-Original Message-
From: Jaap Winius jwin...@umrk.nl
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] SIP account registration fails after
upgrade to 1.8
Date: Fri, 22 Mar
Jim,
Their are many places on the net talking about the 15 minute NAT timeout
issue.
If you are not using this device, well, maybe it has a similar bug.
As I am using a fli4l (Linux Router), this seems to not be the problem. I
cannot see any dropped packets or timeouts in the logfiles of
Jaap Winius, 21.03.2013 17:47:
support IPv6. However, it seems that I can't get it to support both IPv4
and IPv6 at the same time. For example, if in sip.conf I set the bindaddr
variable to '::' it will only listen on IPv6 and none of my IPv4-only
friends and peers will be able to connect
Matthew and list,
thanks for your detailed reply.
This is a little hard to diagnose without seeing the SIP traffic for the
duration of the call. It makes it impossible to tell if the INVITES the
provider is sending are related to the call (i.e. have the same Call-ID
header),
but if they
Hi List,
Try canreinvite=yes in sip trunk
This did not make any difference... -.-
-Original Message-
Hi @ll,
I just moved my Asterisk Box and changed the Provider and Internet Access
to a full IP Access by Deutsche Telekom.
I set up my sip.conf as I found various examples
Hi there
Jakob Hirsch wrote:
This is well explained here: http://serverfault.com/a/39561
In short: In Linux, binding to :: means bind to both ipv6 and ipv4.
Setting /proc/sys/net/ipv6/bindv6only to 1 changes this behaviour, and
Debian has this by default (since squeeze, AFAIK).
On my
On Mar 22, 2013, at 5:22 AM, Florian Wolters flor...@florian-wolters.de wrote:
So I did setup another Extension leading me to a MeetMe conference to at
least listen to some MoH while waiting for the 15 Minutes to exceed. This
showed the same behaviour. After exactly 15 Minutes, the call is
Le 22/03/2013 06:08, Satish Barot a écrit :
I found the problem, it's (I think) a bug with queue command. My
dialplan:
[context]
...
exten = 33123,n,macro(unpauseQueueMembers,q820,104,105,136,,)
exten = 33123,n(back2Queue),Queue(${myQueue},nit,,,14400)
ok thank you so much i use dial(zap/r2) instead of g2 and it works without
problem
now my question i have 2 providers i use g1 for the first and g2 for the
second
if i understand i must use r1 instead of g1 for the first provider and r2
instead of g2 for the second provider in order to use
Ya u r right. Value of 1 in r1 or g1 is group you mentioned in zapata.conf
On Mar 22, 2013 8:54 PM, Salaheddine Elharit salah.elharit...@gmail.com
wrote:
ok thank you so much i use dial(zap/r2) instead of g2 and it works without
problem
now my question i have 2 providers i use g1 for the
What is your provider seeing? Many providers send re-INVITEs at 15 minutes.
Many firewalls have closed their port before this due to UDP timeouts. I have
a whitepaper that I wrote on this subject; I will see if I can dig it up.
-Original Message-
From:
Hello bharat,
ok thank you so much for your help and support now i understand :)
2013/3/22 Bharat Lalcheta bharatlalch...@gmail.com
Ya u r right. Value of 1 in r1 or g1 is group you mentioned in zapata.conf
On Mar 22, 2013 8:54 PM, Salaheddine Elharit salah.elharit...@gmail.com
wrote:
ok
hi,
i think we miss understood you Question?
you need round robin on tdm trunk or on 2 internet connections?
what are you asking about burden-sharing between Wimax and FH?
On Fri, Mar 22, 2013 at 4:24 PM, Salaheddine Elharit
salah.elharit...@gmail.com wrote:
ok thank you so much i use
Mitch Claborn wrote:
Interestingly, using Bria we sometimes see similar, though not exactly
the same, symptoms. That would make me wonder about the TCP stack on
the client machine, or similar.
With a softphone, you're dependent on the entire software stack up to the
softphone and at the
yes i want to use the burden-sharing between Wimax and FH using a diguim
cards
2013/3/22 Asghar Mohammad asghar...@gmail.com
hi,
i think we miss understood you Question?
you need round robin on tdm trunk or on 2 internet connections?
what are you asking about burden-sharing between Wimax
your dialplan nothing to do with bandwidth it dial out to digium card what
ever come in.
1.
if your providers calls come in via digium card and you want send out using
sip or any other tech. then use context defined in group 1 for provider 1
and context defined in group 2 for provider 2.
2.
if
Florian Wolters wrote:
Does it make sense to have a more detailed tcpdump of the SIP session? If
so, how should such a thing been shared without posting too much ASCII
text to the list?
SIP sessions are generally small enough to post right to the list. Otherwise,
you can put them up on a
I've installed 7 Digium D40's over the last 24 hours. They work
flawlessly - no dropped calls, no 1-way audio, sound quality is
noticeably better. If these work out through Monday (our busiest day)
then we'll order a dozen more for the rest of the agents.
The one downside to this approach
Hello Everyone,
We are aware of a2billing and it's LCR functionality. I just wanted to
know what other solutions you may be using. Maybe a tool that is a
self contained module (ie a2billinig - Asterisk - LCR Tool -
Trunk). Is there such a tool? It should be open source as is all good
software.
On Fri, 22 Mar 2013 10:07:57 +0100, Jakob Hirsch wrote:
This is well explained here: http://serverfault.com/a/39561
Indeed, that's the solution!
There is a file, called /etc/sysctl.d/bindv6only.conf, on my system that
sets /proc/sys/net/ipv6/bindv6only to 1 and it's been there since June
On Thu, 21 Mar 2013 16:35:16 +, Jaap Winius wrote:
... For example, if my server sends it a SIP packet with a
register request and a Call-ID that looks like this:
Call-ID: 4991f57656d159925b296e5b3b06496b@[2001:888:abcd:1::a]
... somewhere along they line they end up changing it to
On Fri, 22 Mar 2013 02:46:43 +, Jaap Winius wrote:
Of course, an even better solution would be if Asterisk had a variable
with which to alter the Call-ID string format so that I could omit the
IP address. :-)
It turns out that there in a variable that can do exactly that, and is
Chris,
Thank you for sharing. It will help one day when 11 will become stable
enough to consider it for a production system.
Interestingly, jabber.conf has the same exact setting and the same exact
value and comment in 1.8.20.1:
priority = 1;; Resource priority
Unless the
Bilal,
Here is the respective section from my working 7906 .conf file:
dateTimeSetting
dateTemplateM/D/Ya/dateTemplate
timeZoneCentral Standard/Daylight Time/timeZone
ntps
ntp
name172.29.100.11/name
ntpModeUnicast/ntpMode
/ntp
24 matches
Mail list logo
