Page 176 of Asterisk, the definitive manual, discusses Connecting an
Asterisk system to a SIP provider in the context of, at least the
concept of, trunking.
Previously, I wasn't able to connect to the peer, and attributed that to
a combination of double NAT (plus), and latency and lag due to
If I understand correct you need to increase qualify value.
Regards,
Faisal Hanif
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Tuesday, May 22, 2012 5:02 PM
To: Asterisk Users Mailing List
yeah, put qualify=2000 to ensure that you shall get the latency perfectly.
Regards,
Mitul Limbani,
Chief Architech Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email:
I believe it is set by a character length for formatting the output.
What are you trying to accomplish? Are you just viewing it in the CLI or are
you writing monitoring scripts?
Can you switch names so that they are unique in the beginning?
--E
-Original Message-
From:
Re-compile channels/chan_sip.c because this is what is limiting you
/*! \brief _sip_show_peers: Execute sip show peers command */
static int _sip_show_peers(int fd, int *total, struct mansession *s, const
struct message *m, int argc, const char *argv[])
{
regex_t regexbuf;
int
Perhaps you are running up against the limit of 1024 open files for a
process (I think that is the default number of allowed open files for a
process). You can execute 'ls -l /proc/{PID}/fd | wc -l' (replacing
{PID} with the process ID of asterisk) to get an estimate of how many
files it has open.
Probably another left over word from another message. Is it repeatable?
On 27 Aug 2008, at 13:00, Olivier wrote:
Hello,
On a 1.2 Asterisk / Debian Sarge, I noticed that :
ipbx*CLI sip show peers
Name/username HostDyn Nat ACL Port Status
4201/4201
A closer inspection shows ^@ between on and Name as if these letters came
from a word previously cut (from connexion ?)s o shell command would show
# asterisk -rx sip show peers
on
[EMAIL PROTECTED]/username HostDyn Nat ACL Port
Status
4201/4201
2008/8/27 Steven Howes [EMAIL PROTECTED]
Probably another left over word from another message. Is it repeatable?
At the moment, yes.
Now, I'm looking for a way to flush input/output, to protect shell script
from this type of side effect.
On 27 Aug 2008, at 13:00, Olivier wrote:
Hello,
On 27 Aug 2008, at 13:23, Olivier wrote:
2008/8/27 Steven Howes [EMAIL PROTECTED]
Probably another left over word from another message. Is it
repeatable?
At the moment, yes.
Now, I'm looking for a way to flush input/output, to protect shell
script from this type of side effect.
[EMAIL
I think we're getting closer now as obviously Asterisk's greeting (...UNIX
connection) is mixed with its output.
(I can't understand why this happens now as I never noticed this before and
didn't change anything).
I tried to use asterisk -rx '!sleep 1 sip show peers' to works around but
:
1.
On 27 Aug 2008, at 14:21, Olivier wrote:
I think we're getting closer now as obviously Asterisk's greeting
(...UNIX connection) is mixed with its output.
(I can't understand why this happens now as I never noticed this
before and didn't change anything).
I tried to use asterisk -rx
It does work, here !!
Thanks you very much !!
2008/8/27 Steven Howes [EMAIL PROTECTED]
On 27 Aug 2008, at 14:21, Olivier wrote:
I think we're getting closer now as obviously Asterisk's greeting
(...UNIX connection) is mixed with its output.
