Hi,
in my company we use an asterisk installation with around 50 soft- and
hardphones of all kind.
From time 2 time the users (almost only Softphone users) report some
voice qualities... mostly echoes.
These problems do not occur on all PCs at the same time and since setup
of our PBX almost
Hello everybody
I'm trying to enable the Digium res_fax app at my *11.7 Server.
a fax show stats comes up with
FAX Statistics:
---
Current Sessions : 0
Reserved Sessions: 0
Transmit Attempts: 0
Receive Attempts : 1
Completed FAXes : 1
Failed FAXes : 1
It is really more interesting the receiving part. Can you paste here?
Leandro
2014/1/21 Jakob-Matthias Böttger ja...@j-mb.de
Hello everybody
I'm trying to enable the Digium res_fax app at my *11.7 Server.
a fax show stats comes up with
FAX Statistics:
---
Current Sessions
Hi Duncan,
We have sip set debug on and nothing is shown, even though tcpdump/ngrep
on the same server does. It's very strange.
The output of ip address list is:
[root]# ip address list
1: lo: LOOPBACK,UP,LOWER_UP mtu 16436 qdisc noqueue state UNKNOWN
link/loopback 00:00:00:00:00:00 brd
Hi,
My problem has been solved.In my case because of CPU High load .
On Mon, Jan 20, 2014 at 3:34 PM, mahdieh saeed mahdieh.sa...@gmail.comwrote:
Hi every body
our Calls are begging dropped for no reason and it starts with the sound
quality dropping and then the caller unable to hear our
Hi Larry,
No, they are on separate machines.
On 21 January 2014 17:54, Larry Moore lmo...@omninet.net.au wrote:
Is Kamalio running on the same system as Asterisk?
On 21/01/2014 2:41 PM, David Cunningham wrote:
Hi Larry,
Thanks for the reply. We have all of those settings left out of
I am going to try a Lync server/asterisk integration, so I really
appreciate!
Leandro
2014/1/21 Lincoln King-Cliby linc...@controlworks.com
Ok, so now I just feel kind of stupid. After I got home I decided to play
with this a little more.
After far too long I realized that part of the
Hi
The log i've posted
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
-- Executing [12345678912 tel:%5B12345678912@from-sip:1]
Answer(SIP/abcde-0016, ) in new stack
0x7fd11404cd00 -- Probation passed - setting RTP source
address to 123.456.789.123:17108
--
David,
It seems to me that Asterisk is not seeing/binding to your VPN
interface. You need to debug that first. I would set en explicit bind
statement in sip.conf to the VPN interface address and nothing else.
Then start your asterisk and watch the log messages. It should confirm
that it
On Tue, Jan 21, 2014 at 12:40 AM, David Cunningham
dcunning...@voisonics.com wrote:
Hi Paul,
Using ngrep/tcpdump shows the packet clearly going from the Kamailio server
and arriving at the Asterisk server. This is why it's a mystery that
Asterisk doesn't see the call coming in. We tried
On Tue, Jan 21, 2014 at 10:47 AM, Paul Belanger
paul.belan...@polybeacon.com wrote:
On Tue, Jan 21, 2014 at 12:40 AM, David Cunningham
dcunning...@voisonics.com wrote:
Hi Paul,
Using ngrep/tcpdump shows the packet clearly going from the Kamailio server
and arriving at the Asterisk server.
I am not sure, but try to add a wait(2) as first command. When I want fax
detection, I insert always a small delay for letting the fax detection
routine to detect it.
Leandro
2014/1/21 Jakob-Matthias Böttger ja...@j-mb.de
Hi
The log i've posted
== Using SIP VIDEO CoS mark 6
== Using
i already added a Progess() and Wait(5) and it still does not detect faxes.
Am 21.01.2014 16:53, schrieb Leandro Dardini:
I am not sure, but try to add a wait(2) as first command. When I want
fax detection, I insert always a small delay for letting the fax
detection routine to detect it.
