Displaying 20 results from an estimated 10000 matches similar to: "Zombie SIP channels"
2007 Feb 27
1
Help understanding SIP SHOW CHANNELS
I have a high volume asterisk 1.40 installation and I ran a SIP SHOW
CHANNELS. (see partial output below). My questions are:
1. "wc-l" of the output shows 4000 lines. Does this mean 2000 active calls?
(2 channels per call)
2. The latter part of the output shows "unkn" for Form column. Why does it
not know the codec? Could it be UDPTL? Or are these calls messed up?
3.
2004 Jun 01
1
Stuck SIP channels? -> SIP show channels
Hello all
I've discovered that SIP channels sometimes get stuck in *.
I've read some posts from Fri 29 Aug 2003 which mentions this issue, but
there doesn't seem to be any final answers
I don't know if this is related to the 0001604 bug?
Below is a list from one of the incidents:
I know the (d) means that it is scheduled for destruction but the 10.1.1.45
channel hasn't
2007 Sep 06
1
Dead SIP channels
I am using a2billing as calling card platform with asterisk 1.2.17.
After running for several days, if I issue 'sip show channels' command, I got a lot of dead sip channels although 'show channels' command only show 5 channels. What cause these dead channels? How can I clean out these dead channels? Will they pose any problem to my * server if left alone? What does this (d) mean?
2007 Aug 23
1
channel not hungup (zombie?) so call limit not reset to zero
im having a strange problem related to call-limit for peers. well im not
sure if its related to call-limmit or not. Bottom line is:
I call a user A, from user B. user B hears silence, untill it goes to
voicemail. when user B hangsup. user B's call limit is reset to 0 but user
A's call limit is not reset.strange thing is user A's status on cli is shown
as NOANSWER, while user B did not
2006 Feb 07
1
orphaned sip channels channels?
My sip show channels shows some channels active that I can not make
sense out of, and they have been that way for days, so I am pretty sure
they are orphans.
Is there a way to show active CALLS (instead of channels) to try and
determine the source?
Does the output below provide any clues as to why these channels might
show active?
Anyone aware of related bugs?
The #'s indicate original
2003 Sep 25
4
SIP Problem
I am having a problem when a SIP registration fails. I get the following
messages in the log:
Sep 25 18:51:02 NOTICE[1125329600]: File chan_sip.c, Line 2874
(sip_reg_timeout): Registration for '<user>@fwd.pulver.com@65.39.205.114'
timed out, trying again
Sep 25 18:51:02 NOTICE[1125329600]: File chan_sip.c, Line 5119
(handle_request): Registration from
2007 May 09
10
SIP Problems continue...
SIP channel hang ups are progressively getting worse and I'm really
grasping at straws here trying to find out what the cause is. The
problem start, once a week or so the SIP phones couldn't communicate
with the server, though there was no error message on the server and
everything appeared fine on the server. It's now doing it multiple
times a day and I fear having to go back to our
2008 Feb 13
3
urgent-channels
Hi All
I am using asterisk 1.2.4
Please see the results when I execute Sip show channels
X
X
X
X
x
192.168.8.106(None) 04cddc1f5a0 00101/00000 unkn No
215.96.142.83 (None) caac0846-cf 00101/00000 unkn No
192.168.8.106(None) 94910146-46 00101/00000 unkn No
192.168.8.106(None) 793ed1eb0f2 00101/00000 unkn No
85.219.172.253 (None)
2008 Oct 14
1
SIP channels seem not to close after call is finished
Hello everyone,
I'm getting DIALSTATUS=CHANUNAVAIL when a call is trying to get one of my
queue interfaces, despite the fact it is free at that time, can you give
help?
1. I see many sip channels from that extension:
[root at mysweetpbx]# asterisk -rx "*sip show channels*" |grep 648
Peer User/ANR Call ID Seq (Tx/Rx)
Format Hold
2007 Nov 16
1
channels to destroy
Hello,
In a couple of Asterisks, after type "sip show channels" we have a lot
of these:
IP_PEER dst_number something 00102/00103 unkn No (d) Rx: BYE
IP_PEER dst_number2 something2 00102/00103 unkn No (d) Rx: BYE
We are using ASterisk 1.2.x
When I say "a lot" I mean more than 180, more than 230, etc.
Is it normal?
