Displaying 20 results from an estimated 2000 matches similar to: "Qualify time +2000ms?"
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
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
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
2005 Sep 05
2
USING TWO ACCOUNTS WITH BROADVOICE
Hi,
I have two accounts with broadvoice.
Now, I want to be able to distinguish between them.
I though that this would be simple by adding "/EXTEN" at the end of the
register statement. For example:
register => num1:pass@sip.broadvoice.com/1000
Unfortunately, this is not working.
When I call into my box I hear busy tone.
My config looks like this:
[root@voip asterisk]# cat sip.conf
2005 Mar 14
1
weird outbound problem through broadvoice (new)
Hello,
Have a weird problem when using asterisk (1.0.6). There are certain
numbers I cannot dial when using asterisk with my broadvoice account.
No problems with inbound. With outbound calls, I can call some numbers
(for example broadvoice customer support number) and unsuccessfully with
some. However, when I configure my account directly on x-lite, I dont
see these outbound problems.
Here is a
2005 Mar 12
1
Broadvoice outgoing problems
Hello All,
I'm just getting into *, and trying to use a Broadvoice account. It works inbound, but Outbound fails no matter what sip.conf parameters I try. From the recent posts here I think it could be:
A bad CVS release - I will try to download and build from a new one
Broadvoice not challenging and/or Asterisk not responding with an Authorization: in the INVITE header.
I am
2005 Mar 09
2
Broadvoice latest changes and still not working-An
I've tried everything with the * box after this weekend. I have read
every document on the problems people are having with them after this
weekend as well, but none of them address my problem.
I checked my settings in my sips which I have below as well,
I have changed the host file a few times, but this was new to me and I
never had modified it before. I have and the same results
2004 Jun 01
2
BroadVoice usage?
Hi all,
I've been trying to use BroadVoice as a SIP service provider. They don't
officially
support * but are helpful when it comes to answering questions for setup
parameters. They claim they have no firewalls or access lists that need to be
set up so I can get access to their servers.
However, something's still not quite right when I use the parameters.
It looks like our Asterisk
2005 Feb 26
1
Dial out through Broadvoice
Hello all,
When I call the Broadvoice number all is good.
When I try to call out through DISA on my broadvoice line i get the following:
Executing Dial("SIP/147.135.0.129-0815bc60",
"SIP/16037862111@proxy.bos.broadvoice.com|30") in new stack
-- Called 16037862111@proxy.bos.broadvoice.com
-- Got SIP response 480 "Temporarily Not Available" back from
2005 Mar 10
1
Asterisk@Home, AMP, and Broadvoice
Egad, not again with Broadvoice! Anyhow, I recently installed AAH and
configured my TDM11B and got that and some SIP phones working. I still
have some issues to work out, etc, but my current problem is Broadvoice.
I have checked out all of the online resources, including the recent
list exchange about the recent changes made by Broadvoice. However, the
one thing I have found to be consitent in