[asterisk-users] PJSIP configuration question

[George Joseph]

On Wed, Dec 10, 2014 at 2:03 PM, Dan Cropp <dan at amtelco.com> wrote:

> Thanks George.
>
>
>
> That was the ip address I was given.  Unfortunately, my contact at
> Vitelity is gone for the day so I can’t verify it with him.
>
>
>
> I added the qualify_frequency as you suggested and it does appear that I
> have something configured incorrectly….
>
>
>
> <--- Transmitting SIP request (483 bytes) to UDP:0.0.19.196:5060 --->
>

Well, THAT's not right.  Did you obfuscate the 0.0.19.196 or is that how it
really is?  Are you NATed?



> OPTIONS sip:64.2.142.93 at 5060 SIP/2.0
>
> Via: SIP/2.0/UDP
> xxx.xxx.xx.xxx:5060;rport;branch=z9hG4bKPjcea63914-b8d1-483d-96db-11968abab704
>
> From: <sip:e31d5809-f26a-4219-8365-70931428072b at xxx.xxx.xx.xxx
> >;tag=7cfab3ba-73de-4243-9967-d1e6a5e7b0b4
>
> To: <sip:64.2.142.93 at 5060>
>
> Contact: <sip:e31d5809-f26a-4219-8365-70931428072b at xxx.xxx.xx.xxx:5060>
>
> Call-ID: 7ba766bf-363b-47d0-a388-62a58d1df88d
>
> CSeq: 33778 OPTIONS
>
> Max-Forwards: 70
>
> User-Agent: Asterisk PBX 13.0.0
>
> Content-Length:  0
>
>
>
>
>
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