similar to: no active channel but one active call???

Hi everyone: when I try to run dovecot after modifying the dovecot.conf file I have the following error messages: Mar 23 09:56:52 mailer dovecot: [ID 583609 mail.info] Dovecot v1.2.10 starting up Mar 23 09:56:52 mailer dovecot: [ID 583609 mail.info] Generating Diffie-Hellman parameters for the first time. This may take a while.. Mar 23 09:56:52 mailer dovecot: [ID 583609 mail.error]

----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Wednesday, December 01, 2004 7:07 AM Subject: Asterisk-Users Digest, Vol 5, Issue 6 > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

'morning everybody, Here is the setup: 5126800422 called 3035 (3035 is a Cisco 7960). The call is g729. 3035 presses 'Conference' on her phone and calls 8327549222. This call is ulaw. (65.72.107.2 is our Cisco 7206 SIP->PRI gateway.) asterisk*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format 65.72.107.2 8327549222 1758081f67e

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

vps*CLI> iax2 show channels Channel Peer Username ID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format 0 active IAX channels vps*CLI> core show channels Channel Location State Application(Data) 0 active channels 0 active calls vps*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format

Hi, i've just faced with some bridged calls which could not be hungup just killing the asterisk process solved the problem: Zap/63-1 (incoming s 1 ) Up Bridged Call SIP/2035-e9cb logs say: Jul 22 14:54:12 NOTICE[17161] chan_sip.c: Disconnecting call 'SIP/2035-e9cb' for lack of RTP activity in 6785 seconds Jul 22 14:54:13 NOTICE[17161] chan_sip.c: Disconnecting

Hi! I'm trying an asterisk-FWD connection. I'm using X-Lite OR SIPPS as the IP phone. I 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

Hi all, i'm trouble with codec setup on an asterisk machine 1.4.18 and some Thomson ST2030 as extensions. In the users.conf file for internal extension i have: disallow=all allow=g729 allow=alaw allow=ulaw Without any codec installed (i mean with original g729 of asterisk) all go fine, calling from an extension to one other: Peer User/ANR Call ID Seq (Tx/Rx) Format

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

Hi, one of the sip-extensions we created always returns busy when someone tries to call the phone. The extension itself can place calls. We're using snom360 phones with the latest firmware. On every one of those phones when we register with the sip-extension, we've experienced the same problem. This is the output from sip show channels: Peer User/ANR Call ID Seq

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

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

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

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?

I have a user that just got a broadband connection so she could have an extension off our pbx. The service is DSL and uses a speedstream 5200 dsl router. I sent her a Polycom IP300. At first it would not access the config files via ftp so I had tech support walk her through setting the phone's internal IP to be the dmz. This allowed me to set up the phone using the web interface and now

Hi all, Firsts of all, let me say that I'm new to asterisk. I have some time suscribed to the list reading a lot of your messages and trying to learn a lot. The case is: last week I installed an asterisk server in the following scenario: PBX --- CISCO_ROUTER ---- ASTERISK The calls that are routed within the asterisk work perfect, there is not problem. However, the calls that are

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

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

They are good. I am checking with Release now. 20> Clang :: CodeGenObjC/image-info.m I will investigate it later. ...Takumi vs8 20>Failing Tests (3): 20> Clang :: CodeGenObjC/image-info.m 20> LLVM :: Transforms/SRETPromotion/basictest.ll 20> LLVM-Unit :: support/debug/SupportTests.exe/CastingTest.cast 20> Expected Passes : 8106 20> Expected Failures : 73