subject:"\[asterisk\-users\] asterisk server \- no sound"

Re: [asterisk-users] asterisk server - no sound

2017-06-06 Thread Marcelo Terres

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 
IM: mhter...@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  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 
>> IM: mhter...@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  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 -vvr
>>> 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:
>>>
>>> loLink 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)
>>>
>>> venet0Link 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.

>>>
>>> --
>>> oo.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
>>
>
> --
> oo.io
> bibliotecha.info
>
> --
> _
> --

Re: [asterisk-users] asterisk server - no sound

2017-06-06 Thread Marcelo Terres

Well, based on the config that you sent, your server just have the
localhost IP (127.0.0.1)
Marcelo H. Terres 
IM: mhter...@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  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 
>> IM: mhter...@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  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 -vvr
>>> 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:
>>>
>>> loLink 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)
>>>
>>> venet0Link 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.

>>>
>>> --
>>> oo.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
>>
>
> --
> oo.io
> bibliotecha.info
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] asterisk server - no sound

2017-06-06 Thread andre castro

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 
> IM: mhter...@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  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 -vvr
>> 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:
>>
>> loLink 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)
>>
>> venet0Link 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.
>>>
>>
>> --
>> oo.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
> 

-- 
oo.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

Re: [asterisk-users] asterisk server - no sound

2017-06-06 Thread Marcelo Terres

I think you need to configure the IPs in your server. You just have localhost...
Marcelo H. Terres 
IM: mhter...@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  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 -vvr
> 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:
>
> loLink 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)
>
> venet0Link 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.
>>
>
> --
> oo.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

-- 
_
-- 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

Re: [asterisk-users] asterisk server - no sound

2017-06-06 Thread Marcelo Terres

Looks like it comes com pbx_dundi.c.

Why are you using dundi?

Regards,
Marcelo H. Terres 
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres


On 6 June 2017 at 18:43, Marcelo Terres  wrote:
> Which Asterisk version are you using?
>
> Marcelo H. Terres 
> IM: mhter...@jabber.mundoopensource.com.br
> https://www.mundoopensource.com.br
> https://twitter.com/mhterres
> https://linkedin.com/in/marceloterres
>
>
> On 6 June 2017 at 18:32, andre castro  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 -vvr
>>> 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:
>>>
>>> loLink 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)
>>>
>>> venet0Link 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.

>>>
>>
>> --
>> oo.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:
>>

Re: [asterisk-users] asterisk server - no sound

2017-06-06 Thread Marcelo Terres

Which Asterisk version are you using?

Marcelo H. Terres 
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres


On 6 June 2017 at 18:32, andre castro  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 -vvr
>> 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:
>>
>> loLink 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)
>>
>> venet0Link 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.
>>>
>>
>
> --
> oo.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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asterisk-users

Re: [asterisk-users] asterisk server - no sound

2017-06-06 Thread andre castro

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 -vvr
> 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:
> 
> loLink 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)
> 
> venet0Link 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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Re: [asterisk-users] asterisk server - no sound

2017-06-06 Thread Marcelo Terres

Try to use the echo app. If you can listen your echo, so it is
something in the network.

Regards,
Marcelo H. Terres 
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres


On 6 June 2017 at 14:18, andre castro  wrote:
> hello folks,
> this might be a simple question...
>
> I just installed asterisk in a debian server.
> All seems to be running fine, but the audio sent by the server.
> If I have one of my registered peers call and extension (102) that plays
> back audio (extension.conf and sip.conf coffee-pasted below), Asterisk
> answers and prints no errors.
> Its `sip show channels` prints:
>
> PeerUser/ANRCall IDFormatHoldLast MessageExpiry
>Peer
> peer.ip1001 1...-5060   (ulaw)  No Rx: ACK
>1001
>
> But I hear nothing at the peer's end.
>
> When one peer calls another, sound comes through just fine.
> So my hunch is that is something to do with the audio supplied by the
> server.
> Do I need to have alsa installed??
> Any hint?
>
> sip.conf:
>
> [general]
> context = unauthenticated
> bindport = 5060
> bindaddr = 0.0.0.0
> tcpbindaddr = 0.0.0.0
> tcpenable = yes
> videosupport = no
> textsupport=yes
> alwaysauthreject=yes
> allowguest=no
>
> [1001] ; grandstream 1
> context = home
> type = friend
> callerid = One <1001>
> secret = XYZ
> host = dynamic
> mailbox = 1001
> disallow = all
> allow = ulaw
> transport = udp
> dtmfmode=auto   ; accept touch-tones from the devices, negotiated
> automatically
> nat=force_rport
>
> [1005] ; mobile
> context = home
> type = friend
> callerid = Five <1005>
> secret = XYZ
> host = dynamic
> mailbox = 1005
> disallow = all
> allow = ulaw
> transport = udp
> dtmfmode=auto   ; accept touch-tones from the devices, negotiated
> automatically
> nat=force_rport
>
>
> extensions.conf:
> [home]
> exten = 102,1,Answer()
> same =  n,Wait(1)
> same =  n,Playback(beep)
> same =  n,Wait(1)
> same =  n,Playback(im-sorry)
> same =  n,Wait(1)
> same =  n,Playback(number-not-answering)
> same =  n,Wait(1)
> same =  n,Hangup()
>
> exten => 1001,1,Dial(SIP/1001) ; grandstream phone
> exten => 1005,1,Dial(SIP/1005) ; mobile
>
>
>
>
> --
> oo.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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Re: [asterisk-users] asterisk server - no sound

