Re: [asterisk-users] Asterisk on Ben NanoNote?
At 1:42 PM on 10 Aug 2010, Gilles wrote: > I just read an article on the tiny Ben NanoNote: > > http://en.qi-hardware.com/wiki/Ben_NanoNote > > As CPU, it uses a "JZ4720 366 MHz MIPS compatible processor from > Ingenic Semiconductor Co", and it runs Linux. > > Does someone know if Asterisk has been ported to that platform? The real question is, does it have PCI slots for Digium cards? And where do I get one of those HUGE coke cans?! ;-) -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip add header
At 8:08 AM on 28 Jun 2010, Jerry Geis wrote: > It seems that for local channels (asterisk 1.4.33) the variable > Variable: SIPADDHEADER="Alert-Info: Ring Answer" > (call polycom phones and ring then auto answer) > > Is ignored, Is this just an oversite or is there some reason? > > It works fine with I call the SIP phone directly - however - > when I first call the Local channel - then Dial the SIP phone > the SIPADDHEADER doesnt seem to do anything. Have you tried adding an underscore? Set(_SIPADDHEADER="Alert-Info: Ring Answer") Without the underscore, the variable won't be inherited by the Local channel. Also, look up the /n option to the Local channel. That may affect it, but I can't say how off the top of my head. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can one adjust the voicemail-menu when using VoiceMailMain() ?
At 4:04 PM on 09 Jun 2010, Jonas Kellens wrote:
> I have commented out case 5, case 2 and case 3, leaving case 1,
> 4,6,7,8,9.
>
> But when I press "1" on the menu, I hear: " I'm sorry, I did not
> understand your response"
Looks like someone broke the first rule of Optimization Club[1]. I
think you need to copy these two lines from case 5 into case 1:
cmd = vm_browse_messages(chan, &vms, vmu);
break;
It just so happens that cases 1 and 5 run the same command, so whoever
wrote it took advantage of that, optimizing the size of the binary
while reducing maintainability.
[1]
http://perlbuzz.com/mechanix/2008/02/the-rules-of-optimization-club.html
> if (play_auto) {
> cmd = '1';
> } else {
> cmd = vm_intro(chan, vmu, &vms);
> }
>
> vms.repeats = 0;
> vms.starting = 1;
> while ((cmd > -1) && (cmd != 't') && (cmd != '#')) {
> /* Run main menu */
> switch (cmd) {
> case '1':
> vms.curmsg = 0;
> /* Fall through */
> /* commented out from here
> case '5':
> cmd = vm_browse_messages(chan, &vms, vmu);
> break;
>
>
>
> if (vms.repeats > 3)
> cmd = 't';
> }
> }
> if (cmd == 't') {
> cmd = 0;
> vms.repeats = 0;
> }
> break;
> commented out till here */
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Re: [asterisk-users] pattern containing an asterisk
At 8:04 PM on 12 May 2010, Robert Wagner wrote: > i need to match a number with like 03012345678*0 or 03012345*9 > I tried _X.*X and _X!*X but both are maching 03012345678 too, ignoring > that *X is required at the end. > The interesting part is that like expected _X*X is matching only > numbers like 1*1 and not 11 The . in a pattern is meant only to be used at the end, to match any remaining characters. The *X after the dot in your pattern is just being ignored. Have you checked for warnings in your log? I'm not sure if Asterisk issues a warning on that, but I think it should. One thing you could do is make one pattern for each possible length. e.g.: _XXX*X and _*X If you need it to be variable length, I think you would need to use the Read application instead of standard dialplan matching. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan behaviour
At 4:45 PM on 08 Mar 2010, equis software wrote: > I have this > > [TRONCAL-SIP] > exten=>225/91,1,Answer > exten=>225/91,2,Echo > exten=>225/91,3,Hangup > > exten=>225/92,1,Answer > exten=>225/92,2,Playback(conf-invalid) > exten=>225/92,3,Hangup > [...] > Dont work > > If I add this rule > exten=>225,1,Answer > > Works ok I suspect it's because when the call first comes in, asterisk doesn't have the callerid info yet (it comes after the first ring). So asterisk tries to route the call to a callerid-nonspecific dialplan entry, and simply fails when it doesn't find any. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk listens on all NICs
At 2:50 PM on 16 Feb 2010, Landy Landy wrote: > Hello List. > > I am puzzled and how asterisk listens to calls or connections from > clients. When I do a netstat -nat I don't see asterisk listening on > port 5060. Now, I'm testing a server with three network interfaces: > two to the internet doing load balancing and the other to our LAN. > I would like asterisk to only accept connections coming from our LAN > but, can't find where to configure this. Set bindaddr in sip.conf. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issue with trying to dial two different servers at the same time.
At 9:04 AM on 16 Feb 2010, Steve Anness wrote: [...] > exten => 12109,1,Dial(iax2/castle-rock/109&iax2/colo/17128,20) > > This has one flaw, if for whatever reason his home phone isn't > connected, like it isn't now, than when I call 12109 I get his home > voicemail as it picks up right away. > > How can I program this so that even if the voicemail on 17128 or 109 > pick up it still rings the other phone? I would give app_followme [1] a shot. It requires the callee to hit "1" to connect. That way, an answering machine would never make it through... unless the callee puts the tone for a "1" in his voicemail greeting. :-) You might also consider AMD [2] (answering machine detection), but I don't know much about it. [1] http://www.voip-info.org/wiki/view/Asterisk+cmd+FollowMe [2] http://www.voip-info.org/wiki/view/Asterisk+cmd+AMD -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.2 : global vars not read/set after #include w/ globals
At 6:48 PM on 11 Feb 2010, sean darcy wrote: > OK, now clear on suffix v. prefix ( Doh! ) and having RTFM, > > I have extensions.conf: > > [general] > > #include exts/gvoice.exten.conf [...] > [globals] > pstnline => DAHDI/4 > ... > > and exts/gvoice.exten.conf: > > [globals](+) > test-global => need-a-plus-sign > . [...] > Category addition requested, but category 'globals' does not exist, > line 1 of /etc/asterisk/exts/gvoice.exten.conf Your #include should come after your [globals] section in extensions.conf. From doc/configuration.txt: "The content of the other file will be included at the row that the #include statement occurred." So putting your include before your main [globals] puts the [globals](+) in first. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] billing based on local access number
At 4:02 AM on 10 Feb 2010, umesh maharjan wrote:
>
> Hi all,
>
> I am configuring asterisk as a prepaid calling card. I am getting
> different local rate from my ISDN provider e.g 0.002 for landline
> and 0.13 for mobile etc. In this case I thing I have to say my
> asterisk/a2billing to bill based on local access number. so How can I
> retrieve called number (eg. 03-6832-1040 and 0120-272-060 is our
> ISDN PRI access number) to my asterisk server so i can trigger
> different rates.
