Re: [Asterisk-Users] Asterisk Dynamic DNS
If you're experiencing the same issue I am, there's a less painful solution than restarting asterisk. Asterisk resolves the externip (and, I think, externhost) parameters in sip.conf at startup. If the values are a domain name registered with dyndns.org, and the IP that these domain names point at changes, then you have a problem. It turns out that sip reload will cause it to resolve externip (for sure, and I assume externhost as well) again. So I run a cron job to do a sip reload periodically. (Ideally, I want to trigger this job from my dyndns ip-change script, but I haven't gotten around to that yet. Doing it every 5 minutes is a bit of an ugly hack, but it works for me.) HTH. john Branko Samardzic wrote: That's fine. But there is, obviously, situations where such situation is not welcome. Is it possible to force Asterisk to refresh cache every in a while. Regards, Branko ___ --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 ___ --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 book feedback
The book is a great *starting* point, IMHO. If you've spent a considerable amount of time reading other sources, you probably won't find much new information in the book. OTOH, you may find that its organized approach helps consolidate what you've read. And if it clears up a couple of key concepts about dial plans, AGI, configuration, ZAP, or whatever, which you might be "fuzzy" about, it's probably worth the price. In addition, the appendices are a useful reference guide. Ross C wrote: Just curious what everyone (as in, the people that have read it or use it) thinks about the O'Reilly Asterisk book. I'd really like to delve into the nitty gritty of Asterisk, but I'm getting kinda tired of swimming through forums and Google results. I've been reading the wiki off and on for about a week now, but I'm wondering if a book would be the way to go to get a solid foundation. My IT career for the past 10 years has been based off of learn-as-I-go methods, but I'd really like to learn asterisk the right way. I have a couple Asterisk servers up and running and in use, but they're very small systems (~10 extensions, connected to 3 or 4 pots lines). I have some clients that want to use VOIP, but they're bigger businesses, and I'm not yet comfortable enough to roll out a bigger system. So if there are any other methods for learning Asterisk that I should consider, please do tell! Any opinions (on the book or otherwise) appreciated. Thanks! -ross ___ --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 ___ --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] sharing a line w/multiple extensions
Fascinating discussion. The whole idea of "acceptance" of an asterisk
based system by the rest of the family is probably worthy of its own thread.
I'm in "alpha test" (I switch on asterisk after the wife leaves for
work, switch it back before she gets home ;-) ) of my home asterisk
system, so I've been thinking/worrying about a lot of similar issues.
I'm particularly worried about acceptance of this "shared line" (or lack
thereof) aspect of the system. My wife will "get" the idea of
extensions, transfers, parking, etc. because she uses a PBX at work,
though I worry that the habits of how the phone is "supposed to work" at
home may die hard with her. And the kids are a whole 'nuther story.
I thought that having some "common area" phones share a single extension
(wired into a single ATA FXS port) might ease the transition, but I'm
also afraid it might be confusing ("you can just pick up from these
extensions, but you have to transfer or park to/from these extensions".
Huh?).
The huge selling point, which I'm hoping will overcome any initial
resistance, is the idea that one person will no longer tie up the whole
phone system for the house when they make/take a call. And deploying
one of my free DIDs to give my 16-year-old "his own phone number" that
rings only in his bedroom is the real ace up my sleeve!
Sure, Asterisk will come with a lot of other neat features, but frankly
most of them have more geek appeal (though I have high hopes for my
favorite feature -- announced caller id over the stereo/tivo while we're
making dinner -- to revolutionize the way we deal with (or at least who
answers ;-) ) phone calls at that hour), and in some cases I think may
face similar "that's not the way it's supposed to work" objections. For
example, while they will acknowledge that voicemail is cool, I suspect
they'll miss the simplicity of walking into the kitchen, seeing if the
answering machine is blinking, and just pressing the button.
I'm excited AND anxious about starting a real "beta test" with them!
Maybe that's why I'm already 3 weeks behind my original schedule. ;-)
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Re: [Asterisk-Users] Linksys PAP2 and Asterisk
Hi Jason, I've got several PAP2s working with asterisk. Feel free to e-mail me off-line if you want to compare configurations. Which version of asterisk and which PAP2 firmware are you running? Cheers, john Jason (WeatherServer) wrote: I'm sure this question has been asked before but I can't seem to find any info on it. Is there anything special that needs to be setup on the PAP2 side and the Asterisk side for the PAP2 to work on the asterisk server. I've entered all the settings for my VoIP provider but all I get is Registration State: Can't connect to login server on the PAP2 Thank you. ___ --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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Extension Language -- what's it's "status"?
