Displaying 20 results from an estimated 400 matches similar to: "Peer unreachable after IP change"
2015 Jun 14
0
Peer unreachable after IP change
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On 06/08/2015 01:18 PM, Luca Bertoncello wrote:
> Hi list!
>
> Another day, another problem... I'm checking with Nagios my
> Asterisk at home, and since yesterday I noticed that, after my IP
> changes (Deutsche Telekom drop the DSL-line every 24 hours, so that
> I have a new IP every day), the peer of an VoIP-provider I use is
2015 Jun 13
3
Asterisk and Deutsche Telekom
jg <webaccounts173 at jgoettgens.de> schrieb:
> It doesn't really depend on your sip.conf and Asterisk. Your gateway/router
> will be the major problem. My summer project will be to look at session
Are you sure?
Right now I'm using an italian SIP-Provider (Messagenet), configured in my
sip.conf and I can receive calls without any problem...
So, I don't think, I have to
2015 May 31
6
Signaling incoming call
Hi list!
Finally I got my Asterisk works with my two phones...
It was a problem on my Firewall (for the phone of my wife) and on my Dialplan
(for forwarding calls).
Now all works as expected, at least in the simulation I did with AsteriskNOW.
Hopefully it will work later, when Deutsche Telekom changes my ISDN to VoIP...
Well, now I have some time to spend with "fooling"...
My phone
2012 Oct 24
1
Getting 8139cp (1.3) and 8139too (0.9.28) on Centos 5.8
Subject says it all.
How can I get the 1.3 version and 0.9.28
to compile on CentOS 5.8 ???
When I compile the two as modules I get errors.
My Makefile is:
obj-m += 8139cp.o 8139too.o
all:
make -C /lib/modules/$(shell uname -r)/build M=$(PWD) modules
clean:
make -C /lib/modules/$(shell uname -r)/build M=$(PWD) clean
The errors I get are:
Entering directory
2008 Mar 31
2
alsa 1.016 compile error on latest kernel centos 5.1
Hi all,
I need to compile alsa-project 1.0.16 on the latest centos 5.1 kernel.
I am getting this error. What to do... ?
CC
[M] /home/silentm/MessageNet/alsa-project/alsa-driver-1.0.16/acore/sound_oss.o
CC
[M] /home/silentm/MessageNet/alsa-project/alsa-driver-1.0.16/acore/info_oss.o
In file included
from
2008 Apr 11
1
odd error compiling zaptel-1.4.10
CC [M]
/home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/vpmadt032.o
CC [M]
/home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/GpakApi.o
CC [M]
/home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/../voicebus.o
LD [M]
/home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/wcte12xp.o
2006 Feb 23
3
register => 2345:password@sip_proxy doesn't care about port
Hi,
to register my Asterisk with a SIP provider I use the following
syntax, as shown in the default sip.conf:
register => 2345:password@sip_proxy
where
[sip_proxy]
type=peer
context=from-messagenet
host=sip.messagenet.it
port=5061 <------------- please note this one!!!
5061 is provider's port I have to register to.
This also would work for me:
register =>
2015 May 31
2
Signaling incoming call
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Guenther Boelter <gboelter at gmail.com> schrieb:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA256
>
> On 05/31/2015 02:31 PM, Luca Bertoncello wrote:
> > Hi list!
> >
> > Now all works as expected, at least in the simulation I did with
> > AsteriskNOW. Hopefully it will work later, when Deutsche Telekom
2015 May 28
3
Peer is UNREACHABLE
Darryl Moore <darryl at moores.ca> schrieb:
> Ahh. Seen that before! That suggests to me that you don't have your
> sip.conf records setup right.
>
> What's your sip.conf look like?
Well, here what I wrote in my sip.conf:
register => 00493511111111:MYSECRET at pbxluca/00493511111111
register => 00493512222222:MYSECRET at pbxfax/00493512222222
register =>
2019 Dec 03
4
Delay on speak with Asterisk
Hi list!
I'm using Asterisk 13.14.1 from Debian 9 repositories.
The provider is Deutsche Telekom und Messagenet (just for receive).
I can call and receive calls, but I have a little problem: there is a
"delay" of about 1-1,5 seconds between the time the voice is sent and
the time when the voice is received, so that it happens very often that
the peer does not get my voice and try
2023 Nov 07
2
[Maybe OT]: SIP Provider
Hi all!