The number the caller called to get to you should be passed to Asterisk
as the inbound extension. So, in your incoming context, you can
provide different extensions for the different incoming numbers. Or
you can catch everything with the "_X." pattern and use the ${EXTEN}
variable to check the number in your dialplan.
One thing to note is that it doesn't always pass the whole number. I
have two PRIs from different providers; one of them passes all 10
digits, but the other one only passes the last 4, and for some reason
with one of our numbers that ends in "9977" the PRI passes "2977". You
can either ask your provider what they pass, or you can just make test
calls and log the value of the ${EXTEN} variable with Verbose() calls,
something like this:
[incoming]
exten => _X.,1,Verbose(Incoming call to ${EXTEN});
exten => _X.,n,Playback(welcome);
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http://www.lodgingcompany.com/
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Re: [asterisk-users] Trouble getting feature codes to work
At 9:08 PM on 21 Jan 2010, hugolivude wrote: > The call works fine and the CLI tells me that ** is an active feature: > > Builtin Feature Default Current > --- --- --- > Pickup*8 *8 > Blind Transfer# ## > Attended Transfer *2 > One Touch Monitor *1 > Disconnect Call * ** > Park Call > > When I press ** during a call though, nothing appears in the CLI > (verbosity = 4). I do it very quickly so I don't believe timeout is > an issue. > > I'd be grateful for any troubleshooting tips. Try different values of dtmfmode (rfc2833, inband, info) in sip.conf for the SIP peer that you call in from. Asterisk is probably monitoring the wrong method for DTMF. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] test case with queues and system()
At 3:09 AM on 21 Jan 2010, __ wrote:
> On Wed, Jan 20, 2010 at 10:18 PM, C. Chad Wallace
> wrote:
> >
> > At 5:59 PM on 19 Jan 2010, __ wrote:
> >
> >> Test case:
> >> We have e1 trunk and multi-channel sip line. Clients waiting in the
> >> queue, which can handle 30 clients. They listen mellody and their
> >> position, while waiting. The system can handle only 5 clients at
> >> the moment. As soon as client is the first he hears a background
> >> and then if he inputs any number, asterisk executes system command
> >> like wget example.org/?p= and call terminated.
> >>
> >> I'm reading asteriskbook but can't connect all together right now.
> >
> > I think you'll have to use the Local channel as your queue member,
> > like this (in queues.conf):
> >
> > member => Local/s...@systemcommand
> >
> > And then in your dialplan (extensions.conf) you'd have something
> > like this:
> >
> > [systemcommand]
> > exten => s,1,Background(press-a-key)
> > exten => s,n,Read(INPUT_NUMBER)||1)
> > exten => s,n,System(wget http://example.org/?p=${INPUT_NUMBER})
> > exten => s,n,Playback(goodbye)
> >
> >
> > Please note, these are only examples to get you started, and they
> > probably won't work without some tuning.
>
> Thank you, it helped a lot.
> Now i have only one thing - how can i tell asterisk to work with 5
> clients? I have to make 5 members?
Maybe... But I think the Local channel queue member will accept
multiple callers at the same time, so you could use GROUP_COUNT in your
dialplan to limit it:
[systemcommand]
exten => s,1,GotoIf($[${GROUP_COUNT(systemcommand)} < 5]?continue)
exten => s,n,Busy()
exten => s,n(continue),Set(GROUP()=systemcommand)
exten => s,n,Background(press-a-key)
exten => s,n,Read(INPUT_NUMBER)||1)
exten => s,n,System(wget http://example.org/?p=${INPUT_NUMBER})
exten => s,n,Playback(goodbye)
It returns Busy if there are already 5 calls being serviced.
Also, you could replace the 5 above with a variable, and set that
variable in your globals, so it's easier to maintain later.
--
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The Lodging Company
http://www.lodgingcompany.com/
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Re: [asterisk-users] test case with queues and system()
At 5:59 PM on 19 Jan 2010, __ wrote:
> Test case:
> We have e1 trunk and multi-channel sip line. Clients waiting in the
> queue, which can handle 30 clients. They listen mellody and their
> position, while waiting. The system can handle only 5 clients at the
> moment. As soon as client is the first he hears a background and then
> if he inputs any number, asterisk executes system command like wget
> example.org/?p= and call terminated.
>
> I'm reading asteriskbook but can't connect all together right now.
I think you'll have to use the Local channel as your queue member, like
this (in queues.conf):
member => Local/s...@systemcommand
And then in your dialplan (extensions.conf) you'd have something like
this:
[systemcommand]
exten => s,1,Background(press-a-key)
exten => s,n,Read(INPUT_NUMBER)||1)
exten => s,n,System(wget http://example.org/?p=${INPUT_NUMBER})
exten => s,n,Playback(goodbye)
Please note, these are only examples to get you started, and they
probably won't work without some tuning.
A good resource to learn more about applications is
http://voip-info.org/. Here are a few links:
http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+System
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Read
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Curl
You can also use 'core show application System' and such on the Asterisk
CLI.
GLHF!
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Re: [asterisk-users] Please remove me from the mailing list.
At 2:01 PM on 07 Jan 2010, Dan Journo wrote: > I've never seen that in Outlook. What client do you use? Claws Mail provides a "Mailing-List" sub-menu under the Message menu, which includes Post, Subscribe and Unsubscribe options, among others. It's amazing what paying attention to standards can do for you... > Steve Totaro wrote: > > read your posting and it will tell you haw to remove yourself. > > > > On Thu, Jan 7, 2010 at 10:49 AM, Rick Dean > <mailto:ric.d...@gmail.com>> wrote: > > > > Can I be taken off the mailing list please. > > > > Thanks. > > rick > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > And a proper mail client will also parse the headers and provide > unsubscribe information/buttons based on that... -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Asterisk Dictate?
