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Summary:ASTERISK-02822: have sip not consider port when matching a peer
Reporter:connor (connor)Labels:
Date Opened:2004-11-15 18:09:24.000-0600Date Closed:2011-06-07 14:05:20
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/Interoperability
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:When using a Cisco 26xx gateway, The Cisco passes asterisk it's IP and PORT as a username, or asterisk is interperiting it as the username, thus not matching a peer

Please note these 2 lines in the sip debug below.

Found no matching peer or user for '1.2.3.4:55909'
Looking for 8659342100 in inbound



****** ADDITIONAL INFORMATION ******

Sip read:
INVITE sip:8659342100@198.144.169.28:5060 SIP/2.0
Via: SIP/2.0/UDP  1.2.3.4:5060
From: <sip:8655391933@1.2.3.4>;tag=917D46E8-20A7
To: <sip:8659342100@198.144.169.28>
Date: Mon, 29 Mar 1993 06:01:46 GMT
Call-ID: B552C56A-2B2111CC-951FAF27-2802E4A2@1.2.3.4
Supported: timer,100rel
Min-SE:  1800
Cisco-Guid: 3041900706-723587532-2501685031-671278242
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
CSeq: 101 INVITE
Max-Forwards: 6
Remote-Party-ID: <sip:8655391933@1.2.3.4>;party=calling;screen=yes;privacy=off
Timestamp: 733384906
Contact: <sip:8655391933@1.2.3.4:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 509

v=0
o=CiscoSystemsSIP-GW-UserAgent 8688 8249 IN IP4 1.2.3.4
s=SIP Call
c=IN IP4 1.2.3.4
t=0 0
m=audio 16964 RTP/AVP 0 100 121 101
c=IN IP4 1.2.3.4
a=rtpmap:0 PCMU/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194,200-202
a=rtpmap:121 frf-dialed-digit/8000
a=fmtp:121 0-15
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 192-194,200-202
a=X-cap: 2 image udptl t38

20 headers, 20 lines
Using latest request as basis request
Sending to 1.2.3.4 : 5060 (non-NAT)
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 121
Found RTP audio format 101
Peer audio RTP is at port 1.2.3.4:16964
Found description format PCMU
Found description format X-NSE
Found description format frf-dialed-digit
Found description format telephone-event
Capabilities: us - 0x105(G723|ULAW|G729A), peer - audio=0x4(ULAW)/video=0x0(EMPTY), combined - 0x4(ULAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x5(G723|ULAW), combined - 0x1(G723)
Found no matching peer or user for '1.2.3.4:55909'
Looking for 8659342100 in inbound
list_route: hop: <sip:8655391933@1.2.3.4:5060>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP  1.2.3.4:5060
From: <sip:8655391933@1.2.3.4>;tag=917D46E8-20A7
To: <sip:8659342100@198.144.169.28>;tag=as515e5bbc
Call-ID: B552C56A-2B2111CC-951FAF27-2802E4A2@1.2.3.4
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8659342100@198.144.169.28>
Content-Length: 0
Comments:By: Brian West (bkw918) 2004-11-15 18:10:52.000-0600

just have to make it do one last ditch attempt to match on ip without the port... maybe its just a bug.

bkw
By: Mark Spencer (markster) 2004-11-15 20:25:42.000-0600

This is what insecure=yes is for and/or insecure=very