[asterisk-bugs] [Asterisk 0015890]: 1.6.1.5 - "Ghost" channels
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Mar 30 04:39:19 CDT 2010
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=15890
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Reported By: simonoch
Assigned To:
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Project: Asterisk
Issue ID: 15890
Category: Channels/General
Reproducibility: always
Severity: minor
Priority: normal
Status: acknowledged
Asterisk Version: 1.6.1.5
JIRA: SWP-216
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
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Date Submitted: 2009-09-14 07:41 CDT
Last Modified: 2010-03-30 04:39 CDT
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Summary: 1.6.1.5 - "Ghost" channels
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 at 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 at 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
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Relationships ID Summary
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related to 0015945 [patch] sip session timer: Does not wor...
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(0119997) optisistem (reporter) - 2010-03-30 04:39
https://issues.asterisk.org/view.php?id=15890#c119997
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Hi
I have de same problem with:
- Debian Lenny: Linux asterisk1 2.6.26-2-amd64
https://issues.asterisk.org/view.php?id=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 at 172.16.80.96
asterisk1*CLI>
* SIP Call
Curr. trans. direction: Incoming
Call-ID: 62f2d22a3845b1c61c0cf29f2b730658 at 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 at 172.16.86.2:18934
Need Destroy: 0
Last Message: Rx: INVITE
Promiscuous Redir: No
Route: sip:6102 at 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)
Issue History
Date Modified Username Field Change
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2010-03-30 04:39 optisistem Note Added: 0119997
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