Displaying 20 results from an estimated 10000 matches similar to: "SIP Problems continue..."
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.
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
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
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?
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
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
2004 Jun 03
4
miserable time with Cisco ATA186
I'm having a horrible experience getting a Cisco ATA-186 to work with *.
I can make calls from the ATA with no problems. However, incoming calls
make the ATA ring once, and then the call is disconnected. I have no
problems with my Sipura 2000 or my Grandstream phones.
I am running 2.16.1 sip code on the ATA 186. Neither * nor the ATA is
behind a NAT. They are both on public IP addresses
2003 Sep 27
1
Continuing Budgetone woes
I have spent the morning on this project, still without success.
Summary: Yesterday I inadvertently unplugged my Grandstream phone. I
might add I did a rebuild of my s/w from CVS at the same time. Since
then, the Budgetone seems to talk SIP just fine, but the RTP being sent
to it by asterisk "doesn't make any sound."
It was suggested I do a factory reset of the phone, which I
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)
2003 Sep 25
3
SIP codecs Errors
Hi all:
I recently update a system from CVS (Asterisk CVS-09/25/03-15:58:42), and I receiving the following message:
*CLI> WARNING[1187305408]: File chan_sip.c, Line 1864 (process_sdp): No compatible codecs!
The "show codecs" command shows:
*CLI> show codecs
1 (1 << 0) G.723.1
2 (1 << 1) GSM
4 (1 << 2) G.711 u-law
8 (1 << 3) G.711 A-law
16 (1 <<
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
2003 Dec 20
2
More beginner questions
Using DIAX softphone which seems to be working OK can get to VM/echotest etc
in the demo context
Am trying to setup FWD but get the following problems
Can hear it ringing when dialing FWD no 612 for time. Connects but no sound
from remote end.
Does anyone have any suggestions.
Softphone on 192.168.0.2 asterisk on 192.168.0.3 Netgear RP114 doing NAT to
the internet port 5060 being forwarded to
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,
--
2003 Nov 18
2
ask problem about softphone--asterisk--softphone, Urgent!!!
Hi, all,
I want to use asterisk SIP as a proxy, and two softphone (Ubiquity SIP Phone) as user agent, like below:
Softphone1<-------------->Asterisk SIP<------------>Softphone2
(User Agent) (Proxy) (User Agent)
155.69.xx.xx 155.69.yy.yy 155.69.zz.zz
zhou mysipproxy.com
2008 Jul 29
1
Multiple Asterisk SIP Server/client connections
I have 4 asterisk servers. They all have local phones on their local
network they manage for SIP based conversations. We then have IAX
between them all for inter-asterisk connections.
This setup has worked well for nearly 2 years now, minor problems here
and there but overall very nice.
Recently we acquired some Polycom video conference units. I was able to
setup our main server to host all
2005 Feb 27
3
music on hold trouble
Hi All
I seem to have a small problem with the music on hold button on SJPhone.
I have 2 asterisk installations one from the Rapid distribution and one from the latest CVS.
On the rapid dist when I press the music on hold button on my SJPhone I get music on hold.
When I do the same I get no music on hold just silence.
I create extension like this exten => 1111,1,MusicOnHold(Default),
2006 Jun 22
1
SIP Channel hangup problem with re-INVITE enabled - ugrent
Hi List
I have UAs registered with Asterisk and make outbound calls via ITSP1,
everything is fine without re-INVITE. When people call 178, the actual
number 112233445566 at ITSP1 network will be called.
When UA or called telephone (112233445566) hang up, the call and associated
channels are cleared.
Sip.conf
[general]
canreinvite=no
nat=no
[ITSP1]
type=peer
host=A.B.C.D
2003 Oct 03
4
Iconnect Incomming calls
I have an IconnectHere account with a Inbound number and have setup the sip.conf to register and am recieving the call but When I answer the call it disconnect. I have tried sending the call to from * to a Softphone, Pingtel, and FXS port and all result the same. As soon as I accept the call it disconnects. I believe it may be some type of codec issue but I am not very familiar with that layer.
2006 Apr 12
1
SIP call hangup from asterisk CLI
Hi,
We are using Vicidial and sometime even when agent disconnects, outgoing
call originated by dialer is still active. Since call was initiated by
dialer and then bought into meetme conference of agent and we can't corelate
this call to any agent channel.
When agents are dialing, channels doesn't show calls
vicidial2*CLI> show channels
Channel Location