Re: [asterisk-users] SIP show peers: UNREACHABLE
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 wi-fi. Now that I'm directly connected to the cable modem (well, gateway router and modem combo), the connection is better and I'm able to make outgoing VoIP calls with Jitsi. Am I right in thinking that the very same connection parameters I entered in Jitsi will work fine when entered in Asterisk with syntax like: register = username:passw...@your.provider.tld and by creating the peer entry in sip.conf for the service provider. One concern is that the provider uses: 1. User ID can be any one of your 11-digit babyTEL telephone numbers. Typically your main number but can be any one of your other phone numbers. 2. For your protection the SIP Password field does not reveal your password until you explicitly click on ‘Show password’. 3. If Outbound Proxy is not supported on your system, try one of the following two options: 1. Add the line “198.38.7.34 sip.babytel.ca” to your system’s “hosts” file and configure the SIP Proxy as: “sip.babytel.ca:5065”. This uses the TCP/IP “hosts” file address mapping mechanism to redirect SIP traffic to the Outbound Proxy. 2. Configure the SIP Proxy as: “198.38.7.34:5065”. This replaces the SIP Proxy address with a resolved Outbound Proxy address. On a mac, I added that line to the hosts file -- but I'm not sure it's required. How do I specify the SIP proxy and the outbound proxy? What's the distinction between a SIP proxy and outbound proxy? In Jitsi, I configured as 123456...@sip.babytel.ca for SIP id. In Connection I used sip.babytel.ca for the registrar and the user, 1234567890, as the the authorization name. I put the proxy as nat5.babytel.ca, port 5065 and the preferred transport as UDP. I don't see all those options, particularly surrounding the proxy and outbound proxy. Again, I'm unclear on why there's a proxy specificed, and then a different outbound proxy is specified as well. How do I establish that I've entered the parameters correctly in Asterisk? Or, how do I establish that the parameters are incorrectly entered? Because Jitsi is able to call out and in, I believe that eliminates NAT, firewall or other networking issues. thanks, Thufir -- _ -- 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
Re: [asterisk-users] sip show peers
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 - 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 boxes I am monitoring are also running 1.4.43. Once in a great while 1 of my boxes reports (Unspecified). I am trying to find out why. How can I make the remote boxes have a shorter heart beat to checking more frequently with the server so as not to go (Unspecified). By the time I log in and check its already back connected again. Any other thoughts? Thanks, Jerry -- _ -- 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 -- _ -- 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
Re: [asterisk-users] sip show peers
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...@enterux.in DID: +91-22-61447605 Cell: +91-9820332422 On Tue, May 22, 2012 at 5:50 PM, Faisal Hanif fai...@vopium.com wrote: 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 - 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 boxes I am monitoring are also running 1.4.43. Once in a great while 1 of my boxes reports (Unspecified). I am trying to find out why. How can I make the remote boxes have a shorter heart beat to checking more frequently with the server so as not to go (Unspecified). By the time I log in and check its already back connected again. Any other thoughts? Thanks, Jerry -- _ -- 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 -- _ -- 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 -- _ -- 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
Re: [asterisk-users] sip show peers
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-boun...@lists.digium.com [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 peers to see the FULL Name/Username field??? I have long names and mine are being truncated. Thanks Jerry -- _ -- 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 -- _ -- 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
Re: [asterisk-users] sip show peers
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 havepattern = FALSE;
#define FORMAT2 %-25.25s %-15.15s %-3.3s %-3.3s %-3.3s %-8s %-10s %-10s\n
#define FORMAT %-25.25s %-15.15s %-3.3s %-3.3s %-3.3s %-8d %-10s %-10s\n
char name[256];
int total_peers = 0;
int peers_mon_online = 0;
int peers_mon_offline = 0;
int peers_unmon_offline = 0;
int peers_unmon_online = 0;
const char *id;
char idtext[256] = ;
int realtimepeers;
realtimepeers = ast_check_realtime(sippeers);
if (s) {/* Manager - get ActionID */
id = astman_get_header(m,ActionID);
if (!ast_strlen_zero(id))
snprintf(idtext, sizeof(idtext), ActionID: %s\r\n,
id);
}
switch (argc) {
case 5:
if (!strcasecmp(argv[3], like)) {
if (regcomp(regexbuf, argv[4], REG_EXTENDED |
REG_NOSUB))
return RESULT_SHOWUSAGE;
havepattern = TRUE;
} else
return RESULT_SHOWUSAGE;
case 3:
break;
default:
return RESULT_SHOWUSAGE;
}
if (!s) /* Normal list */
ast_cli(fd, FORMAT2, Name/username, Host, Dyn, Nat,
ACL, Port, Status, (realtimepeers ? Realtime : ));
the 25.25s definition of FORMAT and FORMAT2 means you get 25 characters to
display. You should be able to change the 25.25 to something like 50.50 (I
tried 45.45 and it worked for me).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Tuesday, November 22, 2011 9:01 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 peers to see the
FULL Name/Username field???