Please paste the actual code. First has to be the Wait and then any other
thing.
Leandro
2014/1/21 Jakob-Matthias Böttger ja...@j-mb.de
i already added a Progess() and Wait(5) and it still does not detect
faxes.
Am 21.01.2014 16:53, schrieb Leandro Dardini:
I am not sure, but try to
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Le 20/01/2014 12:03, Jean-Denis Girard a écrit :
Hi list,
I'm looking for the best / recommended solution for automatic discovery
of phone numbers for a multiple Asterisk system. This would be for an
administration, with many branches (~30), but
Hi again,
Some other things missing from the previous message: setting E1/T1/[J1?]
and upgrading.
E1/T1
-
an E1/T1/J1 should be configured to be either E1 or T1.There are a
number of ways to set this. Specifically: quite a few different ways.
With the new DAHDI initialization scheme we
All;
I'm having a problem sending an outbound fax using Asterisk-1.8.15-cert3
and the spandsp fax module using a SIP trunk. I'm seeing hundreds of these:
ERROR[14423]: udptl.c:294 encode_open_type: UDPTL
(SIP/runcentral_outbound-0074): Buffer overflow detected (59 + 134
175)
Has
Hi,
Am running a freepbx install and created trunks, extensions and groups.
Now I'd like to hand out the Asterisk phonebook to the phones (all VVX
310's). Is there an easy way to do this?
Best,
Stanley
--
_
-- Bandwidth and
On 01/21/2014 01:55 PM, Stanley van Dijk wrote:
Hi,
Am running a freepbx install and created trunks, extensions and groups.
Now I'd like to hand out the Asterisk phonebook to the phones (all VVX
310's). Is there an easy way to do this?
Best,
Stanley
Even the old ones could view a webpage.
On Tuesday, January 21, 2014 12:32 PM, Adrian Serafini wrote:
On 01/21/2014 01:55 PM, Stanley van Dijk wrote:
Hi,
Am running a freepbx install and created trunks, extensions and groups.
Now I'd like to hand out the Asterisk phonebook to the phones (all VVX
310's). Is there an easy way to do
Hi Andres,
Thanks for the idea. We did send bindaddr to the VPN address and restarted
Asterisk, but unfortunately that didn't solve the issue. Asterisk didn't
complain, but still the sip set debug on didn't show the packets.
On 22 January 2014 01:40, Andres and...@telesip.net wrote:
David,
Hi Paul,
Thanks, we did try restarting Asterisk after the VPN was up but that didn't
solve the issue either.
On 22 January 2014 02:55, Paul Belanger paul.belan...@polybeacon.comwrote:
On Tue, Jan 21, 2014 at 10:47 AM, Paul Belanger
paul.belan...@polybeacon.com wrote:
On Tue, Jan 21, 2014
(Please don't top-post.)
On Wed, 22 Jan 2014, David Cunningham wrote:
We did send bindaddr to the VPN address and restarted Asterisk, but
unfortunately that didn't solve the issue. Asterisk didn't complain, but
still the sip set debug on didn't show the packets.
Have you confirmed via
On 22 January 2014 09:11, Steve Edwards asterisk@sedwards.com wrote:
(Please don't top-post.)
On Wed, 22 Jan 2014, David Cunningham wrote:
We did send bindaddr to the VPN address and restarted Asterisk, but
unfortunately that didn't solve the issue. Asterisk didn't complain, but
Is there a mapping of AMI versions to Asterisk versions?
eg:
AMI 1.0 = Ast 1.4
AMI 1.1 = Ast 1.6
etc...
