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
CSeq: 1 INVITE
Could someone please help to configure SIP URI in pjsip.conf for urn:service:sos SIP/2.0. I am getting above error.
“urn” is not a supported URI scheme.
Thanks a lot for your reply @jcolp Could you please suggest any alternative to make it work.
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.
Thanks again @jcolp . Is there any documentation or guide to help us to get it there. Any documentation and guide will be highly appreciated.
You have two ish options
- Put an Kamailio in front at use that to convert the headers to somthing asterisk support
- Creat a patch to asterisk / pjsip that enables support for
- 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
Thanks a lot @TheMark for your reply.
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