Displaying 20 results from an estimated 23 matches for "0000ms".
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00000ms
2006 Jan 03
1
IAX2 channels denoted as '(None)'
...Peer Username ID (Lo/Rem) Seq
(Tx/Rx) Lag Jitter JitBuf Format
(None) 111.111.111.111 xxxxxx 00003/00020
00001/00003 00070ms 0001ms 0071ms g729
(None) 111.111.111.111 xxxxxx 00004/00005
00004/00005 00070ms 0000ms 0070ms g729
(None) 111.111.111.111 zzzzzz 00005/00004
00002/00004 00000ms 0007ms 0071ms g729
(None) 111.111.111.111 xxxxxx 00007/00042
00001/00004 00000ms 0001ms 0071ms g729
IAX2/yyyyyy-14 222.222.222.222 yyyyyy 00014/...
2003 Aug 13
1
FWD SIP phone format=2, FWD call format=4, why?
...configured the X-Lite and SIPPS to use GSM codec. Whe I
call FWD, I get this info on the channels when the call has not been
stablished yet:
sip show channels
Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter
Format
192.246.69.223 613 1770bf3430d 00102/00000 00000ms 0000ms 2
150.187.xxx.yyy ildefonso C72ACD25-1A 00101/11482 00000ms 0000ms 2
2 active SIP channel(s)
-- SIP/fwd-161b answered SIP/ildefonso-d2fc
-- Attempting native bridge of SIP/ildefonso-d2fc and SIP/fwd-161b
When it gets stablished, I get:
sip show channels
Peer...
2008 Mar 28
1
IAX user register problem
...144 register_verify: No
registration for peer 'j' (from 203.99.57.80)
advcomm6*CLI> iax2 show channels
Channel Peer Username ID (Lo/Rem) Seq
(Tx/Rx) Lag Jitter JitBuf Format
(None) 203.99.57.80 (None) 00004/15232
00001/00001 00000ms -0001ms 0000ms unknow
(None) 203.99.57.80 jaffaradvc 00005/15233
00004/00004 00000ms -0001ms 0000ms unknow
(None) 203.99.57.80 (None) 00006/18423
00001/00001 00000ms -0001ms 0000ms unknow
3 active IAX channels
Mar 28 03:26:28 DEBUG[2855]: chan_...
2004 Apr 15
1
Asterisk in pass-thru mode
...idged Call SIP/1234-faf1
SIP/1234-faf1 (company1 5001 1 ) Up Dial SIP/22225001|20|r
2 active channel(s)
sip*CLI> sip show channels
Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format
192.168.1.101 22225001 257684717aa 00104/00000 00000ms 0000ms ULAW
210.17.211.5 1234 003094c2-fd 00104/00102 00000ms 0000ms ULAW
2 active SIP channel(s)
Thanks.
Ben
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2004 Nov 28
1
IAX2 and FWD problems?
...alled 526146:secret@iax2.fwdnet.net/612
When the call is trying to connect:-
splat*CLI> iax2 show channels
Channel Peer Username ID (Lo/Rem) Seq
(Tx/Rx) Lag Jitter JitBuf Format
(None) 65.39.205.121 (None) 00002/00000
00001/00000 00000ms 0000ms 0000ms UNKN
IAX2/65.39.205.121:4 65.39.205.121 596146 00003/00000
00002/00000 00000ms 0000ms 0000ms UNKN
2 active IAX channel(s)
And:
splat*CLI> iax2 show registry
Host Username Perceived Refresh State
65.39.205.121:4569 596146...
2008 Mar 28
1
how to register IAX user without password
..._verify: No
registration for peer 'j' (from 203.99.57.80)
advcomm6*CLI> iax2 show channels
Channel Peer Username ID (Lo/Rem) Seq
(Tx/Rx) Lag Jitter JitBuf Format
(None) 203.99.57.80 (None) 00004/15232
00001/00001 00000ms -0001ms 0000ms unknow
(None) 203.99.57.80 jaffaradvc 00005/15233
00004/00004 00000ms -0001ms 0000ms unknow
(None) 203.99.57.80 (None) 00006/18423
00001/00001 00000ms -0001ms 0000ms unknow
3 active IAX channels
Mar 28 03:26:28 DEBUG[...
