Re: [asterisk-users] PJSIP Returning 421 Extension Required
On Wed, Jan 13, 2016 at 12:58 PM, Trey Hilyard wrote: > I am turning up a PJSIP Endpoint and am having problems when they send an > INVITE to my server. Asterisk is returning a 421 Extenstion Required. Since > "extension" means different things in the SIP stack versus Asterisk, I > don't know what it is complaining about. > > I have attached the trace below. Nothing else shows up with core verbose > or core debug enabled, so I am assuming it has to be dying at the PJSIP > module. The INVITE does come from an abnormal UDP Port, which is also shown > in the Via header, but the fact that the PBX is responding makes me think > that isn't the culprit. > > Any thoughts? > > SIP Logger: > INVITE sip:+18165116504@12.4.240.200:5060;user=phone SIP/2.0 > v: SIP/2.0/UDP 10.77.27.103:20065 > ;branch=z9hG4bK0020C575A392E895C39051;oc-accept > Max-Forwards: 70 > t: > f: ;tag=10847511385389740959 > i: 117620342110831512016142@10.77.27.103 > CSeq: 1 INVITE > d: no-fork > Privacy: none > P-Asserted-Identity: > > Require: 100rel > Accept: application/sdp > k: histinfo,resource-priority > c: application/sdp > m: > Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,UPDATE > l: 228 > > v=0 > o=PVG 1452710812870 1452710812870 IN IP4 10.77.160.55 > s=- > c=IN IP4 10.77.160.55 > t=0 0 > m=audio 37700 RTP/AVP 0 101 > b=AS:80 > b=RR:0 > b=RS:0 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > a=maxptime:20 > > <--- Transmitting SIP response (495 bytes) to UDP:10.77.27.103:20065 ---> > SIP/2.0 421 Extension Required > Via: SIP/2.0/UDP 10.77.27.103:20065 > ;rport=20065;received=10.77.27.103;branch=z9hG4bK0020C575A392E895C39051;oc-accept > Call-ID: 117620342110831512016142@10.77.27.103 > From: ;user=phone>;tag=10847511385389740959 > To: ;user=phone>;tag=z9hG4bK0020C575A392E895C39051 > CSeq: 1 INVITE > Require: 100rel > Supported: 100rel, timer, replaces, norefersub > Server: Asterisk PBX 13.3.0-rc1 > Content-Length: 0 > > PJSIP is rejecting the inbound INVITE request as 100rel is required, but is not in the Supported header of the inbound SIP INVITE request. I would suspect that the UAC is doing things incorrectly by placing 100rel in the Require but not in the list of option tags in the Supported header. -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PJSIP Returning 421 Extension Required
I am turning up a PJSIP Endpoint and am having problems when they send an INVITE to my server. Asterisk is returning a 421 Extenstion Required. Since "extension" means different things in the SIP stack versus Asterisk, I don't know what it is complaining about. I have attached the trace below. Nothing else shows up with core verbose or core debug enabled, so I am assuming it has to be dying at the PJSIP module. The INVITE does come from an abnormal UDP Port, which is also shown in the Via header, but the fact that the PBX is responding makes me think that isn't the culprit. Any thoughts? SIP Logger: INVITE sip:+18165116504@12.4.240.200:5060;user=phone SIP/2.0 v: SIP/2.0/UDP 10.77.27.103:20065 ;branch=z9hG4bK0020C575A392E895C39051;oc-accept Max-Forwards: 70 t: f: ;tag=10847511385389740959 i: 117620342110831512016142@10.77.27.103 CSeq: 1 INVITE d: no-fork Privacy: none P-Asserted-Identity: Require: 100rel Accept: application/sdp k: histinfo,resource-priority c: application/sdp m: Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,UPDATE l: 228 v=0 o=PVG 1452710812870 1452710812870 IN IP4 10.77.160.55 s=- c=IN IP4 10.77.160.55 t=0 0 m=audio 37700 RTP/AVP 0 101 b=AS:80 b=RR:0 b=RS:0 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=maxptime:20 <--- Transmitting SIP response (495 bytes) to UDP:10.77.27.103:20065 ---> SIP/2.0 421 Extension Required Via: SIP/2.0/UDP 10.77.27.103:20065 ;rport=20065;received=10.77.27.103;branch=z9hG4bK0020C575A392E895C39051;oc-accept Call-ID: 117620342110831512016142@10.77.27.103 From: ;tag=10847511385389740959 To: ;tag=z9hG4bK0020C575A392E895C39051 CSeq: 1 INVITE Require: 100rel Supported: 100rel, timer, replaces, norefersub Server: Asterisk PBX 13.3.0-rc1 Content-Length: 0 PJSIP Endpoint: zeus*CLI> pjsip show endpoint erc905 Endpoint: I/OAuth: Aor: Contact: Transport: Identify: Match: Channel: Exten: CLCID: = Endpoint: erc905 Invalid 0 of inf Aor: erc905 0 Contact: erc905/sip:10.77.27.103:5060 Avail 32.887 Transport: ngvn udp 0 40 12.4.240.200:5060 Identify: erc905_1/erc905 Match: 10.77.27.103/32 ParameterName : ParameterValue 100rel: required accountcode : aggregate_mwi : true allow : (ulaw) allow_subscribe : true allow_transfer: true aors : erc905 auth : call_group: callerid : callerid_privacy : allowed_not_screened callerid_tag : connected_line_method : invite context : from_pstn cos_audio : 0 cos_video : 0 device_state_busy_at : 0 direct_media : true direct_media_glare_mitigation : none direct_media_method : invite disable_direct_media_on_nat : false dtls_ca_file : dtls_ca_path : dtls_cert_file: dtls_cipher : dtls_fingerprint : SHA-256 dtls_private_key : dtls_rekey: 0 dtls_setup: active dtls_verify : No dtmf_mode : rfc4733 fax_detect: false force_avp : false force_rport : true from_domain : from_user : ice_support : false identify_by : username inband_progress : false language : mailboxes : media_address : media_encryption : none media_encryption_optimistic : false media_use_received_transport : false message_context : moh_suggest : default mwi_from_user : named_call_group : named_pickup_group: one_touch_recording : false outbound_auth : outbound_proxy: pickup_group : record_off_feature: automixmon record_on_feature : automixmon rewrite_contact : false rtp_engine: asterisk rtp_ipv6 : false rtp_symmetric : false sdp_owner : - sdp_session : Asterisk send_diversion: true send_pai : true send_rpid : false set_var
