hello,

you receive this error because the second line of SIP message not can
begin without a Header. You Phone or server (maybe OpenSIPs or Kamailio)
whet quitting a Via Header make some kind of error so the result is you
have the Via Header in two lines instead one.

Regards

---
I'm SoCIaL, MayBe

El 21/02/2018 a las 03:39, Michele Pinassi escribió:
> Hi all, i'm getting this error:
>
> [Feb 21 09:29:09] ERROR[1250]: pjproject:0 <?>:           
> sip_transport.c Error processing 396 bytes packet from UDP
> 193.xxxxx:5060 : PJSIP syntax error exception when parsing '' header on
> line 2 col 1:
> SIP/2.0 480 User 7000 not registered
>
> Via: SIP/2.0/UDP
> 193.xxxxx:5060;received=193.xxxxxx;rport=5060;branch=z9hG4bKPjb092b027-a5b9-4683-8652-c7fefc06ae29
> From: <sip:3000@voip.xxxxx>;tag=3d0a19e7-eabe-4446-84dd-43f02d831033
> To: <sip:7...@voip.xxxx>;tag=24eb447e8d0b8b1e81ba6efb9d8649a2.ea35
> Call-ID: defee3c7-e5ba-41ff-9be7-3c37e62437f2
> CSeq: 22011 INVITE
> Content-Length: 0
>
>
> -- end of packet.
>
> Asterisk 15.2.0 and PJSip 2.7.1
>
> Tnx, Michele
>
>
>

-- 
_____________________________________________________________________
-- 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

Reply via email to