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Summary:ASTERISK-14826: 1.6.1.5 - "Ghost" channels
Reporter:simonoch (simonoch)Labels:
Date Opened:2009-09-14 07:41:08Date Closed:2011-06-07 14:07:21
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:We seems to have ghost channels. We don't understand what's the problem.

# (extract) sip show channels
Peer             User/ANR    Call ID          Format           Hold     Last Message    Expiry
82.241.145.220   103         6cb5e7541370ff7  0x0 (nothing)    No       Init: INVITE                        
82.241.145.220   103         5f8a19a14b48234  0x0 (nothing)    No       Init: INVITE                          
82.241.145.220   103         659369d21a838aa  0x0 (nothing)    No       Init: INVITE              
82.241.145.220   103         434b1ec828bc2e9  0x100 (g729)     No       Tx: ACK                        
82.241.145.220   103         301947390a5111f  0x0 (nothing)    No       Init: INVITE                    
82.241.145.220   103         524b5b7f6289b7a  0x0 (nothing)    No       Init: INVITE                            
82.241.145.220   103         535d242452bdd48  0x0 (nothing)    No       Init: INVITE        

# sip show history 659369d21a838aa

 * SIP Call
1. ReliableXmit    timeout
2. ReliableXmit    timeout
3. ReliableXmit    timeout
4. ReliableXmit    timeout
5. ReliableXmit    timeout
6. ReliableXmit    timeout
7. ReliableXmit    timeout
8. ReliableXmit    timeout
9. ReliableXmit    timeout
10. ReliableXmit    timeout
11. ReliableXmit    timeout
12. ReliableXmit    timeout
13. ReliableXmit    timeout
14. ReliableXmit    timeout
15. ReliableXmit    timeout
16. ReliableXmit    timeout
17. ReliableXmit    timeout
(etc ... etc ...)

# sip show peer 103

* Name       : 103
 Secret       :
 MD5Secret    :
 Context      : from-internal
 Subscr.Cont. :
 Language     : fr
 AMA flags    : Unknown
 Transfer mode: open
 CallingPres  : Presentation Allowed, Not Screened
 Callgroup    :
 Pickupgroup  :
 Mailbox      : 103@default
 VM Extension : 123
 LastMsgsSent : 32767/65535
 Call limit   : 50
 Dynamic      : Yes
 Callerid     : "device" <103>
 MaxCallBR    : 384 kbps
 Expire       : 1229
 Insecure     : no
 Nat          : Always
 ACL          : Yes
 T38 pt UDPTL : No
 CanReinvite  : No
 PromiscRedir : No
 User=Phone   : No
 Video Support: No
 Text Support : No
 Ign SDP ver  : No
 Trust RPID   : No
 Send RPID    : No
 Subscriptions: Yes
 Overlap dial : Yes
 DTMFmode     : rfc2833
 Timer T1     : 500
 Timer B      : 32000
 ToHost       :
 Addr->IP     : XX.XX.XX.XX Port 7374
 Defaddr->IP  : 0.0.0.0 Port 5060
 Transport    : UDP
 Def. Username: 103
 SIP Options  : (none)
 Codecs       : 0x10a (gsm|alaw|g729)
 Codec Order  : (g729:20,gsm:20,alaw:20)
 Auto-Framing :  No
 100 on REG   : No
 Status       : OK (140 ms)
 Useragent    : eyeBeam release 1102u stamp 52344
 Reg. Contact : sip:103@XXX:7374;rinstance=65465ec13147a5e9
 Qualify Freq : 60000 ms
 Sess-Timers  : Accept
 Sess-Refresh : uas
 Sess-Expires : 1800 secs
 Min-Sess     : 90 secs

Does anybody knows what is it ?

Thank you,

Simon
Comments:By: Leif Madsen (lmadsen) 2009-09-18 09:24:01

Can you provide the relative sip debug, sip history, and console output (with debugging enabled) when one (or more) of these channels becomes a "ghost"? We'll need this information in order to determine what is happening.

If you can provide any information about possible scenarios when this happens, that would also be quite useful.
By: David Vossel (dvossel) 2009-09-18 11:40:00

I have seen this before with TCP/TLS. but not UDP, I had an issue for it, issue ASTERISK-13865.  It was resolved by file's patch for issue ASTERISK-14830.  Not sure if this is related or not since the transport for peer 103 appears to be UDP.  Thought it was worth noting though.
By: simonoch (simonoch) 2009-09-18 13:02:49

(sorry for my english)

Actually the plateform receive about 250 calls by a day, and the queue generate a huge log.

So, how to filter this log?

In this time, the plateform is closed and we have :

# sip show channels
Peer             User/ANR    Call ID          Format           Hold     Last Message    Expiry
XX.XX.XX.XX   103         247812e85e1a894  0x0 (nothing)    No       Init: INVITE              
XX.XX.XX.XX   103         48106940381414d  0x0 (nothing)    No       Init: INVITE              
XX.XX.XX.XX   103         6626309529c6fc6  0x0 (nothing)    No       Init: INVITE              
XX.XX.XX.XX   105         76b86216299928b  0x0 (nothing)    No       Init: INVITE              
4 active SIP dialogs

Simon


By: simonoch (simonoch) 2009-09-18 13:05:48

#sip show channel 247812e85e1a894

 * SIP Call
 Curr. trans. direction:  Outgoing
 Call-ID:                247812e85e1a8942323c94ba18143aaa@astserver
 Owner channel ID:      
 Our Codec Capability:   266
 Non-Codec Capability (DTMF):   1
 Their Codec Capability:   0
 Joint Codec Capability:   266
 Format:                 0x0 (nothing)
 T.38 support            No
 Video support           No
 MaxCallBR:              384 kbps
 Theoretical Address:    XX.XX.XX.XX:11198
 Received Address:       XX.XX.XX.XX:11198
 SIP Transfer mode:      open
 NAT Support:            Always
 Audio IP:               astserver (local)
 Our Tag:                as62ecb29f
 Their Tag:              
 SIP User agent:        
 Username:               103
 Peername:               103
 Original uri:           sip:103@XX.XX.XX.XX:11198;rinstance=c452a8553eb4835b
 Need Destroy:           No
 Last Message:           Init: INVITE
 Promiscuous Redir:      No
 Route:                  N/A
 DTMF Mode:              rfc2833
 SIP Options:            (none)
 Session-Timer:          Inactive


By: Leif Madsen (lmadsen) 2009-09-24 09:09:03

Well, now that I've marked 15945 as related here, I'm not so convinced as it appears you're NOT using SIP session-timers, where as the other issue is.

You could attempt the try the patch on the other issue since it is a 1 line fix and see if it helps, and then report back here, or there. Thanks!
By: John Todd (jtodd) 2009-10-05 11:37:25

any update on this?  We're not quite sure what the issue is here, but if you could tell us if you're using SIP session-timers, that would help.  If this is a real problem, it's probably a pretty serious one.
By: simonoch (simonoch) 2009-10-20 07:27:33

Hi,

The problem still occurs :


ip   105         75593b5b0a842d2  0x0 (nothing)    No       Init: INVITE              
ip   105         76772d1f2c2c82d  0x0 (nothing)    No       Init: INVITE              
ip   105         0bec90d27255061  0x0 (nothing)    No       Init: INVITE              
ip   105         5980c7643f174b9  0x0 (nothing)    No       Init: INVITE              
ip   105         5b043b687aea5b4  0x0 (nothing)    No       Init: INVITE              
ip   105         0949659249a3759  0x0 (nothing)    No       Init: INVITE              
ip   105         5935b18d40f7d20  0x100 (g729)     No       Tx: ACK                  


# sip show channel 75593b5b0a842d2

 * SIP Call
 Curr. trans. direction:  Outgoing
 Call-ID:                75593b5b0a842d28586a49d058c92023@server
 Owner channel ID:      
 Our Codec Capability:   266
 Non-Codec Capability (DTMF):   1
 Their Codec Capability:   0
 Joint Codec Capability:   266
 Format:                 0x0 (nothing)
 T.38 support            No
 Video support           No
 MaxCallBR:              384 kbps
 Theoretical Address:    ip:12856
 Received Address:       ip:12856
 SIP Transfer mode:      open
 NAT Support:            Always
 Audio IP:               server (local)
 Our Tag:                as0ddee624
 Their Tag:              
 SIP User agent:        
 Username:               105
 Peername:               105
 Original uri:           sip:105@ip:12856;rinstance=cdd1df5c913246e7
 Need Destroy:           No
 Last Message:           Init: INVITE
 Promiscuous Redir:      No
 Route:                  N/A
 DTMF Mode:              rfc2833
 SIP Options:            (none)
 Session-Timer:          Inactive


