Displaying 20 results from an estimated 500 matches similar to: "peer timeouts and 489s"
2008 Sep 22
1
I can't call my remote users?
Good day to all--
First off let me say that I have been very pleased with the mailing
list. I have learned a ton of stuff just reading other peoples
questions and comments. I really enjoyed the VOIP Conference call on
Friday morning. Still working on figuring out the best approach to
custom voicemail emails (the reason I joined this group); however, we
have more pressing issues. I
2010 Jun 21
1
ISP down internal phones become unavailable
I saw the following lines in the log this morning. From my router logs
I see that the connection went down as my ISP was doing maintenance
for a few minutes last night. I can understand the external
registrations timing out, but why do the phones become unreachable.
They are on the internal lan within the same subnet as the Asterisk
server. Internal DHCP and DNS was functional. If I had a PRI card
2011 Jun 07
1
tls/srtp: sip_xmit error: returned -2
I'm having trouble setting up tls/srtp secure communications on my
Asterisk server- I'm still rather new at working with Asterisk.
I have enabled tls and encryption and I have csipsimple with tls build
on the phone. I'm currently only testing one phone with this capability
so far, and the rest still work in the current state.
My logging looks like this with verbose turned up:
2010 Apr 17
1
Realtime changes not reflected realtime
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
</head>
<body bgcolor="#ffffff" text="#000000">
<font size="-1"><font face="Helvetica, Arial, sans-serif">Hello list,<br>
<br>
Using Asterisk 1.4.25.1<br>
Using realtime sip_buddies<br>
<br>
I notice
2009 Apr 07
1
i have a probleme and my asterisk and ovh
hello every body
my connexion on ovh to pass in UNREACHABLE and not reidentified were not
reboot the server.
[Apr 7 20:17:21] NOTICE[19947]: chan_sip.c:15605
handle_response_peerpoke: Peer 'ovh' is now Lagged. (2067ms / 2000ms)
[Apr 7 20:17:35] NOTICE[19947]: chan_sip.c:19829 sip_poke_noanswer:
Peer 'ovh' is now UNREACHABLE! Last qualify: 2067
but my probleme is the adress
2009 Apr 26
1
Error, Clue to what?
[Apr 26 10:47:01] NOTICE[32151]: chan_sip.c:16223 sip_poke_noanswer: Peer
'3516533812' is now UNREACHABLE! Last qualify: 86
[Apr 26 10:47:11] NOTICE[32151]: chan_sip.c:12723 handle_response_peerpoke:
Peer '3516533812' is now Reachable. (98ms / 2000ms)
[Apr 26 12:08:49] WARNING[32273]: app_dial.c:1242 dial_exec_full: Unable to
create channel of type 'SIP' (cause 20 -
2011 Dec 13
0
[hint state][BLF] Asterisk 1.8.7 does not send RINGING notifications, even with notifyringing=yes
Hi,
I set verbose to 3, but I do not see any RINGING notification in the
CLI. On the contrary, when the phone goes UNREACHABLE I get:
[Dec 13 21:10:06] NOTICE[9988]: chan_sip.c:25533 sip_poke_noanswer: Peer
'152' is now UNREACHABLE! Last qualify: 130
== Extension Changed 152[blf] new state Unavailable for Notify User 154
[Dec 13 21:11:08] NOTICE[9988]: chan_sip.c:20196
2010 Sep 04
0
Global Outage?
Is anyone else using Vitelity right now and having an issue with a global
outage of sorts? Potral/WWW arent accessible and it would appear through
monitoring that the outbound is flapipng like mad. The outbound can be
rerouted, I know, but inbound is a huge problem right now.
[Sep 4 10:26:13] NOTICE[27507]: chan_sip.c:15679 sip_poke_noanswer: Peer
'vitel-outbound' is now UNREACHABLE!
2010 Mar 12
4
Can not enable sip debug because CLI flooded
Hello list,
I have nat=no and qualify=no in my sip peer definition and still my CLI
is flooded with :
[Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985
handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (30ms /
2000ms)
[Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985
handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (24ms /
2000ms)
[Mar 12 10:17:26]
2007 Aug 09
2
Asterisk Help
Asterisk Users,
I am running Asterisk 1.2.13 on Debian Etch with McLeodUSA's T1 service.
I have two Netgear switches on my T1 router, one for VOIP and another for
data.
I use a gigabit switch for all VOIP and a regular 10/100Mbps switch for
all data. This morning I saw this message a few times on the Asterisk
command line. The lagged cause garbled phone calls.
Is my network to
2005 Aug 22
1
Qualify time +2000ms?