2004 May 20
4
Mystery SIP channels
...ent on
both of my asterisk servers. Sometimes they disappear for a few seconds
and then come back. It always has the same Call ID.
voip1*CLI> sip show channels
Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter
Format
192.168.0.102 (None) df92fb1b-8a 00101/03059 00000ms 0000ms
UNKN
2003 Jul 07
0
One-way talk paths (without INVITE?) and other issues
...IP channel disappear after a call is
over. They are always listed as active, even hours later. Here are is
the result of "show sip channels":
> Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format
> 172.28.54.160 m4 478c64565be 00104/00000 00000ms 0000ms 0
> 172.28.54.160 m4 4d330ced01e 00104/00001 00000ms 0000ms 0
> 172.28.54.160 m4 0bb7855f15b 00104/00001 00000ms 0000ms 0
> 172.28.54.160 m4 3db8538b4b4 00104/00001 00000ms 0000ms 0
> 4 active SIP channel(s)
but none of these ca...
2008 Mar 28
1
how to register IAX user without password for any user
..._verify: No
registration for peer 'j' (from 203.99.57.80)
advcomm6*CLI> iax2 show channels
Channel Peer Username ID (Lo/Rem) Seq
(Tx/Rx) Lag Jitter JitBuf Format
(None) 203.99.57.80 (None) 00004/15232
00001/00001 00000ms -0001ms 0000ms unknow
(None) 203.99.57.80 jaffaradvc 00005/15233
00004/00004 00000ms -0001ms 0000ms unknow
(None) 203.99.57.80 (None) 00006/18423
00001/00001 00000ms -0001ms 0000ms unknow
3 active IAX channels
Mar 28 03:26:28 DEBUG[...
2004 Jul 12
1
SIP client to IAXTel 800/888/877/866 dialing thru Asterisk
...e, the SIP side is choosing the wrong codec (though I've tried
putting allow=ulaw first and it still didn't help)
grover*CLI> iax2 show channels
Peer Username ID (Lo/Rem) Seq (Tx/Rx) Lag Jitter
JitBuf Format
69.73.19.178 (None) 00003/00000 00001/00000 00000ms 0000ms
0000ms UNKN
69.73.19.178 phoneboy 00005/00102 00019/00017 00099ms 0000ms
0010ms UNKN
2 active IAX channel(s)
grover*CLI> sip show channels
Peer User/ANR Call ID Seq (Tx/Rx) Format
10.0.0.250 53 2b890d18-49 00101/00103 ILBC
1 active SI...
2003 Jul 08
5
Using multiple iconnecthere accounts
Has anybody out there tried to use two different iconnecthere accounts
with Asterisk?
What I want to do is use a second account if the first is busy.
I have tried the following:
exten=>_91NXXNXXXXXX,1,StripMSD,1
exten=>_1NXXNXXXXXX,2,Dial,SIP/BYEXTENSION@iconnect ;iconnect is the
first account
exten=>_1NXXNXXXXXX,3,Dial,SIP/BYEXTENSION@iconnect2 ;iconnect2 is
the second account
But that
2004 Jun 24
2
How to force G729
...o 'g729' for this call because of ${SIP_CODEC) variable
-- Attempting native bridge of SIP/2016-b119 and SIP/mypstngate-caed
*CLI> sip show channels
Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format
192.168.0.100 0041911234 1f7d34e3642 00102/00000 00000ms 0000ms ULAW
192.168.0.2 2016 4977-4F41-7 00101/00003 00000ms 0000ms G729A
2 active SIP channel(s)
[... after hangup ...]
== Spawn extension (auth-out, 0911234567, 2) exited non-zero on 'SIP/2016-b119'
-- Executing Hangup("SIP/2016-b119", "")...
2004 Sep 07
3
Maximum tollerable lag/jitter for IAX2 w/o jitterbuffer enabled?
