Displaying 20 results from an estimated 6000 matches similar to: "Restricting registration for peer '611' to 60 seconds (requested 1200)"
2008 Feb 19
1
Restricting registration for peer 'iaxmodem0' to 60 seconds
I have setup hylafax today, along with iaxmodem. I'm just starting the
debugging process and see the following message every 60 seconds:
[Feb 19 17:44:16] NOTICE[1996]: chan_iax2.c:6021 update_registry:
Restricting registration for peer 'iaxmodem0' to 60 seconds (requested 300)
Can someone tell me what this means? Why is it there? And how do I get rid
of it!
Thanks,
MD
2008 Feb 19
0
Restricting registration for peer 'iaxmodem0' to60 seconds
There's a #define macro in channels/chan_iax.c that you can modify to make this forced value higher. Just open it up in your favourite editor and search for '60' and you'll find it.
Now if there's an easier way than having to change a source-level macro, I'm all ears...
Cheers!,
--jkinard
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
2006 Feb 06
1
IAX registration expiration
I can't seem to change the default registration for IAX clients:
Feb 6 12:22:52 NOTICE[7883]: chan_iax2.c:5673 update_registry: Restricting
registration for peer 'virbiage' to 60 seconds (requested 3600)
Feb 6 12:23:03 NOTICE[7883]: chan_iax2.c:5673 update_registry: Restricting
registration for peer 'test1' to 60 seconds (requested 1200)
Can this be controlled on a
2013 Jul 04
3
Asterisk + iaxmodem + hylafax makes sometimes wedged for hylafax
Hi, we have a faxserver with Asterisk, IAXModem and Hylafax.
Faxes come from a SIP trunk to Asterisk, then are forwarded throught 5 IAXModems managed with Hylafax.
Hylafax users can also send faxes to these modems and Asterisk send them throught the SIP trunk.
We also have a dedicated modem used only for sending faxes coming from an Hylafax dedicated user.
Sometimes Hylafax reports that a modem
2010 Oct 23
4
Asterisk 1.8 IAX Registration
Hi,
Have just been testing asterisk 1.8.0, 1.8.0-rc5 and a trunk version from about half an hour ago.
IAX Friends (Zoiper Softphones) don't stay registered for more than a few seconds they start out with status unknown and quickly become unreachable, I am using realtime with postgresql, with tables and configuration that have worked fine for IAX in 1.6 and a trunk release from a few months
2006 Jan 27
2
Name/username (sip show peers)
How can I make it more readable?
Name/username
601/601
123456789/123456789
voipbuster/abcd
601 = hotline
123456789 = Peter Pan
only voipbuster/abcd is easy read/understandable!
bye
Ronald Wiplinger
2006 Apr 16
1
[Fwd: Re: voicemail email-from]
Ronald Wiplinger wrote:
> Steve Totaro wrote:
>> Ronald Wiplinger wrote:
>>> kevin ling wrote:
>>>> Hi,
>>>>
>>>> Check the vm_general.inc file
>>>>
>>>>
>>> Where should this file be?
>>>
>>>
>>> bye
>>>
>>> Ronald Wiplinger
>>>
>>>
>> You
2005 Jul 17
2
DNS SRV
I have added in my zone file;
_sip._udp.elmit.com. IN SRV 20 0 5060 vpbx.elmit.com.
As I understand it should mean that any sip connection to
<anyname>@elmit.com should go to the udp port 5060 at the host
vpb.elmit.com.
In Asterisk's extensions.conf I have in the context [default]
exten => ronald,1,Dial(${PHONE_615},60,tr)
exten => ronald,2,Voicemail,u615@office
exten =>
2005 Jul 28
12
Can you caculate with me?
before I accuse somebody to "overbill" I would like you to calculate
with me:
Rate: 0.0189 for calling Taiwan via NuFone
Duration: 930 seconds
Lets vote for the answers: 0.7269 or 0.2929 ???
bye
Ronald Wiplinger
2007 Feb 14
5
Bandwidth shapping device
I have a link to a building (e.g. 10Mb/s) and want to split up the
bandwidth to different users. Each user should get e.g., 512kB/s plus
256kB/s dedicated for VoIP.
