Re: [Asterisk-Users] ASTCC: customer wants 100 accounts
Ronald Wiplinger wrote: He want to use 100 phones at the same time!!! Alas, he won't be able to. Re: ASTCC in-use flag You'll have to disable the in-use flag for his account. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] ASTCC: customer wants 100 accounts
Ronald Wiplinger wrote: I got a request for one customers to set-up 100 accounts. I use usually the Caller-ID as the card number. Is there a way to make it for 100 accounts easier? To generate 100 cards is not a problem, but if it would work with one account number would be even better I could use a different context for this customer and use only his account code as card number. Any advice would be appreciated. I'm not going to ask why the customer needs 100 cards. If he wants to access them all from 1 account, wouldn't he be happier with a single card that has the credits of 100 cards? In short, an account, not a card. Get my drift? Or, try making another brand with a markup of 100% I guess. Never tried that one though. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] ASTCC: customer wants 100 accounts
JP Carballo wrote: Ronald Wiplinger wrote: I got a request for one customers to set-up 100 accounts. I use usually the Caller-ID as the card number. Is there a way to make it for 100 accounts easier? To generate 100 cards is not a problem, but if it would work with one account number would be even better I could use a different context for this customer and use only his account code as card number. Any advice would be appreciated. I'm not going to ask why the customer needs 100 cards. If he wants to access them all from 1 account, wouldn't he be happier with a single card that has the credits of 100 cards? In short, an account, not a card. Get my drift? Or, try making another brand with a markup of 100% I guess. Never tried that one though. Correcting myself...that should be 1000%. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] ASTCC: How to reset periodically all card in use flag back?
Ronald Wiplinger wrote: Nicolás Gudiño wrote: You should install php-pcntl (or compile php to add support for process control functions). The inuse problem will be fixed then. Regards, Can you please give us more info about that? What is php-pcntl? What should it do? How can it be used to be a solution? Nicolas already said and I quote: Replying to myself... I was thinking on a2billing, not astcc, so php-pcntl will make no difference. ASTCC uses Perl-AGI. That should have clued you in. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] ASTCC: How to reset periodically all card in use flag back?
Ronald Wiplinger wrote: ok, How do I check if a particular channel is up? (Wasn't that what I asked above anyway) Try this. Using show channels concise, retrieve the channel names, then loop through each channel using show channel channel_name. Get the channel's UniqueID then compare that to the card's uniqueid. Hth. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] ASTCC: How to reset periodically all card in use flag back?
Maxim Vexler wrote: I would be interesed in your the applications you use in your dialplan to reset this flag. Could you please post here the relevent part of your extensions.conf ? http://lists.digium.com/pipermail/asterisk-users/2006-April/146947.html -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] ASTCC: How to reset periodically all card in use flag back?
Ronald Wiplinger wrote: If a user calls and hangs up before the destination party rings, than the in-use flag remains set! This is one case, but maybe there are many other cases. I have created a number the user can dial to reset this flag. However, that is written in the manual!!! Who reads a manual anyway I want to make to reset all in use flag with a program. Has anybody done it, or has a better idea? My idea is to check every 5 minutes, the database, which cards are set in use and check if this is true, if not reset it. Q: How do I know if a card is in use? Still banging your head over this one? Get the card's uniqueid and use it to check if that particular channel is up. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Asterisk::AGI and DIALEDTIME
Jean-Michel Hiver wrote: Hi List, In one of my AGIs (using DeadAGI) I grab the answered time using: my $res = $agi-exec (DIAL $dialstring); my $answeredtime = $agi-get_variable (ANSWEREDTIME); However this information differs from what's written in the Master.csv file (which happens to be the correct value!) Any ideas why? On my system, answeredtime returns the time elapsed since the call was answered by the destination. The time elapsed stored in Master.csv is from the time the current incoming call (channel) was answered. I'm using asterisk 1.2.7.1 and the lastest asterisk-perl distrib. I'm not using 1.2.7.x but I doubt this would change from earlier versions. P.S. Shouldn't the subject be ANSWEREDTIME? :) -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] WiFi VoIP Handsets..
Lacy Moore - Aspendora wrote: Had to turn my monitor upside down to read them :-) -- Lacy Moore Aspendora, Inc. You must have one of those rotating monitors huh? ;) -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Telephone support charging system with Asterisk?
Mike Dent wrote: My idea was for them to phone or login to a website and create a support account. They can then top this account up with X amount of credits, lets say 1 credit= 5 mins of support. Their account has a PIN associated with it. This is a typical prepaid system at work. When they call to get support they have to enter there number and are told how much credits they have remaining. They then get put through to my office phone, if I am available and pickup, they start to get charged for the duration of their call. If I am unavailable they go to voicemail and dont get charged. A few lines in the dial plan or a macro is all that's needed for this. Does this sound like something which is possible with Asterisk? Anybody doing something similar? I don't do this for support since support is free for our customers. But you can use ASTCC and generate a prepaid card for each support account. Of course, you can call the number a support account number and they won't know the difference. :) You will of course have to take into account your e-commerce system to allow your customers to pay online. ASTPP works with OSCommerce while I've managed to make ASTCC work with Virtuemart/Joomla. There are other applications. Look around. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Remote UNIX connection disconnected over and over
Il Neofita wrote: Hi, I am pretty sure that you already answer to this question, but I was not able to find the solution on the console I have over and over the following msgs -- Remote UNIX connection disconnected -- Remote UNIX connection disconnected -- Remote UNIX connection disconnected -- Remote UNIX connection disconnected -- Remote UNIX connection disconnected -- Remote UNIX connection disconnected -- Remote UNIX connection disconnected -- Remote UNIX connection disconnected -- Remote UNIX connection disconnected -- Remote UNIX connection disconnected -- Remote UNIX connection disconnected -- Remote UNIX connection disconnected Any idea? Someone or something is regularly accessing the manager interface. Nagios perhaps? -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] billing realtime
Dovid Bender wrote: A while back some one posted some code that he used that took out the flag in astcc that kept track if there was a call in progress for that pin or not. Dont know if it wil work for real time though. Dovid I don't know if you were pertaining to what I posted in the message ASTCC: How to reset in-use flag automatically ?. The setinuse() routine already exists in ASTCC. One simply has to use that routine to disable the inuse flag when a call begins and ASTCC will allow multiple calls for the same account. However, I too have no idea if this will work for real time. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] astcc: need partial pin code
Benchev wrote:
astcc is such a neat and stable piece that I would hardly dare to
to mess with it.
You make it sound like it's anathema to modify it :)
My idea was 1) you need a PIN=YES because otherwise pins are
not generated;
You mean to say PINS are not required, right? Because they are generated
upon card creation, whether or not you use them.
2)you need a PIN because most of the guys
use CALLERIDNUM as cardnumber, and that's open to see. I think it was
JP Carballo suggested earlier to authenticate against another variable
i.e.
It's nothing new as everyone here knows that clid can be spoofed.
I have other reasons not to use clid.
One of them is to allow our customers to call from any phone at their
disposal.
Another is to allow me to use the prefix of the card account number as a
code of sorts.
It can denote the geographic area of sale or even the denomination.
Although, PINS are good thing.
Yep. I would suggest however that you change the pin generation routine
a bit to NOT use the same prefix as the card number.
It may be paranoid of me to do so but I've changed the routines a bit to
generate pins with random first digits.
3)I think even if you use quiet=5 i.e.(astcc.agi,${CARDNO},${TEMP},5)
a PIN will be expected(if PIN=YES). That's way a soft link won't work
because you need to comment out the pin validation snippet in your
inner .agi and leave it for you disa .agi
I agree.
Hope I'm not waisting your time with already know things, but I love to
discuss about asterisk :-)
Like everyone else here :)
--
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Re: [Asterisk-Users] astcc: need partial pin code
Ronald Wiplinger wrote: I want to use something like: What is your card number: user keys in the number Enter your pin:user enter a long pin Enter your destination phone number: user enters the destination phone number Is there a code snip available for that? Not that I know of. Just juggle the way the routines are called. Everything you need (or most of them anyway) is in the code. Keyin needs always more time, we need to allow longer spaces between the digits, therefore we need to allow the # to finish the dialstring faster. I wonder if we can use one dialstring for all: cardnumber*pin*destination-number If you need a partial *anything*, be it a pin or a card number, you can use the perl routine substr() to retrieve as many characters of the string as you need. How can a user end the call and dial a new number, without hanging up? In my system, the user just presses * and they get dumped back into the IVR. Since the current cardnumber is stored in a variable, it does not need to be entered again to make another call. The user has usually a desk phone (=card number), and this dialin should work parallel, but of course it assumes still that only one card is in use. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] astcc: need partial pin code
Ronald Wiplinger wrote:
I tried now the examples in the wiki, but they do not fit!!!
If I use in configure Require Pins Yes then everyone needs a pin code!
If I use in configure Require Pins NO then calling in people will
just need to know a valid card number!!!
How can I overcome this?
Modify astcc.agi and the DB so that accounts can be flagged as requiring
a pin check.
Then you have to modify the pin checking routine.
How can I re-write:
exten = _77.,1,Answer
exten = _77.,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN:2},3)
exten = _77.,3,Hangup
sothat the dialstring:
77*123456789012*159753*011886939775516 would be splitted into:
${CARDNUM}=123456789012
${PIN}=159753
${DESTINATION}=0118869397755516
with a mysql lookup of the cardnum in astcc get the pin and compare to
the given pin. If all is ok, than use the dial command
What you have to do is add another argument to astcc.agi
After you've done that, you modify the check for the pin to take the new
argument into account.
