[asterisk-users] asterisk server - no sound

[Marcelo Terres]

Which Asterisk version are you using?

Marcelo H. Terres <mhterres at gmail.com>
IM: mhterres at jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
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https://linkedin.com/in/marceloterres


On 6 June 2017 at 18:32, andre castro <andre at andrecastro.info> wrote:
> Any ideas.
> After configuring  port forwarding on the server (machine making nat) to
> forward connections originated from external clients to the machine
> running asterisk, as explained in
> https://www.voip-info.org/wiki/view/port+forwarding
> My peers were unable to register.
>
>
> And When running Asterisk I am getting:
> 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.
>
> Any advice what to do next?
>
> thanks
> a
>
> On 06/06/2017 05:27 PM, andre castro wrote:
>> 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.
>>>
>>
>
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