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Summary:ASTERISK-14619: Sip reload fails to erase old peers
Reporter:Private Name (falves11)Labels:
Date Opened:2009-08-08 08:27:40Date Closed:2011-06-07 14:00:53
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/CodecHandling
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:I noticed that my Asterisk was rejecting calls, but based on peer definition that no longer existed. There was a peer of that name, with only one codec, and Asterisk was finding it and comparing it to the new call, wrongly, because that peer was not defined after the last sip reload, and furthermore, it does not appear when I do a "sip show peers". This fault will lead to unauthorized calls.
Please notice in the dialog below that peer "g723.208.78.161.210" does not exists at that moment.

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

<--- SIP read from UDP:208.78.161.210:5060 --->
INVITE sip:12818885865@208.78.161.208:5060 SIP/2.0
Record-Route: <sip:208.78.161.210;lr=on;did=92b.141fe2c1>
Via: SIP/2.0/UDP 208.78.161.210;branch=z9hG4bK68f.5943dcb1.0
Via: SIP/2.0/UDP 208.76.155.250;received=208.76.155.250;rport=5060;branch=z9hG4bKQ69BFmNgvU1SF
Max-Forwards: 67
From: "8667065681" <sip:8667065681@208.76.155.250>;tag=8p3F0c31gQ85c
To: <sip:12818885865@208.78.161.210:5060>
Call-ID: 5f2fdf1c-fec0-122c-0b93-00237de89d1c
CSeq: 118750325 INVITE
Contact: <sip:12818885865@208.76.155.250:5060>
User-Agent: YHT Class4
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
Supported: timer, precondition, path, replaces
Allow-Events: talk, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 150
Remote-Party-ID: "8667065681" <sip:8667065681@208.76.155.250>;party=calling;screen=yes;privacy=off
X-Original-IP: 208.76.155.250

v=0
o=mscasheq03 6821918335058750602 4875283860960449008 IN IP4 208.76.155.250
s=sip call
c=IN IP4 208.76.155.250
t=0 0
m=audio 54254 RTP/AVP 0

<------------->
--- (19 headers 6 lines) ---
Sending to 208.78.161.210 : 5060 (NAT)                                                                                      
Using INVITE request as basis request - 5f2fdf1c-fec0-122c-0b93-00237de89d1c
Found peer 'g723.208.78.161.210' for '8667065681' from 208.78.161.210:5060                                                  
Found RTP audio format 0                                                                                                    
Peer audio RTP is at port 208.76.155.250:54254
Capabilities: us - 0x1 (g723), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)                  
[Aug  8 09:15:23] NOTICE[70924]: chan_sip.c:8274 process_sdp: No compatible codecs, not accepting this offer!              
vswitchabe*CLI>                                                                                                            
<--- Reliably Transmitting (NAT) to 208.78.161.210:5060 --->
SIP/2.0 488 Not acceptable here
Comments:By: dant (dant) 2009-08-09 14:49:17

Are you using realtime?
By: Private Name (falves11) 2009-08-09 14:50:45

nope. regular sip.conf.
By: Leif Madsen (lmadsen) 2009-09-17 15:37:53

This revision is pretty old now. Can you try it with something more recent?
By: Leif Madsen (lmadsen) 2009-10-26 09:40:20

Closed due to lack of feedback.