You might also want to add a routine to retrieve a substring from the pin.
--
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Re: [Asterisk-Users] Asterisk as a phone survey system
Jay Milk wrote: Could you kindly let us know what numbers those survey-calls will be coming from, so we can all add them to our blacklists? Thanks! *snickers* -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] billing realtime
random cluster wrote: Now, the question, can I access somehow in a deadagi, or whatever the CDR function in order to update the credit when the call has just finished. Yes, certainly, through deadagi. I just have one question though, why reinvent the wheel? There are prepaid systems that work with asterisk. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Background asynchronous AGI
Matt wrote: Can't you do all of this with the (Absolute) time setting? So if the person has 4,000 minutes left.. set the call length for 4,000 minutes as the absolute max. Alternately... you could probably use screen? Launch an AGI from the main AGI using screen so it goes into the background... Or like astcc does, use the L argument to the dial command. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] billing realtime
Nick Hoffman wrote: Hi Jon. If a customer has 10 minutes of call credit left and he makes 2 concurrent calls, how do you know to cut off the 2 calls at the 5 minute mark rather than cut off both calls after 10 minutes? -- Nick e: [EMAIL PROTECTED] p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it. There's an application (sorry, which one, escapes me at the moment), that gets around this by reserving a certain amount of credit per call. Say the amount is 10 minutes, if you have 30 minutes worth of credit, you can have 3 concurrent calls good for 10 minutes each. The way I understand it, if you only have 15 minutes left in your account, the first call will last for 10 and the next concurrent one for 5 minutes. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] astcc and inwards billing
Ronald Wiplinger wrote:
I (cannot sleep and I) am thinking if there is a way to make inwards
billing easy possible.
To dial out we use something like:
exten =
_9011N.,4,DeadAGI(astcc.agi,${CALLERID(num)},${EXTEN:${TRUNKMSD}},${TARIFF})
(I have an extra field TARIFF, what allows me to use different prices
for different users)
To dial to a phone we use something like:
exten = 8,1,Dial(SIP/6001,20,tr)
Can we use something like:
exten = 8,4,DeadAGI(astcc.agi,${EXTEN},6001,${TARIFF},3)
and use TARIFF with all location as x cents ?
Short answer is Yes.
I use a similar astcc.agi call for in-network calls.
The cost will depend on how you've implemented TARIFF and whether it
adds to or bypasses the cost field for your 600x SIP route.
--
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Re: [Asterisk-Users] How to terminate ringing call before it is answered?
Obelix wrote: Is there a way to terminate a ringing call before it is answered? I am speaking of prepaid card application in which you want to make another call, because the current number it is not being answered, and you don't want to hangup before dialling another number. You mean like pressing * to exit and get dropped back into the IVR to make another call? -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTCC: How to reset in-use flag automatically ?
Ronald Wiplinger wrote:
#
# At this point we have a valid card and pin number.
#
if ($phoneno eq RESET_INUSE) {
setinuse($carddata-{number}, 0);
exit(0);
}
checkexpired($carddata-{number});
checkinuse($carddata-{number});
setinuse($carddata-{number}, 1);
I put this into 682 in the extensions.conf
exten = 681,1,DeadAGI(astcc.agi,${CALLERID(num)},BALANCE,1)
exten = 681,2,Hangup
exten = 682,1,DeadAGI(astcc.agi,${CALLERID(num)},RESET_INUSE,2)
exten = 682,2,Hangup
As soon the flag is set, 682 will also tell you: The card number is in
use, try later !
What do I miss?
There could be a call to checkinuse() before the RESET_INUSE routine.
If the RESET_INUSE flag is set, the routine should exit and not proceed
to the following calls to
checkexpired()
checkinuse() and
setinuse()
I suggest you check that the callerid you are using matches a card in
the astcc db.
--
JP Carballo
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Re: [Asterisk-Users] asterisk box as a voip gateway
Mark Quitoriano wrote: Hi Guys, Im configuring my asterisk box as a voip gateway. I have TE110P which is connected on my PBX and i will be using voip for my outgoing. Here's my config zaptel.conf: span=1,1,0,ccs,hdb3 fxoks=1-32 zapata.conf: context=default signalling=fxs_ks group=1 channel =1-32 -- Regards, Mark Quitoriano, CCNA What seems to be the problem? -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] ASTCC: How to reset in-use flag automatically ?
Ronald Wiplinger wrote:
I have some troubles with ASTCC. TO often the in-use flag
remains set.
I would like to find a solution, where astcc.agi checks automatically
if THIS user is in a call rather than to check the flag.
If that is not possible, I would like to have an extension to dial to,
and it will after hang up, reset the flag!
The in-use flag remains set, if the caller hang up before the gateway
gets the call.
Insert this in astcc.agi; anywhere after the calls for it to load and
connect to the db.
if ($phoneno eq RESET_INUSE) {
setinuse($carddata-{number}, 0);
exit(0);
}
And this in extensions.conf:
exten = s,n,DeadAGI(astcc.agi,${CARDNO},RESET_INUSE,2)
I leave it to you to capture ${CARDNO} :)
I don't enable this in the IVR unless the person has entered a valid
account number, for obvious reasons.
--
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Re: [Asterisk-Users] ASTCC: How to reset in-use flag automatically ?
Ronald Wiplinger wrote:
Insert this in astcc.agi; anywhere after the calls for it to load and
connect to the db.
if ($phoneno eq RESET_INUSE) {
setinuse($carddata-{number}, 0);
exit(0);
}
And this in extensions.conf:
exten = s,n,DeadAGI(astcc.agi,${CARDNO},RESET_INUSE,2)
I leave it to you to capture ${CARDNO} :)
I don't enable this in the IVR unless the person has entered a valid
account number, for obvious reasons.
Wouldn't that totally disable inuse??? It would be possible that a
user uses two or more soft phones and make phone calls on multiple
places!
Nope. I don't want that to happen either.
Because the 2nd argument is normally the phone number to call, the test
will be false and the routine will be skipped if the customer intends to
call.
Besides, if the routine does evaluate to true, it will exit the agi and
not process any calls anyway.
Set this up as a separate extension that you can call if an account is
locked in use.
I've only ever used this when testing new trunks because an account with
the inuse flag set means the previous call ended prematurely.
In my case, I want customers to make one and only one call at a time so
I left the inuse handling mostly intact.
I want it to be anal.
If a customer complains about it, I'm more worried about a trunk failing
than a cheating caller.
--
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Re: [Asterisk-Users] ASTCC: How to reset in-use flag automatically ?
Ronald Wiplinger wrote: I tried now many places to put these lines in. The system still announces This card number is in use. Can you give me a place where to put it in? It's not receiving a card number. Find the following 3 lines: # # At this point we have a valid card number. # Insert the whole routine either just before or after these lines. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] AstCC
Jeremy wrote: Is there anyway for me to authinicate first before givng options You can add an option to astcc to authenticate the account and then drop you back out into the IVR. Or you could use mysql queries in the dial plan to check the account number before passing it to astcc. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] AstCC
Jeremy wrote: My main goalwould be able to drop user back to the main IVR after logging in, but would this still log (bill) total time after auth? I don't use astcc's cdr anymore but from memory, I don't believe it can. *'s cdrs should have the total time, though not total time after auth. You could however, store the current time when authenticating. When the caller hangs up, run another agi that takes this value and subtracts it from the total time stored by *. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] AstCC
Jeremy wrote: Using this same method would I be able to add a cutsom menu in astcc (like call recording), by having it drop back into the IVR and then back to the agi? Of course. It's just a matter of setting things up in the Dialplan before passing the caller to astcc. In my IVR for instance, option 1 is for making calls, option 2 is for checking balance and option 5 is for checking rates for a particular number. When caller presses 1, he is first prompted for the account number and then passed to astcc.agi, which just asked for the number to call. The caller presses * anytime to exit, and is dropped back into the IVR. If he presses 2, his balance is read back immediately since his account number is already in a variable. If he presses 1 again, astcc asks for another number to call. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] AstCC
Il Neofita wrote: Hi, I am wondering if it is possible with astcc to make a second call without hangup and be oblige to re-enter all the codes. Any idea how to do? Thank you Yep, one way is to ask for the account code from the dialplan, save it to a var like CARDNO and pass that to astcc.agi When the person is done with a call, they can press, say, *, exit out to the menu, dial 1 and be prompted for a new number to call. The other way is to modify astcc.agi to save the account code to a var CARDNO. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] PSTN to Asterisk VOIP in Manila
[EMAIL PROTECTED] wrote: Hello, I'm sure you can use the Asterisk as an IP PBX. Good luck Madhawa Matt wrote: Hi list, Does anyone know the legalities of connecting an Asterisk box to the PSTN in Manila or where I can find this info out? I know it is illegal in some countries. thanks -Matt I posted this last year: http://news.inq7.net/infotech/index.php?index=1story_id=57657 VoIP was declared legal last August but subject to the NTC guidelines. Lately, I read that PLDT had lost a case against a company using grey routes. Their argument was that long distance calls were their personal property and hence the act of not using PLDT's gateways constituted theft. It was thrown out by the Supreme court of the Philippines. http://www.manilastandardtoday.com/?page=news06_mar06_2006 -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Call and then play IVR
Dovid Bender wrote: I know there was alk about this before but I cant sem to find it. Anyway to call some one and then play an IVR where they can make choices based on DTMF ? Thanks. Dovid You're looking for information on .call files for dialing out to someone. Just point the callee to a context with your IVR. http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Preferred editor(s) dialplan coding?
