I need to simulate phones which register via port “5260”. A sip.conf section is below.
Note that in the “Registration from logging file”, the “To:” section (To: sip:25001@152.148.200.152>) does not contain the port specification “:5260” which is found in the sip.conf “register=>” (register=>25001:password@152.148.200.152:5260/25001).
The result is a failure in the call. The call log segment is below.
I thank you for you attention and advice.
Registration from logging file:
REGISTER sip:152.148.200.152:5260 SIP/2.0
Via: SIP/2.0/UDP 192.168.15.224:5060;branch=z9hG4bK7321d188
Max-Forwards: 70
From: sip:25001@152.148.200.152;tag=as627fee16
To: sip:25001@152.148.200.152
Call-ID: 56df2bf570f8c5df122662081ab67209@192.168.15.224
CSeq: 104 REGISTER
User-Agent: Asterisk PBX 11.12.1
Authorization: Digest username=“25001”, realm=“switch”, algorithm=MD5, uri=“sip:152.148.200.152:5260”, nonce=“0067D11C-6317-1531-93BE-98C89498AA77”, response=“3ab8612c4d8d39c6844679a204974d3f”, opaque=“737769746368”, qop=auth, cnonce=“363beaba”, nc=00000002
Expires: 120
Contact: sip:25001@192.168.15.224:5060
Content-Length: 0
—
[150417-154816.68062] VERBOSE[25716] chan_sip.c:
<— SIP read from UDP:152.148.200.152:5260 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.224:5060;branch=z9hG4bK7321d188;received=192.168.15.224
From: sip:25001@152.148.200.152;tag=as627fee16
To: sip:25001@152.148.200.152;tag=006D5628-5F07-1526-93BE-98C89498AA77-92420
Call-ID: 56df2bf570f8c5df122662081ab67209@192.168.15.224
CSeq: 104 REGISTER
Server: SIP Server, 2.1.007.01
Expires: 120
Contact: sip:25001@192.168.15.224:5060;expires=120
Content-Length: 0
sip.conf:
[general]
;
; ICM
;
register=>25001:password@152.148.200.152:5260/25001
register=>25002:password@152.148.200.152:5260/25002
;
; OXE
;
register=>46101:password@152.148.200.242:5060/46101
register=>46102:password@152.148.200.242:5060/46102
[authentication]
;
; ICM client sections
;
auth = 25001:password@inse.lucent.com
auth = 25002:password@inse.lucent.com
[25001]
dtmfmode => inband
relaxdtmf => yes
disallow=all
allow=G711
allow=ulaw
allow=alaw
[25002]
dtmfmode => inband
relaxdtmf => yes
disallow=all
allow=G711
allow=ulaw
allow=alaw
Call Segment:
INVITE sip:25001@152.148.200.152:5260 SIP/2.0
Via: SIP/2.0/UDP 192.168.15.224:5060;branch=z9hG4bK3e525815
Max-Forwards: 70
From: “PH_27001_27002_00001” sip:asterisk@192.168.15.224;tag=as4a4ab208
To: sip:25001@152.148.200.152:5260
Contact: sip:asterisk@192.168.15.224:5060
Call-ID: 4b4b1c5428ccd05641dd1d970ebac800@192.168.15.224:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.12.1
Date: Fri, 17 Apr 2015 15:43:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 285
v=0
o=root 513743554 513743554 IN IP4 192.168.15.224
s=Asterisk PBX 11.12.1
c=IN IP4 192.168.15.224
t=0 0
m=audio 12736 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
—
[150417-114320.51776] VERBOSE[17618] chan_sip.c:
<— SIP read from UDP:152.148.200.152:5260 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.15.224:5060;branch=z9hG4bK3e525815;received=192.168.15.224
From: “PH_27001_27002_00001” sip:asterisk@192.168.15.224;tag=as4a4ab208
To: sip:25001@152.148.200.152:5260;tag=006D5628-5F07-1526-93BE-98C89498AA77-90282
Call-ID: 4b4b1c5428ccd05641dd1d970ebac800@192.168.15.224:5060
CSeq: 102 INVITE
Content-Length: 0