similar to: Problem Hangup Help

Hello list, i need help. My problem is that when I want to call (using ISDN phone or internal SIP client) via the Sip provider a real phone number (ISDN phone or internal SIP Asterisk >> SIP ), I get a ring tone. When I now decide to hang up (e.g. if nobody answers), the called telephone continues to ring almost forever. the sip.conf: [2563105] accountcode = 2563105 username =

Hola estoy tratando de configurar mi primera VPN, pero cuando me conecto al servidor VPN Netstat -nat me dice que la coneccion esta en estado TIME_WAIT, por otro lado revisando syslog encuentro lo siguiente: tincd 1.0.2 (Nov 8 2003 20:54:15) starting, debug level 0 Nov 22 08:42:15 woody tinc.vpn[5810]: /dev/net/tun is a Linux tun/tap device (tun mode) Nov 22 08:42:15 woody tinc.vpn[5810]:

Dear collegues, Using maxstat I am getting the following: > blood <- maxstat.test(Surv(SUPER, FV)~ZAP,data=zap70, smethod="LogRank") Error in maxstat(y = structure(c(24.4301369863014, 26.4164383561644, 18.7835616438356, : couldn't find function "cscores" I do not know the meaning of this problem. Could you please help me on dat? Thank you in advance for your

Hi!! I am a new user of R (just trying to analysis microarrays with some packages from the bioconductor project). I would like to import a text-delimeted file containing 20 columns and 22200 rows. I have tried read.table; scan(file="") <matrix(scan("file", n=20*200,20,200, byrow=TRUE)); Doesn't matter what I try I got the next message: error in file (file

just recently I have finished implementing support for saving non-audio chunks in RIFF WAVE and AIFF with the FLAC file in application metadata blocks, and restoring them when decoding. you can read about more about it here: http://www.hydrogenaudio.org/forums/index.php?showtopic=56968 the zip file there has windows binaries; CVS HEAD has everything checked in if you need a different platform.

Hello, I tried to configure a very simple case of Asterisk using SIP userA --- Asterisk server ---- userB sip.conf [userA] type=friend username=userA host=dynamic nat=no context=test [userB] type=friend username=userB host=dynamic nat=no context=test In extensions.conf [test] exten => 1000,1,Dial(SIP/userA) exten => 2000,1,Dial(SIP/userB) I make a call from userA to userB, it works,

--- Erik de Castro Lopo <erikd-flac@mega-nerd.com> wrote: > Hi all, > > If I have code that does this: > > while (FLAC__stream_decoder_process_single (decoder)) > /* Do something. */ ; > > I get an infinite loop. Shouldn't FLAC__stream_decoder_process_single > return false if it gets to FLAC__STREAM_DECODER_END_OF_STREAM? it supposed to be like

*Dear List* Plz help, i am not much experienced with asterisk. i configured it on ubuntu 12.04. no problem when i call any mobile no(0091XXXXXXXXXX) but when i call on my local asterisk no.(101,102 or 105) it is not connecting giving error "Auto fallthrough, channel 'SIP/lucknow-0000006f' status is 'CHANUNAVAIL' *while when i call 200 it is playing audiofile successfully.

Dear list; I'm trying for forward some calls to an others asterisk using IAX2 protocol. But My asterisk can forward some calls and for others it hangs up automaticaly. Before my asterisk was working perfectly, i do not know what is happening!! When i try directly zoiper with my provider's asterisk it works perfectly. Here is the output from the cli when i made a call that asterisk hangs

I am writing a cron script to check if certain extensions are online and if they aren't then Asterisk creates a couple of .call files to notify another set of extensions or external numbers. It works fine except for logging information. What I'm doing in the script is setting a "fake" caller ID (as it's generated by Asterisk, not by a user) and calling out real users. So

Hi all, I have this weird problem. I'm running asterisk 1.0.3 on Debian Sid (official debian package). We have 2 fritz ISDN cards. All is working great. Till I called the bank. It rings one time and then gives me the congestion tone. Here is what I see on the CLI (phone nr obfuscated for privacy reasons): -- Executing Dial("SCCP/michiel-00000004",

Hi I need help This is the error: Really destroying SIP dialog 'NDMxOWRmYTRhMWVkMGFhMjllMzU4YmNmNjQwN2NlM2Y.' Method: SUBSCRIBE -- Executing [00919885497796 at internal:1] Set("SIP/1001-0000000b", "CALLERID(num)=8790771141") in new stack -- Executing [00919885497796 at internal:2] Dial("SIP/1001-0000000b", "SIP/00919885497796 at sonetel")

hi there, i run asterisk 1.4 on my debian machine, which is in my internal 10.x.x.x network, behind my main computer, i cam make call, receive calls, all works fine, with all providers except sipgate.de, there i can receive call and make them, i can hear the other end but they can not hear me, this is only the case with sipgate.de i don#t know how to configure it and thought maybe someone can help

Greetings everyone, I'm having some issues connecting a Asterisk box to a old ISDN PBX. Everything works fine but the undetermined digits rules. For instance, if I have _00X. and I start dialing for instance 0035..., Asterisk just get the 4 first numbers and starts dialing 0035. I've tried adding: stop_tone_after_first_digit = yes append_digits2exten = yes Although I

Hi All, For a 1.4 version asterisk, whats the recommended mechanism for dialling with ENUM lookup? At the moment I user SIPbroker, but am getting tired of it hanging on certain numbers, so I was thinking about implementing it myself. I've seen various vo-ip.info pages (http://www.voip-info.org/wiki/view/Asterisk+cmd+EnumLookup) talking about the func ENUMLOOKUP instead of EnumLookup

hello, I have been using asterisk now for about 2 years now on a RH8.0 it is our main call gateway. I have on the box 3 T1 TDM cards connected to 2 Rhino channel banks (FXS) and 1 CAC Access bank I (FXO) with so many softphones and ATA 186s. It has been working good till today some few hours ago. i just discovered that there were no dialtone on the phones. Asterisk did not spit out any error, it

Hello, I would to allow users to place calls overseas such as India and Malaysia but only with a security code. if they don't have a security code I want to be able to drop the calls. can someone point me to a right direction to achieve this goal? Thanks, Motty -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm having problem with call parking. When I park call, either via transfer to xten or park digit sequence from features.conf, I hear the parking lot number read to me and the user gets transferred. However, MOH stops for the caller the moment user is transferred. The user can be retrieved by dialing the parked extension and voice resumes. If the parked user hangs up, the channel state

Hi, I'm trying to bring up two PRIs from qwest with asterisk and dahdi. I'm using an OpenVox D410E and the drivers are loaded. My system.conf looks like this: # Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" B8ZS/ESF RED span=1,2,0,esf,b8zs bchan=1-24 # Span 2: TE4/0/2 "T4XXP (PCI) Card 0 Span 2" (MASTER) B8ZS/ESF RED span=2,1,0,esf,b8zs bchan=25-47 dchan=48 These

Sipdiscount has replaced their asterisk servers for another thing. Then, no more iax. Ok, but I can't make calls using sip also... I'm getting a "forbidden" error when using sip1.sipdiscount.com. Anybody got it working? -- Alejandro Vargas