similar to: FXO hangup Problem.....

Hello, I would like to use Intel Blade machine for running Asterisk. Is there anyone who already use Intel Blade server for running Asterisk? Can you please explain, how perform Asterisk with Intel Blade machine? I would appreciate for giving me feedback regarding this issue. Regards Nahid -------------- next part -------------- An HTML attachment was scrubbed... URL:

>From couple of weeks i am working on asterisk fax but was not successful. I am able to receive only half fax documents and its ending with blurred lines. I have tried with almost all the versions of spandsp 0.0.2pre4,pre10 etc but of no use. I have digium card to which i have plugged in 2 pstn lines and i have removed echocancellation too(zapata.conf) and also enabled g7111 alaw&Ulaw

Hello, I am facing hangup delay problem using FXO (X100P) card with the following scenario: PSTN Phone--------->BTTB PSTN Gateway------->Asterisk BOX-FXO-------->SIP Endpoint When anyone make call to my Asterisk-BOX FXO, they gets IVR to press their desired SIP extension number. If calling party disconnect the his/her call before pick up the call by the called party, then called

Hello, I've got an Asterisk server with a TDM400P with 2 FXO modules and 2 FXS modules. This server is connected to 2 PSTN lines and 2 analog phones. In my Zaptel configuration, I've defined 2 groups : one for the 2 FXO's and one for the 2 FXS. The asterisk server is just used to add a little IVR and Voicemail service. Eveything works fine, but sometimes the conversation is

Hi, We installed OBSD's ssh package on production and find our batch jobs are hanging on the "ssh" command. The details are included below, for both normal and "hanging" executions. Please let me know if this is a new or existing bug or if you require further info to debug this. The command was executed with the highest debug level (-v -v -v for level 3). This script

Afternoon all, After doing some test on my asterisk box I can successfully receive calls to my Asterisk PBX to a SIP phone from The Telstra PSTN network Dial out from a sip phone is also not an issue, all calls connect and terminate normally. If I call the Asterisk PBX say from PSTN in Zap1-1 and out through ZAP2-2 back to the PSTN (after entering the correct pin off course) the card does not

# lda.r > # > # Author: Amsha Nahid, Jairus Bowne, Gerard Murray > # Purpose: Perform Linear Discriminant Analysis (LDA) > # > # Input: Data matrix as specified in Data-matrix-format.pdf > # Output: LDA plot > # > # Notes: Missing values (if any) are replaced by the half of the lowest > # value in the entire data matrix. >

hello, i already include the error in blue color word. i hope it can help you to understand my question. if not burden you, please give me a guide how to correct the error or maybe you can correct the coding cause error. thank you. > # lda.r > # > # Author: Amsha Nahid, Jairus Bowne, Gerard Murray > # Purpose: Perform Linear Discriminant Analysis (LDA) > # > #

i've this udp daemon which is waiting for an incoming udp datagram. now i want test this daemon for random garbage to test how it behaves. My udp daemon is running because its shows on netstat. problem is if i issue following command for putting udp datagram : $nc -uvvz <host> <port> it does not output any thing. i have straced the udp daemon. which does not returns from recv()

HELLO, anybody... could you help me to check the below coding for volcano. what is the mistake? what the plot could not display? # volcano_plot.r # # Author: Amsha Nahid, Jairus Bowne, Gerard Murray # Purpose: Produces a volcano plot # # Input: Data matrix as specified in Data-matrix-format.pdf # Output: Plots log2(fold change) vs log10(t-test P-value) # #

Hello. My name is Akashah. i work at metabolic laboratory. From my study, i found that volcano plot can help a lot in my section.  i already studied about the volcano plot and get the coding to run in R software, unfortunately, there is may be something wrong with the coding. This is because  no graph appear, but no error (blue color text) was shown on the R console. Below is the coding for

Thanks for the suggestion. I have busydetect=yes in zapata.conf. You refer to "Martin's algorithm". Can you provide more details please? ----- Original Message ----- From: "Soner Tari" <list@kulustur.org> Subject: Re: [Asterisk-Users] FXO module in TDM400P (UK, BT) - Hangup detection failing To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Hi, Im getting mad with this error, I have a x100p installed with wcfxo module loaded perfectly, I can receive incoming calls and detect very good the hangup for incoming calls. But for outgoing calls its a mess. When I place a call for outgoing, i heard the ringing, my cell or phone rings and when I pick up the phone it hangs: -- Called g1/91xxxxxxx -- Hungup

Hi, I'm interested to implement the idea of using search indexes build by Lucene. I have some idea of how to do that. I'm currently studying the index format of both. I have written an application of search using Lucene and now I'll do the same using Xapian. then I'll check the details of those index format and then find how to convert Lucene index to Xapian one. Is there somebody

Dear Developers, I am a Master's student at the ECE department of the University of Florida, USA.​​ For my research project, supervised by Prof. Mark Tehranipoor<http://tehranipoor.ece.ufl.edu/> and Prof. Farimah Farahmandi<http://farimah.ece.ufl.edu/>, I need to use Clang LLVM DataflowSanitizer library in KLEE. However, I have faced some difficulties (explained below) while

My outgoing FXO calls are answered but have no audio in either direction if I have callprogress=no in chan_dahdi.conf. If I change to callprogress=yes then the audio returns. My chan_dahdi.conf file is listed below. Can anyone point-out why callprogress=no isn't working? #cat /tmp/a [trunkgroups] [channels] language=en context=incoming toneduration=40 ;usedistinctiveringdetection=yes

So I made the mistake of buying a Carrier Access channel bank without noticing the page on the wiki about the fact that they don't support disconnect supervision (bastards!). However, apart from that, I do have it working fine for incoming calls. Is there some trick to get asterisk to detect the hangup tones from SBC? I've tried busydetect and callprogress as suggested, but neither

Hi ! I have some issues, i don't know exactly if it's a busy detection issue. When i dial into the Asterisk box, and if i hang up before the Asterisk answers with the IVR Welcome message, the Asterisk goes on with the call. But, if i wait for the Asterisk to answer, and if i hang up, the Asterisk hangs up too. I have this parameters on zapata.conf: busydetect=no

Dear all I have asterisk with TDM808B FXO port with i call in asterisk and i promt IVR then user dial extention for user then my SIP phone rining but i disconnect or hangup my mobile phone but still SIP phone rining and stop rining after timeout is there any PSTN problme or FXO signalling problme i have configuraed singalling=fxs_ks ----PGP Signature-- Satish

I think the problem it's in your dialplan, extensions.conf: ; voicemail management [voicemail] include => misc exten => 6245,1,VoiceMailMain2() exten => 6245,2,Hangup() Check the last line, I have the same problem and was because I wrote wrong the Hangup instruction... Regards! Date: Thu, 10 Jun 2004 09:54:32 -0600 From: Chris Hirsch <chris@base2technology.com> To: