[asterisk-users] no subject
Hello, list. I've set up an outbound alerting system to play a recording when systems go down, etc. and I'm noticing that cellphones tend to answer() and then start ringing the actual handset. So far, I've verified this behavior with Verizon, T-Mobile, and Google Voice (the last produces a SERIOUS delta between bogus answer and actual answer). Has anyone figured out how to detect the actual cellphone answer rather than the bogus one sent by the cell carrier? In the short term, I just have the call play MOH for ten seconds before announcing that all hell has broken loose in the server room, but it¹d be nice to have something a bit more accurate and reliable. Cheers, Mike -- _ -- 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
[asterisk-users] False answer() being sent by cellphone providers
On 7/9/10 9:57 AM, Mike Ely mike...@amyskitchen.net wrote: Hello, list. I've set up an outbound alerting system to play a recording when systems go down, etc. and I'm noticing that cellphones tend to answer() and then start ringing the actual handset. So far, I've verified this behavior with Verizon, T-Mobile, and Google Voice (the last produces a SERIOUS delta between bogus answer and actual answer). Has anyone figured out how to detect the actual cellphone answer rather than the bogus one sent by the cell carrier? In the short term, I just have the call play MOH for ten seconds before announcing that all hell has broken loose in the server room, but it¹d be nice to have something a bit more accurate and reliable. Cheers, Mike Argh, got distracted, here's the version with a Subject: header. -- _ -- 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] False answer() being sent by cellphone providers
Some of the systems blokes might just figure that¹s another collections agent and hang up then ;) On 7/9/10 10:09 AM, Steve Edwards asterisk@sedwards.com wrote: On Fri, 9 Jul 2010, Mike Ely wrote: I've set up an outbound alerting system to play a recording when systems go down, etc. and I'm noticing that cellphones tend to answer() and then start ringing the actual handset. So far, I've verified this behavior with Verizon, T-Mobile, and Google Voice (the last produces a SERIOUS delta between bogus answer and actual answer). Has anyone figured out how to detect the actual cellphone answer rather than the bogus one sent by the cell carrier? In the short term, I just have the call play MOH for ten seconds before announcing that all hell has broken loose in the server room, but it¹d be nice to have something a bit more accurate and reliable. How about a loop with Please press pound to continue? -- _ -- 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] False answer() being sent by cellphone providers
On 7/9/10 10:29 AM, Mike Ely mike...@amyskitchen.net wrote: Some of the systems blokes might just figure that¹s another collections agent and hang up then ;) On 7/9/10 10:09 AM, Steve Edwards asterisk@sedwards.com wrote: On Fri, 9 Jul 2010, Mike Ely wrote: I've set up an outbound alerting system to play a recording when systems go down, etc. and I'm noticing that cellphones tend to answer() and then start ringing the actual handset. So far, I've verified this behavior with Verizon, T-Mobile, and Google Voice (the last produces a SERIOUS delta between bogus answer and actual answer). Has anyone figured out how to detect the actual cellphone answer rather than the bogus one sent by the cell carrier? In the short term, I just have the call play MOH for ten seconds before announcing that all hell has broken loose in the server room, but it¹d be nice to have something a bit more accurate and reliable. How about a loop with Please press pound to continue? Sorry, bad joke. In all seriousness though, is there not a way to detect this behavior and handle the answer() correctly? -- _ -- 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] False answer() being sent by cellphone providers
Thanks for the tip! On 7/9/10 11:35 AM, Faisal Hanif fai...@vopium.com wrote: Do some R D with asterisk function AMD (Answering Machine Detection) if that can help you. Signatures fai...@vopium.com Regards, Faisal Hanif On 7/9/2010 11:24 PM, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Friday, July 09, 2010 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] False answer() being sent by cellphone providers On Fri, 9 Jul 2010, Mike Ely wrote: I've set up an outbound alerting system to play a recording when systems go down, etc. and I'm noticing that cellphones tend to answer() and then start ringing the actual handset. So far, I've verified this behavior with Verizon, T-Mobile, and Google Voice (the last produces a SERIOUS delta between bogus answer and actual answer). Has anyone figured out how to detect the actual cellphone answer rather than the bogus one sent by the cell carrier? In the short term, I just have the call play MOH for ten seconds before announcing that all hell has broken loose in the server room, but it¹d be nice to have something a bit more accurate and reliable. How about a loop with Please press pound to continue? -- It is a DAHDI function that you may or may not get a reliable notification of answer. The best thing to do is to MOH for 7 seconds, then play a message this is a message from the computer room; press 1 to accept. This lets you not waste time on a not real answer. Here is a cliff-note context: [accept] exten = s,1,Answer exten = s,n,WaitExten(7) exten = s,n,Background(important) exten = s,n,WaitExten(5,m) exten = 1,1,backgrounf(message) exten = 1,n,hangup exten = t,1,hangup exten = i,1,hangup exten = *,1,hangup -- _ -- 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] False answer() being sent by cellphone providers
