Question

sip show channels - seems to contain stale info...

Asked by: shaunwingin

When I type show channels there are 0 channels in use yet sip show channels returns the records below. Kindly explain. How do I clear them without a restart of Asterisk? Are they a problem?

Peer             User/ANR    Call ID      Seq (Tx/Rx)  Format           Hold     Last Message
41.221.0.33      074657233   6f637f7e42e  00103/00000  0x0 (nothing)    No       Tx: ACK
196.35.3.82      2782707892  2facd6787ca  00103/00134  0x0 (nothing)    No  (d)  Rx: BYE
196.35.3.82      2782457575  56558878018  00104/00134  0x0 (nothing)    No       Rx: BYE
41.221.0.33      0728730709  21e462e7585  00105/00000  0x0 (nothing)    No       Tx: ACK
196.35.3.82      2783479607  29fe287c7d5  00104/00134  0x0 (nothing)    No       Rx: BYE
196.35.3.82      2783541655  30e954061c7  00104/00134  0x0 (nothing)    No       Rx: BYE
196.35.3.82      2783316728  76877c7c129  00104/00134  0x0 (nothing)    No       Rx: BYE
41.221.0.33      0785107981  387b0ca87be  00105/00000  0x0 (nothing)    No       Tx: ACK
41.221.0.33      0767757333  36649150611  00105/00000  0x0 (nothing)    No       Tx: ACK
41.221.0.33      0836552418  064ae207360  00105/00000  0x0 (nothing)    No       Tx: ACK
196.35.3.82      2778560185  1f350c1f57e  00104/00134  0x0 (nothing)    No       Rx: BYE
41.221.0.33      071864      66abe7593f9  00103/00000  0x0 (nothing)    No       Tx: ACK
41.221.0.33      0852        6a57ddfd689  00103/00000  0x0 (nothing)    No       Tx: ACK
196.15.134.132   RoB6768     78b3fe947b4  00103/00103  0x0 (nothing)    No       Tx: ACK
41.221.0.33      0866513222  67146d952e5  00104/00102  0x0 (nothing)    No       Rx: INVITE
41.221.0.33      0866513201  5e258c9564b  00104/00102  0x0 (nothing)    No       Rx: INVITE
41.221.0.33      0866513201  07159b4230e  00104/00102  0x0 (nothing)    No       Rx: INVITE
41.221.0.33      0866513201  522aff2233a  00104/00102  0x0 (nothing)    No       Rx: INVITE
196.35.3.82      2782312490  6489ff7165a  00104/00134  0x0 (nothing)    No       Rx: BYE
196.35.3.82      2784692008  10a36d6c511  00104/00134  0x0 (nothing)    No       Rx: BYE
41.221.0.33      08429569    161eefda6a0  00103/00000  0x0 (nothing)    No       Tx: ACK
196.15.134.132   RoB6768     7416f9ef7cc  00103/00103  0x0 (nothing)    No       Tx: ACK
41.221.0.33      0791393809  2bf88aef709  00105/00000  0x0 (nothing)    No       Tx: ACK
196.35.3.82      2782371521  120e3bab5e7  00104/00134  0x0 (nothing)    No       Rx: BYE
196.15.134.132   RoB6768     5640b77f014  00103/00103  0x0 (nothing)    No       Tx: ACK
196.25.8.98      07457       6d59f0433f0  00103/00000  0x0 (nothing)    No       Tx: INVITE
196.35.3.82      2782440128  3002c028281  00104/00134  0x0 (nothing)    No       Rx: BYE
196.25.8.98      0842350803  5ea3c911270  00103/00000  0x0 (nothing)    No       Tx: INVITE
196.25.8.98      0737        5acd9224461  00103/00000  0x0 (nothing)    No       Tx: INVITE
196.25.8.98      07637140    6546c44401b  00103/00000  0x0 (nothing)    No       Tx: INVITE
196.35.3.82      2783309869  26e1e9016ed  00104/00134  0x0 (nothing)    No       Rx: BYE
196.35.3.82      2782448078  6bedf8502e2  00104/00134  0x0 (nothing)    No       Rx: BYE
196.25.8.98      0782        65b071b674c  00103/00000  0x0 (nothing)    No       Tx: INVITE
196.25.8.98      0787786     01a555f93f5  00103/00000  0x0 (nothing)    No       Tx: INVITE
196.25.8.98      078563      4495116b59d  00103/00000  0x0 (nothing)    No       Tx: INVITE
196.25.8.98      08353404    0e10482111c  00103/00000  0x0 (nothing)    No       Tx: INVITE
196.15.134.132   RoB6768     72e08f0e14b  00103/00103  0x0 (nothing)    No       Tx: ACK
196.25.8.98      0838562     04d8c06e36e  00103/00000  0x0 (nothing)    No       Tx: INVITE
196.25.8.98      0736        72658f67198  00103/00000  0x0 (nothing)    No       Tx: INVITE
196.35.3.82      2776907792  0fd6a68d08b  00104/00134  0x0 (nothing)    No       Rx: BYE
196.25.8.98      07666       2d7917d2087  00103/00000  0x0 (nothing)    No       Tx: INVITE
196.25.8.98      0723        4023fa87665  00103/00000  0x0 (nothing)    No       Tx: INVITE
196.25.8.98      078204      443bb1751db  00103/00000  0x0 (nothing)    No       Tx: INVITE
196.25.8.98      0827        65e5ad577b0  00103/00000  0x0 (nothing)    No       Tx: INVITE
196.25.8.98      0833        0823279e2a9  00103/00000  0x0 (nothing)    No       Tx: INVITE
196.25.8.98      0721        250f4c0813e  00103/00000  0x0 (nothing)    No       Tx: INVITE
196.25.8.98      07208451    6298807826e  00103/00000  0x0 (nothing)    No       Tx: INVITE
196.25.8.98      08207       7a3e8d3e7e8  00103/00000  0x0 (nothing)    No       Tx: INVITE
196.25.8.98      0827        224004f57cd  00103/00000  0x0 (nothing)    No       Tx: INVITE
196.25.8.98      0731231     53a2cbb3358  00103/00000  0x0 (nothing)    No       Tx: INVITE
196.25.8.98      07914       4cf98b510b6  00103/00000  0x0 (nothing)    No       Tx: INVITE
196.25.8.98      0861500100  056240e4183  00104/00102  0x0 (nothing)    No       Rx: INVITE
196.25.8.98      0114864024  5d720198301  00104/00102  0x0 (nothing)    No       Rx: INVITE
196.25.8.98      0112427243  60d694a0128  00104/00102  0x0 (nothing)    No       Rx: INVITE
196.25.8.98      0821396519  4cffcce173f  00103/00000  0x0 (nothing)    No       Tx: INVITE