[EMAIL PROTECTED] wrote: On Sun, 5 Mar 2006, Michiel van Baak wrote: On 21:22, Sat 04 Mar 06, C F wrote: vi here vim :) Combined with the syntax file for asterisk. http://www.bemroses.net/images/curves.jpg -Dan Rotfl!!! Looks like whoever drew the emacs curve couldn't program himself out of a loop in emacs-lisp ;) -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Info about mp3 which are installed with Asterisk
Zach A wrote:
Hi,
The 3 MP3 files which are installed with asterisk, what is their bit
rate, are they mono and do they have ID3 tags?
Zach A
{192}([EMAIL PROTECTED]:Desktop)# file /var/lib/asterisk/mohmp3/*.mp3
/var/lib/asterisk/mohmp3/fpm-calm-river.mp3: MPEG ADTS, layer III, v1,
128 kBits, 44.1 kHz, JntStereo
/var/lib/asterisk/mohmp3/fpm-sunshine.mp3: MPEG ADTS, layer III, v1,
128 kBits, 44.1 kHz, JntStereo
/var/lib/asterisk/mohmp3/fpm-world-mix.mp3: MPEG ADTS, layer III, v1,
128 kBits, 44.1 kHz, JntStereo
{275}([EMAIL PROTECTED]:Desktop)$ id3tool
/var/lib/asterisk/mohmp3/*.mp3
Filename: /var/lib/asterisk/mohmp3/fpm-calm-river.mp3
No ID3 Tag
Filename: /var/lib/asterisk/mohmp3/fpm-sunshine.mp3
No ID3 Tag
Filename: /var/lib/asterisk/mohmp3/fpm-world-mix.mp3
No ID3 Tag
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm really quite busy.
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Re: [Asterisk-Users] Preferred editor(s) dialplan coding?
Bill Gibbs wrote: Vim for everything -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pete Barnwell Sent: Friday, March 03, 2006 7:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Preferred editor(s) dialplan coding? Emacs... On Sat, 2006-03-04 at 01:35 +0100, adibar wrote: Vim forever ;-) On Fri, Mar 03, 2006 at 03:06:02PM -0500, S McGowan wrote: snip emacs for me :) -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Help with dialplan
Cosmin Prund wrote: I've got a Mobile-to-PBX gateway installed and I want the ability to dial from my mobile phone into my PBX and next dial a land-line from the PBX so I can make cheep mobile-to-land-line calls while on the go. I've contemplated using the WaitExten application but it only seems to wait for ONE digit! Is there a way to put the calling mobile phone into a context and wait for a full-length extension? The documentation on voip-info has just the example you need. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+WaitExten -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Billing inbound calls per minute
[EMAIL PROTECTED] wrote:
On Monday 06 February 2006 09:25, JP Carballo wrote:
snip
ASTCC works fine here. The duration and billseconds fields in my cdrs as
well as ASTCC's cdr are filled.
I don't use the connect fee field though and all are set to 0.
Would you share with me how'd you do billing on a DID
(if you do), and through what Technology?
Anything that goes Local here is ANSWEREDTIME zero.
Thanks,
benchev
That probably explains it.
IIRC, from when I was still testing ASTCC, when calling a Local channel,
the AGI suffers from short term memory loss and forgets the values of
channel variables even if /n is used in the dial string.
I checked my test server logs and while I can verify that ASTCC's CDR
does have blank duration and billsec fields for the Local calls, *'s CDR
records them.
If it's also true for you, you might want to use *'s CDRs for rating.
I do billing based on account number so clients are free to call from
any phone. I don't check callerid.
Since each account is based on the phone number registered by the
client, I can just chop off the 2 digit prefix and set their callerid
with the result.
[makecall]
exten = s,1,Set(CALLERID(num)=${CARDNO:2})
exten = s,n,DeadAGI(astcc.agi,${CARDNO})
exten = s,n,Goto(nf2xsubmenu,s,1)
All my calls are routed to IAX2 or SIP or Zap.
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm really quite busy.
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Re: [Asterisk-Users] SIP Aliases
Darrick Hartman wrote: Is it possible with asterisk to setup aliases for SIP? For example, direct [EMAIL PROTECTED] to [EMAIL PROTECTED] Yep, just name an extension: exten = sales,1,Dial(SIP/Sales1SIP/Sales2) IIRC, we're limited to a single domain at the moment. Feel free to double check and correct me. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Fedora Core 3 or Fedora Core 4? yum update ornot?
Alex Barnes wrote: I think the once it's working, leave it alone advice is very sound indeed :) A similar rule says If it ain't broke, don't fix it. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Fedora Core 3 or Fedora Core 4? yum update ornot?
Jens Vagelpohl wrote: On 8 Feb 2006, at 09:43, JP Carballo wrote: Alex Barnes wrote: I think the once it's working, leave it alone advice is very sound indeed :) A similar rule says If it ain't broke, don't fix it. Until you realize some script kiddie has exploited another Apache/ mod_ssl bug and is now remote-controlling your box. There are no hard and fast recipes here. Neither the automatically apply any and all updates nor the build and never look at it again- policies should be applied without taking the specific situation into account. If your box is on the internet you simply cannot forego updates. Period. If your box is completely walled off from the internet you can be lax about it (unless you have to worry about attacks from the inside). The best policy is probably one that is halfway between the two. There are packages you only ever want to update under parental supervision, like kernels. Then there are packages where you want to grab any update you can get ASAP, like Apache, or PHP, or SSH. Yum allows you to express this in its configuration, you can exclude packages from the automatic update. I personally run a nightly script that uses yum to determine if there are updates. I apply them by hand. However, this is only feasible because it runs on just two machines. That shouldn't be an exception. Anything with an exploit is (at least by my definition), broken. If it's broken, then by all means fix it. Urpmi has that capability too, to skip certain files when you run automatic updates or downloads for manual updating. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Billing inbound calls per minute
Michiel van Baak wrote:
On 00:30, Mon 06 Feb 06, [EMAIL PROTECTED] wrote:
Hi,
Does anyone have a neat idea as how to
bill inbound calls per minute(second) real time?
I've been pplaying with astcc, but while
'billseconds' stays empty, 'billcost' has
strange behavior - either stays ampty
or takes ONCE the Connect fee(if I put one)
and keeps it that way no matter how long
the call is ...( if no Connect fee -stays empty).
i.e.
[inbound]
exten = 1122334455,1,Set(CALLERID(number)=${EXTEN})
exten = 1122334455,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
exten = 1122334455,3,Hangup
DeadAGI is for hungup channels, not for active channels.
That might be a problem.
Try this:
exten = h,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
ASTCC works fine here. The duration and billseconds fields in my cdrs as
well as ASTCC's cdr are filled.
I don't use the connect fee field though and all are set to 0.
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm really quite busy.
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Re: [Asterisk-Users] (newby) Is PING a good indicator of latency?
Garth van Sittert wrote: There is a good utility called iaxping to test IAX latency. Kind Regards Garth Another one is check_asterisk.pl plugin from the Nagios monitoring system. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] (newby) Is PING a good indicator of latency?
Cosmin Prund wrote: Being an ISP I have to disagree - ICMP traffic is rarely prioritized to look better Most large ISPs actually LIMIT ICMP traffic to counter ICMP flood DOS attacks. That does not make it a good indicator of a networks ability to support voip. Any cheap ATA will have jitter, delay, and packet loss counters - hook one up and get the real picture. If you do use ping to pre-qualify a link, grab a copy of pingplotter so you can tune the icmp packet parameters and packet rate, run it for a long time or several times during different timeframes to see if there are periods of congestion. Unfortunatelly good-old Windows based ping is all I can work with before I actually sign the contract with the ISP. I need to assess rather the service might be possible before I start throwing money at it (there's no such thing as an test account: I'll need to set up an account, pay the instalation fees, pay one month in advance for the service then pay for the disconnect!). Sooo... what might I find using ping? Is there an good ping that shows I can use the link for VoIP? I gues there's no such thing as bad ping showing I can't use the link because ISP's are limiting ICMP traffic? You might want to try the tools here: http://www.dnsstuff.com/ and here http://centralops.net/co/ -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Installing the none commercialintelg729codecsinto [EMAIL PROTECTED] 2.2?
ram wrote: Hi how about SIP friend to SIP Friend even it taking gsm ram Check the [general] section of your sip.conf Most likely there is an allow=gsm line there. Just allow=ulaw on your end so you can connect to voipjet. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Live CD?
