similar to: RE: 2 asterisks connected but trying to bridge

Anyone suffering an outage with them right now? I am getting the following from my box when I try to dial using them.... == No one is available to answer at this time W -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050309/e1096325/attachment.htm

Guys.. I have a problem with 2 asterisks connected together but each time one tries to call another I get this: -- Executing Dial("SIP/fozy-dfbc", "IAX2/voip-gw/201|60|mwtWT") in new stack -- Called voip-gw/201 -- Started music on hold, class 'default', on SIP/fozy-dfbc -- Stopped music on hold on SIP/fozy-dfbc -- Call accepted by 10.0.101.1 (format

Is it me... or is it voipjet? This week I've been trying various providers, just can't seem to get voipjet to work. I signed up with voipjet but so far can't get it to work inbound or out bound. I always get 'all circuits busy'. May 12 22:27:05 VERBOSE[2442]: -- Executing [1;36;40mDial[0;37;40m("[1;35;40mSIP/101-ad89[0;37;40m",

Dear List members- I am trying to configure ASTCC (Asterisk calling card application) but having a hard time to configure it properly. My project deadline is approaching and couldn't figure out how to make ASTCC functional. Here are some details what I have done so far. 1) I have installed ASTCC successfully. 2) I can access astcc-admin.cgi script without any problem. 3) I have created

I am using Asterisk CVS from last week and have been using Realtime SIP for a couple weeks now without any problems. Yesterday I decided to turn on Realtime IAX but I am having problems dialing to my long distance providers like Voicepulse, Sixtel or Nufone. I get the following: -- Executing Dial("SIP/2001-3761", "IAX2/password@voicepulse/19566680301") in new stack

When I try to call iaxtel it goes to codec ADPCM even though I have define in iax.conf gsm Call accepted by 69.73.19.178 (format ADPCM) -- Format for call is ADPCM My settings: [general] port=4569 register => xxxx:xxxx@iaxtel.com bandwidth=high jitterbuffer=no tos=lowdelay [voipjet] type=peer host= xxx.xxx.xxx.xx secret= xxx auth=md5 notransfer=yes context=incoming disallow=all ;

It seems to never fail - after 3 to 5 minutes SIP -> IAX calls drop audio on one side. I place a call out through voipjet, and call quality is flawless. However a few minutes later the person who I'm talking to can no longer hear me. I can still hear them. What should I look for to resolve this? Has anyone else had this problem? Using last night's CVS this problem still exists.

Hello List, I've spent the last day trying to find information on how to call multiple sip phones and have them all answer so I page everbody. When I use Dial( ext&ext&ext... ) the first phone that answers gets the page, but none of the others do. Is there a way to get around this? TIA, Steve

Hi, I am Chandramouli from India. I have successfully implemented Intercom (Dialling within my office LAN) and Voicemail features using Asterisk. To implement this, I am using X-Lite Softphone. Now, I wish to make calls to US using VoIP Asterisk. I dont think I need any external hardware to implement pure VoIP solution. Here I am sending my configuration file values: Contents of

Hello, is it possible to create two uncorrelated random vectors for a given distribution. In fact, I would like to have something like the function "rnorm" or "rlogis" with the extra property that they are uncorrelated. Thanks for your help, Luba [[alternative HTML version deleted]]

Hello, I use the function sem the following way sem.mod <- sem(model, mod.cov, N=109) where the variables are modelled: Z -> M Z -> I Z -> R M <-> M I <-> I R <-> R Z <-> Z The output is ... Normalized Residuals Min. 1st Qu. Median Mean 3rd Qu. Max. -7.3300 -0.2750 -0.2670 -0.1290 -0.0369 9.0300 Parameter Estimates Estimate Std Error z value Pr(>|z|)

Hello, is there a similar package like SEM. In fact, I am looking for a package where I as input several given vector variables which are part of an autoregressive model. Then I would like to obtan parameters for so called latent variables. Thank you, Luba [[alternative HTML version deleted]]

Guys. Anybody knows why this is happening? Seems every time I make an internal call, the manager shows this and I don't get the callerid on my identapop but rather the calledid.. Event: Dial Privilege: call,all Source: SIP/intruder1-85f0 Destination: SIP/test-f037 CallerID: 201 CallerIDName: Anton Krall SrcUniqueID: 1117038116.7 DestUniqueID: 1117038116.8 Event: Newchannel Privilege:

I try to load the package lattice by library(lattice). Although I have installed it through CRAN by internet as well as from a local directory using the zip-file, it can't be loaded into the editor. Can anyone give advice, please. I am using R 2.9.1 on Widows XP. Thanks a lot, Luba [[alternative HTML version deleted]]

I try to load the package lattice by library(lattice). Although I have installed it through CRAN by internet as well as from a local directory using the zip-file, it can't be loaded into the editor. Can anyone give advice, please. I am using R 2.9.1 on Widows XP. Thanks a lot, Luba [[alternative HTML version deleted]]

Hello, I tried to build an own R package on Windows XP but get an error which I can't solve. I called my package "pack". It works to creat the file "pack" with package.skeleton(). But when I try to compile it with Rcmd build --binary pack I obtain the following error: IO::Seekable::seek missing at C:/R/R-2.9.0/share/perl/FileHandle.pm line 59. Compilation failed in

Hi List, I'm wondering if someone who uses VoipJet as their termination service would do me a favor. If I call the American Airlines reservation number (1-800-433-7300), the call gets connected, but after 30 seconds asterisk drops the call responding that no one answered. I'm using areskicc2 (calling card app) as an authentication system and I don't know if that is what is

Dear all, I've noticed some significant voice quality deterioration when calling US landline via VoIPjet.com in the last week or so. Before that the quality was pretty good. Has anyone else experienced any voice quality problems with voipjet recently? Thanks, Roman

I've heard good things about VoipJet here, so I was going to set up an account. Then I noticed their Terms of Service here: https://www.voipjet.com/tos.php Several things there are very concerning to me, and I'm interested in what other people here think of them. * The ToS specifically forbids use for any call relating to medical, financial, or government matters -- as well as any

Hi, We found `llvm-config --cxxflags' does not have -fno-exceptions -fno-rtti when using cmake to build LLVM. Does anyone know how to fix it? Best regards, Luba ---------- Forwarded message ---------- From: Luba Tang <lubatang at gmail.com> Date: 2012/11/21 Subject: Re: [MCLinker] Re: Major release - MCLinker 2.0.0 - RockBull To: mclinker at googlegroups.com Hi, We make sure the