asterisk users - Jun 2017 - asterisk server - no sound

Thank you Daniel for pointing out the errors and debug option. Both
fixed and on.
It made no difference. There are no errors printed and still no sound on
ppers

Now to Antony questions:

On 06/06/2017 04:36 PM, Antony Stone wrote:
> On Tuesday 06 June 2017 15:18:32 andre castro wrote:
> 
>> I just installed asterisk in a debian server.
>> All seems to be running fine, but the audio sent by the server.
> 
>> But I hear nothing at the peer's end.
>>
>> When one peer calls another, sound comes through just fine.
> 
> Tell us about your networking arrangement - are both phones and the
Asterisk
> machine on the same network?
Nop. They are in 2 different networks. The phones in one and the
Asterisk machine in another.
> 
> Is there a router in between any of them?
Yes. In the phones network.
> 
> Is there any NAT involved?
Yes in the phones' network. They both have different private IP address
and one public IP.
> 
>> Do I need to have alsa installed??
> 
> No.
So I thought.

Thanks guys!!
> 
> 
> Antony.
> 
-- 
oooooooooo.io
bibliotecha.info

On Tuesday 06 June 2017 16:57:07 andre castro wrote:
> On 06/06/2017 04:36 PM, Antony Stone wrote:
> > 
> > Tell us about your networking arrangement - are both phones and the
> > Asterisk machine on the same network?
> 
> Nop. They are in 2 different networks. The phones in one and the
> Asterisk machine in another.
Okay, that is why you have audio between the two phones, then - they can see 
each other directly, on the same network, and nothing is interfering with the 
traffic between them.
> > Is there a router in between any of them?
> 
> Yes. In the phones network.
> 
> > Is there any NAT involved?
> 
> Yes in the phones' network. They both have different private IP address
> and one public IP.
Okay, I suspect that this NATting router is not passing the UDP packets from 
the server back to the phones correctly, based on the SIP connection (when the 
phone makes the call).

SIP is on UDP 5060; audio is on UDP 10,000 - 20,000.

If it's a Linux router, you need to make sure you are allowing FORWARDed
traffic
which matches ESTABLISHED, RELATED.

If it's not a Linux router, you need to find out how to get it to support
SIP
and RTSP.


Good luck,


Antony.

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Thanks Anthony.

I did it on the server, according to
https://www.voip-info.org/wiki/view/port+forwarding

However after doing it, when running Asterisk I get the following message
sudo asterisk -vvvvvvr
No ethernet interface found for seeding global EID. You will have to set
it manually.
Unable to access the running directory (No such file or directory).
Changing to '/' for compatibility.

How and where can it be set?

My server ifconfig:

lo        Link encap:Local Loopback
          inet addr:127.0.0.1  Mask:255.0.0.0
          inet6 addr: ::1/128 Scope:Host
          UP LOOPBACK RUNNING  MTU:65536  Metric:1
          RX packets:113895058 errors:0 dropped:0 overruns:0 frame:0
          TX packets:113895058 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:0
          RX bytes:36041459269 (33.5 GiB)  TX bytes:36041459269 (33.5 GiB)

venet0    Link encap:UNSPEC  HWaddr
00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00
          inet addr:127.0.0.1  P-t-P:127.0.0.1  Bcast:0.0.0.0
Mask:255.255.255.255
          inet6 addr: ::2/128 Scope:Compat
          inet6 addr: 2a01:488:66:1000:5c33:846e:0:1/128 Scope:Global
          UP BROADCAST POINTOPOINT RUNNING NOARP  MTU:1500  Metric:1
          RX packets:158483849 errors:0 dropped:0 overruns:0 frame:0
          TX packets:272193853 errors:0 dropped:230 overruns:0 carrier:0
          collisions:0 txqueuelen:0
          RX bytes:61233254724 (57.0 GiB)  TX bytes:106403959440 (99.0 GiB)

venet0:0  Link encap:UNSPEC  HWaddr
00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00
          inet addr:server.ip.add.r  P-t-P:server.ip.add.r
Bcast:server.ip.add.r  Mask:255.255.255.255
          UP BROADCAST POINTOPOINT RUNNING NOARP  MTU:1500  Metric:1



On 06/06/2017 05:09 PM, Antony Stone wrote:
> On Tuesday 06 June 2017 16:57:07 andre castro wrote:
> 
>> On 06/06/2017 04:36 PM, Antony Stone wrote:
>>>
>>> Tell us about your networking arrangement - are both phones and the
>>> Asterisk machine on the same network?
>>
>> Nop. They are in 2 different networks. The phones in one and the
>> Asterisk machine in another.
> 
> Okay, that is why you have audio between the two phones, then - they can
see
> each other directly, on the same network, and nothing is interfering with
the
> traffic between them.
> 
>>> Is there a router in between any of them?
>>
>> Yes. In the phones network.
>>
>>> Is there any NAT involved?
>>
>> Yes in the phones' network. They both have different private IP
address
>> and one public IP.
> 
> Okay, I suspect that this NATting router is not passing the UDP packets
from
> the server back to the phones correctly, based on the SIP connection (when
the
> phone makes the call).
> 
> SIP is on UDP 5060; audio is on UDP 10,000 - 20,000.
> 
> If it's a Linux router, you need to make sure you are allowing
FORWARDed traffic
> which matches ESTABLISHED, RELATED.
> 
> If it's not a Linux router, you need to find out how to get it to
support SIP
> and RTSP.
> 
> 
> Good luck,
> 
> 
> Antony.
> 
-- 
oooooooooo.io
bibliotecha.info