|Summary:||ASTERISK-14619: Sip reload fails to erase old peers|
|Reporter:||Private Name (falves11)||Labels:|
|Date Opened:||2009-08-08 08:27:40||Date Closed:||2011-06-07 14:00:53|
|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 "g7188.8.131.52.210" does not exists at that moment.
****** ADDITIONAL INFORMATION ******
<--- SIP read from UDP:184.108.40.206:5060 --->
INVITE sip:firstname.lastname@example.org:5060 SIP/2.0
Via: SIP/2.0/UDP 220.127.116.11;branch=z9hG4bK68f.5943dcb1.0
Via: SIP/2.0/UDP 18.104.22.168;received=22.214.171.124;rport=5060;branch=z9hG4bKQ69BFmNgvU1SF
From: "8667065681" <sip:email@example.com>;tag=8p3F0c31gQ85c
CSeq: 118750325 INVITE
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
Remote-Party-ID: "8667065681" <sip:firstname.lastname@example.org>;party=calling;screen=yes;privacy=off
o=mscasheq03 6821918335058750602 4875283860960449008 IN IP4 126.96.36.199
c=IN IP4 188.8.131.52
m=audio 54254 RTP/AVP 0
--- (19 headers 6 lines) ---
Sending to 184.108.40.206 : 5060 (NAT)
Using INVITE request as basis request - 5f2fdf1c-fec0-122c-0b93-00237de89d1c
Found peer 'g7220.127.116.11.210' for '8667065681' from 18.104.22.168:5060
Found RTP audio format 0
Peer audio RTP is at port 22.214.171.124:54254
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: chan_sip.c:8274 process_sdp: No compatible codecs, not accepting this offer!
<--- Reliably Transmitting (NAT) to 126.96.36.199:5060 --->
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.