At 12:36 PM on 30 Dec 2009, hadi motamedi wrote: > Dear All > Can you please give me more hint on how Asterisk Dictate() works? > Thank you http://lmgtfy.com/?q=asterisk+dictate -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call queue with external numbers??
At 5:01 PM on 22 Dec 2009, Oguzhan Kayhan wrote: > Hello, > Our asterisk is connected to an ericsson pbx by PRI. > What i want is the asterisk clients should call operator numbers by > dialing 0 > > But, when a call is made to ericsson via number 0, it assumes that the > call is made from outside, so it doesnt allow to be dialed. > There are 3 real operator extensions which is grouped by ericsson for > operators. Lets assume 1112 1113. > > What i want to know is, is there a way for me to create such group in > asterisk and add that external extension numbers which should be > dialed by order, or by 3 rings at a time etcso that i can create > that operator group on asterisk side also. > > PS: I can call real extensions on ericsson without a problem. How about this: exten => 0,1,Dial(DAHDI/G1/,18) exten => 0,n,Dial(DAHDI/G1/1112,18) exten => 0,n,Dial(DAHDI/G1/1113,18) ...where DAHDI/G1 is the PRI connected to the ericsson (group=1 in chan_dahdi.conf), and 18 seconds is 3 rings. You might be able to use Queue(), but I'm not sure if you can add a hunt group and external number as a queue member--you might have to use the Local channel for that. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Feature Request: GotoIfTimeWithOffset
At 12:23 AM on 18 Dec 2009, Olivier wrote:
> Hi,
>
> When I was testing an IVR, I realized I miss a function I would call
> GotoIfTimeWithOffset.
>
> Today, this IVR is using function AEL GotoIfTime in several places.
> The problem is if it's 11pm at the moment I'm testing this IVR, I
> can't nicely test the 9am or 2pm branch.
>
> GotoIfTimeWithOffset would get 2 incoming arguments :
> - the first is a time range (just like GotoIfTime),
> - the second is a duration offset which you could delay or "rewind"
> time.
>
> After testing, you would just have to set this offset to 0, to get a
> production-ready dialplan, without changing a line.
It ain't pretty, but this should work (untested):
globals {
TIME_OFFSET=-5;
};
context test-iftime {
s => {
GotoIfTime($[6+${TIME_OFFSET}]:00-$[7+${TIME_OFFSET}]:00|*|*|*?goodmorning);
Playback(hello);
Hangup();
goodmorning:
Playback(goodmorning);
};
};
Basically, just change each of the hours in your time specs to this:
$[+${TIME_OFFSET}]
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Re: [asterisk-users] sequential dialing preferences
At 10:38 AM on 06 Dec 2009, Thomas Perron wrote:
> I am trying to use a simple tool in the Dial plan so that if the first
> number does not connect the logic will go to the second and/or third.
>
> Basically, I want the call to ring and connect to the first number
> Then, if it is not answered I want another number to try to get
> connected Then, if second number does not answer I want the third to
> be tried i only list the scenario for the first two numbers
>
> Here is what I have now which works fine for the one and only
> number...
>
> exten => s,n,Dial(SIP/callwithus/12135551212,120,A(ginger3)) ;
> Service line
>
> so, will this work ... ..
>
> exten =>
> s,n,Dial(SIP/callwithus/12135551212[&SIP/callwithus/12145551212],120,A(ginger3))
> ;
> Service line
>
> Please send comments to make this work.
It'll work without the square brackets. The square brackets that are
shown in "core show application Dial" aren't meant to be put in
literally. They just signify that the stuff inside them is optional.
However, using Dial that way won't do what you're looking for. Instead,
it'll ring both (or all) devices at once, and the first one to answer
will get the call. The others will just be disconnected. If you want
it to ring the second number only after the first one didn't work,
you'll have to do that in your dialplan by checking ${DIALSTATUS} after
Dial.
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[asterisk-users] Restricting transfers between SIP phones
Hello, We are in the process of splitting our phone system into two separate logical systems for our two departments. One of the goals of this switch is to restrict members of one department from transferring calls to the other, but not restrict them from calling that department themselves. So what I need to know is how to detect whether a call from a member of that department is a transfer or an original call. I've looked at the TRANSFER_CONTEXT setting, but that's only for transfers with # and the T and t flags to Dial(). But we use SIP hardphones (Linksys SPA942 & Grandstream GXP2020), which have built-in transfer functions, and we would like to continue using those for transfers, rather than building it into features.conf or dialplan... Because we prefer attended transfers, and the user experience seems more modern. So, does anyone know of a way to detect whether a call from a SIP phone is the first step of an attended transfer or an original call? Thanks! -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can't call through voip provider
What does your Dial command look like? It should be something like
this:
exten => _9.,1,Dial(SIP/voipprovider/${EXTEN:1})
Also, do you have a register statement for voipprovider in sip.conf?
Does sip show registry show that it's registered successfully?
At 2:53 PM on 19 Nov 2009, Landy Landy wrote:
> Nothing. I don't know what in the world is going on with my setup.
>
[...]
> I'm already frustrated with this.
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Re: [asterisk-users] all our circuits are busy now
At 2:58 AM on 21 Oct 2009, B.Masoud @ SH wrote: > I just don't want Asterisk to play (all circuits are busy now) > because by default the call will go to the next route, here is the > debug: > > Where can I find and remove this message from? Edit your dialplan. It's either in extensions.conf or extensions.ael. You probably just have to comment out the Playback line. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linux/Asterisk on game consoles?
