[asterisk-users] SIP and Voice on different nets

[Artem Chekulaev]

Yes, Voice = RTP

Using chan_sip

2017-04-27 15:32 GMT+03:00 Dovid Bender <dovid at 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 <slonikk at 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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>>
>> Check out the new Asterisk community forum at:
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>>
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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
>
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