.3 From: Per Jessen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: 18/10/2007 11:25 a.m. Julian Lyndon-Smith wrote: Apart from religious grounds (!), is there any pros or cons why I should choose one over the other for a new install of asterisk ? I doubt it. A distro is a distro. /Per Jessen, Zürich We use only openSUSE. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] centos 5 vs OpenSuse 10.3
I doubt it. boycottnovell.com/2007/10/02/opensuse-103-release Original Message Subject: Re:[asterisk-users] centos 5 vs OpenSuse 10.3 From: Per Jessen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: 18/10/2007 11:25 a.m. Julian Lyndon-Smith wrote: Apart from religious grounds (!), is there any pros or cons why I should choose one over the other for a new install of asterisk ? I doubt it. A distro is a distro. /Per Jessen, Zürich We use only openSUSE. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL2 Syntax Highlighting
Original Message
Subject: Re:[asterisk-users] AEL2 Syntax Highlighting
From: Tzafrir Cohen [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Date: 13/10/2007 05:24 a.m.
On Fri, Oct 12, 2007 at 05:24:29PM -0500, Perssy Llamosas wrote:
Hi,
I am looking for a syntax highlighter for AEL2. Google is not helping,
so I thought you guys could help me.
I found this vim syntax highlighter for AEL but it doesn't help if you
want to code in AEL2:
http://vim.sourceforge.net/scripts/script.php?script_id=1900
How is AEL2 syntax different from AEL?
Can you please give examples where the above fails for AEL2? (or for
AEL, for that matter)
Well, I am trying to improve that script slowly, I admit I knew nothing
about writing vim highlighting files before so this is a good
opportunity to learn...
Some examples where the above fails:
//-example: No ; after brackets.
globals {
}
/* Anything below fails */
context failed1 {
};
//-example: No ; after brackets.
context failed2 {
1 = {
Hangup();
}
/* Anything below fails */
2 = {
Hangup();
};
};
//-example: Inline if else while for random
context failed3 {
1 = {
if(1) NoOp(This fails);
};
2 = {
if(1) {NoOp(This also fails);}
};
};
//-example: bug
context failed4 {
1 = {
if (1) {
} else {
}
/* Anything below fails */
};
2 = {
NoOp(This fails);
};
};
//-example: bug
context failed5 {
1 = {
switch(1) {
}
/* Anything below fails */
};
2 = {
NoOp(This fails);
};
};
//-example: Hints
context failed6 {
hint(Sip/1) 2 = {
NoOp(This fails);
};
};
//-example: Next line bracket
context failed7
{
1 = {
NoOp(This fails);
};
};
//-example: Switches and eswitches
context failed8 {
switches {
IAX2/abox;
};
/* Anything below fails */
1 = {
};
};
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[asterisk-users] AEL2 Syntax Highlighting
Hi, I am looking for a syntax highlighter for AEL2. Google is not helping, so I thought you guys could help me. I found this vim syntax highlighter for AEL but it doesn't help if you want to code in AEL2: http://vim.sourceforge.net/scripts/script.php?script_id=1900 Cheers, PLL. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] . (period): Wildcard match; matches one or more characters
Hello, No, in this case wildcard means a symbol that stands for one or more unspecified characters, used especially in searching text and in selecting multiple files or directories. There is no relation with the card which is just a name. PLL. Original Message Subject: [asterisk-users] . (period): Wildcard match;matches one or more characters From: bilal ghayyad [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: 28/09/2007 07:55 a.m. Hi List; In the outbound, I read in the documents the Wildcard match by using the . (period), but I did not understand how Wildcard will work (like what)? As I know that Wildcard is a term used with the Diguim TDM card (FXO and FXS), so what is the relation between such cards and the matching in the dial plan? Any help? Regards Bilal Shape Yahoo! in your own image. Join our Network Research Panel today! http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 NAT issues
Hello, I am playing around with IAX2 and I have encountered a problem trying to setup an asterisk box through NAT using IAX2. This is the problem: Asterisk box = Advanced Firewall = Internet = User's router = User The user can register, the server can answer, calls can be made. Asterisk box = Very simple router = Internet = User's router = User User's packet reach the server, the server cannot reply because the udp connection is lost, several RX retry TX retry, user is unable to call. In both cases the firewall and the router are forwarding the port 4569 to Asterisk, user's router is not forwarding anything, the user has qualify=yes to maintain the connection open but the very simple router will drop the connection before Asterisk can reply to the packet. So I ask the list: Is there a way to overcome this problem? Udp connection timeout in Asterisk? Should I get a new router? Thanks, PLL. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Died message
You are using safe_asterisk, it will restart automatically Asterisk after it crashes. Original Message Subject: [asterisk-users] Asterisk Died message From: Nitesh Divecha [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: 04/09/2007 11:23 a.m. Hello All, Anyone knows what does this error message means and where to check for the cause and why it happened? Asterisk on hyperion exited on signal 11. Might want to take a peek. But when I check Asterisk, its running fine... Cheers, Nitesh ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hints and Noop
Hello, I want to get rid of bunch of useless notices in the logs when the hint is not found, does setting the hint to noop for everything breaks anything? exten = _X.,hint,NoOp So far it did what I wanted. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] priorityjumping not working, Dial goes to n+1 not n+101
I think n+101 worked in Asterisk 1.2.x but it doesn't work in Asterisk 1.4.x
use ${DIALSTATUS} if you want Asterisk to act depending the result of Dial()
I read that the variable has been disabled in SVN to be replaced by the
DEVSTATE function, I need to confirm that.
Well... an example:
exten = 1337,1,Dial(SIP/zytek,5,Ttj)
exten = 1337,n,Goto(${DIALSTATUS},1)
exten = CONGESTION,1,Congestion
exten = CANCEL,1,Hangup
exten = BUSY,1,Busy
exten = CHANUNAVAIL,1,NoOp(I can't find it)
exten = CHANUNAVAIL,n,Busy
Although it would look a lot nicer if you create a macro that acts upon
the result of Dial.
Perssy Llamosas
Original Message
Subject: [asterisk-users] priorityjumping not working,Dial goes to
n+1 not n+101
From: Jakub Głazik [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: 20/07/2007 04:45 a.m.
Priorityjumping is totally ignored by my asterisk (tested 1.4.4 and
1.4.7.1 on FreeBSD 6.2)
[general]
priorityjumping=yes
With n+101:
exten = 1337,1,Dial(SIP/zytek,5,Ttj)
exten = 1337,102,Dial(SIP/zytek,${RINGTIME},${OPTIONS})
exten = 1337,n,Hangup
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/113-087a3000,
SIP/zytek|5|Ttj) in new stack
-- Called zytek
-- SIP/zytek-087b9000 is ringing
-- Nobody picked up in 5000 ms
== Auto fallthrough, channel 'SIP/113-087a3000' status is 'NOANSWER'
With n+1:
exten = 1337,1,Dial(SIP/zytek,5,Ttj)
exten = 1337,2,Dial(SIP/zytek,${RINGTIME},${OPTIONS})
exten = 1337,n,Hangup
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/113-087c8000,
SIP/zytek|5|Ttj) in new stack
-- Called zytek
-- SIP/zytek-087da000 is ringing
-- Nobody picked up in 5000 ms
-- Executing [EMAIL PROTECTED]:2] Dial(SIP/113-087c8000,
SIP/zytek|720|Ttm) in new stack
-- Called zytek
-- Started music on hold, class 'default', on channel
'SIP/113-087c8000'
-- SIP/zytek-087b6000 is ringing
Why?
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