At 4:48 PM on 16 Oct 2009, Adam Moffett wrote: > Out of curiosity why would you want to? Because he can? or, "Because it's there." http://www.askoxford.com/worldofwords/quotations/quotefrom/mallory/ ...but hopefully the OP doesn't end up bricking his console. > > I don't know much about game consoles, and I was wondering if > > someone had successfully ported Linux and Asterisk to the current > > hardware, ie. Nintendo Wii, Sony PS3, or Microsoft XBox360? -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues with unavailable members
At 7:35 PM on 16 Oct 2009, Benny Amorsen wrote: > "C. Chad Wallace" writes: > > > Also, if there is another agent available, the caller would be > > connected immediately, and it wouldn't have to make any more > > attempts. With the Wait() solution, that caller would be waiting > > for 30 seconds regardless of whether there's anyone else > > available. > > This bit is solved by the ringall strategy. > > > Of course, I don't know your business case, so you'll have to decide > > which of the two problems is worse. > > I'm fairly happy with the Wait(1000) solution for now. We'll see if > testing shows any problems with it. Oh yeah, I hadn't even considered the ringall strategy! With that, your Wait() solution sounds perfect to me. Congrats! -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues with unavailable members
At 11:23 AM on 16 Oct 2009, Benny Amorsen wrote: > I was going in the same direction at the end of my first mail, but I > hadn't written any code. There is a problem though: The Queue > application will keep sending calls to the Local channel, which have > to be rejected, over and over. > > Would it perhaps work to simply Wait(30) if the call is rejected by > the phone? If the Queue assumes that the phone is busy for those 30 > seconds, I have accomplished my goal. It's worth a shot. It would only be trying one agent at a time for each waiting queue member... I don't know how expensive it is to open and close a Local channel and do a DB lookup, but I wouldn't expect it to be a real problem. You are at least avoiding multiple calls out to the cellular network. Also, if there is another agent available, the caller would be connected immediately, and it wouldn't have to make any more attempts. With the Wait() solution, that caller would be waiting for 30 seconds regardless of whether there's anyone else available. Of course, I don't know your business case, so you'll have to decide which of the two problems is worse. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues with unavailable members
At 11:32 AM on 15 Oct 2009, C. Chad Wallace wrote:
> At 3:37 PM on 15 Oct 2009, Benny Amorsen wrote:
>
> > Perhaps the problem could be restated in a different way: After a
> > queue member rejects a call (instead of just not answering), the
> > queue should wait X amount of time before sending the next call.
> > Queues.conf has a million settings, but I can't find one which does
> > this.
>
> To pause an agent, store the "unpause time" per agent in the AstDB.
> Then when you're deciding whether to give out a call (in the Local
> channel), look up "${DB(AgentPaused/)}" and compare it to the
> current time. If there is no record or the time has passed, put the
> call through; otherwise, skip that agent.
>
> Sorry, no example code yet... I just wanted to get the idea out
> there.
OK, I decided to write it up in AEL. It's incomplete and untested, but
it probably gets the idea across a little better.
context agentcalls {
_2XX => {
Set(AGENT=${EXTEN}); // Assuming agent ID is extension.
if (${EPOCH}>${DB(AgentPaused/${AGENT})}) {
// Let the call through to the cell phone
Dial(...);
if () {
// Flag agent as paused for the next 30 seconds.
Set(DB(AgentPaused/${AGENT})=$[${EPOCH}+30]);
};
}
else {
// Agent still paused.
};
};
};
--
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Re: [asterisk-users] Queues with unavailable members
At 3:37 PM on 15 Oct 2009, Benny Amorsen wrote:
> Perhaps the problem could be restated in a different way: After a
> queue member rejects a call (instead of just not answering), the
> queue should wait X amount of time before sending the next call.
> Queues.conf has a million settings, but I can't find one which does
> this.
To pause an agent, store the "unpause time" per agent in the AstDB.
Then when you're deciding whether to give out a call (in the Local
channel), look up "${DB(AgentPaused/)}" and compare it to the
current time. If there is no record or the time has passed, put the
call through; otherwise, skip that agent.
Sorry, no example code yet... I just wanted to get the idea out there.
--
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Re: [asterisk-users] Config Files
At 10:39 AM on 14 Oct 2009, Matt wrote: > Greetings, > I have a fresh asterisk installation. When I install I get all of the > config files. What is the best way to get a 'stripped' down system > with just the bare config files I would need to do a sip connection? In my experience, this was a bit of a procedure. But I thought it was worth it to reduce clutter. Basically, you need to find out which modules you need, and set "autoload=no" in modules.conf, and then load (with "load=xxx.so") each module individually, making sure you've also loaded each module's dependencies. Finding these dependencies was a bit of leg work. I don't remember where I figured that out... sorry. Then you need to find out which modules need which configuration files, and then delete the rest of them... To get you started, I've attached my modules.conf (with sparse comments), and here's a list of the files in my /etc/asterisk directory: asterisk.conf cdr.conf cdr_custom.conf extensions.ael extensions.conf features.conf indications.conf logger.conf modules.conf musiconhold.conf queues.conf sip.conf voicemail.conf zapata-channels.conf zapata.conf YMMV, HTH, HAND. :-) -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 modules.conf Description: Binary data signature.asc Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] limit concurrent calls on trunk supporting multiple DID
At 7:16 AM on 17 Sep 2009, Patrick wrote:
> I've one SIP trunk that support multiple DID. Only the trunk is
> documented in sip.conf (called DID is taken from the sip-header in
> real time).
> I would like to limit the number of simultaneous calls on each DID. Is
> there a way to achieve this ?
I think you could use GROUP() and GROUPCOUNT() for that. I do that for
Queue calls currently, so each agent only gets one call at a time. It
would go something like this (entirely untested):
[incoming]
exten => _X.,1,Set(DID=${EXTEN})
exten => _X.,n,GotoIf($[GROUP_COUNT(${DID})=0]?accept)
exten => _X.,n,Busy()
exten => _X.,n(accept),Set(GROUP()=${DID})
; Now let the call through as usual...
exten => _X.,n,Goto(mainmenu,s,1)
That puts each call into a group named by the DID, and returns Busy
if there is another call on the same DID.