Although I'm convinced that Broadvoice doesn't have the most stable of
ping times, it seems like I get ping results that are approximately the
ping time +2000ms at times. Has anyone experienced this problem with
qualify on a SIP connection before?
So here, was the ping 20ms or 2020ms as reported?
Aug 22 06:39:49 NOTICE[6964]: chan_sip.c:8481 handle_response_peerpoke:
Peer
2006 Dec 12
0
Disregistering Constantly - message: chan_sip.c:11564 sip_poke_noanswer: Peer 'provider-13052181000' is now UNREACHABLE! Last qualify: 0
Hi guys,
I configure one Fedora Core Linux 5 for use with asterisk as gateway
using Digium TE110P interconected in Alcantel 4100
I've set up it to register 100 voip numbers on my provider.
All calls on Alcatel is send to asterisk.
In some periods of day i receive this messages on asterisk console:
Dec 12 17:49:30 NOTICE[11565]: chan_sip.c:11564 sip_poke_noanswer:
Peer
2009 Jul 20
0
No subject
And after reload ALL your phones are unreachable for 2 minutes!
Imagine you have several thousands devices unreachable for 2 minutes.
How much calls will fail during that time?
Regards,
Mindaugas Kezys
Kolmisoft UAB=20
VoIP Billing Solutions
e-mail: info at kolmisoft.com
URL: http://www.kolmisoft.com
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com =
2008 May 19
1
DHCP Failure screws up system
Maybe someone could point in the right direction.
I have a small facility that's running around 40 Polycom 301/501 phones,
Asterisk 1.4.18 running under Mandriva 2007.1.
The phones were assigned a DHCP address in the 10.10.10.x range. Today,
the DHCP server failed and to get them back online, I loaded the
dhcp-server onto another system (Also running Mandriva) and copied the
dhcpd.conf
2015 Mar 29
0
Help! How to make Asterisk support ICE in public network
Hi friends,
I am just starting use asterisk for our VoIP server. It works fine in LAN. But when it is deployed in public network(with a public IP), the SIP clients in different NAT fails to communicate with each other. I have set 'icesupport' to 'yes' in sip.conf and set STURN and TURN server in rtp.conf. It still fails!
Hope someone to help me out! Thanks in advance:)
This
2004 Jun 22
5
CISCO 7960 Goes missing
I've got a number (10) Cisco 7960's connected to my network. All the phones
work perfectly except one.
The asterisk console keeps throwing up:
Jun 22 15:39:15 NOTICE[-1147470928]: chan_sip.c:5887 sip_poke_noanswer: Peer
'4001' is now UNREACHABLE!
Jun 22 15:39:27 NOTICE[-1147470928]: chan_sip.c:4925 handle_response: Peer
'4001' is now REACHABLE!
Jun 22 15:42:08
2019 Nov 16
2
problem with logger
Hello,
I am logging directly into file and also to syslog.
Here is snippet from my /etc/asterisk/logger.conf:
messages => notice,warning,error,verbose
syslog.local0 => notice,warning,error,verbose
But the logs look different:
VERBOSE[7609][C-00000013] pbx.c:
NOTICE[3042] chan_sip.c: Peer '1111' is now UNREACHABLE!
vs.
VERBOSE[7609][C-00000013]: pbx.c:2925 in
2013 Mar 15
0
No subject
[Mar 21 23:24:18] NOTICE[9931]: chan_sip.c:26242 sip_poke_noanswer: Peer
'1000' is now UNREACHABLE! Last qualify: 110
[Mar 21 23:24:18] NOTICE[9931]: chan_sip.c:26242 sip_poke_noanswer: Peer
'patton' is now UNREACHABLE! Last qualify: 20
I also get errors for connections to SIP servers for which I have
"register" entries in the [general] section of sip.conf. The
2020 Jun 02
0
problem with logger: syslog vs. file
> On 2019-11-16 03:29, Fourhundred Thecat wrote:
> Hello,
>
> I am logging directly into file and also to syslog.
> Here is snippet from my /etc/asterisk/logger.conf:
>
> messages => notice,warning,error,verbose
> syslog.local0 => notice,warning,error,verbose
>
> But the logs look different:
>
> VERBOSE[7609][C-00000013] pbx.c:
>
2005 Sep 28
3
cisco phones problems
hi folks.
we recently deployed 10 Cisco 7960G w/ SIP firmware 7.3 on our network and
we start having problems of dropping calls (actually the calls wasn't dropped
it just the sound was muted for about 5-10 seconds, but most users will think
the call dropped and hangup/redial). i've check the console output.
there was a lot of messages like the following:
Sep 28 15:00:49 NOTICE[8182]: