Displaying 20 results from an estimated 500 matches similar to: "Can not enable sip debug because CLI flooded"
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
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
2016 Sep 17
2
Ast 13.11.2 : bridgepeer variable empty ?
Hello
a call goes out and is answered :
[Sep 17 11:29:52] VERBOSE[23420][C-00000051] app_dial.c:
SIP/myprovider-0000010b is making progress passing it to
SIP/mysippeer-00000108
[Sep 17 11:30:05] VERBOSE[23420][C-00000051] app_dial.c:
SIP/myprovider-0000010b answered SIP/mysippeer-00000108
[Sep 17 11:30:05] VERBOSE[23522][C-00000051] bridge_channel.c: Channel
SIP/myprovider-0000010b joined
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
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
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
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 -
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
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
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
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
2016 Sep 19
2
Ast 13.11.2 : bridgepeer variable empty ?
Hello
I can confirm that the variable DIALEDPEERNAME contains the information
that I would expect in the variable BRIDGEPEER.
But I read nowhere that DIALEDPEERNAME has replaced BRIDGEPEER as of
Asterisk version 13 ?!
So if this is not the intention, then yes this is probably a bug and
should be reported.
Kind regards.
Jonas.
On 18-09-16 19:58, Ludovic Gasc wrote:
> Hi,
>
>
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!
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.
2009 May 11
1
Problems with res_odbc
Good morning,
I'm having suddenly cut-offs and I don`t know why. It's been hapenning since
I enabled cdr_odbc/func_odbc in my system.
We use func_odbc to register some queue member's events (login, pause, etc.)
at an external DB ('remoto' connector) and to uptade local tables at a local
DB ('local' connector).
Currently we are usind cdr_odbc to Postgresql and cdr_addon
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.