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Re: [asterisk-users] Duplicate DTMF
Oops! I missed the part where you said you use a SIP trunk. My experiences and comments are entirely irrelevant to your case. Sorry! At 12:02 PM on 10 Sep 2009, C. Chad Wallace wrote: > > At 10:22 PM on 09 Sep 2009, John A. Sullivan III wrote: > > > Hello, all. I've come across a nasty problem just as we are ready > > to put our first system into production. During our final testing, > > we were plagued with several "invalid extension" or "password > > incorrect" messages when we knew the information entered was > > correct. Upon investigation, we saw that DTMF signals were > > occasionally but not consistently duplicated. We might dial > > extension 1234, see 1234 on the phone from which we dialed, but see > > 112334 on the Asterisk console. > > > > We have seen this from cell phones calling via the PSTN (we use a > > SIP trunking carrier and do not handle the PSTN interface > > ourselves); we've seen it from land line phones via the PSTN, and > > have even seen it internally from our own Snom SIP phones. > > > > dtmfmode=auto but we have also tried setting it to rfc2833 and we > > have tried relaxdtmf set to both yes and no. > > > > We are running Asterisk 1.6.1.6 on CentOS 5.3. We really don't know > > what more to do to fix it. Googling shows that others have had this > > problem but I haven't seen a clear resolution other than playing > > with relaxdtmf. How do we solve this problem? Thanks - John > > When we had that problem, it turned out it was caused by the txgain > being too high (10.0). I dropped it down to 5.0, and the DTMF > problems went away. > > Have you tuned your gains using a milliwatt line and ztmonitor? > > I followed this howto: http://www.mattgwatson.ca/?p=14 > > But that led to the ridiculous txgain setting of 10.0 on all my ports, > which I later clawed back to 5.0 to fix the DTMF issue. Maybe I did > it wrong? Anyway, the rxgain settings I got from that procedure > seemed to improve our call quality. We've since moved to a partial > PRI instead of those analog lines, so we don't have to worry about > that anymore. :-) -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Duplicate DTMF
At 10:22 PM on 09 Sep 2009, John A. Sullivan III wrote: > Hello, all. I've come across a nasty problem just as we are ready to > put our first system into production. During our final testing, we > were plagued with several "invalid extension" or "password incorrect" > messages when we knew the information entered was correct. Upon > investigation, we saw that DTMF signals were occasionally but not > consistently duplicated. We might dial extension 1234, see 1234 on > the phone from which we dialed, but see 112334 on the Asterisk > console. > > We have seen this from cell phones calling via the PSTN (we use a SIP > trunking carrier and do not handle the PSTN interface ourselves); > we've seen it from land line phones via the PSTN, and have even seen > it internally from our own Snom SIP phones. > > dtmfmode=auto but we have also tried setting it to rfc2833 and we have > tried relaxdtmf set to both yes and no. > > We are running Asterisk 1.6.1.6 on CentOS 5.3. We really don't know > what more to do to fix it. Googling shows that others have had this > problem but I haven't seen a clear resolution other than playing with > relaxdtmf. How do we solve this problem? Thanks - John When we had that problem, it turned out it was caused by the txgain being too high (10.0). I dropped it down to 5.0, and the DTMF problems went away. Have you tuned your gains using a milliwatt line and ztmonitor? I followed this howto: http://www.mattgwatson.ca/?p=14 But that led to the ridiculous txgain setting of 10.0 on all my ports, which I later clawed back to 5.0 to fix the DTMF issue. Maybe I did it wrong? Anyway, the rxgain settings I got from that procedure seemed to improve our call quality. We've since moved to a partial PRI instead of those analog lines, so we don't have to worry about that anymore. :-) -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sticky Park
At 11:52 AM on 27 Aug 2009, Mat Murdock wrote:
> [parkedcallstimeout]
>
> exten => _SIP011XX,1,Answer()
> exten => _SIP011XX,n,NoOp(Call Parked on: ${PARKINGSLOT})
> exten => _SIP011XX,n,NoOp(This is who parked us: ${EXTEN})
> exten => _SIP011XX,n,Set(PARKINGEXTEN=${PARKINGSLOT})
> ;This sets the PARKINGEXTEN to the parking slot we were parked in.
> exten =>
> _SIP011XX,n,Dial(SIP/${EXTEN:4:4},${RINGTIMER},${INTERNAL_DIAL_OPTIONS})
> ;This send the call back to the person who parked it. There are a
> couple of global variables I use here. Nothing unusual here.
>
>
> So what is the problem? Well the problem is that the PARKINGEXTEN
> variable gets reset after the dial command in parkedcallstimeout.
> That makes it so I cannot find out where that call was originally
> parked If I can find out how to get that little bit of information
> when the call is re-parked then I think this will work. If anyone
> has any suggestions on how to accomplish this I would be grateful.
Have you tried prefixing PARKINGEXTEN with '__' (two underscores) on
the Set call in parkedcallstimeout? That makes it a persistent
variable, which will be inherited by sub-channels, like after a Dial.
exten => _SIP011XX,n,Set(__PARKINGEXTEN=${PARKINGSLOT})
You might only need one underscore.
For more info, see 'core show application set'.
--
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Re: [asterisk-users] Need to now my "Asterisk User ID"
At 9:04 AM on 24 Aug 2009, Steve Edwards wrote: > Un-top-posting... > > > jonas kellens wrote: > >> > >> How do I know my "Asterisk User ID" ?? > > On Mon, 24 Aug 2009, bails wrote: > > > Type 'id asterisk' at your command line. It should return uid gid > > and all groups the asterisk user belongs to. > > This assumes you have a user named asterisk. Also assumes that > Asterisk is running as the user named asterisk. > > There's probably a "more proper" way, but this works: > > ~$ ps -ef | grep /sbin/asterisk | grep -v grep > > You should get something like: > > root 12477 12476 0 Aug03 ?00:02:09 /usr/sbin/asterisk -f > -g -n -p -q $ ps -fC asterisk Or for the uid: $ ps --no-headers -o uid -C asterisk -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting up Outgoing Trunk
At 3:32 PM on 06 Aug 2009, kumarshantanu wrote: > I have a genuine problem in Asterisk setup. > I have three inbound trunks in my asterisk box, everything is > working fine but the only problem is when any user make an out- > going call through his/her extension it goes with same number labeled > on this. > > Can we set each of these lines to have fixed outgoing numbers > > like if extn: 201 make an outgoing call the recipient should get > different no and if extn: 202 make an outgoing call the recipient > should > get different one. I'm not sure I understand your question... Are you saying that you want the outbound caller ID to be your public phone number? Or do you want the outbound caller ID to change based on what internal extension initiated the call? If you want your public phone number (the one your telco sends by default) for all outgoing calls, you need to add hidecallerid=yes to your zapata.conf or chan_dahdi.conf for the channels in those trunks. Then Asterisk won't send the caller ID it receives from the internal extensions, and your telco should then insert your phone number as the caller ID on all outbound calls. If you want it set to something other than what your telco puts as the default, you'd have to have hidecallerid=no (or not there) for those channels and set CALLERID(all) in your dialplan for outgoing calls. But this requires that your telco allows you to override the default caller ID. Some do, and some don't--you would have to talk to them. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk dont detects hangup by phone