I've been playing with ael a little bit and wondering how much time to devote to it. Originally, when I read about it, I thought I'd try to convert my extensions.conf files to ael. But after playing a little bit, I'm not sure that's a good idea. A couple of observations and questions: 1. As far as I can tell, there's limited interoperability between existing (extensions.conf) and new (extensions.ael) constructs. For example, and please correct me if I'm wrong, but I don't believe there's a way to access an ael macro from an extensions.conf dial plan. To me, this seems to make dabbling with ael harder, since I can't just incrementally add some stuff in ael and use it in my dial plan. 2. I can't see a way to organize my ael dial plan using #include's. Again, for me, this makes experimentation harder since, especially in conjunction with the first point, I effectively have to maintain my entire old conglomerate dial plan in one gigantic ael file. Again, please correct me if I'm missing something. All in all, for me, these two points have discouraged me from doing more than dabbling with ael. I assume the point of including it in 1.2 is to get people to try to use it in a serious "beta test" manner. I wonder if support for these kind of capabilities is planned? I really would like to use ael as it seems to have some clear advantages over the existing "language", but I'm reluctant to be too far out on the bleeding edge without a little more insight into where this feature is headed. Thanks for any input. Oh, and this is my first post here, so forgive me if I put this in the wrong place. Does this belong on the developers list? ___ --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 Extension Language -- what's it's "status"?
John Biundo wrote: A couple of observations and questions: 1. As far as I can tell, there's limited interoperability between existing (extensions.conf) and new (extensions.ael) constructs. For example, and please correct me if I'm wrong, but I don't believe there's a way to access an ael macro from an extensions.conf dial plan. I'm an idiot. And on my first post. Way to go, John. So the problem is that my #includes weren't working (after my first mistake, I'll refrain from saying they DON'T work (actually hoping someone proves me wrong again and explains that they do, indeed, work)), so my ael macros weren't getting defined in the first place. Once I moved them into extensions.ael, they were, as expected, fully available to my extensions.conf dial plans. This is goodness, to me, because I think I'll enjoy learning to write loops and conditionals MUCH MORE in ael than in the old style. ___ --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
[Asterisk-Users] Invalid/Timeout handlers in ael?
Does anybody know how to code invalid and timeout handlers in ael macros?
I tried the following, but no luck.
=
macro call-screen() {
NoOp(Macro call-screen);
Background(privacy-screening-unidentified-calls);
tryagain:
Playback(pls-rcrd-name-at-tone);
Set(SCREEN_FILE=/tmp/screen-${EPOCH});
Record(${SCREEN_FILE}.wav,6,25);
catch i {
NoOp(invalid?);
goto tryagain;
}
catch t {
NoOp(timeout?);
goto tryagain;
}
};
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[Asterisk-Users] How to do "asynchrononous" Dial?
Are there any features in asterisk that might be used to effect a background dial task? I want to be able to start a Dial task running, in chich the execution can be automated via DTMF tones (e.g., voicemail retrieval), but not block the originating thread of execution. The dial plan or AGI containing the Dial command should proceed on to the next step. I'll arrange some kind of rendezvous later on. As far as I can tell, I'm on my own to build this capability through some sort of daemon and the manager API or my own IPC mechanism. I'd really hoped for some higher level of abstraction from within Asterisk to accomplish this. Am I missing something in Asterisk that will make this easier to accomplish? Are there any publically available examples of this kind of application? Anybody got any ideas for nifty tricks to get this done? Thanks in advance. John ___ --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] How to do "asynchrononous" Dial?
Thanks for the suggestion. Sorry if my requirements weren't clear. I need a general purpose "run this in the background" capability. The voicemail was simply an example, and probably a bad one. I want the Dial command to go off on a separate thread so the main thread can continue. What happens in that second thread is arbitrary. I'll deal with getting "data" in and out of that second thread as appropriate - perhaps by dropping stuff in a database or some other synchronization method. The key is that while that second "background task" is executing, the initial thread of execution doesn't block and can continue running arbitrary Asterisk applications. See sample.call in the Asterisk source directory and .call files on the Wiki. Sounds like you want Asterisk to call a person when a voicemail is left and allow the person to get their voicemail during the notification call. ___ --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] NAT setup
Tom Rymes wrote: To connect to an Asterisk box that sits behind NAT, you need to forward ports 5060 and 1-2 too the asterisk box Follow up question: I can't forward 1-2 with my router. So I used rtp.conf to narrow the band of ports down to something like 14000-14030 and forwarded those ports That seems to work fine. Am I asking for trouble down the line with this approach? ___ --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
[Asterisk-Users] Problem with Zap/1 picking up on OUTBOUND calls from analog extensions
Problem: Zap/1 channel answers when trying to make OUTBOUND call on
analog phone.
Configuration:
-Asterisk: Running CVS Head from Nov. 1
-X100P card attached to PSTN line; same PSTN line routed to other analog
phones throughout the house
-Linksys PAP2 ATA with analog phone connected to Asterisk via LAN
See attached zaptel info, zapata.conf and asterisk console log
Details: I can receive and handle inbound calls via the PSTN line
normally. My custom IVR, voicemail, call-screening, etc. all works. I
can make outbound calls to my ITSP via the analog phone and ATA or
softphones.
