When I issue a 'sip show peers' command the left most column is titled
'name/username'. Some lines show one item in the column like 123, others
show bob/123. Can someone explain the difference? (What does does name vs
username mean)
And why does 'sip show users' not show a name column title?
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
I'm trying to configure SIP trunking. Now, I'm referencing Asterisk
the definitive guide, 4th ed. While I don't have the page handy, I was
reading the suggestion to try SIP to SIP before proceeding to outside
connectivity. I'm aware that SIP trunking is a construct, but am,
obviously,
I have a process that runs on a server and does a simple 'asterisk -rx
sup show peers' /tmp/peers
and then looks for any (Unspecified) items and reports them as having
lost connection.
My server is running 1.4.43 and the two boxes I am monitoring are also
running 1.4.43.
Once in a great while
- Non-Commercial Discussion
Subject: [asterisk-users] sip show peers
I have a process that runs on a server and does a simple 'asterisk -rx sup
show peers' /tmp/peers
and then looks for any (Unspecified) items and reports them as having lost
connection.
My server is running 1.4.43 and the two
[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 - Non-Commercial Discussion
Subject: [asterisk-users] sip show peers
I have a process that runs on a server and does a simple 'asterisk -rx sup
show peers' /tmp
Is there a way with the command (1.4.42) for sip show peers to
see the FULL Name/Username field???
I have long names and mine are being truncated.
Thanks
Jerry
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[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Tuesday, November 22, 2011 10:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] sip show peers
Is there a way with the command (1.4.42) for sip show
-users] sip show peers
Is there a way with the command (1.4.42) for sip show peers to see the
FULL Name/Username field???
I have long names and mine are being truncated.
Thanks
Jerry
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Hello everybody,
When I execute the sip show peers command in the asterisk console I
always get the following notice, repeated 15 times after the sip show
peers output.
[Dec 21 03:38:31] NOTICE[12693]: utils.c:1074 ast_wait_for_output:
Timed out trying to write
This happens on a freshly
-boun...@lists.digium.com] On Behalf Of Alejandro
Recarey
Sent: Monday, December 21, 2009 4:20 AM
To: Asterisk Users Mailing List
Subject: [asterisk-users] sip show peers returns several notices
Hello everybody,
When I execute the sip show peers command in the asterisk console I
always get
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 192.168.100.111 D 5060 OK (8 ms)
4200/4200 192.168.100.110 D 5060 OK (8
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
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
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
What happened to sip show peers in 1.4.13?
Jerry
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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
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On Friday 02 November 2007 15:45:21 Tony Plack wrote:
htmlheadmeta name=Generator content=PSI HTML/CSS Generator/
style type=text/css!--
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Could I get you to
Hello guys,
Is there anyone who could explain me some stuff about sip show peers ?
108/10810.1.1.40 5060 OK (1 ms)
107/10710.1.1.246 D 51074OK (101 ms)
The port seems different here, and the main difference is
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
, September 14, 2006 10:44 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] sip show peers
Hello guys,
Is there anyone who could explain me some stuff about sip show peers ?
108/10810.1.1.40 5060 OK (1
ms)
107/107
Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] sip show peers
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?
bye
Ronald Wiplinger
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?
bye
Ronald Wiplinger
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: Monday, 31 October 2005 7:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] sip show peers
Sip show peers includes the line:
602/602(Unspecified)D N 0UNKNOWN
However, I can call it? Should not peer means
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
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 Dyn Nat Status
200/200 192.168.1.150 D Unmonitored
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
Hey Everyone,
This is not an operational issue, and to my knowledge only effects the
look of the command, but when I issue a sip reload then a sip show
peers I see all of the actual usernames I have assigned in my sip.conf.
However, five minutes later I reissue the sip show peers and all of the
Hello
How can i see the sip show peers if I use sipfriends database.
I see only the peers who are in the sip.conf.
Thanks
Sjaak
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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
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
discovers the firewall no problem and connects to Asterisk without a
problem. After connecting the agent shows up properly in sip show
peers with the
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
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