Have a look at the latest blacklist sample in dahdi trunk
http://svnview.digium.com/svn/dahdi/tools/trunk/blacklist.sample?view=log
file: blacklist.sample
...
# Some mISDN drivers may try to attach to cards supported by DAHDI. If you
# have a card which is *not* supported by DAHDI but supported
2012/6/21, Richard Mudgett rmudg...@digium.com:
My previous message was incomplete.
On thing to note is I had to forbid hfcmulti in modprobe.d in the
second box to comply with a warning from dahdi. Without that, I could
see this line in the output of lsmod:
mISDN-core hfcmulti
1. What
2012/6/22, Alec Davis siva...@paradise.net.nz:
Have a look at the latest blacklist sample in dahdi trunk
http://svnview.digium.com/svn/dahdi/tools/trunk/blacklist.sample?view=log
file: blacklist.sample
...
# Some mISDN drivers may try to attach to cards supported by DAHDI. If you
# have a
Am 21.06.2012 11:30, schrieb [Digital^Dude] ®:
Asterisk 1.8.7.1 built by root on a x86_64 running Linux.
CentOS release 5.5 (Final)
RAM: 4 GB
CPU: Dual Xeon 2.66 GHz
Asterisk is running as root
data seg size (kbytes, -d) unlimited
file size (blocks, -f) unlimited
Am 21.06.2012 um 18:05 schrieb Richard Mudgett:
My previous message was incomplete.
On thing to note is I had to forbid hfcmulti in modprobe.d in the
second box to comply with a warning from dahdi. Without that, I could
see this line in the output of lsmod:
mISDN-core hfcmulti
1.
hello,
I just installed a2billing, I did all the config, at least I guess ..
but I still can not integrate sip accounts that I had created (with sip.conf
) in a2billing to apply their billing ..
could someone tell me how to make the junction between asterisk and
a2billing??
I also noticed that the
Gorguez Ka писал 22.06.2012 15:31:
hello,
I just
installed a2billing, I did all the config, at least I guess ..
but I
still can not integrate sip accounts that I had created (with sip.conf)
in a2billing to apply their billing ..
could someone tell me how to
make the junction between
I confess that I had already generated sip accounts but I just delete them.
it remains in the dialplan a2billing and RateCard I created ..
Please can you tell me how to complete the setup?
2012/6/22 Mikhail Lischuk mlisc...@itx.com.ua
**
Gorguez Ka писал 22.06.2012 15:31:
hello,
I just
Hi,
I recently configured T.38 on Asterisk 10.4.2. When I send the fax to
Asterisk, it gives the errors as listed below;
WARNING[25986]: app_fax.c:442 transmit_audio: channel
'SIP/192.168.1.69-' refused to negotiate T.38
WARNING[25986]: app_fax.c:174 span_message: WARNING T.30 ECM
Does your VoIP provider support t.38?
Sent from my iPad
On Jun 22, 2012, at 11:05 AM, Ahmed Munir ahmedmunir...@gmail.com wrote:
Hi,
I recently configured T.38 on Asterisk 10.4.2. When I send the fax to
Asterisk, it gives the errors as listed below;
WARNING[25986]: app_fax.c:442
Here is my setup;
Fax machine - PSTN - Cisco Voice GW - IP cloud - Asterisk. As on Cisco
Voice GW, T.38 fax already configured on SIP protocol.
Does your VoIP provider support t.38?
Sent from my iPad
On Jun 22, 2012, at 11:05 AM, Ahmed Munir ahmedmunir...@gmail.com wrote:
Hi,
I
Hello,
Which one of these ensures that SIP packets are sent and received in a
secure format so that users using public wifi don't allow MITM type of
attacks or others can't read the plaintext SIP packet info. VPN is not an
option. Looking for 2nd most secure to VPN.
P.S. Are both options part of
Is there anything specific in the plaintext SIP packets you want to secure?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B
Sent: Friday, June 22, 2012 1:57 PM
To: Asterisk Users Mailing List -
On 06/22/2012 12:05 PM, Ahmed Munir wrote:
Here is my setup;
Fax machine - PSTN - Cisco Voice GW - IP cloud - Asterisk. As on
Cisco Voice GW, T.38 fax already configured on SIP protocol.
Apparently your configuration of the 'Cisco Voice GW' was not
successful, as it refused to accept a
On 06/22/2012 12:56 PM, Bruce B wrote:
Which one of these ensures that SIP packets are sent and received in a
secure format so that users using public wifi don't allow MITM type of
attacks or others can't read the plaintext SIP packet info. VPN is not
an option. Looking for 2nd most secure to
ADTRAN has some interesting Voice Quality Monitoring built into their switches,
routers, etc: http://adtran.com/web/url/vqm
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stefan at WPF
Sent: Wednesday, June 20, 2012 2:05 PM
To:
We have been quite disappointed by the Adtran VQM. It often shows calls which
had audio issues as being close to perfect. It also often shows calls which
sound perfect as having significant quality issues.
We don't allow reinvites so this might be part of the issue. I don't have a
lot more
Thanks. Want to secure everything and anything possible.
1- Can both SIP over TLS and SRTP work in conjunction to each other?
2- Is SIP over TLS a package or added on module that can be installed from
Digium Asterisk repository?
3- SRTP takes care of the RTP and makes it secure so that MITM
Dears;
One of the problems I faced with Polycom is the voice volume and ring volume,
it is low.
When it rings, even if it is maximum volume, still it is weak.
When I talk and I set the volume to the maximum, I still feel the voice volume
is low and would if to increase it.
The volume is
Tried that. That gave me the dreaded username mismatch:
chan_sip.c:14807 check_auth: username mismatch, have office_outgoing,
digest has s
NOTICE[1738]: chan_sip.c:23250 handle_request_invite: Failed to
authenticate device
sean
On 06/20/2012 04:26 PM, Warren Selby wrote:
On Wed, Jun 20,
On 6/20/2012 8:24 AM, Darren Sessions wrote:
I just finished replying to your direct email (which you can disregard
now as this seems to be a different problem). I'm pretty sure I know
what the issue is, but I'll have to get back to you later this evening (my
time).
I have a different
both would be appreciated.
if you can send me a backtrace, that'd be great
On Jun 22, 2012, at 8:06 PM, Jeremy Kister wrote:
On 6/20/2012 8:24 AM, Darren Sessions wrote:
I just finished replying to your direct email (which you can disregard
now as this seems to be a different problem). I'm
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