Currently I'm using Messagenet, a SIP-Provider in Italy, to have an
italian number via VoIP, _to receive calls only_.
I use it to allow my friends and parents in Italy to call me in Germany
without paying too much.
This service was free of charge in the last years.
Now will Messagenet beginning from end of november, to cancel this free
service and only offer paying services (for
2015 May 28
4
Peer is UNREACHABLE
Hi list!
I have a problem and I hope someone can help me...
I configured an Asterisk on a VM to serve more accounts and act as a proxy to
other SIP-providers.
The first account running on my phone works without any problem.
A second account, running on the phone of my wife, is always UNREACHABLE.
I can just see in the log:
[May 28 21:48:46] NOTICE[3646]: chan_sip.c:22933 sip_poke_noanswer: Peer
2013 Mar 15
2
app_rtsp.c ported to Asterisk 11.x
Hi,
If anyone is interested, I have ported app_rtsp.c to Asterisk 11.x. I have
tested it with GStreamer RTSP server and a C920 webcam streaming H264 SVC
video from one machine to another machine running Linphone. Contact me at
this e-mail address robkrakora at messagenetsystems.com for source code.
Best Regards,
--
Rob Krakora
MessageNet Systems
101 East Carmel Dr. Suite 105
Carmel, IN 46032
2006 Nov 09
2
register suddenly fails
Hi everybody,
I've got a very strange problem:
As far as I remember I didn't change anything on my Asterisk side. I
have 2 SIP providers to which I can place outbound calls.
Today I noticed that outbound calls to provider "inode" fail (and
inbound from this provider too). On the CLI I get every 20 seconds
following messages:
Nov 9 20:01:07 NOTICE[952]: chan_sip.c:5422
2020 Jun 13
3
Voice "broken" during calls
Am 13.06.2020 um 22:09 schrieb Antony Stone:
Hi Antony
> You are *assuming* that it's the codec causing the difference.
Well, I really don't know what I can think, now...
> We don't know that.
>
> Let me get this clear, to make sure I understand (differences emphasised):
>
> 1. You use *a VoIP softphone app* on your mobile, which is registered by SIP,
> to
2020 Jun 15
1
Voice "broken" during calls
On Monday 15 June 2020 at 18:55:23, Luca Bertoncello wrote:
> Absolutly *no changes* on the behaviour compared with my Thomsons...
Okay, I'm glad we can rule out the specific make / model of phone - that would
have been bizarre.
> I try to summarize:
>
> 1) Phones are not the problem, since 3 phones of 2 different
> companies/model have the same issue.
Good (if you see
2015 May 27
3
Asterisk as "Proxy" and more device for a number
Hi list!
I'm very new in Asterisk and VoIP, and of course I have a problem... :)
Well, my problem is, that Deutsche Telekom wants me to change my ISDN
to VoIP... :(
I must do that, since I have no alternative.
Well, I have now two VoIP-phones (Thomson ST2022 and KE1020A). I can
configure my two numbers by Deutsche Telekom and I got now an extra
number from Messagenet.it.
Now the
2015 Apr 23
2
Sample Docker images for Asterisk available
Hello all,
I created a set of Docker images running Asterisk and exposing AMI /
ARI ports that i found to be quite useful for ARI / AMI development
and regression.
As they are based on Docker with whaleware, adding new configuration
files to roll your own dialplan / queues / voicemail etc is pretty
easy. And you can run quite a lot on the same box to simulate
clusters.
There is no SIP / RTP
2015 May 29
0
Calling from "extern"
Hi list!
Finally I got my wife's phone working in my Asterisk.
Unfortunately I have some problems, too...
Current situation:
- AsteriskNOW with 4 Accounts (00493511111111, 00493512222222,
00493513333333, 5678). This is "for test" and it will be replaced by "the
real world", when I got my Asterisk to work...
- A second Asterisk (Ubuntu-PBX) on another VM, logging in
2020 Jun 13
0
Voice "broken" during calls
Am 13.06.2020 um 08:28 schrieb Luca Bertoncello:
> Hi!
>
> I have a Asterisk installation to manage my phones at home (provider is
> Deutsche Telekom).
> It works, but very often the voice is "broken"...
> Yesterday during a call it was very difficult to understand what my
> partner sayd...
>
> It can NOT be a problem of other downloads/uploads, since in that