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory
When I issue a 'core show channels' command I notice that long usernames (and
channel number) are truncated. For example, if the username is
FONEMITEL1234567890 for a trunk, then it will show
SIP
Privilege: Command
Channel Location State Application(Data)
On Tue, Jan 21, 2014 at 3:39 PM, Michelle Dupuis mdup...@ocg.ca wrote:
When I issue a 'core show channels' command I notice that long usernames
(and channel number) are truncated. For example, if the username is
FONEMITEL1234567890 for a trunk, then it will show
SIP
Privilege: Command
On Tue, Jan 21, 2014 at 5:18 PM, David Cunningham
dcunning...@voisonics.com wrote:
On 22 January 2014 09:11, Steve Edwards asterisk@sedwards.com wrote:
(Please don't top-post.)
On Wed, 22 Jan 2014, David Cunningham wrote:
We did send bindaddr to the VPN address and restarted Asterisk,
On Tue, Jan 21, 2014 at 5:51 AM, Jakob-Matthias Böttger ja...@j-mb.de wrote:
Hello everybody
I'm trying to enable the Digium res_fax app at my *11.7 Server.
a fax show stats comes up with
FAX Statistics:
---
Current Sessions : 0
Reserved Sessions: 0
Transmit Attempts
At this point in time, you'll need to show us a .pcap on the Asterisk
box, when you make a call to it via Kamailio.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter:
Hello,
Perhaps you need to have directmedia=no set for the channel, the call
doesn't appear to have been answered hence asterisk won't be able to
hear any tones to determine for itself if the call is an incoming fax.
Larry.
On 21/01/2014 6:51 PM, Jakob-Matthias Böttger wrote:
Hello
Sorry, I missed the line showing the call had been answered.
On 22/01/2014 8:11 AM, Larry Moore wrote:
Hello,
Perhaps you need to have directmedia=no set for the channel, the call
doesn't appear to have been answered hence asterisk won't be able to
hear any tones to determine for itself if the
On 1/21/14, 4:38 PM, David Cunningham wrote:
Hi Andres,
Thanks for the idea. We did send bindaddr to the VPN address and
restarted Asterisk, but unfortunately that didn't solve the issue.
Asterisk didn't complain, but still the sip set debug on didn't show
the packets.
Ok, that is progress
I'm having some trouble turning with trunk monitoring while using an
outbound proxy.
While all other sip messaging (e.g. calls) respects the outboundproxy
setting, Options packets from setting qualify=yes do not. Asterisk
tried to send the Options message directly to the host= IP, instead
of the
Are you absolutely sure you need to use the outboundproxy setting rather than
using a peer/friend?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Lemberger
Sent: Tuesday, January 21, 2014 7:53 PM
To:
Hello,
Is there anyway to encrypt or scramble a bit the secret used to register
with a provider? Im talking about the
register = fromuser@fromdomain:secret@host
directive in
sip.confhttp://www.voip-info.org/wiki/view/Asterisk+config+sip.conf
This clever dude modified the code back in 1.4:
Hello All,
Asterisk: 1.8.13.0
Dahdi : 2.6.2
Kernel : 2.6.32-431.3.1.el6.i686 #1 SMP Fri Jan 3 18:53:30 UTC 2014 i686
i686 i386 GNU/Linux
OS : CentOS 6.4
When I show meetme room details using meetme list command it shows Minus
in activity column.
Any Idea.
meetme list
Conf Num Parties
Solved
On Wed, Jan 22, 2014 at 12:44 PM, Chandrakant Solanki
solanki.chandrak...@gmail.com wrote:
Hello All,
Asterisk: 1.8.13.0
Dahdi : 2.6.2
Kernel : 2.6.32-431.3.1.el6.i686 #1 SMP Fri Jan 3 18:53:30 UTC 2014 i686
i686 i386 GNU/Linux
OS : CentOS 6.4
When I show meetme room details
Hello,
Le 16/01/2014 14:20, Darryl Moore a écrit :
Yup. That's what i do. The CLI version of linphone set to autoanswer,
with the audio jacks tied to our exernal sound system. Works well. The
echo cancellation in linphone helps a lot for speakerphones.
Indeed, works well :-) Thanks.
On
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