What kind of device can I use for that ? (managing switch ??? which one?)
bye
Ronald Wiplinger
2006 Apr 26
6
I am looking for a webphone on MY SITE
I am looking for a way of not to install a softphone, preferable as a
link on a web site to a webphone on MY SITE !!!
Has anybody an idea for that? AJAX?
bye
Ronald Wiplinger
2006 Mar 31
4
How to check if a phone / line is used?
In the past I used SetGroup and CheckGroup to figure out if my allowed
providers lines are all used or not.
Since most of my provider have given me a single line anyway, I wonder
if there is a way to check if this (provider) line is taken already.
How can I do that?
Same is with the phone. How can I see in CLI if a phone is now in use or
not?
"Sip show peers" shows me just if it is
2008 Apr 04
1
rxfax crashes Asterisk (segmentation fault)
Hi,
I am using spandsp-0.0.4, tiff-3.8.2, and agx-ags-addon with Asterisk
1.4.18.
Everytime rxfax executes, Asterisk crashes:
-- Executing [fax at phones:1] Set("Zap/2-1",
"FAXFILE=/var/spool/asterisk-fax/1207322398.0.tif") in new stack
-- Executing [fax at phones:2] RxFAX("Zap/2-1",
"/var/spool/asterisk-fax/1207322398.0.tif") in new st ack
[Apr 4
2006 Jun 04
3
transfer & other features
*CLI> show features
Builtin Feature Default Current
--------------- ------- -------
Pickup *8 *8
Blind Transfer # ##
Attended Transfer *2
One Touch Monitor *1
Disconnect Call * *0
Dial option is tTwWr
I tried to call from 601 to 615
601 keys in *0
2006 Jun 24
5
ASTCC: How to reset periodically all "card in use" flag back?
If a user calls and hangs up before the destination party rings, than
the in-use flag remains set! This is one case, but maybe there are many
other cases.
I have created a number the user can dial to reset this flag. However,
that is written in the manual!!! Who reads a manual anyway!!!!
I want to make to reset all in use flag with a program. Has anybody done
it, or has a better idea?
My idea
2005 Jul 23
2
ASTCC gives me only the time, but no cost
I try to track down an error that causes that Astcc just reports the time, but not the costs.
I could narrow the problem down into this sub routine:
sub calccost() {
my ($adjconn, $adjcost, $answeredtime, $increment) = @_;
eval { my $adjtime = int(($answeredtime + $increment - 1) / $increment) * $increment };
my $cost;
print STDERR "Adjusted time is $adjtime, cost is $adjcost with
2005 May 17
4
Is SKYPE a threat or should we do something (together)
Skype is very succesfsfull and get more and more users, ... we can
ignore them, accept them or do something,...
My suggestion is that we try to do something, ...
If we would peer to each other, than we get soon also a great amount of
users together, and than our service becomes more valuable, ...
Let's discuss advantages and disadvantages!
bye
Ronald
--
Ronald Wiplinger (CEO of
2007 Feb 14
2
moving WiFi phone
Can anybody tell me how I can set-up multiple access points with
overlapping coverage, so that a moving WiFi phone user can continuesly
use the phone.
bye
Ronald Wiplinger
2005 Jul 23
1
astcc timestamps
The time stamps in ASTCC are useless as they are now:
Fri Jul 22 15:06:25 2005
Wouldn't it be better to use something like:
2005-07-22 15:06:24 Fri
I want to sort the records by date, but with the format now it is
impossible... or do I miss something?
bye
Ronald Wiplinger
2005 Aug 11
2
list in asterisk cli is getting too long
How can I use something like |more in CLI ?
The lists are getting too long, like sip show users
bye
Ronald Wiplinger