# sip show history

 * SIP Call
1. ReliableXmit    timeout
2. ReliableXmit    timeout
3. ReliableXmit    timeout
4. ReliableXmit    timeout
5. ReliableXmit    timeout
6. ReliableXmit    timeout
7. ReliableXmit    timeout
8. ReliableXmit    timeout
9. ReliableXmit    timeout
10. ReliableXmit    timeout
11. ReliableXmit    timeout
12. ReliableXmit    timeout
13. ReliableXmit    timeout
14. ReliableXmit    timeout
15. ReliableXmit    timeout
16. ReliableXmit    timeout
17. ReliableXmit    timeout
18. ReliableXmit    timeout
19. ReliableXmit    timeout
By: simonoch (simonoch) 2009-10-20 07:28:38

I think i'm not using SIP session-timers, actually I don't know what is it. I search and I tell you
By: Silvia Gallego Gonzalez (optisistem) 2010-03-30 04:39:19

Hi

I have de same problem with:
- Debian Lenny: Linux asterisk1 2.6.26-2-amd64 #1 SMP Thu Nov 5 02:23:12 UTC 2009 x86_64 GNU/Linux
- Asterisk: 1.4.24
- Useragent    : eyeBeam release 1100l stamp 46320

> sip show channels
172.16.86.3 6118 1139e7863f2 00102/00003 0x0 (nothing) No Rx: INVITE
172.16.86.13 6129 093bbc22033 00102/00003 0x0 (nothing) No Rx: INVITE
172.16.86.11 6130 78b480350fe 00102/00003 0x0 (nothing) No Rx: INVITE
172.16.86.2 6102 62f2d22a384 00102/00003 0x0 (nothing) No Rx: INVITE
172.16.86.13 6129 1e30cd62160 00102/00003 0x0 (nothing) No Rx: INVITE
172.16.86.13 6129 52bfc19113c 00102/00003 0x0 (nothing) No Rx: INVITE
172.16.86.11 6130 679519bd3bd 00102/00003 0x0 (nothing) No Rx: INVITE
172.16.86.4 6111 2914855570c 00102/00003 0x0 (nothing) No Rx: INVITE
172.16.86.7 6124 17e8843b0bc 00102/00003 0x0 (nothing) No Rx: INVITE

asterisk1*CLI> sip show channel 62f2d22a3845b1c61c0cf29f2b730658@172.16.80.96
asterisk1*CLI>
 * SIP Call
 Curr. trans. direction: Incoming
 Call-ID: 62f2d22a3845b1c61c0cf29f2b730658@172.16.80.96
 Owner channel ID: <none>
 Our Codec Capability: 1835272
 Non-Codec Capability (DTMF): 1
 Their Codec Capability: 8
 Joint Codec Capability: 8
 Format: 0x0 (nothing)
 MaxCallBR: 384 kbps
 Theoretical Address: 172.16.86.2:18934
 Received Address: 172.16.86.2:18934
 SIP Transfer mode: open
 NAT Support: RFC3581
 Audio IP: 172.16.80.96 (local)
 Our Tag: as791c5369
 Their Tag: 6c6dd529
 SIP User agent: eyeBeam release 1100l stamp 46320
 Username: 6102
 Peername: 6102
 Original uri: sip:6102@172.16.86.2:18934
 Need Destroy: 0
 Last Message: Rx: INVITE
 Promiscuous Redir: No
 Route: sip:6102@172.16.86.2:18934;rinstance=cdbe390f14aeb5e0
 DTMF Mode: rfc2833
 SIP Options: (none)

I think the problem could be related to the eyeBeam softphones.
Since I'd change the rtptimeout value en sip.conf and restart the asterisk, the problem disapears.

> sip.conf
[general]
rtptimeout=30


(sorry about my google-english)
By: simonoch (simonoch) 2010-03-30 08:19:04

Yes I think so, i put rtptimeout to 30 weeks ago, and changed other settings. Since these modifications, there is no problem.