Seems I responded the same time as Josh. Follow what he has suggested.

On Thu, Apr 27, 2017 at 8:41 AM, Artem Chekulaev <slon...@gmail.com> wrote:

> Yes, Voice = RTP
>
> Using chan_sip
>
> 2017-04-27 15:32 GMT+03:00 Dovid Bender <do...@telecurve.com>:
>
>> By voice do you mean RTP? Are you using chan_sip or pjsip?
>>
>>
>> On Thu, Apr 27, 2017 at 8:10 AM, Artem Chekulaev <slon...@gmail.com>
>> wrote:
>>
>>> ​I have connection with two networks (by VoIP provider setup)
>>> 1 - 10.10.10.0/24 = SIP
>>> 2 - 10.10.11.0/24 = Voice
>>>
>>> How to tell Asterisk send / receive voice traffic not on SIP network.
>>> When I look into dumps, I see Asterisk trying to use SIP net for voice
>>>
>>> Unfortunately, I _need_ to use two networks instead of one​
>>>
>>> --
>>> _____________________________________________________________________
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>>>
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>>
>>
>> --
>> _____________________________________________________________________
>> -- 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
>
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