At 5:24 PM on 06 Aug 2009, ABBAS SHAKEEL wrote: > Hello > I have configured TDM400P with asterisk . > The problem is that when i make a call to server. and while going on > it dont detects call hang up. > > ie i called the Asterisk server and it start playing files that i > indicated to do so in extensions.conf > i suddenly put down the phone. now the server must detect that phone > is hangup but it dont. > > How can i make server to detect this You may need to enable busydetect in chan_dahdi.conf or zapata.conf. However, if you do enable it, I would recommend you also tune it with the busycount and busypattern options so you don't get false positives (asterisk hanging up in the middle of a call). You can read the comments in the sample configs or on voip-info.org for more information: http://www.voip-info.org/wiki/view/Asterisk+config+chan_dahdi.conf http://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf.sample -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail feature: enable or disable the ability to leave a message
At 11:24 PM on 31 Jul 2009, Emrah wrote: > Doug, > > Thanks for the suggestion. > I know there are plenty of workarounds there, I am not asking how to > do it because I know how to do it too. > What I am saying is that it could be an embedded feature in the > Voicemail application, like the recent ability to flag a message as > "urgent". Maybe the work being done (been done?) through Google Summer of Code 2009[1] will net some progress in this direction. MiniVM looks like a pretty good idea... but it's got a long way to go. I seriously doubt app_voicemail itself will get any new features, regardless of how many good ideas get thrown around. Just like what happened with AgentCallbackLogin, I think instead of augmenting the incumbent system, Digium will probably replace VoiceMail and VoiceMailMain with an equivalent dialplan solution using MiniVM. It's much more flexible that way. [1] http://lists.digium.com/pipermail/asterisk-dev/2009-April/038028.html > Doug Lytle wrote: > > Emrah wrote: > > > >> Mark, > >> > >> I think you did not understand my message. > >> I am accustomed to have the option to allow or disallow the > >> recording of a message in my voicemail, even my mobile carrier > >> provides it. E.g.: I > >> > > > > The simplest thing to do is to allow users to set a flag, maybe > > using mysql or the astdb, if they want that option. > > > > And, in your dial plan, check for the existance of that flag. If > > it's there, then don't jump to the voice mail app, just jump to > > your context that would play back an audio file that the user has > > pre-recorded > > > > Doug -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reading/Writing the Astdb
At 8:48 PM on 26 Jan 2009, cbbs...@hotmail.com wrote: > > Tzafrir; >That's a really good idea, however, I am having problems getting > it to work. I tried the following: > > echo -n "asterisk -rx \"database put FOO BAR 1\" " | socat > - /var/run/asterisk/asterisk.ctl > > and > > echo -n "asterisk -rx \"database put FOO BAR 1\" " | socat - > UNIX-CONNECT: /var/run/asterisk/asterisk.ctl > > both look like there was a connection to asterisk on the CLI, but no > update. What am I missing? BTW; the difference between connecting to > asterisk by doing a asterisk -rx "" and doing it this way was > > 500ms the first way and 7ms the second. Awesome. Just guessing, but try it without the "asterisk -rx": echo -n "database put FOO BAR 1" | socat - UNIX-CONNECT:/var/run/asterisk/asterisk.ctl And if that doesn't work, maybe omit the "-n" so that it sends a newline. > > Date: Sat, 24 Jan 2009 23:23:40 +0200 > > From: tzafrir.co...@xorcom.com > > To: asterisk-users@lists.digium.com > > Subject: Re: [asterisk-users] Reading/Writing the Astdb > > > > On Sat, Jan 24, 2009 at 11:00:58AM -0500, cbbs...@hotmail.com wrote: > > > > > > All; > > >I have a question regarding the Astdb. When reading more than > > > a few values, it can take quite a while to grab several > > > values in the astdb using say, asterisk -rx "database show" > > > > output.txt and work with that and then set a new value such as > > > asterisk -rx "database put $key $value". The whole process can > > > take over 1 second for EACH ENTRY which adds up for more than a > > > few keys. > > > > Either do that through the manager interface, or (if you want to > > batch commands) send them directly over the unix-domain socket > > asterisk.ctl . -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP to IAX?
At 8:29 AM on 11 Sep 2008, John Millican wrote: > > Not directly on-topic for this list, but I'd not heard of OpenSIPS > > before, so I had a look at the website. It looks to be a fork of > > OpenSER. Does that mean OpenSER development has slowed/ceased, or > > has the OpenSER project itself morphed into OpenSIPS? > > > > Regards, > > > > Chris > > > via a quick google:OpenSER is now OpenSIPS > www.opensips.org OpenSER continues via OpenSIPS A new name, same > project Uhhh, I thought that was Kamailio: www.kamailio.net ...I'm confused. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 Debian Hint #21: If your Debian box is behind a slow network connection, but you have access to a fast one as well, check out the apt-zip package. signature.asc Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Probably very simple... call a number and play a sound?
At 2:29 PM on 11 Sep 2008, Mike Johnson wrote: > Hey hey... > > I'd like to create a new feature code in asterisk so when a user > dials... say... *00, it would then call some other extensions and > play a sound file to them. > > So far, this is what I have... > > [testing-custom] > exten => *00,1,Wait(1) > exten => *00,2,Playback(beep) > exten => *00,3,Playback(beep) > exten => *00,4,AGI(festival-script.pl|I will now attempt the call) > exten => *00,6,Set(CALLERID(all)=Notify <9000>) > exten => *00,7,NoOp() > exten => *00,8,Dial(SIP/302,15}) > exten => *00,9,Wait(2) > exten => *00,10,Playback(demo-congrats) > exten => *00,15,Answer() > exten => *00,20,Hangup() > > I must be missing something glaringly obvious as this doesn't sound > like a tough thing to accomplish. For some reason however, I cannot > figure it out... > > Can anyone offer some ideas or assistance? Your dialplan is missing priority 5. The prioritites MUST be sequential, incrementing by 1, with no gaps. So the Set(CALLERID... line should be *00,5, and the rest will have to be adjusted accordingly. Consider using the 'n' priority instead: [testing-custom] exten => *00,1,Wait(1) exten => *00,n,Playback(beep) exten => *00,n,Playback(beep) exten => *00,n,AGI(festival-script.pl|I will now attempt the call) exten => *00,n,Set(CALLERID(all)=Notify <9000>) exten => *00,n,NoOp() exten => *00,n,Dial(SIP/302,15}) exten => *00,n,Wait(2) exten => *00,n,Playback(demo-congrats) exten => *00,n,Answer() exten => *00,n,Hangup() TTYL... -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 Debian Hint #8: If you have problems with Debian that you can't solve by reading the manuals and documentation, try asking on the Debian Users mailing list ([EMAIL PROTECTED]). signature.asc Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Discover connected Zap lines
At 9:55 PM on 20 May 2008, Vinz486 wrote: > 2008/5/17 bilal ghayyad <[EMAIL PROTECTED]>: > > So no way to discover the status of FXO if a cable ^^^ Foxtrot X-ray *Oscar* > > pluged or not? > > Did you read my previou msg > > Hookstate (FXS only): Offhook <--Cable plugged > > Hookstate (FXS only): Onhook <--Cable unplugged ^^^ Foxtrot X-ray *Sierra* When it says "FXS only", I think it's reasonable to assume that FXO is excluded. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 Debian Hint #19: If you're interested in building packages from source, you should consider installing the apt-src package. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No-mobo PC for USB Drives Enclosure?