Thczv F. Thczv wrote: I would love to run Asterisk on an old laptop, in a mostly solid state configuration, with no HD. The laptop is slow (Pentium 233), and I need PCMCIA support (for my network card). Are any of you aware of a live CD that might work? Thanks, Dave Take your pick: http://www.voip-info.org/wiki-Asterisk+Bootable+CDROM -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Connecting the two servers
satish Ahalawat wrote: Hi All, I want to setup the interconnectionm between two servers, both having sip clients behind firewalls. I want the calls from any of the servers to land on any of SIP clients on the other. I am looking for dial out plans with the sample configuration files . Thanks,. satish It is generally trickier to set up dual servers when both of them are behind firewalls. See here: http://www.voip-info.org/wiki-Asterisk+-+dual+servers -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Installing the none commercialintelg729codecsinto [EMAIL PROTECTED] 2.2?
ram wrote: Hi as per the list people guidence i have downloaded the Codec and installe my Pc is P4, but i have downloaded the P2.so file and copied in specific directory whe i see show translation i could able to see 30 i have configure AAH for VOIP JET connection when i try to make call out, its using only GSM even though i mention g729 in top list whats wrong ?? ram IMHO voipjet only allows g.711 ulaw. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Access Codes
Ronald Wiplinger wrote: Dakota wrote: I would like to setup Asterisk as follows: When users make inter-office calls they can dial the extensions, however if they want to make an external call, that they enter a code on their phone, before they external call can go through. We would like to give each user an access code, this way we can limit certain employees from making certain calls to certain places. What's the best recommendation for this? Use astcc (prepaid card system) with pin! You can give the rates all to free if you want. Give each one who is allowed a balance, if they do not have, they cannot call. You also would have a nice statistic for each call, bye Ronald Wiplinger Nice one Ronald. Keep in mind though that ASTCC has an issue with free calls. There's a workaround in the wiki. Not a show stopper though, even for what he wants. Why not use Authenticate()? You can then set each person to be under a certain context based on password or extension. Under each person's context, include other contexts that allow/limit their capability to call. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] RoadRunner
Dean Collins wrote: Yep I use iax and sip with time warner cable new york. Works fine. Dean IAX2 and SIP used here. All systems green. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] ASTPP
Ronald Ramos wrote: Hi, Has anyone implemented astpp? I'm configuring one right now and I have a problem on the pricelist. I followed the steps here http://www.astpp.org/index.php?n=ASTPP.Installation and created tables using http://www.astpp.org/index.php?n=ASTPP.Structure, but i didn't see there a query on creating pricelist table, can anyone help me on this please? Thank You Regards, Ronald Under Rates click on - Pricelists then Add... -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Nagios and Asterisk
Darrell Long wrote: Is anyone using Asterisk (and Festival) to make calls to appropriate persons (techs, etc. ) when Nagios generates a particular type of alert? If so, I would love to hear how people are doing it. Thanks, It's as simple as defining a host or service notification command consisting of a .call file generating script. The context it drops to will contain your festival routine. You'll find the pertinent info on voip-info.org You then give Nagios the proper contact host or service notification options so that voice won't nag you when everything's fine. I admit I no longer have festival running. It's a good idea at first but after you hear the voice one too many times for an error message you will find yourself switching to email/sms :) I just keep the option activated for the possibility that my internet provider will fail, but I use my own recorded message now. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] CDR reporting between two Asterisk servers
[EMAIL PROTECTED] wrote: Damon Estep wrote: Use cdr_mysql Log your CDRs to a common database Query as needed from either server using realtime() or from an external app Yeah, I thought about that. If it works how I think it would work though I would have two CDR records for one call though. I would have one record from the remote server and one from the local. Correlating one record with another could be a pain. There are several fields you can use to sync records. You could also set the accountcode and/or userfield to a value you generate per call.. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Nagios and Asterisk
Patrick wrote: snip I'm in Europe and pagers died here (if they ever lived) when bellbottoms went out of style but perhaps this link is of help: http://www.qpage.org/ Regards, Patrick Lol! Pagers were being improved upon and were still pretty much in use in Asia 4 years ago. The last one I played with had mp3 capabilities and games and a tiny thumbpad. Their extinction was caused by the widespread adoption of prepaid SIM cards for cellphones and SMS. It's funny though that the last time I saw a bellbottom there was about the same time. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Re: Anyone using verizon fios ftth for analog voice?Any echo?
LJ wrote: I am using Verizon FIOS to my home. I subscribe to a 5 MB down 2 MB up data package. I continue to pay for a standard voice line in addition to the broadband connection only for local calling, fax and emergency 911 use. I would if I could. I'm paying more to Time Warner for a fraction of the speed Verizon provides. Until Verizon is done with their tedious wiring of affluent neighborhoods, I can only dream. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Re: Anyone using verizon fios ftth for analog voice?Any echo?
Chris Mason (Lists) wrote: JP Carballo wrote: I would if I could. I'm paying more to Time Warner for a fraction of the speed Verizon provides. Until Verizon is done with their tedious wiring of affluent neighborhoods, I can only dream. Rich bastards. Come the revolution... lol! It's all relative. I know of an island in Asia where the people have to ride a ferry for an hour to the next island so they can call their relatives here in the USA. The internet cafe they use is paying the same amount in dollars as a typical yahoo dsl client for a line that's around a third of the speed. I'm pretty sure they're saying the same about us Chris. :) -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Asterisk Prepaid Solution
Ronald Ramos wrote: Hi All, Any solution on how I can implement prepaid billing on asterisk? But not the calling card type, just a simple Custome rwill buy credit, consume then buy again. Also, is there a solution for that when you combine asterisk with ser? Regards, Ronald Hi Ronald, Check the prepaid applications here for ideas: http://www.voip-info.org/wiki/view/Asterisk+Prepaid+Applications ASTPP, which is based on ASTCC is highly recommended. http://www.aleph-com.net/astpp Myself, I've implemented what you aim to do using ASTCC hooked to the shopping cart Virtuemart/Joomla. Customers register through Virtuemart/Joomla, then a card is created on ASTPP. When they buy a refill card through the store, their account is credited. As for * on ser, you may want to visit : http://www.voip-info.org/wiki-SIP+Express+Router -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Asterisk Prepaid Solution
Darren Wiebe wrote: JP Carballo wrote: Ronald Ramos wrote: Hi All, Any solution on how I can implement prepaid billing on asterisk? But not the calling card type, just a simple Custome rwill buy credit, consume then buy again. Also, is there a solution for that when you combine asterisk with ser? Regards, Ronald Hi Ronald, Check the prepaid applications here for ideas: http://www.voip-info.org/wiki/view/Asterisk+Prepaid+Applications ASTPP, which is based on ASTCC is highly recommended. http://www.aleph-com.net/astpp Myself, I've implemented what you aim to do using ASTCC hooked to the shopping cart Virtuemart/Joomla. Customers register through Virtuemart/Joomla, then a card is created on ASTPP. When they buy a refill card through the store, their account is credited. That's cool! I have been working on integrating ASTPP with oscommerce. I had hoped to have a release out for Jan 1st but I am behind. Check out the astpp demo. www.astpp.org Thanks Darren! The backend is really a hybrid of ASTCC and ASTPP's calling card part. I remember telling you about a month ago that I had osCommerce setup and was trying to get either ASTCC or ASTPP to work with it. I gave up, lol. I was spending far too much time patching osCommerce. Imho, Virtuemart/Joomla is much easier to customize and maintain. The only hurdle I see is Virtuemart's current inability to handle recurring charges or monthly payments. Once a module exists for that, I can use ASTPP's postpaid capabilities. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Asterisk Prepaid Solution
Adrian Carter wrote: Hey JP, I have just built a Joomla shopcart system for our general ordering system, and am scratching my head how to handle the backend post the order being created... can I ask, would you mind sharing some more detail on that particular aspect of your prepaid soultion ? Im not particularly interested for PrePaid, we run a Wholesale internet provider (ISP for ISP's) and want to just use Joomla/Virtuemart to manage orders from the ISPs for DSL tails and the like.. and maybe... asterisk stuff eventually (one thing at a time...) Thanks for your time Adrian Not at all, although this is OT, so I'll keep it simple. Goto /administrator/components/com_virtuemart/classes/ps_order.php Look for the order_status_update function Add your routines, based on $curr_order_status and $d[order_status] In my case, the routine I added to the function checks if an order is confirmed, then the relevant variables are passed on to ASTCC/PP on the * server. For further details, feel free to email me. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Re: Ben Higley Can you send us your AGI CDR logging application?
