similar to: Hung SIP channels

I am a fairly new user of Asterisk and I am generally impressed with its features. I have some questions about the SIP channel support: 1. I have noticed that even when there are no active calls, there is a list of active SIP channels. This appears to be a bug. Has anyone seen this? 2. If there are stuck SIP channels, how can one clear them without re-starting Asterisk? 3. One time a

I have some stuck channels that I think I'm going to have to bounce Asterisk to get rid of, but am curious to know what they are and how they've managed to accumulate. The show up with a channel identifier of '(None)' as in the output below, and do not show up in the soft hangup list, and so can't be cleared by that method. Here is the output from iax2 show channels:

hi, i want to call PC2PC between to IAX client without authentication i want to allow every user to use PC2PC no any password required. Please let me know what i have need to do in IAX.conf or any other file to allow any user to call Pc2Pc. My IAX.conf [guest] type=user context=default callerid="Guest IAX User" My extensions.conf [default] exten=>_.,1,Dial(IAX2/${EXTEN})

hi, i want to call PC2PC between to IAX client without authentication i want to allow every user to use PC2PC no any password required. Please let me know what i have need to do in IAX.conf or any other file to allow any user to call Pc2Pc. My IAX.conf [guest] type=user context=default callerid="Guest IAX User" My extensions.conf [default]

Dear Sanjay, Sorry sanjay i miss to explain completely. My PC2PC mean is Dialer2Dialer i want to allow call between Dialer with out any registry and authentication through IAX. so i need to setup Asterisk accept calls from any user and users can call to each other without any password and registration. please help how can i configure Asterisk using IAX in this regards. thanks, Asif Message: 9

Can someone tell me how to clear hung SIP channels in asterisk without restarting? Currently I have 62 channels and only show 10 in use.. this is some of the sip show channels output.. xxx.xxx.xxx.xxx 00xxx24xxx 04240xxxxxx 00103/00001 UNKN (d) ?How can I remove these? from * without rebooting? ? .o-------------------------------------------------------o. Brian Fertig Network

Hi I have two Asterisk Server: The first server "A", all phone are connected The Second server "B" only route call to a lot of SIP supplier the server A sent: ; Destination: Non connu dans le DialPlan - Apparaitra en UNKNOW dans le CDR exten => _X.,1,Set(CDR(CodeTier)=BUS-UNKNOW) exten => _X.,2,Dial(IAX2/SERVERB/${EXTEN},180,rt) exten =>

I have a data frame that reads client ID date transcations 323232 11/1/2010 22 323232 11/2/2010 0 323232 11/3/2010 missing 121212 11/10/2010 32 121212 11/11/2010 15 ................................. I want to order the rows by client ID and date and using a black-box forecasting method create the data fcst(client,date of forecast, date for which forecast applies). Assume that I

Hey Folks, This is a quick release that fixes a major bug. I forgot to require the timeout library properly in mongrel.rb so people using Mongrel outside of Rails would see pauses. 0.3.3 fixes this all up. The 0.3.3 release also has a small change to the examples/simpletest.rb file with some gzip response using Ruby''s zlib support. Curious what people think about this and whether it

Dear all I have problem with calculate probability, I have data x1,...,x10, I want to calculate probability x11 given x1,...,x10 with two conditions. 1. x is normal 2. unknow distribution How I can do this. Many Thanks. Jumlong -- Jumlong Vongprasert Assist, Prof. Institute of Research and Development Ubon Ratchathani Rajabhat University Ubon Ratchathani

Compiled the 3.0.0 version of the source code , then tried lli --use-mcjit irfile.txt On both windows and linux, I got: LLVM ERROR: Unknow object format. If I omit the -use-mcjit option, the command works well. It seems to me that something about MCJIT is broken in the 3.0.0 version. Also tried to initialize an ExecutionEngine from code, got errors like "Target does not support MC

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 Asterisk 1.4.18 with 70 various Polycoms, 12 analog, and 18 Spectralink wireless IP phones. Most of the Spectralink phones have entries in 'sip show channels' that do not go away. None of the other phones do this. Is there anyway to remove these entries without restarting Asterisk? Any ideas on what could be done to prevent this? Example output: xxx.xxx.xxx.xxx 541

Hi On my FC3 box with asterisk 1.0.9....MusicOnHold is not working. It starts and stops immediately... An unknow option mono comes...from where it is originating.?? As there is nothing written in .conf file. Console output is below: I am using mpg123 version 0.59r. Although I am able to play music with mpg123 but why it is on No-cooperation movement against asterisk ? Need help..any

Hi, I use guestfs.part_disk to partition one disk with 'mbr' and formated it to ntfs, then attach this disk to one windows xp virtual machine. But this windows xp can't recognise that partition, it says 'uknown partition' Why it is so, and how to solve it? May someone help me with this problem? Thanks. -------------- next part -------------- An HTML attachment was

Hi, I was looking at some bugs to play with, and I started with http://llvm.org/bugs/show_bug.cgi?id=18606 As I commented there, a loop is unrolled and exhibit this pattern: %mul.1 = mul i32 %mul, %mul %mul.2 = mul i32 %mul.1, %mul.1 .... With an unroll factor of 32, the last multiply has 2^32 terms in its SCEV expression. (I mean I expect it would have those terms if I was patient

Hi I've 3 iax connections to my provider , each of them have own DID , PH1<----| | \/ PH2<-->|-----| <---------------------------> |----|<-- DID1 | A1 | <---------------------------> |ISP |<-- DID2 PH3<-->|-----| <---------------------------> |----|<-- DID3 I had iax phone on each of this connection , but now I want to terminate all

On 2/7/14, 10:24 AM, Andrew Trick wrote: > > On Feb 5, 2014, at 12:54 AM, Mehdi Amini <mehdi.amini at silkan.com > <mailto:mehdi.amini at silkan.com>> wrote: > >> Hi, >> >> I was looking at some bugs to play with, and I started with >> http://llvm.org/bugs/show_bug.cgi?id=18606 >> >> As I commented there, a loop is unrolled and exhibit

Hello all, When I do 'sip show channels' I have seen a lot of entries where these calls has already been terminated. Some of these channels are bolong to calls being made 2 days ago but still showing from the CLI. They look like 10.223.51.173 0022676583 130b36625fc 00102/00103 unknow(d) Rx: BYE 10.223.51.173 0022676583 5533069e578 00102/00103 unknow(d) Rx: BYE 10.223.51.173

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, --