At 11:45 PM on 13 May 2008, Matthew Rubenstein wrote: > The drives are 750GB drives, each one a different related set > of apps from a different Asterisk machine. I've consolidated them all > into a single Asterisk server. And I already have the existing PC > chassis and power supply, as well as the $10 each SATA/USB adapters. > If I can just figure out how to power them from the PC power supply > without plugging in a useless motherboard, I'll have it done without > spending any money (other than whatever cheap part tells the power > supply to run without a mobo). What I do to power up a supply without a mobo is short the green wire to a black one (on an ATX 20-pin connector) with a small piece of metal--like a staple straightened and then bent in half, or a piece of a paper clip. As soon as you plug the supply into AC, it powers up. Not sure if this is very safe... but it works for me every time. I guess you might want to avoid letting the shunt contact the case... however, given that the black wires are ground, I wouldn't worry too much about it. Anyway, this advice comes with no warranty... Use it at your own risk. If anything breaks, you get to keep both parts. ;-) -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 Debian Hint #22: Wondering which Debian mirror is best for you? Check out the apt-spy and netselect-apt packages, which can give you information about how various mirror sites perform. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Dialplan: Best Practice in using Context - Do not use Default??
At 9:43 AM on 13 May 2008, Lee, John (Sydney) wrote: > In "The future of Telephony", it says "... We should also note for > security's sake you should always make sure that your [incoming] > context never allows outbound dialing. (If by chance it did, people > could dial into your system and make outbound toll calls that would > be charged to you!) > > The book was demonstrating using a PSTN environment and the > zapata.conf was something like: > context=internal > signaling=fxo_ks > channel=>1 > > context=incoming > signaling=fxs_ks > channel=>2 > > In PRI environment, does it mean that we have to purposely separate > the say ISDN 20 channels into [internal] and [incoming] as well? > This would not make sense to me as ISDN uses a one port card to > contain multiple channels while the ports of a say TDM400P refer to > each channel. > > If I just define a [default] context for a PRI environment, is this > insecure? > > Can someone please enlighten me on this? In the example you quoted, channel 1 is an FXS port, which would be an internal extension--a phone--from which someone would be allowed to make an outbound call. Channel 2 is an FXO port, which is connected to the PSTN, and would take incoming calls from "the wild". So in that example, you wouldn't want the "incoming" context to be allowed to make outbound calls. In your case, I'm guessing all your Zap channels come from the PRI, which is connected to the PSTN. If so, then you're right--you just need one context for your zapata.conf which you would use on all your ISDN channels. Just don't let that context dial out. I don't know if you'd want to call that context "default"... because that one seems to be "special" in Asterisk. But maybe I'm just being superstitious. :-) -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 Debian Hint #14: If you would like to follow things happening to a package (for example, if you want to see bug reports, release notices, and other similar things), consider subscribing to it on the Package Tracking System. You can find out more about the PTS at: http://www.debian.org/doc/manuals/developers-reference/ch-resources.en.html (Section 4.10) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zap Channels Collide (Incoming & Outgoing)
At 5:22 PM on 08 May 2008, Forrest Beck wrote: > I have a client that is using the Sangoma A200DE with two phone > lines attached. > > The problem is: > > They use their phone (Grandstream GXP2020) to dial out of the system. > Instead of getting ringing, there is someone on the other end of the > line that happened to dial in at the exact same moment. > > So now they are stuck talking with this person, instead of the one > the originally called. > > The ZAP channels are in a dial plan context that instructs it to > just dial the office phones. > > [zap1] > exten => s,1,Dial(SIP/1001&SIP/1002&SIP/1003) > exten => s,n,Voicemail([EMAIL PROTECTED]) > > Anyone know how to get around this? This is known in the telephony world as "glare", and there's not much you can do about it, especially if you only have one line. If you have multiple lines on an over-ring (or hunt group or whatever you call it), the best thing to do is find out which way the telco assigns calls to those lines wrt how they are assigned to the Asterisk box. And then allocate outgoing calls in the other direction. On our installation, the calls are allocated from the first FXO port (Zap/25) up. So we set Asterisk to dial out starting from the last FXO port in the group by calling Dial(Zap/G2) (capital G means dial down from last, lowercase g means dial up from first). That minimizes glare. But, as I said before, if you only have one line, you can't do that... -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 Debian Hint #19: If you're interested in building packages from source, you should consider installing the apt-src package. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Order of queue member list
We just recently upgraded from Asterisk 1.2 to 1.4, and quickly noticed a change in the behaviour of the queues--a change that we cannot live with. We've used AddQueueMember/RemoveQueueMember to manage logging into and out of our queues for over a year now with Asterisk 1.2, and in that version the queue members were sorted in such a way that the person who had been logged in the longest would be the first one to get a call. But when we deployed 1.4 last week, we noticed that the member list was no longer sorted based on login time. It seemed to have a pre-set order that members were always placed into. After looking at the code (apps/app_queue.c), I found the cause of this. In 1.2, the members were stored in a linked list, so when someone logged in, they were placed at the end, and when calls were handed out, it was done starting at the front of the member list (or vice-versa, but either way, it has the same effect). In 1.4, the member list was changed to an "ao2_container", which apparently uses a hash table, and iterates through the list in a fixed order, meaning one of our agents is always the favourite for a call, and it is quite unfair. Now, I know that the ordering of members in the queue in 1.2 was not documented, and it may not have even been intentional, but it was very appropriate for our business model, and we'd like to find a way to get it back. Is there a way to control the order in which the ao2_iterator returns the items? Even a random distribution would be better than the current--which always favours some agents over others. And before anyone mentions the "strategy" setting in queues.conf, I should say that we use "leastrecent", but because of the ratio of agents to queues in our business, the strategy doesn't come into effect immediately. With many agents answering each queue, it takes a while for each of them to get a call. Until then (which usually takes about half of each day), the calls are distributed based on the ordering of the member list. We have switched to the "rrmemory" strategy for now, but we've yet to notice what effect that has--and our ideal would be to use "leastrecent" along with the behaviour that Asterisk 1.2 exhibited. Thanks! -- cc -Wall The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: OpenPGP digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Queues
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Floyd wrote: > Hi everyone: > > I've searching for a while and haven't found what i > need. > The thing is that i have a tdm422p with the two fxo > ports connected to the pstn. I want my sip users to be > able to call other numbers(any number) in the pstn > through my zap fxo channels. I have a big number of > sip users so as you can imagine there will be > congestion when some of them(more than two!!) want to > call outside, that is why i want to be able to put > those outgoing calls in a queue. For example if i want > to call someone in the pstn and the fxo port is > already in use, i want to be placed in a queue and > when the channel is free my call is routed to the > aproppiated destination. As far as i have read the > queues are not for this kind of stuffs, there are > just agents or extensions that attend the calls in the > queue and nothing more. am i wrong??? > Any help will be useful. > thanks in advance!! You could probably do this using the Local channel. You'd create a context, say outbound, to take calls from the queue and connect them to a Zap channel, with 2 extensions in that context--one for each channel. Then you add each of those extensions as members of the queue: member => Local/[EMAIL PROTECTED]/n member => Local/[EMAIL PROTECTED]/n Make sure your dialplan in outbound returns Busy if the Zap channel is busy. The tricky part would be passing the dialed number through... But if you set an inheriting channel var, it should go through the queue and into the Local channel to your outbound extension. Sorry I don't have any code for you... I haven't done it yet; I'm just putting the idea out there. Hope this helps! Good luck. - -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGjWCTKeSNHCYiCKARAtdgAKCVUs6OF2KIpjbpwQFrwr2E4NatVACfWh6I 9XwYqQ7cc5gwVznybIglBGs= =miEL -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GotoIf Dialplan inquiry
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Steve Finkelstein wrote:
> Hi all,
>
> I have the following in my extensions.conf:
>
> exten => s,4,GotoIf($["${CALLERID(number)}" = "8585979857" |
> "8585970327"]?15:5)
>
> The numbers listed above are known spammer numbers. However, when I call
> from any other CALLERID, it still directs me to s,15 which is the
> Hangup() application. Here are logs from the asterisk CLI:
You have two problems there:
1. You have CALLERID(number), which I'm pretty sure should be CALLERID(num).
2. You are trying to combine two values with the | operator to check
them both against CALLERID(num). It can't be used that way. You have
to have two sets of comparisons.
Here's how the line should look:
exten => s,4,GotoIf($["${CALLERID(num)}" = "8585979857" |
"${CALLERID(num)}" = "8585970327"]?15:5)
TTYL.
> -- Executing [EMAIL PROTECTED]:1] Macro("IAX2/lime-3",
> "forward|SIP/8995&SIP/31337|15|8995|IAX2/[EMAIL PROTECTED]/13476681546") in
> new stack
> -- Executing [EMAIL PROTECTED]:1] Zapateller("IAX2/lime-3",
> "answer|nocallerid") in new stack
> -- Executing [EMAIL PROTECTED]:2] PrivacyManager("IAX2/lime-3", "")
> in new stack
> -- CallerID Present: Skipping
> -- Executing [EMAIL PROTECTED]:3] Wait("IAX2/lime-3", "1") in new stack
> -- Executing [EMAIL PROTECTED]:4] GotoIf("IAX2/lime-3",
> "("8585970327")?15:5") in new stack
> -- Goto (macro-forward,s,15)
> -- Executing [EMAIL PROTECTED]:15] Hangup("IAX2/lime-3", "") in new stack
> == Spawn extension (macro-forward, s, 15) exited non-zero on
> 'IAX2/lime-3' in macro 'forward'
> == Spawn extension (macro-forward, s, 15) exited non-zero on 'IAX2/lime-3'
> -- Hungup 'IAX2/lime-3'
- --
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.skihills.com/
OpenPGP Public Key ID: 0x262208A0
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Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
iD8DBQFGbuUBKeSNHCYiCKARAlPVAKCBYdUm4nRQd4clYphLg4bOzjeRrgCgxIgs
inueeqMYByPtpNDFgypNgLo=
=Z7qz
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Re: [asterisk-users] Zaptel kernel module load order
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Mitch Jackson wrote: > Evening, > > My latest asterisk box is having a difficult problem. It is > configured with one TE210P and TDM400P with four FXO modules. I'm > running FC6. > > The TE210P only has a single PRI. > > When the system boots, it is completely random what order the zaptel > modules will get loaded in. Sometimes zttool shows the FXO as the > last span, sometimes as the first. When it does load as the first, > which happens more often, nothing will initialize properly. When this > happens, I have to unload all the zaptel modules, and re-load them > over and over again, until the hardware comes up in the correct order. > The order it is loaded is in no way related to what order I load the > modules on the command line. This problems makes it unlikely that > asterisk will start properly if the system is rebooted. > > Is there something I can do to ensure the modules get loaded in the > correct order? If you use udev (and subsequently modprobe), you can override the install command for the TDM card to load the T1 card first: Create a file in the /etc/modprobe.d directory, and put the following line in it: install wctdm4xxp modprobe wcte21xp && modprobe --ignore-install wctdm4xxp $CMDLINE_OPTS && /sbin/ztcfg (Make sure it's all one line; your mail reader might break it.) This method works for me, with a TE110P and a TDM2400P in the same box. However, I am using Debian, and I'm not sure if modprobe and udev work the same way in FC6. TTYL. - -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGN7bcKeSNHCYiCKARAoefAKDAh/V2W3cwd/ASfHH5JsMOdj3wOgCgnexb yMh5TUnMZzHdM572J67oxmU= =pHCB -END PGP SIGNATURE- ___ --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
