[asterisk-users] Asterisk Billing
Network Configurations Block D, Surrey Park, Barham Road, Westville, 3610 Helpdesk: (086) 163-8266 Tel: (031) 266-1563 Fax: (031) 266-4206 Hi List. I'm in need of something that will allow me to analyze cdr details either via .csv or mysql that will give me call durations as well as call costs. This is so that we can see in what areas/staff are costing what per month/week on outbound phone calls. Can anyone recommend a system? I've looked at Asterisk CDR and while this works perfect it doesn't allow for actual call costs. I'm also looking at Astbill but not so sure if it will suit this application as that seems more for a provider - end user and Astbill wants to control the workings/creating of users/peers or am I mistaken? Thanks, Richard . Best Regards Richard Soderblom Network Configurations Cell: E-Mail: [EMAIL PROTECTED] Number of Attachments: 0 This message (and any associated files) is intended only for the use of the individual or entity to which it is addressed and may contain information that is confidential, subject to copyright or constitutes a trade secret. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this message, or files associated with this message, is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. Messages sent to and from us may be monitored. Any views or opinions presented are solely those of the author and do not necessarily represent those of the company. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] chan_misdn
Network Configurations Block D, Surrey Park, Barham Road, Westville, 3610 Helpdesk: (086) 163-8266 Tel: (031) 266-1563 Fax: (031) 266-4206 Sounds like call waiting on the telco side of things. I have it here as well but asterisk just shows that there is an incoming call on the isdn line but doesn't actually ring any phones until a channel is available. Best Regards Richard Soderblom Network Configurations Cell: E-Mail: [EMAIL PROTECTED] Number of Attachments: 0 This message (and any associated files) is intended only for the use of the individual or entity to which it is addressed and may contain information that is confidential, subject to copyright or constitutes a trade secret. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this message, or files associated with this message, is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. Messages sent to and from us may be monitored. Any views or opinions presented are solely those of the author and do not necessarily represent those of the company. -Original Message- From: Tiziano Martelli [mailto:[EMAIL PROTECTED] Sent: 29 March 2007 07:43 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] chan_misdn This is my problem. I don't even know it this is the right site to ask for it, but let's go. I've a Asterisk box with 1.4.1 version version installed. It's equipped with a TDM400 with 2 FXO modules and a HFC based ISDN BRI card. Everything goes OK, except for the following scenario: on the PMP S0 bus of the ISDN telecom NT adapter I've connected both a ISDN modem and the HFC card. Now: 1. when the modem IS NOT connected, I could receive (and obviously answer) up to two calls and the (eventually) third one get a busy signal (right). 2. when the modem IS connected (it's set up to use only 1 channel) I could receive (and answer) to only one call (right). But, if a third call comes in, the phone designed to ring in extension.conf rings (wrong!) and the calling party ears the ringing tone! If I try to answer, both ends ears the busy signal and the other conversations (the first I've responded and the modem connection still run). What's wrong? Thanks for any help ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR Billing
Network Configurations Block D, Surrey Park, Barham Road, Westville, 3610 Helpdesk: (086) 163-8266 Tel: (031) 266-1563 Fax: (031) 266-4206 Hi everyone. I'm looking for a postpaid billing application that will pull call records either from the cvs file or from a mysql table (something like ASTERISK : CDR ANALYSER http://www.areski.net/asterisk-stat-v2/about.php but just with billing). Ideally it mustn't modify any .conf files, we don't need it to control or activate/suspend SIP accounts, just report on their usage and bill accordingly with the destinations dialed. We do fairly complex call routing in our extensions.conf file based on what SIP dialed what SIP and the status of the termination of the dial app and where to route unsuccessful calls based on the SIP id. So I wouldn't like it to modify these, but if it has to then so be it. I'm not interested in the price however oss does appeal more to me. The closest one I have found so far is MOR from http://www.kolmisoft.com/index.php?option=com_frontpageItemid=1 however it doesn't seem to support mISDN (using the B410P BRI cards). Does anyone have any experience with these or others? Regards, Richard. . Best Regards Richard Soderblom Network Configurations Cell: E-Mail: [EMAIL PROTECTED] Number of Attachments: 0 This message (and any associated files) is intended only for the use of the individual or entity to which it is addressed and may contain information that is confidential, subject to copyright or constitutes a trade secret. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this message, or files associated with this message, is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. Messages sent to and from us may be monitored. Any views or opinions presented are solely those of the author and do not necessarily represent those of the company. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dial application timeout
Network Configurations
Block D, Surrey Park, Barham Road, Westville, 3610
Helpdesk: (086) 163-8266
Tel: (031) 266-1563
Fax: (031) 266-4206
Hi people.
