Displaying 20 results from an estimated 9000 matches similar to: "SIP Responsecodes"
2003 Nov 12
7
SoftFax question
Hi,
I am looking at using the softfax that Steve Underwood has developed.
It's very straight forward when you assign an extension for the fax.
A function that several pbx's has is that they listen for the 'faxtone'
for 5 seconds
after 'answer' in the menu where you can enter your local extension number,
it's normally done in parallel with the DTMF detection. I think
2014 Nov 22
2
High resident memory with 11.14.0 ?
>
> Its up to 5.8G of resident memory with 28321 calls processed.
> The OOM killer is going to kill this soon at this rate (8GB RAM machine).
> This seems like a pretty serious problem.
> It looks like I'll need to restart asterisk every night....
Hi the number of cpu cores that you see with top times 512Mbyte is the
level of ram that's needed
e.g. a hp-gen8 with 2 octo
2003 Nov 13
1
RE: Aculab SS7/ISUP (new subject)
>Freddi Hansen wrote:
>> with boards from Aculab, we are replacing Aculab boards with Digium
>> boards BUT we would need more
>> Digium boards IF we could use both Digium and Aculab cards in the same
>> server. The reason being that
>> TE410P doesn't support SS7-ISUP so we continue using only Aculab cards
>> in the servers that must support
>>
2005 Jul 13
2
OT: proliant fedora asterisk
HP doesn't support Fedora on Proliant hw so you can't just install their
ILO and get access
to hw info like cpu/mb/temperature,powersupply status,fan info aso.
I used the link below to get that access, which enabled me to write a
small script that sends
snmp-traps to hp-ovo.
I did spend quite some time myself until I found this link.
2006 Feb 14
5
Multiple AGI Issues
I've got several issues with AGI/FastAGI
1. When an AGI script sends a command to Asterisk via stdin, why does Asterisk block and not return a result until the command is complete? Specifically, the dial command. If I send a Dial command to Asterisk, I don't get a return result until AFTER the call is HUNG UP. Not when it's ringing, not when the call is connected, but when it's
2008 Oct 15
2
Zaptel compile error after make update.
Hi,
I started to get some Zaptel compile errors after a 'make update'
I did a clean zaptel install with:
svn co http://svn.digium.com/svn/zaptel/branches/1.4 zaptel
I am still getting the error, is this someelse seeing this ?.
CC [M] /usr/src/zaptel/kernel/zaptel-base.o
/usr/src/zaptel/kernel/zaptel-base.c: In function 'zt_reallocbufs':
2006 May 23
3
AGI ?
Hi All,
I have been attempting to get an AGI LCRdialout script to work.
Basically what I need to have happen is when someone dials out a number
the script check to see if it is local if so, go out the ZAP channel. If
the ZAP channel is busy, go out the IAX channels, if IAX is all busy, go
out the SIP channels. Here is a sample of what I have in my script.
#!/usr/bin/perl
use strict;
use
2014 Nov 24
2
High resident memory with 11.14.0 ?
Also, how big does the cache in frame.c grow to?
I've recompiled with MALLOC_DEBUG on that server:
asterisk -rx "memory show summary"
....
1780466242 bytes (1780181594 cache) in 2352909 allocations in file
frame.c
...
Seems like a ridiculous cache.
On Mon, Nov 24, 2014 at 9:02 AM, James Lamanna <jlamanna at gmail.com> wrote:
> cat /proc/cpuinfo lists 4 cores.
>
2005 May 19
1
Re: Grandstream ATA 286 and ilbc (Anton Krall)
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta http-equiv="Content-Type" content="text/html;charset=ISO-8859-1">
<title></title>
</head>
<body text="#000000" bgcolor="#ffffff">
<blockquote type="cite"><br>
<pre
2006 Apr 19
1
Sending SIP NOTIFY / How to get remote SIP port?
try,
database get SIP/Registry/<peername>
it gives you a string which contains the info, then pass it to CUT to
extract ip-adr and port
Freddi
> To do that you need to get the remote ip address and port of the sip peer!
>
> I found the function:
>
> ${SIPPEER(exten:ip)
>
> But how can I get the port???
>
>
2004 Jan 08
4
2nd call leg status?