(I can't understand why this happens now as I
2 maj 2008 kl. 16.51 skrev Jerry Geis:
When doing a sip show peers I might see something like:
Name/username HostDyn Nat ACL Port
Status
devcentos5x64_to_mmfirepa 192.168.1.177 5060
Unmonitored
devcentos5x64_to_bt610tMM 192.168.1.159
/ When doing a sip show peers I might see something like:
// Name/username HostDyn Nat ACL Port
// Status
// devcentos5x64_to_mmfirepa 192.168.1.177 5060
// Unmonitored
// devcentos5x64_to_bt610tMM 192.168.1.159 5060
// Unmonitored
//
Jerry Geis schrieb:
/ When doing a sip show peers I might see something like:
// Name/username HostDyn Nat ACL Port
// Status
// devcentos5x64_to_mmfirepa 192.168.1.177 5060
// Unmonitored
// devcentos5x64_to_bt610tMM 192.168.1.159 5060
On Friday 02 May 2008 12:35:48 Philipp Kempgen wrote:
Yeah, sometimes it would be helpful if asterisk -rx 'command' didn't
pretty print but instead fall back to an easily parseable output
format (like TSV with cslashes) if stdout isn't connected to a tty
(isatty()).
The CLI is intended to be
-Commercial Discussion
Subject: Re: [asterisk-users] sip show peers
On Friday 02 May 2008 12:35:48 Philipp Kempgen wrote:
Yeah, sometimes it would be helpful if asterisk -rx 'command' didn't
pretty print but instead fall back to an easily parseable output
format (like TSV with cslashes) if stdout
On Friday 02 May 2008 14:50:38 Ed Nunez wrote:
Anyone has any good ideas on how to parse the CDR events and QUEUEs log
events from AMI connection?
There is a cdr_manager module, for generating CDRs directly to AMI. Queue
events are also sent, as a matter of course.
--
Tilghman
2 maj 2008 kl. 21.31 skrev Tilghman Lesher:
On Friday 02 May 2008 12:35:48 Philipp Kempgen wrote:
Yeah, sometimes it would be helpful if asterisk -rx 'command' didn't
pretty print but instead fall back to an easily parseable output
format (like TSV with cslashes) if stdout isn't connected to
Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] sip show peers
Anyone has any good ideas on how to parse the CDR events and
QUEUEs log
events from AMI connection?
Thank you
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED
Jerry Geis wrote:
What happened to sip show peers in 1.4.13?
Connected to Asterisk 1.4.13 currently running on indy (pid = 8236)
Verbosity is at least 5
indy*CLI
Bogus*CLI sip show peers
Name/username HostDyn Nat ACL Port Status
52/52
Gotta admit, it works for me. I am on SVN 88329 which is post 1.4.13, but still, should work.
Are you sure that chan_sip is loaded?
What happened to "sip show peers" in 1.4.13?
Jerry
___ --Bandwidth and
Colocation Provided by
On Friday 02 November 2007 15:45:21 Tony Plack wrote:
htmlheadmeta name=Generator content=PSI HTML/CSS Generator/
style type=text/css!--
body{font-family:'Tahoma';font-size:10pt;font-color:'#00';}
LI{display:list-item;margin:0.00in;}
p{display:block;margin:0.00in;}
Could I get you to
Response below
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Eric Rousse
Sent: Thursday, September 14, 2006 10:44 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] sip show peers
Hello guys,
Is there anyone who
Hi Andrew,
Thanks for the response. Interesting.
But one thing though, both extensions are softphones actually.
The one on 108, is actually VoiceGenie that I'm testing with Asterisk.
But I'm trying to explain why I'm getting some glitch with the systems
sometimes with my softphone,
and I
Mark Edwards wrote:
This indicates that 602 is a dynamic host. It must therefore register
with the pbx so that the pbx knows where to send data.
In this state it is unregistered so it will be unlikely you can call it.
That was I expected, that I cannot call it, but I could
That gives
This indicates that 602 is a dynamic host. It must therefore register
with the pbx so that the pbx knows where to send data.
In this state it is unregistered so it will be unlikely you can call it.
Regards,
Mark
-Original Message-
From: Ronald Wiplinger [mailto:[EMAIL PROTECTED]
Sent:
On Mon, 2005-10-31 at 16:33 +0800, Ronald Wiplinger wrote:
Sip show peers includes the line:
602/602(Unspecified)D N 0UNKNOWN
However, I can call it? Should not peer means if it is reachable?
