similar to: SIP channels UNKWN

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

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

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?

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)

Hi List, I've got a 7960 that's behind NAT (nat_enabled: 1 and nat_received_processing: 1) and for whatever reason doesn't seem to register, or at least hold a registration. If both the phone and the router (netgear) are rebooted, the phone will register, take a few incoming/outgoing calls no problems, then a few hours later, it drops the registration and never re-registers. If the

Has anyone seen this before? This channel is consistently present 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

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

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.

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

Having just run the command "sip show channels" I get a list of channels even though there is no one on the phone (we only have 4 so it's easy to tell). Here is what I get: Peer User/ANR Call ID Seq (Tx/Rx) Format 192.168.0.22 (None) 4c81ac8e90c 00101/00000 UNKN 192.168.0.22 (None) 984ee48048d 00101/00000 UNKN 192.168.0.22

Behavior on Asterisk 1.2.12, 1.2.15, 1.2.16 "sip show channels" Always tends to show 100+ lines such as 192.168.1.241 (None) 2e2872da-1d 00101/21507 unkn No Rx: REGISTER Never seem to go away 198 total peers on this server All devices are behind NAT Registration expirations between 30secs to 2 minutes to help keep NAT open Should I extend the

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]

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

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

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

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 <<

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, --

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

Dear asterisk users, A few weeks ago I've been attacked by a DOS on REGISTER that I've solved with a fail2ban script. Now, since a few hours, I have my asterisk 1.4.21.2 running at 100% CPU again. I've checked the log and it shows nothing related to failed register or whatever. It just tells me that some of my peers are lagged, even with a verbosity of 10000 I've made a

== Parsing '/etc/asterisk/adsi.conf': Found -- Accepting call from '890003' to '185' on channel 27, span 1 -- Executing Answer("Zap/27-1", "") in new stack -- Executing Record("Zap/27-1", "soundexampless:mp3") in new stack -- Playing 'beep' WARNING[360468]: File translate.c, Line 128