There's an perl AGI CDR logging routine in ASTPP as well. Ben Higley wrote: I have placed the custom-cdr-V1.0.tar for download http://www.itsngroup.com/software/asterisk/downloads/ Thanks Dear Ben, I've also the problems as Chris Mason, Can you send us your own AGI CDR logging application? Best regards, Jian Hong Guan France www.directcentrex.com ' I have written my own AGI CDR logging application. I set certain variables, and then extract them out of the channel using the agi. Run what I need with it, and then post it. Upon hangup (run a deadagi) and the channel still has the variables in it, so i finish up. It has worked rock solid. ./ben -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Getting Yoda unit to register all four ports
Hi Kevin, Now that makes perfect sense! Good call. I never got around to checking but the last time I worked with one of their tech help, he had an accel.com.tw email addy. No wonder the firmware commands were surprisingly familiar. As you probably know by now, they have never used *. The tech first tried to run * in their lab and failing that, later decided to remotely test around my unit. The units I bought were all originally H.323 before I switched to * I got the latest SIP firmware from Yoda and upgraded them myself. If you need help, let me know. I can re-enable my tftp server in Asia for you to use. The image file is it4mcs330.imz Consoleshow version Internet Telephony Gateway Version: gs020200140ena_0406mc Boot Loader Version: 4.13 RTOS Version:2.5.0/BE SIP Stack Version: 3.0.4.1 DSP image Version: 8.1.2.1. TSG Version: R8.0 Gateway (Build 4) Console Did you follow these same steps? Welcome to minicom 2.1 OPTIONS: History Buffer, F-key Macros, Search History Buffer, I18n Compiled on Oct 27 2004, 16:57:58. Press CTRL-A Z for help on special keys Incorrect password Password? Incorrect password Password? *** Consoleconfig erase The system configuration data will be erased from non-volatile storage permanently. Are you sure to erase it (y/n)? [n] y System configuration records erased from flash Consoledownload == WARNING == * Entering download mode will hang up all telephone connections * * and all the configuration settings will lose. * * Be certain all the configuration settings have been saved. * === Do you want to enter download mode now (y/n)? [n] y Boot loader V4.13 Mem 16b 16M Loading s/w upgrade utility. ** Internet Telephony Gateway TFTP Loader Ver 5.00 ** EITGLoaderstart Allocated 0x730200 bytes = 7360 KB for downloading files IP address of the TFTP server? [24.199.11.42] File name? it4mcs330.imz Starting download file: it4mcs330.imz 728K bytes read Download complete, file size = 751636 Application code downloaded successfully Do you want to write downloaded image to flash EEPROM (y/n)? [y] Press Enter to start flash EEPROM programming Flash EPROM programming on-going, BE CERTAIN NOT TO TURN POWER OFF... Programming Application code Sector 12 of 12 programmed Flash programming completed All sectors programmed successfully Download another file (y/n)? [n] EITGLoaderquit Do you want to restart the system now (y/n)? [n] y kevin ling wrote: Hi, I download the guide from yoda site. It's seems the original vendor is Accel AmiGate Elite 400 (http://www.accel.com.tw/frame/frame_age400.htm) I have one H.323 model and can't upgrade to SIP firmware. So what is your firmware version? Regards, Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists) Sent: Friday, December 30, 2005 5:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Getting Yoda unit to register all four ports I have a sample of the Yoda VG400 and I am having a devil of a time trying to get all four channels to register to Asterisk. I have an Asterisk 1.2.1 server. I have tried adding one at a time and rebooting it, but it stops after the first. http://www.yoda.com.tw/model.php?type=Enterprise_VoIPpname=VG400 Anyone had success with this? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] SoCal Users Group Meeting Schedule
SFVLUG also has a couple of * aware members. Sent a copy. Mike Fedyk wrote: Forwarded to OCLUG, LUGIE UUASC which have members that have expressed interest in asterisk. Mike Kerry Garrison wrote: The SoCal Asterisk Users Group will be meeting at the Heritage Park Public Library on the corner of Walnut and Yale in Irvine on the 3rd Thursday every month. The following dates are already secured: Thurs Jan 19 Thurs Feb 16 Thurs Mar 17 Irvine Heritage Park Library (949) 936-4040 14361 Yale Ave Irvine, CA 92604 Google Directions: http://tinyurl.com/9vq3e -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Configuration of two Asterisk server
Mantu Jha wrote: Hi I am have two Asterisk server at two different location one is having static ip 203.101.42.14 and other is having static ip 10.42.16.1 how can i integrate both so that i can use the others dial plan. It's all here on this page. http://www.voip-info.org/wiki-Asterisk+-+dual+servers You can use the switch statement on server 203.101.42.14 to make the server 10.42.16.1's dial plan available. You can then dial extensions registered on 203.101.42.14 from 10.42.16.1 You cannot use the switch statement on both servers though. Only one. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Configuration of two Asterisk server
JP Carballo wrote: Mantu Jha wrote: Hi I am have two Asterisk server at two different location one is having static ip 203.101.42.14 and other is having static ip 10.42.16.1 how can i integrate both so that i can use the others dial plan. It's all here on this page. http://www.voip-info.org/wiki-Asterisk+-+dual+servers You can use the switch statement on server 203.101.42.14 to make the server 10.42.16.1's dial plan available. You can then dial extensions registered on 203.101.42.14 from 10.42.16.1 You cannot use the switch statement on both servers though. Only one. Doh, slight correction. I should have said that you could then dial extensions registered on 10.42.16.1 from 203.101.42.14 -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Anybody having trouble terminating calls at Voxee? eom
Tom Lynn wrote: Their trunk works fine as of the time this email is sent. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Experience sharing on Planet VIP-450 + Asterisk
Jason Chan (jasonOfficial) wrote: What's next? Well... thanks to the buggy firmware and imcompatable standard with Asterisk... First of all, I can't deny that Planet VIP-450 does a good job in packetizing voice stream, the voice quality is really good and delay is really small. Also the hardware itself is quite robust, it seldom halt.. (the machine has been up for a few days). Also it is quite feature-rich, I can say. BUT I think there is quite a number of BUGS in the firmware! In order to see which kind of DTMF Relay it is using, I have done a packet analysing. When I try to pass SIP INFO type DTMF band to VIP-450, it replies "501 Unimplemented". Also when I try to pass DTMF from my POTS phone via the FXO port, only RTP payload can be seen in the packet captures. I DID suspect that it is RFC2833, because as far as I know RFC2833 did have the DTMF textx inside the RTP packet somewhere (seems header). But asterisk just simply did not regconize them (of coz I have set DTMFmode=rfc2833)! It is pretty strange that the user manual states "VIP handles DTMF Relay per SIP specification". So VIP-450 actually is using what kind of SIP specification? Sounds familiar.See below. How about using its Inband DTMF relay? This will certainly generate strange warning just like my case : improper ilbc frame size and tell me to use u-law to do DTMF even if I AM using G.711 u-law. It is seems that the DTMF tone generated by VIP-450 generate is kinda strange... So the final solution is, SIMPLY SWITCH OFF THE DTMF RELAY IN VIP-450. Please try to type "show coding" in console mode and you will see a lot of coding (codec) profiles. Most of them are with DTMF relay. Just switch off them by "set coding profile id dtmf_relay off" (please check with the manual). If you want to stop certain codec, just simply make that coding profile unusable in voice. For example, "set coding profile id voice off". If youonly turn on the profile withu-law, the SIP header it issues will just consist of 0x4 (ulaw) codec, not 0x105. This is what got my attention.Take a look at the commands that I use for the Yoda VG-400. If I'm not mistaken, they're exactly the same for the Planet.Same firmware libraries I presume. http://lists.digium.com/pipermail/asterisk-users/2005-August/120588.html If you notice, I also set dtmf_relay off.Too bad you didn't post any commands earlier, you would have saved a lot of time. Looks like we should compare notes. In mypoint of view, Planet isexpectingthis deviceisconnected to another VIP-450, not really for Asterisk or anything else, even not fora soft phone. Certainly this is not enough for everyone, at leastI can't do any IVR and something what a PBX should have (just like what I can do in Asterisk). I hope my experience will help anyone who is using VIP-450 with Asterisk, just like me. I have done Googling for 3 days but I can search for nothing related to this issue. Sorry for my poor written English. Cheers, Jason Chan, Hong Kong This is exactly what Yoda wants as well. I remember planning to buy a Planet unit during my love-hate relationship with Yoda and Asterisk but I'm now glad I didn't. It seems I would have had to tackle the same problems. I'm just happy the units I have work well. Thanks to Yoda's support. They're set on GnuGK rather than Asterisk so it was a first for me as well as them. Xie Xie -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Astcc help
Insider KT wrote: Hi. I am having some problem with Astcc. It works, but I would like to have IAX and SIP_Friends to work also. Hope someone here can help. In the web admin interface: I have put YES in : Enable Iax/Sip Friends DB (YES/NO) Then I have pressed create database and all is fine. The database is made, but without Iax_friends or Sip_Friends in it. If I then press Users_configure or Iax_Friends it says NOT CONFIGURED. I think I am missing something important here, but I don't know what. I find nothing after 3 hours of searching google. Does anyone here know what I am doing wrong ? Nothing. Both of them have never been fully implemented. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Fax detected, but no fax extension
[EMAIL PROTECTED] wrote: OK, Is Asterisk able to switch incoming calls according to fax or voice to the right extension . Which function detect incoming signal ? If you have faxdetect enabled in zapata.conf, (the default is off), asterisk listens for a fax tone when a call comes in. Now if you Answer() the call before you Dial(), it will switch to the fax extension. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Re: [Asterisk-biz] Help with learning Asterisk for the real world..
Robert Webb wrote: Hi all, hope this is not too off topic, but thought it fit better here than any of the other lists. I am in the IT field with some telecommunications background knowledge. Mainly learning from when my father worked for a telco and from keeping up the old Nortel PBX here at work. My issue is that where I am employed they have no plans on the drawing board of every trying to go VoIP, espicially from an open source solution as they are Microsoft finatics, because they will not justify the cost nor the resources to research by buying hardware and testing. So here is my delimma. I am running asterisk in my home running [EMAIL PROTECTED] I have no need for any of the advanced dial plans and setups like there are out there in the real world. But I want to gain more knowledge in setting up asterisk and programming it dial plans. I cannot afford T1 cards and other expensive gear off my persoanl budget. Can anyone give some insight as to a good direction to go to try and learn all this. I will be finishing my MIS degree in August 2006 and then want to concentrate on learing some C programming so I can understand what is going on behind the scenes. It is just lately between school and work I have not had time to pick up trying to learn C and asterisk. Sorry if too off-topic and no, I am not trying to get someone here to offer me a job. Only some insight into understanding and being proficient in setting up some real systems. Regards, Robert Webb Not to be crass but cross posting to the asterisk-users list anyway. This falls under the category of read the fine manual :) You're a step up from those of us who started with no shred of knowledge of the telco world. I recommend you look for an X100 clone on ebay. Should cost you less than the price of a movie ticket. Go to voip-info.org and read up the rest of what you need to do there. For show stoppers, you can ask live on #asterisk on freenode.net and of course, the asterisk-user's list. You can also refer to the asterisk CLI for help, the installed docs, the asterisk documentation project and of course, google. While learning C and becoming proficient enough to debug and contribute would be welcome, there's lots of testing going on and many other areas need improvement. On the day you convince your employer that they can have a voip setup for probably less than the price of their yearly MS license, we would be glad to hear about it here. Good luck! -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Realtime Replication of a Single File
[EMAIL PROTECTED] wrote: This sounds like a prime candidate for a database implementation. That way you can get very near real-time stats without the overhead of frequent cronjobs or polling. You number crunching computer would then just grab the data and crunch away. I'm just now getting started on using Asterisk in the more advanced modes (ie Realtime) so I do not know how to implement this, but I'm sure that it could be done. Ryan http://dev.mysql.com/doc/refman/4.1/en/replication.html -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Unable to compile zaptel / ztdummy
Insider KT wrote: Hi. I just upgraded my asterisk server to a better P4 2.6 Ghz. I thought everything went smooth until someone tried the Meetme. I seems the ztdummy won't compile on the new server. I am running Mandriva 2006 on the new server. After downloading in /usr/src/ I uncommented the #ztdummy to ztdummy in the Makefile and tried to modprobe ztdummy and got this error: FATAL: Module ztdummy not found. FATAL: Error running module install command for ztdummy I thought maybe it was something to do with the usb modules, but I have them. I did a lsmod|grep usb and got this: usblp120960 usbcore111612 4usblp,ehci_hcd,uhci_hcd Any ideas what I am doing wrong ? I'll presume you're using 1.2.0 There are 1.2.0 zaptel srpm packages on cooker that build fine. Haven't tried loading ztdummy itself though. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Include a variable from another file in configfiles
amaury BOSSE wrote: Thanks for your answer but I don't want to include a file, I only want to include a variable. Is it possible to execute linux commands like grep or top in a .conf file in order to parse a file and get a variable? Look into the System() command: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+System -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Include a variable from another file in configfiles
JP Carballo wrote: amaury BOSSE wrote: Thanks for your answer but I don't want to include a file, I only want to include a variable. Is it possible to execute linux commands like grep or top in a .conf file in order to parse a file and get a variable? Look into the System() command: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+System Oops, I missed the get a variable part. Your best bet is to use AGI. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Re: missing libpq.so.4
hrishikesh shrivastaw wrote: [cdr_pgsql.so]Dec 4 11:56:02 WARNING[3839]: loader.c:258 ast_load_resource: libpq.so.4: cannot open shared object file: No such file or directory Dec 4 11:56:02 WARNING[3839]: loader.c:440 load_modules: Loading module cdr_pgsql.so failed! Check that you've installed libpq. That's the package that includes the shared library libpq.so.4, at least on my system. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] sixtel
Bill Michaelson wrote: Just curious... Is there anyone out there who has given this outfit money and actually received any service from them? I only have a 1800 number from them, but no problems so far. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] IAX Configuration
Alejandro Vargas wrote: 2005/12/2, ram [EMAIL PROTECTED]: if you are using AMP go to trunk and start regitering your account Humm... what I'm trying to do, and what is this thread subject, is to connect asterisk-to-asterisk. Then... I go to trunks, create a new iax trunk, invent some user/password, use the ip of the other asterisk server, etc. Then I go to the other server... I supose I must also create an iax trunk, but... where do I create the user that I placed in the first server in order to validate it? Also I want to link the two asterisk boxes in order to be able to call extensions in each other and use the external lines, then I supose I must place it in from-internal context. -- Alejandro Vargas Perhaps you were thinking of doing this? http://www.voip-info.org/wiki-Asterisk+-+dual+servers To transparently call extensions on the second server, use switch in extensions.conf It's all there on that page. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] FW: CDR issues
Michaël Gaudette wrote: My biggest grip is I don't know where to troubleshoot this. Any log files I can look at? The message log in var/log/asterisk only shows that I am using simple CDR. Enable MySQL logging in /etc/my.cnf Restart mysqld. Then use your favorite tail program, (like multitail) to watch /var/log/mysqld/mysqld.log You should see the inserts into the cdr table or lack thereof. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] CallerID
Eric ManxPower Wieling wrote:
My point is that CALLERID(number) is ALWAYS the same as ${CALLERIDNUM}
so setting one to the other is pointless. It's like setting 2=2.
Same with the CallerIDName stuff.
___
Point taken.
Well, between our posts, (and a few minutes testing), I believe there
was enough info to figure out what was needed.
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm really quite busy.
___
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Re: [Asterisk-Users] Asterisk MySQL CDR - MySQL starting too late
Dennis Gilmore wrote:
snip
ok a little back round on runlevels.
Linux allows for up to 10 runlevels, 0-9, but usually only some of these are
defined by default. Runlevel 0 is defined as ``system halt''. Runlevel 1 is
defined as ``single user mode''. Runlevel 6 is defined as ``system reboot''.
Other runlevels are dependent on how your particular distribution has defined
them, and they vary significantly between distributions. Looking at the
contents of /etc/inittab usually will give some hint what the predefined
runlevels are and what they have been defined as.
ok so when you turn mysqld off on run level 3 and thats what you system runs
as mysqld will never start. the services selected for that run level are ran
when you enter that run level.
the order that they are run at is defined by a priority system. so you need
to make sure the priority of asterisk is such that is it started after
mysqld.
on my system mysqld has a priority of 64 and asterisk is 99 look
in /etc/rc3.d the files starting with a S are for startup and K for
shutdown. they start with lowest number up through highest number. that
last thing ran is /etc/rc.local so you could always put in
there /etc/init.d/asterisk restart to make sure its the last thing done.
Here's mine:
([EMAIL PROTECTED]:~)# ll /etc/rc.d/rc3.d/*{asterisk,mysqld}
147052 lrwxrwxrwx 1 root root 16 Nov 22 20:32 /etc/rc.d/rc3.d/S11mysqld
- ../init.d/mysqld*
135866 lrwxrwxrwx 1 root root 18 Nov 22 20:31
/etc/rc.d/rc3.d/S40asterisk - ../init.d/asterisk*
and the chkconfig default levels:
([EMAIL PROTECTED]:asterisk)# chkconfig --list | grep asterisk\|mysqld
mysqld 0:off1:off2:on3:on4:on5:on6:off
asterisk 0:off1:off2:on3:on4:on5:on6:off
just as they are defined in:
([EMAIL PROTECTED]:asterisk)# grep chkconfig /etc/init.d/{asterisk,mysqld}
/etc/init.d/asterisk:# chkconfig: 2345 40 60
/etc/init.d/mysqld:# chkconfig: 2345 11 90
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm really quite busy.
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Re: [Asterisk-Users] CallerID
Eric ManxPower Wieling wrote:
JP Carballo wrote:
Nitesh Divecha wrote:
Hello All,
My Asterisk is configured like a Soft Switch, I have many incoming
and outgoing traffic to different providers.
How can I forward CallerID to my providers. My providers are
complaining big time as the CallerID is set to Zero...
Please help!!!
Thanks,
Neal
In your provider's outgoing context, add the following before calling
out:
exten = _X.,1,Set(CALLERID(number)=${CALLERIDNUM})
exten = _X.,n,Set(CALLERID(name)=${CALLERNAME})
Why would he want to do that? It's pointless. It won't work. If the
provider is not getting Caller*ID then no amount of the above lines
will fix it.
You want to set the callerid using callerid=Robert Dobbs 666 in the
sip.conf, zaptel.conf, iax.conf, or whereever the calling device is
configured. You'll notice the lack of quotes. That's deliberate.
Ok, so? That's an 'If alright. I'm not an oracle. He didn't give us any
examples of what he wanted. I just grabbed the first thing I saw in my
extensions.conf and pasted it.
Granted, that wasn't very bright, but the answer he needs might not even
be what we think. He did mention incoming and outgoing, so my example
could be wholly wrong.
But so could yours. He might not be able, (or might not want), to
statically assign callerid.
How do we know he doesn't actually need to do exten =
_011.,1,Set(CALLERID(number)=(${EXTEN}) ?
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm really quite busy.
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Re: [Asterisk-Users] Asterisk MySQL CDR - MySQL starting too late
Eric Bishop wrote: I have: [EMAIL PROTECTED] ~]# chkconfig --list | grep mysql mysqld 0:off 1:off 2:off 3:on4:off 5:off 6:off [EMAIL PROTECTED] ~]# chkconfig --list | grep asterisk asterisk0:off 1:off 2:on3:on4:on5:on6:off What would you suggest I do? snip rant Holy crap, this kind of replying is getting me dizzy! Up, down, what next? Left and right? Why can't we just agree to delete all previous text, anyway we all have threaded readers...don't we? /rant chkconfig --level 3 mysqld off chkconfig --level 2 mysqld on chkconfig --level 2 asterisk off -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk MySQL CDR - MySQL starting too late
JP Carballo wrote: Eric Bishop wrote: I have: [EMAIL PROTECTED] ~]# chkconfig --list | grep mysql mysqld 0:off 1:off 2:off 3:on4:off 5:off 6:off [EMAIL PROTECTED] ~]# chkconfig --list | grep asterisk asterisk0:off 1:off 2:on3:on4:on5:on6:off What would you suggest I do? snip rant Holy crap, this kind of replying is getting me dizzy! Up, down, what next? Left and right? Why can't we just agree to delete all previous text, anyway we all have threaded readers...don't we? /rant chkconfig --level 3 mysqld off chkconfig --level 2 mysqld on chkconfig --level 2 asterisk off I forgot to add that you should get this: ([EMAIL PROTECTED]:asterisk)# chkconfig --list | grep asterisk\|mysqld mysqld 0:off1:off2:on3:off4:off5:off 6:off asterisk 0:off1:off2:off3:on4:off5:off 6:off -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID
JP Carballo wrote:
Ok, so? That's an 'If alright. I'm not an oracle. He didn't give us
any examples of what he wanted. I just grabbed the first thing I saw
in my extensions.conf and pasted it.
