[asterisk-dev] Asterisk RPID

[Klaus Darilion]

Gregory Boehnlein wrote:
> Hello,
> 	I'm tracking down an issue w/ RPID between an Asterisk 1.4
> (SVN-branch-1.4-r186229) and an Audiocodes gateway.
> 
> The Audiocodes gateway is choking on the following header:
> 
> Remote-Party-ID: "anonymous" <sip:@207.166.192.174>;privacy=off;screen=no
                                    ^^^^

This is a bug. If username is empty there must not be @.

Actually if the userid is missing does it make sense to add RPID header 
at all?



regards
klaus

> 
> Seems to me that the URI is malformed as it leaves Asterisk. The Audiocodes
> complains thusly:
> 
> 13d:0h:26m:31s ( sip_stack)(9533756 ) AcSIPParser: Problem in SIP Message
> Headers
> 13d:0h:26m:31s ( sip_stack)(9533757 ) !! [ERROR] AcSIPParser: Parse Error.
> Top label must start with ALPHA
> 13d:0h:26m:31s ( sip_stack)(9533758 ) !! [ERROR] Message type: INVITE
> 13d:0h:26m:31s ( sip_stack)(9533759 ) !! [ERROR] Source header:
> 13d:0h:26m:31s ( sip_stack)(9533760 ) !! [ERROR] Line: 10. Column: 35
> 
> The invite as received at the Audio Codes:
> 
> INVITE sip:4408354800 at 69.4.36.4 SIP/2.0
> Via: SIP/2.0/UDP 207.166.192.174:5060;branch=z9hG4bK40fa1acf;rport 
> From: "anonymous" &lt;sip:asterisk at 207.166.192.174&gt;;tag=as2a56ff37 
> To: &lt;sip:4408354800 at 69.4.36.4&gt; 
> Contact: &lt;sip:asterisk at 207.166.192.174&gt; 
> Call-ID: 0dcb31f37e27465325b2a9246f90b84f at 207.166.192.174 
> CSeq: 102 INVITE 
> User-Agent: N2Net-Univoice-1.4 
> Max-Forwards: 70 
> Remote-Party-ID: "anonymous"
> &lt;sip:@207.166.192.174&gt;;privacy=off;screen=no 
> Date: Mon, 06 Apr 2009 16:18:39 GMT 
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY 
> Supported: replaces 
> Content-Type: application/sdp 
> Content-Length: 291
> v=0 o=root 3039 3039 IN IP4 207.166.192.174 s=session c=IN IP4
> 207.166.192.174 t=0 0 m=audio 17404 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000
> a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16 a=silenceSupp:off - - - - a=pt
> 
> Not sure if this is a bug, a feature, or a strict implementation on the
> Audiocodes side. Comments?
> 
> 
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
> 
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-dev