Displaying 20 results from an estimated 53 matches for "messagenet".
2012 Oct 24
1
Getting 8139cp (1.3) and 8139too (0.9.28) on Centos 5.8
...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 `/usr/src/kernels/2.6.18-308.4.1.el5-i686'
CC [M] /home/silentm/MessageNet/realtek/8139cp.o
/home/silentm/MessageNet/realtek/8139cp.c: In function ?cp_rx_skb?:
/home/silentm/MessageNet/realtek/8139cp.c:430: error: ?struct
net_device? has no member named ?stats?
/home/silentm/MessageNet/realtek/8139cp.c:431: error: ?struct
net_device? has no member named ?stats?
/home/si...
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 /home/silentm/MessageNet/alsa-project/alsa-driver-1.0.16/acore/../alsa-kernel/core/info_oss.c:29,
from /home/silentm/MessageNet/alsa-project/a...
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/wcte12x...
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 => 2345:password@sip.messagenet.it:5061
but I need the other syntax 'cause I *have* to specify a different
context for incom...
2015 May 31
2
Signaling incoming call
...at's a good news...
Currenty I configured my sip.conf so:
register => 00493511111111:MYSECRET at pbxluca/00493511111111
register => 00493512222222:MYSECRET at pbxfax/00493512222222
register => 00493513333333:MYSECRET at pbxanika/00493513333333
register => 4444444444:MYVERYSECRET at messagenet/4444444444
[pbxluca]
type=peer
defaultuser=00493511111111
secret= MYSECRET
dtmfmode=rfc2833
host=172.16.34.132
context=luca_incoming
outboundproxy=172.16.34.132
port=5060
fromuser=00493511111111
fromdomain=172.16.34.132
usereqphone=yes
canreinvite=no
insecure=invite
qualify=yes
qualifyfreq=600
[...
2015 Jun 08
3
Peer unreachable after IP change
...her...
Yesterday I solved changing the Port from 5061 to 5060, but I don't want to
change the port every day... :)
On the Asterisk CLI I see:
OpenWrt*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
messagenet:5060 N 5406013068 105 Unregistered Mon, 08 Jun 2015 00:30:27
and every couple of seconds:
[Jun 8 07:17:52] WARNING[30003]: chan_sip.c:13784 transmit_register: Probably a DNS error for registration to 5406013068 at messagenet, trying REGISTER again (af...
2005 Sep 04
0
Messagenet.it
Hi to all,
I need help to setting up messagenet.it <http://messagenet.it> account in
Asterisk.
My * is connected with static IP to the net.
No other cards at the moment, just the network-card.
I'm able to receive call on the geographical number, but I'm not able to
setup the outgoing calls.
All that it does is:
I dial the number(...
2015 May 29
0
Calling from "extern"
...t 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 AsterikNOW **AND** Messagenet
- 2 VoIP phones, logged into Ubuntu-PBX (my phone, my wife's phone)
- A Twinkle instance on my PC, logged into AsteriskNOW
On AsteriskNOW:
localhost*CLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Desc...
2020 Jun 13
0
Voice "broken" during calls
...oads/uploads, since in that moment
> there were no ones...
Hi again!
Just a detail: I tried an internal call (from my phone, to my wife's
phone) and it works wonderful, no broken, no delay, top quality.
So the problem _MUST_ be in the settings of the communication with
Deutsche Telekom and MessageNet (the providers I used).
The settings for Deutsche Telekom are:
[pbxluca]
type=peer
defaultuser=<mylogin>-0001
secret= <myverysecretpassword>
dtmfmode=rfc2833
host=tel.t-online.de
context=luca_incoming
outboundproxy=tel.t-online.de
port=5060
fromuser=0351xxxxxxx
fromdomain=tel.t-online...