                                  
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Asked On
2008-12-07 at 04:34:58ID23963971
Tags

Digium

,

Asterisk

,

1.4.21.2

Topics

Voice Over IP

,

Asterisk Open Source Telephony

Participating Experts
3
Points
500
Comments
16

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Answers

 

by: feptiasPosted on 2008-12-08 at 01:28:41ID: 23119347

This is a guess, but I wonder if these CLI commands might be relevant:
sip history
sip history off

I have never used the commands and the wiki seems to have no info about them, but I would guess that they somehow cause Asterisk to retain information about old SIP calls. Perhaps you turned on "sip history" without realising what it would do. This would make sense because some of the calls in your list show that BYE was the last message, so it looks like Asterisk is aware that they had finished.

 

by: shaunwinginPosted on 2008-12-08 at 01:35:54ID: 23119376

I tried it but it doesn't help. Tx for the suggestion.

 

by: xuserx2000Posted on 2008-12-15 at 15:41:18ID: 23179282

Try SIP SHOW PEERS

What you are looking at is not stale... it's the channel id for each registration.


http://www.voip-info.org/wiki/view/asterisk+cli+command+sip+show+peers

 

by: xuserx2000Posted on 2008-12-15 at 15:43:41ID: 23179313

Also try:   sip show inuse

 

by: xuserx2000Posted on 2008-12-15 at 15:46:30ID: 23179349

 

by: shaunwinginPosted on 2008-12-16 at 13:38:55ID: 23188524

sip show inuse sometimes shows insue as 4/2 30 (30 is the max no. of calls) but what is the 4/2 for example please?

 

by: shaunwinginPosted on 2008-12-16 at 13:38:59ID: 23188525

sip show inuse sometimes shows insue as 4/2 30 (30 is the max no. of calls) but what is the 4/2 for example please?