2017-06-06 Thread andre castro

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 -vvr
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:

loLink 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)

venet0Link 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.
> 

-- 
oo.io
bibliotecha.info

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Re: [asterisk-users] asterisk server - no sound

2017-06-06 Thread Antony Stone

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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play on words.

   Please reply to the list;
 please *don't* CC me.

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Re: [asterisk-users] asterisk server - no sound

2017-06-06 Thread andre castro

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.
> 

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Re: [asterisk-users] asterisk server - no sound

2017-06-06 Thread Administrator TOOTAI


Le 06/06/2017 à 16:25, Daniel Tryba a écrit :

On Tue, Jun 06, 2017 at 03:18:32PM +0200, andre castro wrote:

extensions.conf:
[home]
exten = 102,1,Answer()
same =  n,Wait(1)


If this is copy and paste, then your dialplan is broken (= should be =>)


Well, no. = or => are the same.

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Re: [asterisk-users] asterisk server - no sound

2017-06-06 Thread Antony Stone

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?

Is there a router in between any of them?

Is there any NAT involved?

> Do I need to have alsa installed??

No.


Antony.

-- 
Perfection in design is achieved not when there is nothing left to add, but 
rather when there is nothing left to take away.

 - Antoine de Saint-Exupery

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Re: [asterisk-users] asterisk server - no sound

2017-06-06 Thread Daniel Tryba

On Tue, Jun 06, 2017 at 03:18:32PM +0200, andre castro wrote:
> extensions.conf:
> [home]
> exten = 102,1,Answer()
> same =  n,Wait(1)

If this is copy and paste, then your dialplan is broken (= should be =>)

But to debug, enable logging (core set verbose 5), when needed debugging
(core set debug 5) and sip logging (sip set debug on / pjsip set logger
on).

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[asterisk-users] asterisk server - no sound

2017-06-06 Thread andre castro

hello folks,
this might be a simple question...

I just installed asterisk in a debian server.
All seems to be running fine, but the audio sent by the server.
If I have one of my registered peers call and extension (102) that plays
back audio (extension.conf and sip.conf coffee-pasted below), Asterisk
answers and prints no errors.
Its `sip show channels` prints:

PeerUser/ANRCall IDFormatHoldLast MessageExpiry
   Peer
peer.ip1001 1...-5060   (ulaw)  No Rx: ACK
   1001

But I hear nothing at the peer's end.

When one peer calls another, sound comes through just fine.
So my hunch is that is something to do with the audio supplied by the
server.
Do I need to have alsa installed??
Any hint?

sip.conf:

[general]
context = unauthenticated
bindport = 5060
bindaddr = 0.0.0.0
tcpbindaddr = 0.0.0.0
tcpenable = yes
videosupport = no
textsupport=yes
alwaysauthreject=yes
allowguest=no

[1001] ; grandstream 1
context = home
type = friend
callerid = One <1001>
secret = XYZ
host = dynamic
mailbox = 1001
disallow = all
allow = ulaw
transport = udp
dtmfmode=auto   ; accept touch-tones from the devices, negotiated
automatically
nat=force_rport

[1005] ; mobile
context = home
type = friend
callerid = Five <1005>
secret = XYZ
host = dynamic
mailbox = 1005
disallow = all
allow = ulaw
transport = udp
dtmfmode=auto   ; accept touch-tones from the devices, negotiated
automatically
nat=force_rport


extensions.conf:
[home]
exten = 102,1,Answer()
same =  n,Wait(1)
same =  n,Playback(beep)
same =  n,Wait(1)
same =  n,Playback(im-sorry)
same =  n,Wait(1)
same =  n,Playback(number-not-answering)
same =  n,Wait(1)
same =  n,Hangup()

exten => 1001,1,Dial(SIP/1001) ; grandstream phone
exten => 1005,1,Dial(SIP/1005) ; mobile




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