*Sometimes*, when I try to make an outbound call on one of the other
house phones (one of the phones on the same PSTN line as the X100P),
asterisk answers the zap channel as if it had an incoming call. This
seems to happen about 80% of the time. Sometimes, pressing just the *
key will cause the line to be answered. Other times, it's only after an
outbound call has been connected (ringing tone heard on the line) that
the zap channel picks up.
The situation seems to have deteriorated over the last day or two,
though I haven't changed anything in the zaptel.conf or zapata.conf
files for several weeks. The frequency of occurrence seems to have gone
up signficantly in 48 hours. My extensions.conf has changed quite a bit
in that time, but I'm not sure how that should matter.
FWIW, I have the house lines (including the one going to the x100p) all
connected via a punch-down block that I *believe* is wired correctly,
though I'm not an expert, so it's possible something's going on in that
area -- I just don't know what to look for there. These lines haven't
changed since before asterisk was installed. They test properly for
polarity, and line quality is good.
Has anyone got any suggestions on how to track this problem down? I'm
really scratching my head over it!
Thanks in advance.
Cheers,
john
=
Output from ztcfg:
$ ztcfg -vv
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
1 channels configured.
=
zapata.conf:
[trunkgroups]
; define any trunk groups
[channels]
usecallerid=yes
hidecallerid=no
callwaiting=yes
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes
musiconhold=default
faxdetect=incoming
callreturn=yes
immediate=yes
busydetect=yes
callprogress=yes
progzone=us
; define channels
context=incoming
signalling=fxs_ks
channel => 1
==
CLI Output:
-- Starting simple switch on 'Zap/1-1'
Nov 15 09:24:55 WARNING[816]: chan_zap.c:6088 ss_thread: CallerID
returned with error on channel 'Zap/1-1'
-- Executing Macro("Zap/1-1", "incoming") in new stack
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[Asterisk-Users] Read() application behavior change: bug or feature?
Bug or feature? I'm fairly sure this behavior changed between CVS HEAD Nov 1 and RC2. In the earlier release, I could successfully execute the Read() application immediately following answering a (SIP) channel. In the current RC2, Read times out without detecting user input, unless I precede it in the dial plan with Background, as below. exten => s,1,Answer exten => s,n,Background(silence/1) exten => s,n(readopt),Read(ACCEPT,,1,,,10) ___ --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] Problems with Read() in outgoing calls
I posted the following a couple of days ago. My problem was inbound, but the workaround might be worth a try: == Bug or feature? I'm fairly sure this behavior changed between CVS HEAD Nov 1 and RC2. In the earlier release, I could successfully execute the Read() application immediately following answering a (SIP) channel. In the current RC2, Read times out without detecting user input, unless I precede it in the dial plan with Background, as below. exten => s,1,Answer exten => s,n,Background(silence/1) exten => s,n(readopt),Read(ACCEPT,,1,,,10) = Chris Cahill wrote: I have used Read() in many inbound context (ie. when a user dials me). I have an outbound call between asterisk and a user, initiated by a call file in the outgoing directory, but Read() does not seem to take any input in this situation. Is there anyway of getting round this? Scouse ___ --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] Allowing Called user to accept call before transfer
I'm still fairly wet behind the ears, so I hope one of the experts will
chime in, but I think I'm doing what you want, so let me try to describe it.
When a call comes in without caller id, I "screen it" (record the
caller's name), then dial one of the internal extensions and play back
the recording. I provide the callee the options to accept the call,
send it to voicemail, or do other nastier things.
The way I do it is to call a macro in the Dial command. For example:
exten => s,n,Dial(${ALLHOUSE},20,TtmM(play-screen^${ANNOUNCE}))
In the play-screen macro, I play the screen, and accept DTMF tones to
accept the callee's desired outcome. Based on the tone, I then set
the MACRO_RESULT variable appropriately. See the dial command on the
wiki for more on this. I use something like:
exten => s,n,SetVar(MACRO_RESULT=GOTO:sendfamilyvmail^s^1)
When the macro ends, if the value of MACRO_RESULT is undefined, the call
is bridged. This is the key point: if you want the call to be bridged,
do not define MACRO_RESULT to anything. If the value is set to GOTO...,
the call is NOT bridged at the end of the macro, and execution continues
at the target of the GoTo.
Hope that helps.
DEEZED wrote:
I was wondering if its possible to have a asterisk dial a number, and then
when it is answered, asterisk will playback a message asking the called
person to press a button if they would like to accept the call, and then if
asterisk hears a DTFM tone, it will then bridge the 2 calls together.
If this is possible, how could this be done. Does asterisk have a command to
merge channel A with channel B?
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