Hello,
I’m having an issue where I can’t make outbound SIP calls (getting 401 Unauthorized). The same phone can make internal calls with no issues and it can receive external calls. It’s only outbound dialing and I’m at a loss as to what the issue could be. A debug from a test call is below. I would be grateful for any help you could offer! Thank you!!
Jason
<— SIP read from UDP://173.15.212.49:1046 —>
INVITE sip:918772504705@72.233.54.194:5060 SIP/2.0
Via: SIP/2.0/UDP 173.15.212.49:1046;branch=z9hG4bK6c0a6d2f2a5e5301f.ef6a39172a45a7c0b
Max-Forwards: 70
From: “Jason Harper” sip:5110@72.233.54.194:5060;tag=dfc94a4b09
To: “918772504705” sip:918772504705@72.233.54.194:5060
Call-ID: 185231504d865062
CSeq: 4733 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Contact: “Jason Harper” sip:5110@173.15.212.49:1046;transport=udp;+sip.instance="urn:uuid:00000000-0000-1000-8000-00085D1ABA76"
Supported: gruu, timer, 100rel, replaces
User-Agent: Aastra 55i/2.4.1.37
Content-Type: application/sdp
Content-Length: 592
v=0
o=MxSIP 0 0 IN IP4 10.0.2.61
s=SIP Call
c=IN IP4 173.15.212.49
t=0 0
m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:106 BV16/8000
a=rtpmap:107 BV32/16000
a=rtpmap:113 L16/16000
a=rtpmap:110 PCMU/16000
a=rtpmap:111 PCMA/16000
a=rtpmap:112 L16/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:115 G726-32/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:on - - - -
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
— (14 headers 25 lines) —
== Using SIP RTP CoS mark 5
Sending to 173.15.212.49 : 1046 (no NAT)
Using INVITE request as basis request - 185231504d865062
Found user ‘5110’ for '5110’
Astro*CLI>
<— Reliably Transmitting (NAT) to 173.15.212.49:1046 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 173.15.212.49:1046;branch=z9hG4bK6c0a6d2f2a5e5301f.ef6a39172a45a7c0b;received=173.15.212.49
From: “Jason Harper” sip:5110@72.233.54.194:5060;tag=dfc94a4b09
To: “918772504705” sip:918772504705@72.233.54.194:5060;tag=as174ad2ee
Call-ID: 185231504d865062
CSeq: 4733 INVITE
User-Agent: Asterisk PBX 1.6.0.9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="210f952c"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘185231504d865062’ in 32000 ms (Method: INVITE)
Astro*CLI>
<— SIP read from UDP://173.15.212.49:1046 —>
ACK sip:918772504705@72.233.54.194:5060 SIP/2.0
Via: SIP/2.0/UDP 173.15.212.49:1046;branch=z9hG4bK6c0a6d2f2a5e5301f.ef6a39172a45a7c0b
Max-Forwards: 70
From: “Jason Harper” sip:5110@72.233.54.194:5060;tag=dfc94a4b09
To: “918772504705” sip:918772504705@72.233.54.194:5060;tag=as174ad2ee
Call-ID: 185231504d865062
CSeq: 4733 ACK
User-Agent: Aastra 55i/2.4.1.37
Content-Length: 0