...even though g.729 is also an 'allowed' codec
(w/5 licenses installed). During an average call 'iax2 show channels'
provides:
Peer Username ID (Lo/Rem) Seq (Tx/Rx) Lag Jitter
JitBuf Format
10.0.40.140 astpbx-woo 00002/00002 00005/00006 00040ms 0036ms
0000ms GSM
Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District
2009 Nov 13
1
destroy zombie session
...have this scenario (CLI output of command "iax2 show channels")
IP-AM-PBX*CLI> iax2 show channels
Channel Peer Username ID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format
(None) 10.229.47.113 REMOTE_SER 06818/14174 00002/00002 00000ms -0001ms 0000ms unknow
1 active IAX channel
IP-AM-PBX*CLI>
Here I can't issue "soft hangup" command because I haven't a channel to specify ("None" is not a choice :) ).
Now the question is: is there a way to drop this (zombie) channel off and release frozen resourc...
2003 Sep 13
2
MusicOnHold (MOH) silent on BudgeTone-100 only.
...xited non-zero on 'SIP/202-351f'
While the BudgeTone is 'ringing', Asterisk appears to think it has a
live connection:
*CLI> sip show channels
Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format
192.168.42.56 202 26e8973e-e3 00101/06079 00000ms 0000ms ULAW
1 active SIP channel(s)
Has anyone else tried MOH from a Grandstream BudgeTone extension? It
would mean a lot just to hear that it's my local config and not some
weird BT problem. If someone could post a working BudgeTone SIP
definition, that might help too.
Thanks!
-Steve
2003 Sep 25
4
SIP Problem
...5 18:51:02 NOTICE[1125329600]: File chan_sip.c, Line 4571
(handle_response): Failed to authenticate on REGISTER to
'<sip:<user>@fwd.pulver.com>;tag=as75fc26a2'
That continues until
asterisk*CLI>sip show channels
65.39.205.114 <usr>@fwd. 3ff9e23356a 00103/00000 00000ms 0000ms UNKN
65.39.205.114 <usr>@fwd. 3ff9e23356a 00102/00000 00000ms 0000ms UNKN
504 active SIP channel(s)
tail -4 /var/log/asterisk/messages
Sep 25 18:31:00 WARNING[1125329600]: File rtp.c, Line 708 (ast_rtp_new): Unable
to allocate socket: Too many open files
Sep 25 18:31:00 WARN...
2004 Dec 14
0
Codec "Uknown" with IAX connection
...auth=md5
secret=mypassword
disallow=all
allow=ulaw
qualify=5000
linux-home*CLI> iax2 show channels
Channel Peer Username ID (Lo/Rem) Seq (Tx/Rx)
Lag Jitter JitBuf Format
IAX2/teliax@teliax/7 66.234.228.170 teliax 00007/00309 00001/00000
00000ms 0000ms 0000ms unknow
linux-home*CLI> iax2 debug
IAX2 Debugging Enabled
linux-home*CLI>
Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE
Timestamp: 00017ms SCall: 00005 DCall: 00000 [63.218.7.228:4569]
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno:...
2003 Aug 21
0
No audio in either direction, sip channels hanging, asterisk will not shut down.
...62.254.245.18",response="466777cb9c995d5c862fc5310d33477d",nonce="09f50874",algorithm=md5
10 headers, 0 lines
*CLI> sip show channels
Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format
62.254.245.14 3046 171697661ab 00102/00000 00000ms 0000ms 4
192.168.101.186 2001 000ab714-51 00101/00103 00000ms 0000ms 4
2 active SIP channel(s)
*CLI>
2003 Jul 09
2
sip jitter buffer
This is kind of a repost of one part of a previous question I have had.
Peer Username Call ID Seq (Tx/Rx) Lag Jitter
Format
213.137.73.178 xxxxxxxxxx 3705df0a5f7 00103/00000 00000ms 0000ms
4
1 active SIP channel(s)
I see that there is 0ms Jitter set. How can I set a Jitter buffer
for use with sip channels?
I can't seem to find any documentation about this.
Any help is always appreciated.