Hi,
okay heres what I want to do .. simple ivr, we take a call, answer it, play a
menu, dial out based on options. No problems so far.
The CDR always shows the call as answered as I answer the 1st leg to play the
prompts, I am actually more interested in if the 2nd leg - the outbound part -
has been answered or not before the call is hungup. How can I get this and
record the information in
2014 Aug 13
2
Better info on call failure
Hey everyone,
Currently, I've got a PBX that is emailing me on call failures to an
international SIP provider of ours.
I'm doing this with exten => 1,1,System(mail -s "Call from
${CALLERID(num)} to ${DNID} Failed with DialStatus ${DIALSTATUS}"
nick at flhsi.com < /dev/null)
This works fine, However it's a little lacking. For Instance,
Our INTL SIP
2006 Feb 16
3
AGI Flakyness *sigh*
Well, I'm about ready to throw Asterisk across the room.
Can someone tell me WHY, when you've sent a Dial command to Asterisk via AGI, if the callee hangs up the call, Asterisk sends a return code, but if the caller hangs up, it does not???
This means if an agi script services a call, and after the two parties have finished speaking, the person who initiated the call hangs up, the agi
2009 Jan 14
8
evaluate SIP response codes in dialplan
Hi!
Is it somehow possible to evaluate the SIP response code inside the
dialplan?
I have an Asterisk server which forwards requests to various PSTN
gateways with SIP. If the Dial() attempt is not successful I want to
differ at least these 3 options:
- called destination is busy (486): e.g. activate auto-redial
- called destination does not exist, unassigned number (404)
- gateway is broken,
2009 May 12
1
enum agi interesting problem
Hi,
I am having a strange problem with enum and AGI.
Here is what happens:
I have in my agi something like that:
foreach my $resolver ("e164.arpa", "e164.info", "e164.org") {
my @enums = get_enums($phone, $resolver);
foreach my $enum (@enums) {
$dialstring = $enum .
2009 Sep 23
3
Simple dialplan issue
I have an issue where a particular dialplan works but another doesn't. I'm
not sure why. To me they look identical and it has me stumped.
This works:
[to-test]
exten => _X., 1, SetCallerPres(allowed)
exten => _X., 2, Monitor(wav,/tmp/test-${UNIQUEID},mb)
exten => _X., 3, Ringing
exten => _X., 4, Dial(SIP/9330 at a-test,20,ro)
exten => _X., 5,
2018 Nov 27
2
PJSIP add header on forwarded call
Le 27/11/2018 à 12:13, Joshua C. Colp a écrit :
> On Tue, Nov 27, 2018, at 5:49 AM, Administrator TOOTAI wrote:[...]
>>
>> [TOOTAiAudio]
>> ;
>> ; Call our gateway
>>
>> exten = s,1,Set(PJSIP_HEADER(add,X-TOOTAiAudio-CALLED)=${ARG1})
>> same = n,Dial(PJSIP/${ARG1}@TOOTAiAudio,,T)
>> same = n,Return
>>
>> exten = h,1,NoOp()
2010 Nov 30
2
Correct operation of timout parameter for dial application
Hi All,
I'd just like to verify what the correct operation of the timeout parameter is for the dial application. I'm not sure if I've encountered a bug or a configuration issue.
When a sip phone is not responding to invites on an outbound call, the dial application still waits the duration of timeout before continuing with dialplan execution. I was under the impression that app_dial
2018 Mar 14
2
DIALSTATUS vs HANGUPCAUSE
Thanks Dovid!
Indeed looks a bug but regardless of this, this problem made me think that
the HANGUPCAUSE could be used for this purpose with benefits.
I couldn't find an explanation about when DIALSTATUS would actually be
better.
The HANGUPCAUSE was reworked in version 11 (
https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause) but I didn't find
someone actually stating it is a better
2018 Mar 14
2
DIALSTATUS vs HANGUPCAUSE
Hello list,
Hope all doing well!
I've been checking some cases when a Dial fails and dialplan execution
continues to handle this. I am finding it a little confusing how we should
handle the DIALSTATUS and the HANGUPCAUSE in this situation....
More specifically, I am facing a case in version 13.6.0 where I am getting
a DIALSTATUS=BUSY and HANGUPCAUSE=19 after receiving a 480 SIP error. Seems