[asterisk-users] asterisk server - no sound
Marcelo Terres
mhterres at gmail.com
Tue Jun 6 14:28:22 CDT 2017
And it is worst (and that could be the reason of the error).
127.0.0.1 is configured in 2 interfaces (lo and venet0), just with
different network masks.
Marcelo H. Terres <mhterres at gmail.com>
IM: mhterres at jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres
On 6 June 2017 at 18:54, andre castro <andre at andrecastro.info> wrote:
> I am using version: 14.5.0
> No, Im not using Dundi.
> Can you a bit more informative when you say I "need to configure the IPs
> in your server"?
> thanks!
> a
> On 06/06/2017 07:47 PM, Marcelo Terres wrote:
>> I think you need to configure the IPs in your server. You just have localhost...
>> Marcelo H. Terres <mhterres at gmail.com>
>> IM: mhterres at jabber.mundoopensource.com.br
>> https://www.mundoopensource.com.br
>> https://twitter.com/mhterres
>> https://linkedin.com/in/marceloterres
>>
>>
>> On 6 June 2017 at 16:27, andre castro <andre at andrecastro.info> 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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>>>
>>> --
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>>
>
> --
> oooooooooo.io
> bibliotecha.info
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
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