2015 May 28
3
Peer is UNREACHABLE
...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 => 00493513333333:MYSECRET at pbxanika/00493513333333
register => 4444444444:MYSECRET at messagenet/4444444444
[pbxluca]
type=peer
defaultuser=00493511111111
secret= MYSECRET
dtmfmode=rfc2833
host=172.16.34.132
context=luca_incoming
outboundproxy=172.16.34.132
port=5060
fromuser=00493511111111
fromdomain=172.16.34.132
usereqphone=yes
canreinvite=no
insecure=invite
[pbxfax]
type=peer
defaultuser...
2020 Jun 13
4
Voice "broken" during calls
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 moment
there were no ones...
I already had the problem in the past, solved it enabling the
2007 Jul 17
0
help with sip configuration for sipgate.de on asterisk 1.4
...udgetphone.nl/number
register => user:pass at sipgate.de/number
register => user:pass at sip.voiparound.com/number
register => user:pass at sip.webcalldirect.com/
register => user:pass at ixcall.net/number
register => user:pass at freedigits.net/number
register => user:pass at sip.messagenet.it:5061/number
context=inbound
bind=0.0.0.0
nat=yes
fromdomain=sshn.net
localnet=10.0.0.0/255.255.255.0
externip=195.xxx.xxx.xxx
srvlookup=yes
[authentication]
[eutelia-out]
;maxexpirey=360000
;defaultexpirey=180000
type=friend
allow=alaw
context=inbound
username=xxxx
secret=xxxxx
fromuser=numbe...
2015 May 28
0
Peer is UNREACHABLE
...; Well, here what I wrote in my sip.conf:
>
> register => 00493511111111:MYSECRET at pbxluca/00493511111111
> register => 00493512222222:MYSECRET at pbxfax/00493512222222
> register => 00493513333333:MYSECRET at pbxanika/00493513333333
> register => 4444444444:MYSECRET at messagenet/4444444444
>
> [pbxluca]
> type=peer
> defaultuser=00493511111111
> secret= MYSECRET
> dtmfmode=rfc2833
> host=172.16.34.132
> context=luca_incoming
> outboundproxy=172.16.34.132
> port=5060
> fromuser=00493511111111
> fromdomain=172.16.34.132
> usereqphone=ye...
2003 Oct 31
1
Problems with SIP
I'm new to Asterisk, but, Managed to get it working for outound calls from
my ATA --> Asterisk --> Cisco 2620 using SIP. However, I'm having problems
with Inbound calls from the Cisco.. Cisco 2620 --> Asterisk --> ATA .. In
fact, voice mail won't even work..
This is a snippet of what I'm getting when I try to call the ATA
-- Executing
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 to repeat the que...
2015 May 28
4
Peer is UNREACHABLE
...51111111/00493511111 192.168.200.11 D 5060 UNREACHABLE
0049351222222/00493512222 192.168.200.10 D 5060 OK (17 ms)
0049351333333 (Unspecified) D 5060 UNKNOWN
1234 (Unspecified) D 5060 UNKNOWN
messagenet/1234567890 212.97.59.76 5061 Unmonitored
pbxanika/00493511111111 172.16.34.132 5060 Unmonitored
pbxfax/00493513333333 172.16.34.132 5060 Unmonitored
pbxluca/00493512222222 172.16.34.132 5060 Unmonitored
8...
2015 May 28
4
Peer is UNREACHABLE
Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb:
> The phone you gave your wife is really old. Are you sure it supports SIP
> OPTIONS? Can you make a call in or out to it? If you can, it is more
> likely that it just doesn't support that and you can't use a qualify
> statement.
No, I'm not sure.
And no, I can't make any call, right now... At least,
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...
2020 Jun 13
2
Voice "broken" during calls
Am 13.06.2020 um 18:06 schrieb Michael Keuter:
> So the call used Alaw as Codec.
Yes, so seems it to be...
It should has the better quality... But the calls done using my mobile
phone in VoIP with the Asterisk have better quality as the calls done
using the normal VoIP-telefon...
I'm really puzzled...
Luca Bertoncello
(lucabert at lucabert.de)
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 expect problem on my NAT (anymore... initially I
had some problems...).
Regards
Luca Bertoncello
(lucabert at lucabert.de)