asterisk users - Apr 2017 - SIP and Voice on different nets

?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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On Thu, Apr 27, 2017, at 09:10 AM, Artem Chekulaev
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?
Both the chan_sip and chan_pjsip modules have a "media_address" option
which can be used to specify the address to place in the SDP for media.
In the case of chan_pjsip there is also a "bind_rtp_to_media_address"
option which can be used to guarantee that RTP leaves from that same
address as well.

Cheers,

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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?
>
> --
> _____________________________________________________________________
> -- 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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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?
>>
>> --
>> _____________________________________________________________________
>> -- 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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