I am having a really bad day trying to get incoming calls to work
on Asterisk 13 with PJSIP. We just migrated from Asterisk 1.8 where
everything was working but there seems that something got lost in
translation. No matter what I try I always get a 401 Unauthorized
message when receiving a call from the PSTN provider. I can make calls
and the registration is working. I have tried to set the identify to an
endpoint that does not have an auth defined. Anyone using Alestra SIP
trunks in Mexico?
Here is what I get on the cli:
<--- Received SIP request (1092 bytes) from UDP:200.94.59.150:5060 --->
INVITE sip:[email protected]:5060;line=qooanvj SIP/2.0
Via: SIP/2.0/UDP 200.94.59.150:5060;branch=z9hG4bKnvnkof007gngrp80d2g1.1
From:
<sip:[email protected];user=phone>;tag=866455524-1512253376938-
To: "MEXICO USERNAME"<sip:[email protected];line=qooanvj>
Call-ID: [email protected]
CSeq: 212444374 INVITE
Contact: <sip:[email protected]:5060;transport=udp>
Supported: 100rel,timer
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Recv-Info: x-broadworks-client-session-info
Accept: application/media_control+xml,application/sdp,multipart/mixed
Min-SE: 90
Session-Expires: 900;refresher=uac
Max-Forwards: 9
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 287
v=0
o=BroadWorks 26026640 1 IN IP4 200.94.59.152
s=-
c=IN IP4 200.94.59.152
t=0 0
m=audio 5470 RTP/AVP 18 0 8 100
a=rtpmap:18 G729/8000
a=fmtp:18 annexb:no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=sendrecv
a=maxptime:40
<--- Transmitting SIP response (588 bytes) to UDP:200.94.59.150:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
200.94.59.150:5060;received=200.94.59.150;branch=z9hG4bKnvnkof007gngrp80d2g1.1
Call-ID: [email protected]
From:
<sip:[email protected];user=phone>;tag=866455524-1512253376938-
To: "MEXICO USERNAME"
<sip:[email protected];line=qooanvj>;tag=z9hG4bKnvnkof007gngrp80d2g1.1
CSeq: 212444374 INVITE
WWW-Authenticate: Digest
realm="asterisk",nonce="1512253376/546618e0645f233990bd70d97691ddba",opaque="3b5f610b33037ba2",algorithm=md5,qop="auth"
Server: Asterisk PBX 13.18.3
Content-Length: 0
<--- Received SIP request (434 bytes) from UDP:200.94.59.150:5060 --->
ACK sip:[email protected]:5060;line=qooanvj SIP/2.0
Via: SIP/2.0/UDP 200.94.59.150:5060;branch=z9hG4bKnvnkof007gngrp80d2g1.1
CSeq: 212444374 ACK
From:
<sip:[email protected];user=phone>;tag=866455524-1512253376938-
To: "MEXICO
USERNAME"<sip:[email protected];line=qooanvj>;tag=z9hG4bKnvnkof007gngrp80d2g1.1
Call-ID: [email protected]
Max-Forwards: 9
Content-Length: 0
My identify is:
=============================================
endpoint : Alestra
match : 200.94.59.150/255.255.255.255
match_header :
srv_lookups : true
It does not matter if I use the original endpoint or an endpoint with no
auth. Asterisk will still reject the call. Any tips? How can I make
sure that the identify is being used?
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
dCAP #1349
+52-(55)8116-9161
--
_____________________________________________________________________
-- 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
