PJSIP Returning 421 Extension Required

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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: ;tag0010847511385389740959
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 065;received.77.27.103;branch=z9hG4bK0020C575A392E895C39051;oc-accept Call-ID: 117620342110831512016142@10.77.27.103
From: ;tag0010847511385389740959
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 :
srtp_tag_32 : false sub_min_expiry : 0
t38_udptl : false t38_udptl_ec : none t38_udptl_ipv6 : false t38_udptl_maxdatagram : 0
t38_udptl_nat : false timers : yes timers_min_se : 90
timers_sess_expires : 1800
tone_zone :
tos_audio : 0
tos_video : 0
transport : ngvn trust_id_inbound : false trust_id_outbound : false use_avpf : false use_ptime : false user_eq_phone : false

One thought on - PJSIP Returning 421 Extension Required

  • 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.