On Thu, Oct 13, 2016 at 12:06 PM, wrote:
> > I have Asterisk running well inside our network. I did some
> > experiments exposing it to internet but had some issues:
> > 1. NAT issues (voice one way, etc). From what I understand double-
> > NAT users will always
A few years ago I ran into something similar. Using TLS seemed to fix
it, but it was a while ago so I might be wrong.
On 10/14/2016 11:35 AM, Greg Woods wrote:
On Fri, Oct 14, 2016 at 9:06 AM, Dovid Bender > wrote:
Changing your port
On Fri, Oct 14, 2016 at 9:06 AM, Dovid Bender wrote:
> Changing your port should fix all your worries.
>
That may work if you control both ends of the SIP connection.
--Greg
--
_
-- Bandwidth and
On Fri, Oct 14, 2016 at 7:55 AM, Jerry Geis wrote:
> Apparently Verizon is blocking or changing packets on port 5060 so my
> softphone from my hotspot will not work.
>
Sounds like you are another victim of SIP ALG. I ended up having to change
to a VOIP provider that would
Changing your port should fix all your worries.
On Fri, Oct 14, 2016 at 11:00 AM, Greg Woods wrote:
>
>
> On Fri, Oct 14, 2016 at 7:55 AM, Jerry Geis wrote:
>
>> Apparently Verizon is blocking or changing packets on port 5060 so my
>> softphone from
iptables -t nat -A PREROUTING -i inboundinterface -s sourceip -d destinationip
-p udp --dport 5070 -j DNAT --to destinationip:5060
Or something similar. You may however found that the provider's filtering is
application based rather than port based.
--
Sent from my cellphone.
--
They don't like competition ;)
On Fri, Oct 14, 2016 at 9:55 AM, Jerry Geis wrote:
> Apparently Verizon is blocking or changing packets on port 5060 so my
> softphone from my hotspot will not work.
>
> How do I set asterisk (11.23.0) to run default 5060 for all other
In article
Apparently Verizon is blocking or changing packets on port 5060 so my
softphone from my hotspot will not work.
How do I set asterisk (11.23.0) to run default 5060 for all other devices I
have - BUT for my software run on a different port like 5070? I'm using
linphone and is easy to change the
hello,
i recently purchased a Wildcard AEX800 digium card. Ive installed
asterisk 13 and all prerequistses on ubuntu serv14.04 LTS. Dahi is the
driver am using; ive configured all but when i call from PSTN through
fxo port an not getting anything in logs or to extensions. below are
error messages
See below output;
[root@abc ~]# lsof -u root | wc -l
5116
From: Dovid Bender
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Subject: Re: [asterisk-users] Openfile Issue
> Message-ID:
>
See below;
[root@abc asterisk]# lsof -u 50771 | wc -l
0
BTW, I'm using CentOS 6.5
> From: Dovid Bender
>
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>
>> Subject: Re: [asterisk-users] Openfile Issue
>>
On 13 October 2016 at 13:18, Tony Mountifield wrote:
> exten => _X,1,NoOp(Matching single digit)
> exten => _X.,1,NoOp(Matching multiple digits)
> exten => _X!,2,SayNumber(${EXTEN})
> exten => _X!,3,Etc..
Thanks - I appreciate the idea, but it matches more than 2 digits.
13 matches
Mail list logo