I'm hoping someone has come across this problem with version 1.2.14
In my dial plan I call various SIP phones using the following little
macro:
exten = s,1,Set(TIMEOUT(absolute)=14400)
exten = s,2,MixMonitor(from-${CALLERID}-to-${ARG2}-at-${TIMESTAMP}.wav)
exten = s,3,Dial(SIP/${ARG1},20,tTmwW)
exten = s,4,Goto(s-${DIALSTATUS},1)
Currently the caller hears music on hold when the called extension is
ringing. This all works fine.
When I remove the m option to stop the moh from playing then things
start falling apart. Everything goes through and the call is connected
when the other phone is answered. However after 20 seconds (set by the
,20, option) of the conversation the call is terminated and both users
are sent to the s-NOANSWER extension and end up in each others
voicemail.
After I replace the m in the option string it works as per normal
again but with the moh playing.
Thanks,
Richard
Best Regards
Richard Soderblom
Network Configurations
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RE: [asterisk-users] Callback/ringback
Network Configurations
Block D, Surrey Park, Barham Road, Westville, 3610
Helpdesk: (086) 163-8266
Tel: (031) 266-1563
Fax: (031) 266-4206
Excellent little script. Thanks, Yehavi.
Best Regards
Richard Soderblom
Network Configurations
Cell:
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-Original Message-
From: Yehavi Bourvine +972-8-9489444 [mailto:[EMAIL PROTECTED]
Sent: 18 January 2007 07:40 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Callback/ringback
Enclosed bellow is the fragment from extenstions.conf which does two
things:
*41 - Does the ring-back staff.
*42 - Calls back the last one who called you.
Regards, __Yehavi:
; regular local extensions:
; The flow is: If not available or no answer send to mailbox if exists,
; send busy if no mailbox. Same for busy.
; We try to avoid the n+101 rule whenever possible, but it is not always
; possible as HasVoiceMailbox() does only n+101 jump.
exten = _999XX,1,Set(_To=${EXTEN}) ; Save the original extension
dialled.
exten = _999XX,n,Set(_From=${CALLERID(num)}) ; Save the caller.
; Save the caller number at the called extension for *42 usage.
exten = _999XX,n,Set(DB(${To}/LastCaller)=${From})
; Where we called for *41
exten = _999XX,n,Set(DB(${From}/LastCalled)=${To})
; Now dial the extension.
exten = _999XX,n,Dial(SIP/${EXTEN},20,) ; Dial the phone for 20
seconds.
; No answer or busy
exten = _999XX,n,GoTo(s-${DIALSTATUS},1) ; Jump according to the
failure mode
exten = _999XX,n,Hangup() ; Just to be sure...
; No answer:
exten = s-NOANSWER,1,MailboxExists(${To}|j); Has a mailbox?
exten = s-NOANSWER,n,Busy(); No maibox = play busy.
exten = s-NOANSWER,102,VoiceMail(u${To}) ; Has mailbox - send the call
to there
; Busy:
exten = s-BUSY,1,MailboxExists(${To}|j); Has a mailbox?
exten = s-BUSY,n,Busy(); No maibox = play busy.
exten = s-BUSY,102,VoiceMail(b${To}) ; Has mailbox - send the call to
there
; Unavailable channel - act as busy:
exten = s-CHANUNAVAIL,1,Goto(s-BUSY,1);
; Called here when the call is successfull and the user hanged the
phone.
; Check whether the user has a waiting callback queued on him/her
exten = h,1,NoOp(${From} ${To} ${EXTEN})
exten = h,2,Set(tmp=${DB(${From}/CallBack)}) ; Get who is waiting for
us
exten = h,3,NoOp(${From} ${tmp})
exten = h,4,GotoIf($[ ${tmp} ]?5:103) ; Anyone waiting for us?
exten = h,5,DBdel(${From}/CallBack); And delete it...
