Hi Ernie,
When one-way audio appear (no matters if there is a VPN or NAT server on
the diagram) I simply :
* Enable SIP debug on Asterisk server. Excecute 'sip set debug ip x.x.x.x'
on Astrisk CLI, where x.x.x.x is the IP of the phone or SIP peer you want
to debug.
* Make a test call and replica
On 4/18/2017 7:40 PM, Ernie Dunbar wrote:
Server network: 192.168.0.0/24
OpenVPN network: 10.8.0.0/24
Asus network: 192.168.1.0/24
The Asterisk SIP registration appears to be responding properly to
this - this is what I see when I do a 'sip show peer' for an Aastra
phone that's connecting thro
rs@lists.digium.com>
Sent: 19-Apr-17 10:25:59 AM
Subject: [asterisk-users] SIP connections over OpenVPN
connection get one-way voice.
Hi everyone. I'm having some trouble with an
OpenVPN tunnel that
l Discussion'"
>>
>> Sent: 19-Apr-17 10:25:59 AM
>> Subject: [asterisk-users] SIP connections over OpenVPN connection get
>> one-way voice.
>>
>>> Hi everyone. I'm having some trouble with an OpenVPN tunnel that isn't
>>> working *qu
rs@lists.digium.com>
Sent: 19-Apr-17 10:25:59 AM
Subject: [asterisk-users] SIP connections over OpenVPN
connection get one-way voice.
Hi everyone. I'm having some trouble with an
OpenVPN tunnel that isn't working *quite* as well as we&#
te table.
Originalmeddelande
Från: Ernie Dunbar
Datum: 2017-04-19 00:25 (GMT+01:00)
Till: 'Asterisk Users Mailing List - Non-Commercial
Discussion'
Rubrik: [asterisk-users] SIP connections over OpenVP
zarre routing problems when states appear in the state table.
Originalmeddelande Från: Ernie Dunbar
Datum: 2017-04-19 00:25 (GMT+01:00) Till: 'Asterisk Users Mailing List -
Non-Commercial Discussion' Rubrik:
[asterisk-users] SIP connections over OpenVPN connection get
-- Original Message --
From: "Ernie Dunbar"
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Sent: 19-Apr-17 10:25:59 AM
Subject: [asterisk-users] SIP connections over OpenVPN connection get
one-way voice.
Hi everyone. I'm having
Hi everyone. I'm having some trouble with an OpenVPN tunnel that
isn't working *quite* as well as we'd hoped.
First, here's our technical details:
The OpenVPN server (v2.3.4-5+deb8u1) is a Debian 8 box behind a NAT
router. The router has UDP port 1194 forwarded
I am building a server that accepts audio over SIP and returns
pre-recorded audio files based on the result of a recognizer. However,
I am not sure exactly how to establish and collect data over and SIP
connection. I literally only need a program (socket?) to establish the
connection and buffe
, as seen in the log, but the SIP
connection of 3. does NOT hangup.
Regards,
Florian
PS: Believe me, I'm searching for over one week in the whole internet for a
solution, but did not find it.
- Original Message -
From: "Jean-Yves Avenard" <[EMAIL PROTECTED]>
To: <[EMAIL
TED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, July 30, 2004 11:46 PM
Subject: Re: [Asterisk-Users] SIP connections do not hang up
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> If you just bothered to search this list in the past 12 hours, you
> would have found a
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
If you just bothered to search this list in the past 12 hours, you
would have found a solution around that:
to summarize:
Add in zapata.conf:
busydetect=yes
busycount=6
The maximum it will take for asterisk to see the person hanged-up is
after 6 busy
Hi everybody,
I have strange problem:
I'm calling from inside (either X-Lite using SIP channel or a ISDN telephone
using Zap Channel) using sipgate to a number in public network.
When I'm hanging up before the other person picked up the phone, the line is
not closed correctly.
The phone keeps on
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