Trey Hilyard
2016-Jan-13 18:58 UTC
[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 at 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: <sip:+18165116504 at 12.4.240.200;user=phone>
f: <sip:+18165116504 at
10.77.27.103;user=phone>;tag=000010847511385389740959
i: 117620342110831512016142 at 10.77.27.103
CSeq: 1 INVITE
d: no-fork
Privacy: none
P-Asserted-Identity: <sip:+18165116504;oli=62;rn=+1229218 at
10.77.27.103:20065
;user=phone>
Require: 100rel
Accept: application/sdp
k: histinfo,resource-priority
c: application/sdp
m: <sip:10.77.27.103:20065>
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 at 10.77.27.103
From: <sip:+18165116504 at 10.77.27.103
;user=phone>;tag=000010847511385389740959
To: <sip:+18165116504 at 12.4.240.200
;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 Endpoint:
zeus*CLI> pjsip show endpoint erc905
Endpoint: <Endpoint/CID.....................................>
<State.....> <Channels.>
I/OAuth:
<AuthId/UserName...........................................................>
Aor: <Aor............................................>
<MaxContact>
Contact: <Aor/ContactUri...............................>
<Status....> <RTT(ms)..>
Transport: <TransportId........> <Type> <cos>
<tos>
<BindAddress..................>
Identify:
<Identify/Endpoint.........................................................>
Match: <ip/cidr.........................>
Channel: <ChannelId......................................>
<State.....> <Time(sec)>
Exten: <DialedExten...........> CLCID:
<ConnectedLineCID.......>
========================================================================================
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 : <unknown>
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
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Matthew Jordan
2016-Jan-18 18:52 UTC
[asterisk-users] PJSIP Returning 421 Extension Required
On Wed, Jan 13, 2016 at 12:58 PM, Trey Hilyard <kctrey at gmail.com> 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 at 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: <sip:+18165116504 at 12.4.240.200;user=phone> > f: <sip:+18165116504 at 10.77.27.103;user=phone>;tag=000010847511385389740959 > i: 117620342110831512016142 at 10.77.27.103 > CSeq: 1 INVITE > d: no-fork > Privacy: none > P-Asserted-Identity: > <sip:+18165116504;oli=62;rn=+1229218 at 10.77.27.103:20065;user=phone> > Require: 100rel > Accept: application/sdp > k: histinfo,resource-priority > c: application/sdp > m: <sip:10.77.27.103:20065> > 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 at 10.77.27.103 > From: <sip:+18165116504 at 10.77.27.103 > ;user=phone>;tag=000010847511385389740959 > To: <sip:+18165116504 at 12.4.240.200 > ;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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160118/625b2edb/attachment.html>