RURI using SIP 2.0 in PJSIP

Asterisk Asterisk SIP
1

sip_transport.c Error processing 2970 bytes packet from UDP xxxxxxxxxx:5060 : PJSIP syntax error exception when parsing ‘Request Line’ header on line 1 col 23:

INVITE urn:service:sos SIP/2.0

Via: SIP/2.0/UDP XXX.XX.XX.XX:5060;branch=pkpkksdewd+4p8369b88cd2b3f9z318778efb8d98234+sip+1+a65f6552

From: sip:XXXXXXXX@XXX.XX.XX.XX:5060;tag=XXX.XX.XX.XX+1+4c69b85f+42ac9458

To: sip:sos@XXXXXXXXXXXXXX:5060

CSeq: 1 INVITE

Expires: 180

Content-Length: 1376

Could someone please help to configure SIP URI in pjsip.conf for urn:service:sos SIP/2.0. I am getting above error.

2

“urn” is not a supported URI scheme.

3

Thanks a lot for your reply @jcolp Could you please suggest any alternative to make it work.

4

The code doesn’t support it, so you’d either have to modify the code to support it or not have such URIs in use to Asterisk.

5

Thanks again @jcolp . Is there any documentation or guide to help us to get it there. Any documentation and guide will be highly appreciated.

6

You have two ish options

  1. Put an Kamailio in front at use that to convert the headers to somthing asterisk support
  2. Creat a patch to asterisk / pjsip that enables support for
  3. Wait for someone else to add it (will take long time)

unless you know a lot about Asterisk / Pjsip and are a strong coder I recoment option 1
as these kind of URI is intended for providers and not endusers
so you should probaly also talk with you provider on where they can handel them for you

7

Thanks a lot @TheMark for your reply.