Granted, that wasn't very bright, but the answer he needs might not
even be what we think. He did mention incoming and outgoing, so my
example could be wholly wrong.
But so could yours. He might not be able, (or might not want), to
statically assign callerid.
How do we know he doesn't actually need to do exten =
_011.,1,Set(CALLERID(number)=(${EXTEN}) ?
This is bad form replying to oneself, but that was a typo I previously
posted: exten = _011.,1,Set(CALLERID(number)=${EXTEN})
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm really quite busy.
___
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Re: [Asterisk-Users] CallerID
Nitesh Divecha wrote:
Hello All,
My Asterisk is configured like a Soft Switch, I have many incoming
and outgoing traffic to different providers.
How can I forward CallerID to my providers. My providers are
complaining big time as the CallerID is set to Zero...
Please help!!!
Thanks,
Neal
In your provider's outgoing context, add the following before calling out:
exten = _X.,1,Set(CALLERID(number)=${CALLERIDNUM})
exten = _X.,n,Set(CALLERID(name)=${CALLERNAME})
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm really quite busy.
___
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Re: [Asterisk-Users] IAX and Firewall
Joseph wrote: VOIPJET - one of their servers was down, so switching the server IP solved the problem. Is it possible to specify two or three alternate server in iax.conf just in case once goes down so the system will try alternate server connection? Yes you can. The logic for cascading servers is in extensions.conf. However, registering the same account at the same time on different servers is something you'll have to ask VOIPJET about. You could keep your current account on one server, open another account and register it on the other server. I've tried connecting to Philippines from both providers and it appear to me something must be wrong with the line to the Philippines, the connection will not go through. the connection will not go through can mean a failure at the Philippine end or at your voip provider. You might want to be a little more specific. What area codes are you calling? I'll see if I can test them from my end. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UK Pounds and pence prompt wanted
Obelix wrote: Is there a .gsm file for announcing UK pounds and pence after the credit remaining prompt, besides the dollar and cents file? /Obelix http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+international I peeked into the archives from: http://www.desktop2door.com/asterisk/ and http://www.g7ltt.com/VoIP/vmfiles.html Found pound and pounds but no pence. I could have missed it though. You could also add your own voice to the UK male voice archive. That's what I did when I didn't find philippine(s).gsm My voice is nowhere near Allison's though. O.T. Is the Asterisk the Gaul comics still in circulation? It's been years since I read the series... -- JP Carballo http://www.netfone2x.com Bringing the world closer. Programmers confuse Christmas and Halloween because DEC 25 = OCT 31. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] emacs syntax/keyowrd highlighting: asterisk-mode
Dave Grey wrote: Speaking of glaringly unworkable, like a numb-skull I edited and tested the thing with default my green-on-black color scheme. I happened to open something up in a raw black-on-white xterm and realized that I had created a nightmare. I have made the appropriate changes, so if you looked at it and said, Hey, this is horrible..., then my apologies and give it another look. Same url and filename above, updated version noted in the comments. lyd Is it? :) I work in a myriad of colored terms. green-on-black is the default for my local machines, other schemes are used identify remote machines... Not a show stopper for me. I have a few screen sessions open for ages. Sometimes, the only time I see a prompt is when I do M-x shell :) I'll check out your changes the next time I need to edit a .conf file. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] emacs syntax/keyowrd highlighting: asterisk-mode
Good work Dave! I suggest you post this in www.voip-info.org for future emacs/asterisk users. Dave Grey wrote: Hi all. I've just begun learning *, and as my dialplans and macros have gotten more complex I started wishing for a way to more easily follow the flow of various Goto, GotoIf, Dial, and etc. commands, especially when trying to use priority n and labels rather than numbered priorities. Toward this end, I hacked together an emacs general/minor mode for basic font-locking (syntax shading) support. Feel free to grab it here: http://homepage.mac.com/lydanynom/asterisk-mode.el.zip Feel even freer to suggest or make changes, or point me to your own major or minor asterisk-mode. A quick search before I started did not find any such thing, but in retrospect I probably should have just asked here rather than trying to reinvent the wheel. Lisp regexps give me a headache, and this was my first and likely only foray into writing emacs modes, so don't expect too much. For now, though, it seems to do what I want it to, which is, as I commented in the file: ;; Assumes applications and functions will be in the same case they appear ;; in the documentation: ;; Dial, BackGround, DISA, CALLERID, Playback, etc. ;; Assumes comma (rather than pipe) will be used as the argument delimiter, ;; and that arguments to applicatins will always be enclosed in (): ;; Goto(context,123,1) ;; ;; The principle goal of this mode is to make it easier to follow the flow ;; of a complicated dialplan with many Goto and other context-jumping apps ;; geing called. Extensions, priorities, and labels are handled in a ;; way that attempts to higlight the (potentially) operative bits while ;; subdueing the bits that can't be directly addressed. Secondarily, the ;; applications, functions, variables, and config keywords are highlighted as well, ;; each in their own face. It is intended primarily for use with extensions.conf, but it is effective in all of the configuration files. Of course, it seems that this is rapidly become moot with the advent of ael, but I haven't ventured there yet, myself. Anyway, use and enjoy, comment, or criticize as you feel is warranted. Big kudos to this list and the * community as a whole. I have learned a ton in the last two weeks, thanks to all of you. lyd -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sixtel DID
Yu Safin wrote: has anybody tried to register with Sixtel to obtain a DID? I signed up 9/27 and I am yet to receive my DID. Also, how do I change my aix.conf to connect to Sixtel? I have a userid and password but I don't have details about all the parameters. Log in to your account at http://control.sixtel.net and click on DIDs You should then see 3 sections: Numbers assigned to you: Telephone Numbers (DID) and Toll Free numbers Click on the link under Telephone Numbers to get a DID assigned to you. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTCC - INUSE Flag
As soon as the number is entered. Consider the scenario where two people dial in with the same card number. Once one person has entered a valid number, you want to let the other party know as soon as possible that the account is in use. If the card isn't actually in use, it's still best to notify the caller at the earliest. This shouldn't happen of course. Darren Wiebe wrote: Thanks. I have a question for the mailing list in general. Where should the card get marked as in use? Should it be as soon as you enter the number or should it be when it dials? I don't know for sure. snip -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTCC - INUSE Flag
It works fine here. I'm using 1.0.8 HEAD from 20050529, MySQL 4.1.11, and perl-asterisk-perl-0.08 Darren Wiebe wrote: Any developers out there that would like to look at this one? It works fine on Asterisk CVS-v1-0-09/22/05-22:23:34 on a i686 running Linux but it does not work on the 1.2 betas. I agree that the number should be set aside then. I wonder what the problem is. Darren JP Carballo wrote: As soon as the number is entered. Consider the scenario where two people dial in with the same card number. Once one person has entered a valid number, you want to let the other party know as soon as possible that the account is in use. If the card isn't actually in use, it's still best to notify the caller at the earliest. This shouldn't happen of course. Darren Wiebe wrote: Thanks. I have a question for the mailing list in general. Where should the card get marked as in use? Should it be as soon as you enter the number or should it be when it dials? I don't know for sure. snip snip -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calling Card Application
Il Neofita wrote: I am looking for a calling card application which is able to advise me during a call when the credit is almost finish. For examples 1 minute before the end of the credit. Thank you I and quite a few others use AstCC: http://www.voip-info.org/tiki-index.php?page=ASTCC -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] forward incoming analog call to SIP?
