[Freeswitch-users] Call send from Freeswitch to Asterisk

[Brian :]

You probably need insecure directive in your FS peer in asterisk

https://www.voip-info.org/wiki/view/Asterisk+sip+insecure

On Tue, Oct 31, 2017 at 11:46 AM, Uday kumar <udy786 at gmail.com> wrote:
> Hello All,
>
> I am using Freeswitch on Debain 8. Configured few DIDs. Calls coming from
> provider and I am trying to send Call to a Asterisk server but getting an
> error "MANDATORY_IE_MISSING".
>
> DID Provider (3.3.3.3)  => FS (192.168.1.70) => Asterisk (192.168.1.92) =>
> Ext 8001
>
> FS dialplan =>
>
> <extension name="public">
>   <condition field="destination_number" expression="(.*)">
>     <action application="bridge" data="sofia/external/$1 at 192.168.1.92"/>
>     <action application="hangup"/>
>   </condition>
> </extension>
>
> Added DID Provider IP and Asterisk IP in ACL.
>
> When I checked on Asterisk, got an error in sngrep, "401 Unauthorized".
> Attached image is of error.
>
> On Asterisk =>
>
> SIP.conf
>
> [freeswitch_1]
> type=peer
> host=192.168.1.70
> port=5080
> disallow=all
> allow=ulaw
> trustrpid=yes
> sendrpid=yes
> context=from-internal
>
> EXTENSIONS.conf
>
> exten => _XXX.,1,Dial(SIP/8001,30,oT)
> exten => _XXX.,n,Hangup
>
> Tried on port 5080 and 5060. Error both time same.
>
>
> My point is, this is Asterisk issue or Freeswitch? Please advice.
>
>
> --
> Thanks & Regard
> Uday.
> Mobile:- +91-9377579349
>
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