How we can remove it?
Thank you very much,
--
2006 Jun 22
3
Showing Current Calls
Can someone recommend the best way to view current calls in progress on the Asterisk console?
Neither the 'show channels' or 'sip show channels' commands are easy to read.
hestia*CLI> show channels
Channel Location State Application(Data)
SIP/2944093-f9e2 (None) Up Bridged Call(SIP/2944079-e7f2)
SIP/2944079-e7f2
2004 Jul 12
1
SIP client to IAXTel 800/888/877/866 dialing thru Asterisk
Through my Asterisk server, I am trying to use IAXTel to dial 800-type
numbers (yes, I know I can do the same thing with FWD and others via
SIP, but I wanted to play with IAX a little). It appears I'm running
into some sort of a codec mismatch or something because it's not working
right. The SIP client is a SPA-3000.
In iax.conf, I have something like the following:
[General]
2009 Sep 27
1
Peers Listed in "sip show channels"
Hi,
I am using Trxibox 2.6 latest ISO install.
Following is the output of : "sip show channels"
[trixbox ~]# /usr/sbin/asterisk -rx "sip show channels"
Peer User/ANR Call ID Seq (Tx/Rx) Format
Hold Last Message
212.53.40.40 0218245 6cfb845d050 09011/00000 0x0 (nothing) No
192.168.1.116 (None) YTc4ZmM3NjV 00101/00006 0x0
2009 Oct 28
1
Clear pending SIP channels
Hi all,
I have a question regarding pending (zombie) SIP sessions: on Asterisk CLI, with command 'sip show channels' , I see two channels in use with callID and other infos detailed; also 'sip show inuse' give me same result (in terms of channels usage):
Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message
xx.xx.xx.79 209
2010 Sep 14
5
sip show channels
Hi,
I'm trying to view a list of the active calls to see if I can restart Asterisk.
When I do 'sip show channels', I get a huge list like this (just a sample pasted):-
92.110.7.210 (None) 198827f2469 00102/00000 0x0 (nothing) No Init: OPTIONS
92.110.7.210 (None) 6b211bb04ac 00102/00000 0x0 (nothing) No Init: OPTIONS
92.108.34.153
2007 Aug 17
4
Call Limits
Hi all,
Some of my asterisk users have used their maximum call limit for incoming
calls (peers). There incoming call limit should automatically reset to zero
after hangup but its not happening and they no longer can recieve any calls
as their allowed limit is already full. So is there any way to reset the
call limit on peers by commands or do i have to restart my asterisk server?
--
Best Regards
2006 Apr 19
0
Re: new_callback_call and conf disconnect
We are using G711 for phones to talk to Asterisk and G729 licenses at
asterisk to talk to ITSP
Could you please suggest transcoder to use from G711 and G729 and which is
comptible with Asterisk. We will like to avoid using TDM if possible
Also i remember that initially we didn't have G729 and were using only 711
for with vicidial but then also we had same problems. at that time it was
only 2
2003 Oct 23
0
WAS: Call pickup (*8) on SIP devices. Bug #116
I've attached two SIP debugs in reference to bug #116. They are from
today's CVS build.
1. pickup.txt is a call from SIP(1) to SIP(2) with SIP(3) picking up the
call. After which, SIP(2) rings for about 30 seconds then stops.
2. hangup.txt is a call from SIP(1) to SIP(2) with SIP(1) hanging up before
the call is answered.
SIP(1&3) are Cisco 7960's and SIP(2) is a Polycom
2004 Sep 16
1
SIP channel stuck after registration
Quick question for the experts: I'm seeing stuck SIP channel(s)
scheduled for destruction stay open. This appears to happen
after (apparently successfully) registering with a SIP peer.
Any ideas where to start digging into this? I'm running today's
CVS, however the problem existed before and does persist.
Thanks,
Kai-Uwe
Example: (registration with iconnecthere.com)
ast1000*CLI>
2004 Jul 16
1
SIP channels UNKWN
I'm having an oddball issue with a Polycom SoundPoint IP 500. As you
can see below Asterisk thinks there are 2 SIP channels active, but show
channels tells me there are no calls active. Anyone have any idea why
this is happening? The Polycom occasionally stops accepting calls and
requires a power cycle.
fs-1*CLI> sip show channels
Peer User/ANR Call ID Seq