Re: [asterisk-users] PJSIP issue - Syntax error exception when parsing
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.x: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.x:5060;received=193.xx;rport=5060;branch=z9hG4bKPjb092b027-a5b9-4683-8652-c7fefc06ae29 > From: ;tag=3d0a19e7-eabe-4446-84dd-43f02d831033 > To: ;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
Re: [asterisk-users] PJSIP issue - Syntax error exception when parsing
This error is caused by the phone sending an erroneous sip header not by asterisk pjsip stack. solution would be to change the phone. On Wed, Feb 21, 2018 at 8:39 AM, Michele Pinassi wrote: > 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.x: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.x:5060;received=193.xx;rport=5060;branch=z9hG4bKPjb092b027-a5b9-4683-8652-c7fefc06ae29 > From: ;tag=3d0a19e7-eabe-4446-84dd-43f02d831033 > To: ;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 > > -- > Michele Pinassi > Responsabile Telefonia di Ateneo > Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di > Siena > tel: 0577.(23)5000 - central...@unisi.it > > Per trovare una soluzione rapida ai tuoi problemi tecnici consulta le FAQ di > Ateneo, http://www.faq.unisi.it > > > > -- > _ > -- 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 -- _ -- 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
[asterisk-users] PJSIP issue - Syntax error exception when parsing
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.x: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.x:5060;received=193.xx;rport=5060;branch=z9hG4bKPjb092b027-a5b9-4683-8652-c7fefc06ae29 From: ;tag=3d0a19e7-eabe-4446-84dd-43f02d831033 To: ;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 -- Michele Pinassi Responsabile Telefonia di Ateneo Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di Siena tel: 0577.(23)5000 - central...@unisi.it Per trovare una soluzione rapida ai tuoi problemi tecnici consulta le FAQ di Ateneo, http://www.faq.unisi.it signature.asc Description: OpenPGP digital signature -- _ -- 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
