Displaying 20 results from an estimated 200 matches similar to: "Global Outage?"
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
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
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
2007 Jun 19
0
peer timeouts and 489s
Hi All,
I'm wondering if anyone can share any info on why I frequently get peer
timeouts like below, and receive 489 messages from another A*k server on
the same LAN.
For the peers, we've one L2 switch. ICMP is <1ms. The CPU of the main
A*k server is usually < 2%. So I can't see why we'd get such large
delays. The phones are all Cisco 7940s (SIP 2xx)
The 489 originate
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
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:
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
2011 May 17
1
Name or service not known
Hi, my log is full of errors from this mobile user:
-- Registered SIP '0010106' at 212.93.97.135:7759
[2011-05-17 17:44:05] NOTICE[21456]: chan_sip.c:19804
handle_response_peerpoke: Peer '0010106' is now Reachable. (381ms /
10000ms)
[2011-05-17 17:44:06] ERROR[21456]: netsock2.c:245
ast_sockaddr_resolve: getaddrinfo("212.93.97.135:7759", "7759", ...):
Name
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
2011 May 12
0
log full of Name or service not known
Hi!
Here's a user with mobile phone - however why does it treat this as ERROR ?
I have a log full of that ---
-- Registered SIP '0010106' at 212.93.100.181:3698
[2011-05-12 16:07:57] NOTICE[30258]: chan_sip.c:19679
handle_response_peerpoke: Peer '0010106' is now Reachable. (212ms /
10000ms)
[2011-05-12 16:07:57] ERROR[30258]: netsock2.c:245
ast_sockaddr_resolve:
2007 Jan 15
0
Asterisk Realtime and MD5 authentication
Hi,
I've troubles with setting up Asterisk Realtime and MD5 authentication.
With clear text passwords everything is working fine.
-- Registered SIP 'edwin' at 10.0.0.37 port 5060 expires 600
-- Saved useragent "Cisco-CP7940G/8.0" for peer edwin
[2007-01-15 10:18:12] DEBUG[28528]: res_config_mysql.c:651 mysql_reconnect:
MySQL RealTime: Everything is fine.
2008 Feb 27
1
SPA3102 registration problem
Hi list,
After failing to get a Sipura/Linksys SPA3000, which I've configured
as a PSTN gateway, to pass on the Caller ID, I decided to try my luck
with a Linksys SPA3102 after hearing some promising stories.
Unfortunately, I've run into a completely new problem: it seems
Asterisk won't let this device register.
I went about configuring the SPA3102 in much the same way as I
2009 Sep 12
1
E65 fails registration, soft phone works
Hey folks,
I am trying to get an E65 to connect to asterisk, and I would really
appreciate a second set of eyes. The SIP dialog completes fine, but
the phone subsequently says "Registration failed".
I am in a network that has what seems to be a SIP-capable NAT
gateway, but the asterisk is configured nat=yes anyway. Using
a softphone (twinkle), I can connect just fine, SIP and RTP work.
2008 Oct 15
1
Cisco 7960 not always receiving incoming calls
I've searched around and found a few similar situations where the
phone will call out when using a Asterisk server but not receive
inbound calls. My issue is a little stranger. If I call out from the
phone then the phone will receive the next inbound call. The phone
will not receive another inbound call until a call out again from it
first. Any ideas?
I am using SIP and am using the latest
2008 Oct 09
2
Menu for call forwarding or voicemail
I would like to create a simple menu that would allow a caller to
decide whether they want to leave a message or be forwarded to another
number (i.e cell phone). Thanks in advance for any insight.
Here's my current extension.conf
[general]
static=yes
writeprotect=yes
[globals]
[default]
exten => 101,1,Dial(SIP/101,20)
exten => 101,n,Voicemail(101 at default)
;This automatically
2008 Aug 11
1
Asterisk Realtime Unregister
Hi,
I'm running asterisk realtime, i had prob when a user does not
unregister properly.
I tested with SPA942 and a PAP2, when phone is registered, i call using
the SPA using x-lite no problem, but when i unplugged the power, it does
not unregister properly, so asterisk think SPA942 is still registered,
when i call using x-lite, asterisk tries to call it.so it gets stuck at
[Aug 11