I have long names and mine are being truncated.
Thanks
Jerry
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Re: [asterisk-users] sip show peers returns several notices
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. I've modified my asterisk init script to modify that
value with ulimit. My system has 192 DAHDI channels and 227 SIP peers.
When a lot of channels are in use, the number of open files climbs.
-Chris
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-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 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 installed 1.6.1.12 and a 1.6.1.6 box that I
am running. Both of them use Debian Linux (lenny) on Dell PowerEdge
1950. My list of SIP peers is quite large (3000+).
I have not noticed anything wrong with the asterisk installation apart
from this notice, but it is worrying as the error seems to crop up in
other bug reports as a precursor to crashes.
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Re: [asterisk-users] sip show peers from shell or from CLI
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 192.168.100.111 D 5060 OK (8 ms) 4200/4200 192.168.100.110 D 5060 OK (8 ms) but from shell, I've got # asterisk -rx sip show peers on Name/username HostDyn Nat ACL Port Status 4201/4201 192.168.100.111 D 5060 OK (6 ms) 4200/4200 192.168.100.110 D 5060 OK (9 ms) I never noticed this on word before. Can anyone explain ? I'm using Asterisk 1.2.17-BRIstuffed-0.3.0-PRE-1y-e Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip show peers from shell or from CLI
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 192.168.100.111 D 5060 OK (6 ms) 4200/4200 192.168.100.110 D 5060 OK (9 ms) When passing this to grep, grep replies it got binary data. Strange, isn't ? 2008/8/27 Olivier [EMAIL PROTECTED] 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 ms) but from shell, I've got # asterisk -rx sip show peers on Name/username HostDyn Nat ACL Port Status 4201/4201 192.168.100.111 D 5060 OK (6 ms) 4200/4200 192.168.100.110 D 5060 OK (9 ms) I never noticed this on word before. Can anyone explain ? I'm using Asterisk 1.2.17-BRIstuffed-0.3.0-PRE-1y-e Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip show peers from shell or from CLI
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 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 ms) but from shell, I've got # asterisk -rx sip show peers on Name/username HostDyn Nat ACL Port Status 4201/4201 192.168.100.111 D 5060 OK (6 ms) 4200/4200 192.168.100.110 D 5060 OK (9 ms) I never noticed this on word before. Can anyone explain ? I'm using Asterisk 1.2.17-BRIstuffed-0.3.0-PRE-1y-e Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip show peers from shell or from CLI
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 PROTECTED] asterisk]# asterisk -rx sip show peers -- Remote UNIX connection Name/username HostDyn Nat ACL Port Status I get that on mine, every time. Guess its your machine not catching up in time to print that bit.. Might be possible to suppress the output of that somehow? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip show peers from shell or from CLI
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. !sleep is not valid when issued from shell CLI (it's ok from Asterisk CLI) 2. constructions like 'foo foo' are not accepted by asterisk Beside using sed to remove 'on^@', I can't imagine any smarter workaround ... If anyone is inspired, please do not hesitate ... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip show peers from shell or from CLI
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 '!sleep 1 sip show peers' to works
around but :
1. !sleep is not valid when issued from shell CLI (it's ok from
Asterisk CLI)
2. constructions like 'foo foo' are not accepted by asterisk
Beside using sed to remove 'on^@', I can't imagine any smarter
workaround ...
If anyone is inspired, please do not hesitate ...
asterisk -rx'sip show peers' | grep -a '('
Bit hacky but works...
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Re: [asterisk-users] sip show peers from shell or from CLI [SOLVED]
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 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. !sleep is not valid when issued from shell CLI (it's ok from
Asterisk CLI)
2. constructions like 'foo foo' are not accepted by asterisk
Beside using sed to remove 'on^@', I can't imagine any smarter
workaround ...
If anyone is inspired, please do not hesitate ...
asterisk -rx'sip show peers' | grep -a '('
Bit hacky but works...
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Re: [asterisk-users] sip show peers
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 5060 Unmonitored devcentos5x64_to_am2mm/de 192.168.1.178 5060 Unmonitored Where the Name/username is truncated. Is there a method to display this information and to NOT have that truncated? The manager interface is an excellent source, or sip show peer I don't think you use the username so in most cases it's not important. We should propably remove it from sip show peers not to confuse people. It did really confuse me when I started to use Asterisk many years ago. Regards, /Olle ___ -- 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] sip show peers
/ 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 // devcentos5x64_to_am2mm/de 192.168.1.178 5060 // Unmonitored // // Where the Name/username is truncated. // // Is there a method to display this information and to NOT have that // truncated? / The manager interface is an excellent source, or sip show peer I don't think you use the username so in most cases it's not important. We should propably remove it from sip show peers not to confuse people. It did really confuse me when I started to use Asterisk many years ago. Olle, I am using the first field to lookup information about my client asterisk connections. Problem is the FULL name is not given. I am using it to look up IP address of the client. So I need the FULL name in the first column and the IP address of course. Is there another way to get it? Is there a way not to truncate the name/username? Thanks, Jerry ___ -- 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] sip show peers
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 // Unmonitored // devcentos5x64_to_am2mm/de 192.168.1.178 5060 // Unmonitored // // Where the Name/username is truncated. // // Is there a method to display this information and to NOT have that // truncated? / The manager interface is an excellent source, or sip show peer I don't think you use the username so in most cases it's not important. We should propably remove it from sip show peers not to confuse people. It did really confuse me when I started to use Asterisk many years ago. Olle, I am using the first field to lookup information about my client asterisk connections. Problem is the FULL name is not given. I am using it to look up IP address of the client. So I need the FULL name in the first column and the IP address of course. Is there another way to get it? Is there a way not to truncate the name/username? 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()). Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ -- 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] 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 isn't connected to a tty (isatty()). The CLI is intended to be used by a human. If you want machine parseable output, I would suggest using AMI, as that's what it's meant for. -- Tilghman ___ -- 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] 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] On Behalf Of Tilghman Lesher Sent: Friday, May 02, 2008 3:31 PM To: Asterisk Users Mailing List - Non-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 isn't connected to a tty (isatty()). The CLI is intended to be used by a human. If you want machine parseable output, I would suggest using AMI, as that's what it's meant for. -- Tilghman ___ -- 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] sip show peers
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 ___ -- 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] sip show peers
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 a tty (isatty()). The CLI is intended to be used by a human. If you want machine parseable output, I would suggest using AMI, as that's what it's meant for. To clarify: We made a decision about this earlier this year and deprecated all the CLI commands made for parsing (concise), to be able to focus on the CLI for humans and AMI for applications. Now, when you can reach AMI both over TCP and over HTTP/TCP and HTTP/TLS we have many alternatives. All CLI commands will somehow be adjusted for display on a terminal. /O ___ -- 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] sip show peers
You'll also find a ton of libraries that know how to do parsing from the AMI -- see http://www.voip-info.org/wiki/view/Asterisk+manager+API. Under see also, you'll find links to Java and PHP libraries. There's also many examples at http://www.voip-info.org/wiki/view/Asterisk+manager+Examples. Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Nunez Sent: Friday, 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 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Friday, May 02, 2008 3:31 PM To: Asterisk Users Mailing List - Non-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 isn't connected to a tty (isatty()). The CLI is intended to be used by a human. If you want machine parseable output, I would suggest using AMI, as that's what it's meant for. -- Tilghman ___ -- 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 ___ -- 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] sip show peers in 1.4.13
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 (Unspecified)D 0Unmonitored 51/51 (Unspecified)D 0Unmonitored 50/50 192.168.103.32 D 5060 Unmonitored 4190/4190 10.10.10.187 D 5060 OK (40 ms) 4178/4178 (Unspecified)D 0UNKNOWN -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] sip show peers in 1.4.13
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.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] sip show peers in 1.4.13
On Friday 02 November 2007 15:45:21 Tony Plack wrote:
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body{font-family:'Tahoma';font-size:10pt;font-color:'#00';}
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Could I get you to please turn off HTML when posting to the list? It's a
significant increase in bandwidth, and it doesn't add anything to the
discussion.
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RE: [asterisk-users] sip show peers
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 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 that the extension 108, is a server with a fixed IP 107, is a client with a softphone (X-Lite) and a dynamic IP. Why the diffrence in the port ? And why the difference in the reponse time ? We are on the same physical network, a ping is giving me a response of 1ms for each. Is it because the softphone is with a dynamic IP and Asterisk is treating this differently ? Thanks, SNIP A SIP ping and an ICMP ping are two different entities. The SIP ping operates at a higher level in the OSI stack than a simple ICMP ping. This means that whatever is receiving the ping has to do more work to decode it, and respond. I wouldn't worry about the latency difference as the SIP Ping is prioritized a bit more by a computer which is multitasking than by a hard phone which is not. As for the port, they simply chose to negotiate on a higher port. You might check your X-Lite settings, but I don't think this will break anything! Andrew ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip show peers
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 thought the response time of 101ms, was the answer. Is there anything I can do to improve that response time ? Thanks, Andrew Kirch a écrit : 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 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 that the extension 108, is a server with a fixed IP 107, is a client with a softphone (X-Lite) and a dynamic IP. Why the diffrence in the port ? And why the difference in the reponse time ? We are on the same physical network, a ping is giving me a response of 1ms for each. Is it because the softphone is with a dynamic IP and Asterisk is treating this differently ? Thanks, SNIP A SIP ping and an ICMP ping are two different entities. The SIP ping operates at a higher level in the OSI stack than a simple ICMP ping. This means that whatever is receiving the ping has to do more work to decode it, and respond. I wouldn't worry about the latency difference as the SIP Ping is prioritized a bit more by a computer which is multitasking than by a hard phone which is not. As for the port, they simply chose to negotiate on a higher port. You might check your X-Lite settings, but I don't think this will break anything! Andrew ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip show peers
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 me more the hint, that sip show peers is not telling always the truth It also did not come up at the moment I called. bye Ronald Wiplinger Regards, Mark -Original Message- From: Ronald Wiplinger [mailto:[EMAIL PROTECTED] Sent: 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 if it is reachable? bye Ronald Wiplinger ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com http://voip.elmit.com http://e-paper.elmit.com Tel. (M) +886.939.775.516 (O) +886.2.2835.7765 (ENUM) or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sip show peers
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: 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 if it is reachable? bye Ronald Wiplinger ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip show peers
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 is a language issue (ie english is not your first language). Anyway, I will try to answer. The UNKNOWN refers to the ping time to that peer. To enable that you have to have a 'qualify=yes' in your configuration. The Unspecified means that there isnt an IP address specified for that peer. Which would seem odd given that you say you can call it. I dont know enough about how you have it set up, if you have it such that you set the IP address it can try when a call comes it and succeed but it doesnt show becuase you didnt register one with the other. Does that answer your question? -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip show peers
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 from-internalNo No 210 from-internalNo Always 310 from-internalNo Always sip show peers Name/user Host Dyn Nat Status 200/200 192.168.1.150 D Unmonitored 210/210 84.36.36.140 D N OK (305 ms) 310/310 71.180.126.60 D N Unmonitored __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip show peers
--- 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 Dyn Nat Status 200/200 192.168.1.150 D Unmonitored 210/210 84.36.36.140 D N OK (305 ms) 310/310 71.180.126.60 D N Unmonitored As you can see, ext 310 has qualify=yes and and Unmonitored Status !!! __ Yahoo! Music Unlimited Access over 1 million songs. Try it free. http://music.yahoo.com/unlimited/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sip show peers
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 Dyn Nat Status 200/200 192.168.1.150 D Unmonitored 210/210 84.36.36.140 D N OK (305 ms) 310/310 71.180.126.60 D N Unmonitored Does anyone know why ext 210 the only one has a ping status OK (305 ms) and the others are Unmonitored Regards ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sip show peers
--- 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 N Unmonitored __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip show peers
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 qualify=yes This is the result of sip show peers: Name/user Host Dyn Nat Status 200/200 192.168.1.150 D Unmonitored 210/210 84.36.36.140 D N OK (305 ms) 310/310 71.180.126.60 D N Unmonitored Does anyone know why ext 210 the only one has a ping status OK (305 ms) and the others are Unmonitored Regards __ Yahoo! Music Unlimited Access over 1 million songs. Try it free. http://music.yahoo.com/unlimited/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip show peers
--- 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/r/hs ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip show peers
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 __ Start your day with Yahoo! - Make it your home page! http://www.yahoo.com/r/hs ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip show peers
--- 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 Host Dyn Nat Status 200/200 192.168.1.150 D Unmonitored 210/210 84.36.36.140 D N OK (305 ms) 310/310 71.180.126.60 D N Unmonitored __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [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 show peers MySQL Database 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sip show peers MySQL Database
You will see the peer from the database if you do 'sip show peer name. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Wednesday, October 13, 2004 6:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [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 show peers MySQL Database 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip show peers - disappearing
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 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 IP address of their firewall, etc. They can receive calls no problem. After some time goes by... they don't show as registered with * any more in the sip show peers. They can still make outbound calls, but can not receive the inbound ones. Anyone have any ideas on this one? Thanks, Jonathan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sip show peers - disappearing
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. 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 IP address of their firewall, etc. They can receive calls no problem. After some time goes by... they don't show as registered with * any more in the sip show peers. They can still make outbound calls, but can not receive the inbound ones. Anyone have any ideas on this one? Thanks, Jonathan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip show peers - disappearing
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 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 IP address of their firewall, etc. They can receive calls no problem. After some time goes by... they don't show as registered with * any more in the sip show peers. They can still make outbound calls, but can not receive the inbound ones. Anyone have any ideas on this one? Thanks, Jonathan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Go to http://www.digium.com/index.php?menu=documentation and look at the Unofficial Links section. This section has links to a wide variety of 3rd party Asterisk related pages. My page is the Asterisk Resource Pages. BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip show peers - disappearing
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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
