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I have an asterisk installation at a client, it's quite simple. Basically it's an asterisk downloaded from CVS about a week ago, with 3 Zaptel FXO cards (the digium ones) and 10 Grandstream Budgettone SIP phones ... Every now and then, especially when a call is ringing and not picked up immediately, Asterisk quits with a segmentation fault error. IT seems quite inexplicable, my dialplan

..." to configure call parking ? I haven't managed to find a direct way to transfer a call from phone to phone except using blind transfer and I want the person initiating the transfer to speak to the receiving person before actually passing the call. can anybody help please ? cheers Dave A Caruana

Hello, I'd need to compute the calibration error of posterior class probabilities p(y|x) estimated by using rpart as classification tree. Namely, I train rpart on a dataset D and then use predict(... type="prob") to estimate p(y|x). I've found the possibility to do that in the ROCR package, but I cannot find a link to a paper/book which explains the details of the

...efore forwarding the calls to a gateway in USA. If anyone has unlimited patience and feels like helping, it's more than appreciated. Finally, are there any good books on the subject ? I'm ok with IP networks and the likes, but pretty green when it comes to telephony. many cheers, Dave A. Caruana Malta -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030513/3c3ed06a/attachment.htm

...n walk in, pick up a phone & use a calling card, they are routed through hardware boxes made by Quintum (tenor A800) so i'm thinking of something in a similar vein but taking calls off the PSTN. Anyone who has any suggestions/sample configs/help etc. is more than welcome!! cheers Dave Alan Caruana Malta -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030527/95bc2414/attachment.htm

i've just lost $2000 dollars or so on my first commercial asterisk installation .. i'm running a PIV class server, three Digium Wildcard FXO cards, and 10 Grandstream Budgettone SIP phones. The system was to be a PBX for a small company. After over 2 months of pissing about, the client has had his fill of asterisk problems, and asked me to take my equipment out of the building. Obviously,

I have just compiled & installed the latest oh323, on a fresh asterisk installation however using a previously working oh323.conf file. When I try to dial an outbound oh323 call I get the following error : -- Going to extension s|1 because of immediate=yes -- Executing Wait("Zap/1-1", "1") in new stack -- Accepting call from '21382890' to 's'

just three "stupid" questions I need to ask .. 1. what's the sequence to press on a SIP phone to transfer a call to another extension. 2. what's the same thing if you want to hold an incoming call, speak to the other extension, then pass the call? 3. what's the extensions.conf syntax to dial two SIP extensions at once? many thanks Dave

well .. good news :) i've just put in txgain=1.0 rxgain=1.0 in my zapata.conf and upgraded the Grandstream Budgettones i'm using to version 81 of the software and all seems fine .. there is still an echo but after the first couple of seconds of call it vanishes, as the echocancelling kicks in .. so far my client is happy :) now .. i have one slight problem left .. although most of my

Dear all please ,I need to write a function in R to estimate the parameters of negative binomial distribution and then calculate the loglikelihood amount for given data.Is there any one to help me. thank you very much for any help Best regards [[alternative HTML version deleted]]

Hi all, i use the DISA app for giving the user a trunk after a authentication through PGSQL as follows .... auth via PGSQL exten => s,1,DISA,no-password|test I think the user is now in context "test" and he could dial any number if the extension-conf in "test" is for example exten s,1,Dial,OH323/<myip> But if the user dial one digit the call build up

I am trying to get asterisk to dial this address : sip:723@216.52.153.207 Using a softphone on my PC (217.168.168.49) it dials immediately and I get a voice prompt .. I have configured an extension, 1303 on asterisk, modifying the demo configuration : exten => 1303,1,Dial(SIP/723@216.52.153.207) When from my softphone I dial sip:1303@217.168.168.51 on the console I get : -- Executing

i've been having a problem getting two SIP phones to bridge running through asterisk, actually one is a SIP softphone, SJ Phone, and the other is the Go2Call calling gateway. Someone suggested that I don't have the right codecs. How do I find out which codecs are installed, and how can I install further codecs? Any suggestions which would be the right one? I think hte problem is from the

I added the line "allow G723.1" in my sip.conf general config, and from a bridge connection which gives silence, I have progressed to the error message below, and the call gets rejected. help!! Dave ps. 217.168.168.49 : soft sipphone, i'm trying SJphone & Pingel Instant Expressa 723@216.52.153.207 : Go2Call SIP gateway -- Executing

hi, does anyone have a problem with OH323 crashing with a segmentation fault whenever anything tries to connect to it ??? are the current CVS versions OK? Would like to speak to someone with a bit of OH323 experience, so if u're in a good mood to help, please do :) cheers Dave

not really asterisk related this, but is it normal for a mail to take so long to be resent through the mailing list server? i'm speaking about 20 minute + delays here .. (or it it only me ?) cheers Dave

My system is an asterisk machine, with an E1 card (functioning) and forwarding calls to a remote SIP address .. when a call connects I am getting the following error : NOTICE[1240577216]: File rtp.c, Line 330 (ast_rtp_read): Unknown RTP codec 19 received can anybody tell me what this means (& how I may fix it ?) cheers Dave ps. the service i'm connecting to uses G723

hi .. maybe someone can help me, I seem to have lost the sheet of paper that comes with an E100P card and tells you how to compile the stuff it requires to run. I'm trying to move my Asterisk to a different box and at this time totally stuck. Could someone be kind enough as to mail me a PDF of it ?? many thanks Dave

what is the format for an h323 entry in the dialplan? can I use chan_h323 without compiling anything else or should I compile oh323? basically what's the best way :) cheers Dave

hi .. when placing a SIP call to a sip host in the states every few seconds I get an RTP codec 19 error. I know this is related to comfort noise, and the call goes through OK ... how can I suppress the error message ? Also, many times I get "Invalid CSeq Number" back from 216.52.153.207 (which is the host i'm calling) and the call drops.. is there a solution for this ? cheers Dave