Displaying 20 results from an estimated 1000 matches similar to: "[hint state][BLF] Asterisk 1.8.7 does not send RINGING notifications, even with notifyringing=yes"
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
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 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
2014 May 23
1
BLF and notifyringing in Asterisk 11
I am trying to get something working that is just not doing quite what I
want. It may not be possible, but I figured it was worth asking about.
The details:
Asterisk 11.6.0
Polycom SoundPoint IP650 phones running 4.03 firmware.
We have a queue with 4 phones in it. ringinuse is set to yes and the
stategy is ringall. In sip.conf, we have notifyringing set to yes as well.
Asterisk is sending
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 Apr 27
2
Fwd: Re: xen-qemu-dm does not build with backported xen-4.1
Il 27/04/2011 14:37, Ian Campbell ha scritto:
> Which specific revision of qemu-xen-4.1 are you using?
I'm using a tarball from
http://bits.xensource.com/oss-xen/release/4.1.0/xen-4.1.0.tar.gz
It seems Thomas used the tools/ioemu-qemu-xen subdirectory as source for
xen-qemu-dm. At a first glance I didn't notice
debian/mypatches/add-rules.patch: it adds Rules.mk from the tools
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
2010 Sep 20
1
Confused about notifyringing in sip.conf
Hello list,
I read this in sip.conf :
notifyringing = no ; Control whether subscriptions already
INUSE get sent RINGING when another call is sent (default: yes)
What does this mean ?!
Does this mean that when I mark this as "yes", a phone that already has
taken a call will be send a second and third call ?!
I want that if a phone is in use (calling), the phone does not
2009 Mar 04
1
What's the use of sip.conf's notifyringing ?
Hello
With 1.4.23.1, I can't really see any difference between setting this value
to yes or no.
Can you explain ?
Regards
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2009 Apr 09
2
notifyringing=no does not work
"
Hello,
I have been trying to get my Grandstream GXP2000 phones to stop showing ringing state on monitored extensions. But no matter where I put notifyringing=no asterisk still sends the ringing state to the phones. Is this a bug I should report or is there another way around it.
Here is how i have my subscriptions setup:
extensions.conf
[demo]
exten => 6100,hint,SIP/100
exten =>
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
2009 Oct 17
4
how to limit the calls leaving a queue?
Hi,
I explain what I want to do..
All the operators share their phones. The number of the operator isn't
constant, so it's possible that two operators share all the phones.
They need to move all around, so they pick up the first phone they find.
If there are only few operator is very annoying for them to ear the
other phones ringing while they are at the phone!
So I'dd like to limit