 

by: feptiasPosted on 2008-12-17 at 01:59:44ID: 23191858

That is a good question. I just tried "sip show inuse" on my local asterisk box and, purely from experimentation, got these results:

1) SIP phone 4003 is calling 4004 (during ringing)
* Peer name           In use        Limit
4004                      1/1             4
4003                      1/0             4

2) After the call is answered (4003 is talking to 4004)
* Peer name           In use        Limit
4004                      1/0             4
4003                      1/0             4

I also tried googling for more information about "sip show inuse", but couldn't find anything meaningful.

By the way, I would hazard a guess that the problems you are seeing on your sip calls are due to incorrect routing of packets or firewall/NAT problems. It is just a guess, but your list for "sip show channels" has too many INVITE and BYE's shown in the "Last message" column. On my Asterisk servers most of the "Last message" entries are ACK's or OK's. If your routing or firewall policies are wrong, it is possible for one end point to send a SIP message, but the other end point to then not be able to respond to it - at least it tries to send a response, but the response is not received. Usually, when this is happening you will see the first end point re-send the request several more times. For INVITE's and BYE's there is always an ACK or an OK at the end of the dialogue so that is what you are most likely to see in the "Last message" column.

 

by: shaunwinginPosted on 2008-12-19 at 04:25:26ID: 23211773

Thanks this is most informative.

The iptables looks like this - does it look inorder?

-A RH-Firewall-1-INPUT -d 41.2.9.6 -m state --state NEW -m tcp -p tcp --dport 5060:5082 -j ACCEPT
-A RH-Firewall-1-INPUT -d 41.2.9.6 -m state --state NEW -m udp -p udp --dport 5060:5082 -j ACCEPT
#-A RH-Firewall-1-INPUT -d 41.2.9.6 -m state --state NEW -m tcp -p tcp --dport 8000:20000 -j ACCEPT
-A RH-Firewall-1-INPUT -d 41.2.9.6 -m state --state NEW -m udp -p udp --dport 8000:20000 -j ACCEPT
-A RH-Firewall-1-INPUT -d 41.2.9.6 -m state --state NEW -m udp -p udp --dport 4569 -j ACCEPT

 

by: feptiasPosted on 2008-12-19 at 04:43:15ID: 23211876

The rules look ok as far as I can tell. It may not just be firewall rules. It may depend on the IP addresses embedded within the SIP packets (The Contact header and Via's especially) and also on your network routing.

I think you would have to capture some SIP packets to confirm if the problem is like I suggested. Especially look for the tell-tale resending of the same request. I use the built-in packet logging mechanism in Asterisk. CLI commands: "sip debug" (v1.2) or "sip set debug" (v1.4). Type "help sip" for details of all the debug commands.

 

by: shaunwinginPosted on 2008-12-19 at 05:24:54ID: 23212092

Tx. The SIP packets to seem to indicate the problem as you suggest.
The ip address is a public ip - but ICMP ping is dissabled. The interface is attached directly to a router and is hosted at an isp.
How can I try and solve the problem please?

 

by: feptiasPosted on 2008-12-19 at 05:37:53ID: 23212157

A diagram would help.

Is NAT involved? If it is, then check the advice on my web page here:
http://www.smartvox.co.uk/astfaq_configbehindnat.htm

ICMP ping is very unlikely to be relevant to SIP.

 

by: shaunwinginPosted on 2008-12-19 at 05:56:19ID: 23212265

I like you web site tx for the link.
Nat's not involved.
VSP<-->Asterisk Server<-->Clients Asterisk Server
All 3 are not behind a NAT.
Any ideas please?
By the way when I try and send to an alternate VSP I get the error
frame.c:203 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end

I have confirmed that "Silence supresion" is off on their side. They say other * servers send to them fine. Their devices are Quintums.
Perhaps this can give a clue.
I don't get this error to any other VSP I send traffic too.

 

by: feptiasPosted on 2008-12-19 at 07:34:49ID: 23212916

Can you capture a SIP packet and post it back here as an attached snippet. I just need to see one example of a packet that is getting sent multiple times - probably an INVITE or a BYE. In particular, I want to see what is in these three headers (they contain IP addresses/port used for returning SIP messages or responses):
Via
Contact
Record-Route

 

by: shaunwinginPosted on 2009-02-11 at 00:39:26ID: 23609194

Feptias, Sorry for the delay - hope to get to do this soon....

 

by: ctarbetPosted on 2010-01-06 at 08:35:07ID: 26191809

20120131-EE-VQP-002

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