Dave Williams wrote: I'm trying to setup a demo where my Asterisk box with a TDM01B (FXO) answers an incoming call and forwards that call to a SIP softphone (X-lite.) Seems all is built/installed okay: # ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. I'm pretty new at this and the extensions.conf file is eating my lunch. Here are my various config files - maybe someone will take pitty on me and point me in the right direction. Needless to say, Asterisk pukes on my dialplan when I try and startup. . (zapata.conf) context=analog signalling=fxs_ks language=en channel = 1 (sip.conf) [sip_proxy] For incoming calls only. Example: FWD (Free World Dialup) type=user context=sip [xlite1] Transmit Silence=YES type=friend regexten=1234 ; When they register, create extension 1234 username=xlite1 callerid=Jane Smith 5678 host=dynamic allow=ulaw allow=alaw (extensions.conf) [general] static=yes writeprotect=no [analog] include=test include=local [sip] include=test include=local [test] 611,1,echo_test [local] exten = 1237,1,Dial(SIP/xlite1,10,t) try this: [analog] exten = s,1,Answer exten = s,2,Dial(SIP/xlite1,10,t) exten = s,3,Hangup of course, this is an unconditional forward... -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PC for 8 line system
Chris Gamble wrote: I have 2 TDM04b cards currently running in an asterisk at home box that I am ready to replace with the CVS version of asterisk. What I am looking for is thoughts / recommendations. I want to move this to a small form factor ( shuttle ) machine and was wandering what expeience / advice there was for this? I have seen the incompatible motherboard list at digium ( and in fact I think my current machine is on the list ! ), but wanted to know what others are doing for small form factor tdm setups? Thanks, I'm curious, what shuttle model has 2 pci slots? I have a TE110P running on a shuttle SB61G2 2.8GHz P4 w/ 512 RAM and * 1.0.8. I haven't put it through its paces yet, but I will as soon as our remote office gets their server. I guess the usual pointers apply (i.e. don't share interrupts, etc.) and the wiki was informative when I first set mine up. Didn't even have to ask then and it has been functioning well since. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P FXO channel hookstate always Offhook outbound digits sent before provider dialtone
Rich Adamson wrote: I have an [EMAIL PROTECTED] 1.3 server (Asterisk 1.0.9) and recently added a TDM400P with (1) FXO card on port 4. Inbound calls are always successful but outbound calls fail 75% of the time with intercept messages from my dial tone provider that include we're sorry, your call did not go through, and we're sorry, when placing a local call it is now necessary to dial an area code and the 7-digit number. Add a w in the dial string within extensions.conf. The w adds a short delay before sending the dtmf to the telco, which is likely happening too fast for the telco switch. Is that undocumented? The wiki entry for the Dial() command says: w: Allow the /called/ user to start recording after pressing *1 or what defined in features.conf (Asterisk v1.0.x) To add a short delay to one of my remote FXO lines, I added A(silence/2) to the dial string. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Announcement to called party
Steven Hall wrote:
I am trying to send an announcement to the called party using the A(x)
parameter in Dial, however, the message is not being played. There is
a pause between the Dial command being executed and the call being
connected to the calling party of the same length as the announcement
.gsm file, but the message itself is not being played. (I have tried
this and timed it with different announcement files). The problem
occurs whatever the called party technology.
I am running Asterisk version 1.0.8. A typical exten line in
extensions.conf is
exten = s,1,Dial(MGCP/aaln/[EMAIL PROTECTED],20,mA(${MES}))
where MES is a global variable containing the name of the .gsm file in
/var/lib/asterisk/sounds. Putting the actual name in makes no
difference and the same problem occurs with ZAP and SIP channels.
Does anyone have a solution to this problem? Is there possibly
something wrong with my configuration
Steven Hall .
[EMAIL PROTECTED]
What does the CLI say?
Does it show Playing 'value-of-MES' (language 'en')?
I'm using 1.0.8 here and I have no problems using A(x) in my dial
strings in either ZAP or SIP channels.
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm really quite busy.
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Re: [Asterisk-Users] *** Asterisk 2.0 Stable release out now
Kevin Walsh wrote: Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote: r: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Use with care and don't insert this by default into all your dial statements as you are killing call progress information for the user. Really, you almost certainly do not want to use this. Asterisk will generate ring tones automatically where it is appropriate to do so. r makes it go the next step and additionally generate ring tones where it is probably not appropriate to do so. I think you might have replied to the wrong article. That's his signature. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Yoda VG-400 and Asterisk Settings
reg add 204 500 111.222.333.444 5060 204user 204pass OK 111.222.333.444 is your * server IP Consoleset port 0 cid number 201 OK Consoleset port 0 cid name FXO1 OK Consoleset port 0 cid number 202 OK Consoleset port 0 cid name FXO2 OK Consoleset port 0 cid number 203 OK Consoleset port 0 cid name FXO3 OK Consoleset port 0 cid number 204 OK Consoleset port 0 cid name FXO4 OK Consoleset sip auto_reg on OK Lets delete the default codec order on ports 0-3 Consoleset port all prof_bit all 0 OK And add/prioritize codec 6 (G.711 PCMU) Consoleset port all prof_bit 6 1 OK Disable DTMF Relay for codec G.711 Consoleset coding 6 dtmf_relay off OK Then tell ports 0-3 to use G.711 Consoleset port all voice_prof 6 OK Consoleset port all rxgain -3 OK Activate the current configuration... Consoleconfig activate OK Then store all of the changes in NVRAM Consoleconfig store Restart the unit... Consolenet reset == WARNING == * Restarting the system will hang up all telephone connections* * and all the configuration settings will lose. * * Be certain all the configuration settings have been saved. * === Do you want to restart the system now (y/n)? [n] y Upon rebooting...check registration by typing: Consoleshow sip SIP Addr Configuration: SIP Signaling port = 5060 RTP Voice port = 2070 reg_num: 201 Registrar_ID 1: Registered registrar: 111.222.333.444 5060 expires: 500 name: 201userpasswd: 201pass reg_num: 202 Registrar_ID 2: Registered registrar: 111.222.333.444 5060 expires: 500 name: 202userpasswd: 202pass reg_num: 203 Registrar_ID 3: Registered registrar: 111.222.333.444 5060 expires: 500 name: 203userpasswd: 203pass reg_num: 204 Registrar_ID 4: Registered registrar: 111.222.333.444 5060 expires: 500 name: 204userpasswd: 204pass 201user201pass 202user202pass 203user203pass 204user204pass Domain name server = 11.22.33.44 Info switch is off nat_call is off auto_reg is on outboundproxy : None stunserver : None stun_call is off alt_registrar : None allow_num : None. Console Call in from 202/203/204, dial 0112011234567 to route the call to *, (optionally do whatever you want) then have * pass 1234567 to 201... -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 error when trying to start Asterisk
When running ldconfig -v, did you see it find the files under the directory /usr/local/lib? If not, edit /etc/ld.so.conf with your favorite editor and add /usr/local/lib in a new line. Then rerun ldconfig -v. Check that the libpt* files were seen. kurt turner wrote: yes.. i have the following IPD:/usr/local/lib# ls firmwarelibpt_linux_x86_d.so.1.5 libpt_linux_x86_r.so.1 python2.3 libpt_linux_x86_d.solibpt_linux_x86_d.so.1.5.2 libpt_linux_x86_r.so.1.5 libpt_linux_x86_d.so.1 libpt_linux_x86_r.so libpt_linux_x86_r.so.1.5.2 I found ldconfig under root /sbin/ldconfig when you say run ldconfig what are you saying? ldconfig -v .. right? if so I did that and I still get the h323 error listed below when firing up * anymore ideas? */Derek Whitten [EMAIL PROTECTED]/* wrote: does libpt_linux_x86_r.so.1.5.2 exist on your machine? maybe try running ldconfig or if that file is in a non-standard location, maybe add that path to ld.so.conf and then run ldconfig again On Wed, 2005-08-10 at 08:09, kurt turner wrote: Asterisk has been working fine for me for several weeks using MGCP to a Adit600 for intra office calling. I have recently loaded h323 and the following errors occurs when starting asterisk. [chan_h323.so]Aug 10 09:09:18 WARNING[7824]: loader.c:258 ast_load_resource: libpt_linux_x86_r.so.1.5.2: cannot open shared object file: No such file or directory Aug 10 09:09:18 WARNING[7824]: loader.c:440 load_modules: Loading module chan_h323.so failed! Ouch ... error while writing audio data: : Broken pipe I loading the following : Open H.323 v1.12.2, PWLib v1.5.2, expat-dev-1.95, expat-1.95, openssl-devel-0.9.6b, openssl-0.9.6b - per the read me files I did try the advsie previously give here -- http://lists.digium.com/pipermail/asterisk-users/2003-April/011019.html but that didn't work for me. I thinking I may have loaded these in the incorrect directories.. here is where they are located in (slash root) - is the following openh323 and pwlib located in root - /root/usr/src/asterisk/channels is the following - chan_h323.c - h323 - Would these being in different areas be the cause? Should I move these or remove them then reinstall them? Sorry for my noobness as I'm learning Linux for Asterisk.. I'm really a class 5 voice guy tryin to keep up!! Thanks, Kurt -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [OT] Yoda Communications' VG-400 (4 Port FXO and 2 Port FXO/2 Port FXS)
Chris Mason (Lists) wrote: I replied to your query earlier Re: FXO Gateways with a quote. Feel free to write to me for specifics. I did see where you mentioned a price of $325, that's fine, can I get two units please? Ship fedex international economy to Anguilla, British West Indies What's the voltage in the British West Indies? -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk to PSTN
Edwin Lam wrote: hi folks. i'm planning to connect * to 120 POTS line. i've done some research on FXO cards but unfortunately most manufacturers only make 4 ports/card. the most i've found is 12 ports. so do i have to get 10 of these cards and setup 3 Asterisk servers (assuming each have 4 free PCI slots) link them together with some insane dialplan? or is there an easier way? any suggestions? comments? remarks? parameters? thx. Get a T1/E1 card. They come in 4 port (TE4xxP series) and 1 port (TE110P) flavors. You then connect each of those ports to a 24 FXO port channel bank. So, to do the math, get a 4-port card and a 1-port card taking up only 2 PCI slots. You attach 5 24-port channel banks to them. And no, you don't need to punch a gaping hole in your wall snaking 120 phone lines into your office. I'm sure there are better scenarios than this. I just scaled up a setup I have. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