; Create the callfile and then move it to the spool directory to make
the call.
exten = h,6,System(echo Channel: SIP/${tmp} /tmp/test.tmp${To})
exten = h,7,System(echo WaitTime: 20 /tmp/test.tmp${To})
exten = h,8,System(echo Extension: ${From} /tmp/test.tmp${To})
exten = h,9,System(echo CallerID: Callback \\\${tmp}\\\
/tmp/test.tmp${To})
exten = h,10,System(mv /tmp/test.tmp${To}
/var/spool/asterisk/outgoing/)
exten = h,103,NoOp(Nothing to call)
; *42: Get the last number who called us, say it and call it.
exten = *42,1,Set(tmp=${DB(${CALLERID(num)}/LastCaller})
exten = *42,n,SayDigits(${tmp})
exten = *42,n,Goto(${tmp},1)
; *41: Camp on the last extension dialled
exten = *41,1,Set(tmp=${DB(${CALLERID(num)}/LastCalled)})
exten = *41,n,SayDigits(${tmp})
; Save it so when the other side hangs it will see it and dial us.
exten = *41,n,Set(DB(${tmp}/CallBack)=${CALLERID(num)})
exten = *41,n,Hangup()
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[asterisk-users] Callback/ringback
Hi. Has anyone had any success in implementing a callback or ringback function in Asterisk? I've had a look at the callback-voicemail example on voip-info.org http://www.voip-info.org/wiki/view/Asterisk+tips+callback However it won't quite work for me. I need it for local SIP users which most of them don't have voicemail. If one SIP user calls another SIP user and the second user is busy or unavailable then Asterisk should inform the first user that the number they dialed is busy and hangup the call. Once the second caller is available again then Asterisk should initiate a call back to both the users and connect them. Any ideas on how to achieve this will be appreciated. Thanks, Richard . ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium Wildcard B410P
Network Configurations Block D, Surrey Park, Barham Road, Westville, 3610 Helpdesk: (086) 163-8266 Tel: (031) 266-1563 Fax: (031) 266-4206 Does anyone know if misdn and the B410P is working yet in kernel 2.6.18/19? Best Regards Richard Soderblom Network Configurations Cell: E-Mail: [EMAIL PROTECTED] Number of Attachments: 0 This message (and any associated files) is intended only for the use of the individual or entity to which it is addressed and may contain information that is confidential, subject to copyright or constitutes a trade secret. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this message, or files associated with this message, is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. Messages sent to and from us may be monitored. Any views or opinions presented are solely those of the author and do not necessarily represent those of the company. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: 04 January 2007 03:50 PM To: asterisk-users@lists.digium.com Subject: asterisk-users Digest, Vol 30, Issue 11 Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of asterisk-users digest... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and outlook
Network Configurations Block D, Surrey Park, Barham Road, Westville, 3610 Helpdesk: (086) 163-8266 Tel: (031) 266-1563 Fax: (031) 266-4206 Hi list. Has anyone used any commercial or open source application to integrate Asterisk into MS Outlook 2003 which can be used to place calls directly to contacts from Outlook? And if so how well does it work? Thanks, Richard Best Regards Richard Soderblom Network Configurations Cell: E-Mail: [EMAIL PROTECTED] Number of Attachments: 0 This message (and any associated files) is intended only for the use of the individual or entity to which it is addressed and may contain information that is confidential, subject to copyright or constitutes a trade secret. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this message, or files associated with this message, is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. Messages sent to and from us may be monitored. Any views or opinions presented are solely those of the author and do not necessarily represent those of the company. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] promotional info in music on hold
Network Configurations Block D, Surrey Park, Barham Road, Westville, 3610 Helpdesk: (086) 163-8266 Tel: (031) 266-1563 Fax: (031) 266-4206 Hi. Is it possible to have asterisk insert various audio files into the playback with the music on hold if they are holding on for an extension or in a queue? Something like the following: | V Welcome to ABC | V Music on hold for 30s | V Please remember to | V More music on hold for 30s or so | V More voice overs And so on. Could one put these files into a separate folder and the have asterisk randomly play them back with the moh? It would be easy because then we could just update the files every month or whenever we need to. Thanks, Richard Best Regards Richard Soderblom Network Configurations Cell: E-Mail: [EMAIL PROTECTED] Number of Attachments: 0 This message (and any associated files) is intended only for the use of the individual or entity to which it is addressed and may contain information that is confidential, subject to copyright or constitutes a trade secret. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this message, or files associated with this message, is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. Messages sent to and from us may be monitored. Any views or opinions presented are solely those of the author and do not necessarily represent those of the company. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
