[Asterisk-Dev] Support for SIP message 300 multiple choices (SONUS)

[John Todd]

At 4:20 PM +0200 on 6/9/04, Olle E. Johansson wrote:
>Umar Sear wrote:
>>Hi does asterisk support SIP message 300 multiple
>>choices.
>No, but I would love to see a SIP Debug with such a packet.
>I've never seen it actually being used.
>
>/O

Oh, you'll see lots of interesting things that SONUS boxes send back that nobody else uses. 

The most confusing to date is stuff like this (from SONUS to a SIP proxy).  Take a close look at the INVITE line; the address that Asterisk (or any other SIP proxy) parses this as is "12021234444;npdi=yes".  So, I have to run every INVITE through an additional pass to chop out that junk before it can be properly parsed.  Note that this is totally legal, apparently, with the RFC's, but it REALLY makes my job difficult when gateways put that stuff in the user portion of the URI and not the host-portion of the URI.

http://lists.cs.columbia.edu/pipermail/sip-implementors/2001-July/001537.html


> 5 May 05:33:26/GLOBAL/ser: RECEIVED message from 10.10.22.34:5060:
> INVITE sip:12021234444;npdi=yes at 128.151.224.17 SIP/2.0
> Via: SIP/2.0/UDP 10.10.22.34:5060;branch=z9hG4bK026vib3048igq1c1a6u0sr
> From: <sip:3015551313-mmvoip-pajbsqpvq0em0 at 10.10.22.34;isup-oli=0;otg=DC_DMS_100_SS7>;pstn-params=808482;tag=SD404ve01-72
> To: <sip:12021234444 at 128.151.224.17:5060>
> Call-ID: SD404ve01-49790f708b92c43acd06b572762be2fa-06a30i1
> CSeq: 20078 INVITE
> Allow: OPTIONS, SUBSCRIBE, INVITE, CANCEL, ACK, PRACK, INFO, REFER, NOTIFY, BYE
> Accept: multipart/mixed, application/sdp, application/isup, application/dtmf, application/dtmf-relay
> Contact: <sip:3015551313-mmvoip-pajbsqpvq0em0 at 10.10.22.34:5060;transport=udp>
> Remote-Party-ID: <sip:3015551313-mmvoip-pajbsqpvq0em0 at 10.10.22.34>;privacy=off
> Content-Length: 200
> Content-Disposition: signal;handling=required
> Content-Type: application/sdp
> Max-Forwards: 70
>
> v=0
> o=Sonus_UAC 20077 26921 IN IP4 10.10.22.34
> s=SIP Media Capabilities
> c=IN IP4 10.10.22.34
> t=0 0
> m=audio 6156 RTP/AVP 0 0
> a=fmtp:0 ptime:20
> a=fmtp:0 ptime:20
> a=sendrecv
> a=ptime:20

JT