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
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: mi...@ent
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 -
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 have
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: asterisk-users-b
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.
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
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
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 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.
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:
>
> >
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 1
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
May 02, 2008 3:51 PM
> To: 'Asterisk Users 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
&g
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 connec
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
-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
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
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 192.168.1.1
>/ 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
/
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
On Friday 02 November 2007 15:45:21 Tony Plack wrote:
>
>
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 http://www.ap
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
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 though
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 a
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
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
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:
--- 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
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
--- 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
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
--- 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/u
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:
> 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!
http://www.yahoo.com/
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 qua
erisk-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,
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
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 wrote
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
>
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.
Alfred
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
> u
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