Displaying 20 results from an estimated 1000 matches similar to: "[Maybe OT]: SIP Provider"
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
2003 Feb 03
2
Hashing for short pathnames
Hi,
using samba 2.2.5-UL (on a SuSE-SLOX-System) we have to mangle long pathnames
to short ones. We need this for some of our apps which generate batch-files
(*.bat) for compilation.
Normally this works correctly, exept for a directory named
"Only_for_generation". This directory is mangled into "Only_~%0" (it' s NULL
at the end). And this is the problem. In
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
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
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 Jun 08
3
Peer unreachable after IP change
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
UNREACHABLE.
Yesterday I though it was a problem on the line, but today is the same, so I
think it is something other...
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
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
2007 Jan 31
1
Q on configuring shared mboxes
Hello!
I have the following three types of mailboxes on my server:
1) Regular user mail /var/mail/<user> in mbox format
2) Suspected spam in /var/tmp/spamprobe in mbox format
3) Archived mailing list in /home/mks/mksarch in mbox format
I want the 1) to be only accessible to the respective <user>, obviously (POP3
and/or IMAP4).
I wanted the 2) to be accessible to all users --
2011 Mar 01
2
two questions regarding incoming call
Hello,
I want to make an agi script to match incoming DIDs with usernames.
I tried to do such entry in incoming trunk.
[DID_diddw]
include = from-didww
[from-didww]
exten = 3130XXXXXXX,1,AGI("did.php")
exten = 3130XXXXXXX,n,DIAL(SIP/${yup_no},20)
but when i run the rule it says
chan_sip.c:20152 handle_request_invite: Call from '81.85.224.41' to extension
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
2020 Oct 28
4
PJSIP tight loop on auth failure
Hi,
We're using Asterisk 13.17.0 with PJSIP 2.8 bundled.
I've found an issue when Asterisk tries to make a SIP call out using
auth, but has the wrong credentials and keeps getting returned a SIP
407, in this example to an OpenSIPs server requiring user auth.
Basically this happens:
1. Asterisk sends plain INVITE to OpenSIPs
2. OpenSIPs responds with SIP 407 auth required with a
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
2020 Jun 18
3
CallerID fail with Voicetrading operator
Hello,
does some people here use https://voicetrading.com which is a Dellmont
service from Netherlands. At the high begining they were also known as
Finarea (CH and DE mixed Co)
Anyway, after moving from Asterisk13/chan_sip to Asterisk16/PJSIP our
callerID is no more seen by them. We use
Set(CALLERID(num)=+331234356789) and Set(CALLERID(name)=Co name) or
equal to CALLERID(num). We tried
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
2016 May 09
4
VoipRaider is true for FREE calls?
VoipRaider the site, says calls to landlines in Brazil is FREE within
the freedays period. Log in to the website and hire the service, it
says that I have 90 days of freedays paying for cheaper service is $
10.. That is from what I understand, I will pay 10 dolares for
unlimited call in landlines for a period of 90 days? Is that it? Has
anyone tested it there? How many simultaneously calls can