Displaying 20 results from an estimated 1100 matches similar to: "Asterisk Help"
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
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]
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 Feb 01
1
SIP Challenge response bug?
Ok, here's an odd one. I would have opened a bug on this but last time I
tried that I got flamed.. :)
Problem: When proxy requests digest challenge (SIP) Asterisk responds
normally with the exception that for some reason it changes the FROM:
(Also changes Contact: )to what's in the original TO: line. Why on earth
is it doing this?! It must be a bug, I've gone over my extensions.conf
2005 Mar 17
2
Netlogic inbound DID issue
Anyone out there using NetLogic DIDs? And have inbound working? I got
outbound working, but no joy so far with inbound. Here are the relevant
parts from my conf files:
iax.conf
[general]
tos=lowdelay
jitterbuffer=no
register => username:secret@zoot.netlogic.net
[netlogic]
type=friend
host=dynamic
context=sourcekit-main
auth=plaintext
username=
secret=
disallow=all
allow=ulaw
allow=all
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
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:
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
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
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
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
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
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 -
2005 Mar 18
0
IAX Peer/auth issues WAS: Netlogic inbound DID issue
Has something changed in the recent modifications to Asterisk that would
break dialing of the IAX peer? We're getting these authority failures
everywhere.
Everything is configured just the way it was half a year ago, this is
the message we're getting on the most recent vers of asterisk. Wiki says
nothing, nor does the ast-dev list..
-lost
Mar 18 12:55:23 NOTICE[3479]: chan_iax2.c:6545
2005 Mar 15
6
Realtime config
Having problems getting realtime working, I'm trying to use odbc for all
of this. I've got Fedora 3 and have been fighting with odbc for a day
now. I think I got it working correctly, however I can't seem to get the
realtime portion working. In asterisk 'odbc show' shows it connected, I
see it on my (odbc) mysql server connected and all, it connects and just
idles. So, without
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
2006 Mar 02
3
Child PID's
All, I'm not sure how to word this question but we're noticing a lot of
our asterisk boxes no longer have multiple asterisk child processes.
i.e. doing a 'ps ax' reveals only 1 asterisk PID when normally I'm used
to seeing 8+ .. There is no rhyme or reason to it, and we're using the
safe_asterisk script which has always worked in the past. Ast 1.2.4, zap
1.2.4, naturally..
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
2004 Dec 09
0
Can asterisk accept cleartext auth (uri user:pass) via SIP
Does anyone know if Asterisk can accept cleartext auth (SIP), as in it
recv's a call destined to:
1234:blah@har.har.com
The problem I'm having is simply for faxing, normal calls come in as
g729 and of course we need ULAW for faxes.
sip.conf snippet
[sipfarm]
insecure=very
host=blahblah.netlogic.net
type=peer
context=sip-out
username=+18165551212
secret=blah
canreinvite=no
disallow=all
2004 Dec 15
2
chan_sccp compile problem w/ CVS head?
Any ideas? I edited the Makefile as instructed, ty.
Now compiling .... sccp_channel.c 279 lines
sccp_channel.c: In function `sccp_channel_send_callinfo':
sccp_channel.c:48: structure has no member named `callerid'
sccp_channel.c:49: structure has no member named `callerid'
sccp_channel.c:49: structure has no member named `callerid'
sccp_channel.c:49: structure has no