I dont quite understand the question, I think there
you get ping time in the status page if your extension.conf has qualify=yes
Quoting Samy Antoun [EMAIL PROTECTED]:
--- Sergey Okhapkin [EMAIL PROTECTED] wrote:
Hmm.. What is the output of sip show users and sip show peers?
sip show users
Username Def.Context ACL NAT
200
--- Jonathan Lin [EMAIL PROTECTED] wrote:
you get ping time in the status page if your extension.conf has
qualify=yes
Setup
# Device Location options
200 Sipura local
210 Sipura remote nat=yes qualify=yes
310 eyebeam remote nat=yes qualify=yes
sip show peers
Name/user Host
They do not have NAT option.. and they do not have qualify...
Hi,
I have 3 SIP extensions, setup as follows:
# Device Location options
200 Sipura local
210 Sipura remote nat=yes qualify=yes
310 eyebeam remote nat=yes qualify=yes
This is the result of sip show peers:
Name/user Host
--- Goran Skular [EMAIL PROTECTED] wrote:
They do not have NAT option.. and they do not have qualify...
Ext 310 HAS nat=yes AND qualify=yes
# Device Location options
310 eyebeam remote nat=yes qualify=yes
sip show peers:
Name/user Host Dyn Nat Status
310/310 71.180.126.60 D
Are the devices at 200 and 310 set up to register with your asterisk?
On Sat, 2005-10-15 at 11:42 -0700, Samy Antoun wrote:
Hi,
I have 3 SIP extensions, setup as follows:
# Device Location options
200 Sipura local
210 Sipura remote nat=yes qualify=yes
310 eyebeam remote nat=yes
--- Sergey Okhapkin [EMAIL PROTECTED] wrote:
Are the devices at 200 and 310 set up to register with your asterisk?
Yes, they are registered and I can call them
__
Start your day with Yahoo! - Make it your home page!
Hmm.. What is the output of sip show users and sip show peers?
On Sat, 2005-10-15 at 12:30 -0700, Samy Antoun wrote:
--- Sergey Okhapkin [EMAIL PROTECTED] wrote:
Are the devices at 200 and 310 set up to register with your asterisk?
Yes, they are registered and I can call them
--- Sergey Okhapkin [EMAIL PROTECTED] wrote:
Hmm.. What is the output of sip show users and sip show peers?
sip show users
Username Def.Context ACL NAT
200 from-internalNo No
210 from-internalNo Always
310 from-internalNo Always
sip show peers
Name/user
That's all your gonna see..
Matthew
- Original Message -
From: Sjaak Nabuurs [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Wednesday, October 13, 2004 4:38 PM
Subject: [Asterisk-Users] sip show peers MySQL Database
Hello
How
: [Asterisk-Users] sip show peers MySQL Database
That's all your gonna see..
Matthew
- Original Message -
From: Sjaak Nabuurs [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Wednesday, October 13, 2004 4:38 PM
Subject: [Asterisk-Users] sip
The registry expires after sime time. You can set the default expirey and
max in sip.conf. It's up to your phone/sip device to reregister after the
registration expires.
Martin
On Mon, 22 Dec 2003, Jonathan Tew wrote:
We have people connecting to an asterisk box over the internet. They're
My guess would be that the NAT firewall times out and closes the port.
Reopening it from the inside is no problem, but access from the outside gets
blocked.
In order to keep the path open both ways, the client needs to send some kind
of messages with the proper IP/port in regular intervals.
Their firewall may be timeing them out. Try adding qualify=60 to each
of the entries in sip.conf
On Mon, 2003-12-22 at 10:26, Jonathan Tew wrote:
We have people connecting to an asterisk box over the internet. They're
using the x-lite client behind linksys firewalls. The X-Lite client
I think we've having some luck with this setting. Of course we had to
crank it up higher so that it didn't consider the clients LAGGED. When
the clients were LAGGED they couldn't receive any calls for some
reason. So like a setting of 200ms seems to work fine for everyone.
Eric Wieling
43 matches
Mail list logo
