Sip.conf "register=>..." :port is not in REGISTER To:

Asterisk Asterisk Support
1

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

2

It is a strange registrar that expects a port number in an address of record. In any case, I would expect the registration to fail if the registrar required a port number.

3

I’m a bit confused here. I thought that the “:port” was an acceptable portion of a URI specification. What can we do when some of our registrars expect registrations via port 5260? Our previous simulator supported that fact by providing suitable registrations such as the example below:

REGISTER sip:152.148.200.152:5260 SIP/2.0
From: <sip:25001@152.148.200.152:5260>;tag=009059FC-5E72-14E7-BEE5-E00FA8C0AA77-1
To: <sip:25001@152.148.200.15:5260>
Call-ID: 009059B6-5E72-14E7-BEE5-E00FA8C0AA77-1@192.168.15.224
CSeq: 1 REGISTER
Content-Length: 0
Via: SIP/2.0/UDP 192.168.15.224:5069;branch=z9hG4bK00905A1A-5E72-14E7-BEE5-E00FA8C0AA77-1
User-Agent: Epi(SIP)phone 8.1.000.44 ray.smith 12256
Contact: <sip:25001@192.168.15.224:5069>
Expires: 1800
4

The registrar has accepted the URI without the port.

5

I appears to me that the registration does accept the port(5260)l. The subsequent INVITE, shown below, specifies the “To:” with a port of 5260:
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

This is rejected (404 Not found) because the REGISTER lacks the port specification in the To: header:
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

The sip.conf includes the port:
register=>25001:password@152.148.200.152:5260/25001

The end result is we are trying to simulate phones registering on a server that the required port is NOT 5060 but 5260 instead. The phones are looking like they are registering, but when we try to call from one phone to another phone, it does not work (404 not found). I think this is because the To: line in the registration does not include the port 5260. If I am wrong on my thought process, please let me know as well.

I hope you can help us.
Thanks