On 7/9/10 3:20 PM, Gordon Henderson gordon+aster...@drogon.net wrote: On Fri, 9 Jul 2010, Mike Ely wrote: On 7/9/10 9:57 AM, Mike Ely mike...@amyskitchen.net wrote: Hello, list. I've set up an outbound alerting system to play a recording when systems go down, etc. and I'm noticing that cellphones tend to answer() and then start ringing the actual handset. So far, I've verified this behavior with Verizon, T-Mobile, and Google Voice (the last produces a SERIOUS delta between bogus answer and actual answer). Has anyone figured out how to detect the actual cellphone answer rather than the bogus one sent by the cell carrier? In the short term, I just have the call play MOH for ten seconds before announcing that all hell has broken loose in the server room, but it¹d be nice to have something a bit more accurate and reliable. Cheers, Mike Wow. So presumably you start to pay for the call before the mobile phone actually rings and you answer the mobile phone? So you're charged even if the mobile phone user doesn't answer? Are you sure? Although I guess it's a country specific thing - if they tried that over here I think it'd be pitchforks and flaming torches at their UK HQ offices... Gordon (off list) Yes indeed we do. The telcos here are absolutely abhorrent, to the point that much could be written about how horrible they are but nobody would want to read such depressing material. And consumer protections? Hah! The devotees of Ayn Rand have written most consumer law here. Don't get me started. -- _ -- 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] False answer() being sent by cellphone providers
-Original Message- From: asterisk-users-boun...@lists.digium.com on behalf of Steve Edwards Sent: Fri 7/9/2010 5:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject:Re: [asterisk-users] False answer() being sent by cellphone providers On Fri, 9 Jul 2010, Mike Ely wrote: (off list) Continuing to veer off-topic... Yes indeed we do. The telcos here are absolutely abhorrent, to the point that much could be written about how horrible they are but nobody would want to read such depressing material. And consumer protections? Hah! The devotees of Ayn Rand have written most consumer law here. Don't get me started. Maybe you should re-read Atlas Shrugged. Your laws may have been written by the people Ayn Rand wrote about: the government increasingly asserts control over all industry, while society's most productive citizens, led by the mysterious John Galt, progressively disappear,* not devotees of Ayn Rand and her philosophy. *) http://en.wikipedia.org/wiki/Atlas_Shrugged Well, so much for my off list attempt. Perhaps I should learn how to use email before I take on anything so complex as a PBX. At any rate, Steve, you have it completely backwards: in the US and many other countries, it is industry asserting control over government, not the other way around. Walk down K Street in Washington, D.C. and you'll see my point. And no thanks: I've already read that execrable book, and found it to be nothing more than overwrought claptrap written to give people with a huge inferiority complex (witness all the carping on about mediocrity) some smug self-justification when they abandon all ethics in favor of their reptile-brain, base instincts. Disgusting. Cheers! -- _ -- 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't dial out through AMI
On 7/8/10 5:07 AM, Paul Belanger paul.belan...@polybeacon.com wrote: On Wed, Jul 7, 2010 at 2:46 PM, Mike Ely mike...@amyskitchen.net wrote: Maybe I missed something here? SIP users configured within Asterisk can dial out just fine through the trunk. It's just when I try to use AMI that it fails. The far end is rejecting your call; SIP/2.0 401 Unauthorized. If you can dialout without using AGI, then capture a 2nd debug log, and post it. We can then compare why one works and the other does not. Got it. The issue was in the Channel directive in my AMI script. Before, it looked like this: Action: Originate Channel: SIP/ShoreTel Exten: 7979 Variable: Data=testing1 Context: accept Priority: 1 When it works, it looks like this: Action: Originate Channel: SIP/ShoreTel/7979 Variable: Data=testing1 Context: accept Priority: 1 Thanks for your help! Mike -- _ -- 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't dial out through AMI
On 7/6/10 8:44 PM, Mike Ely mike...@amyskitchen.net wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com on behalf of Paul Belanger Sent: Tue 7/6/2010 5:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject:Re: [asterisk-users] Can't dial out through AMI On Tue, Jul 6, 2010 at 8:00 PM, Mike Ely mike...@amyskitchen.net wrote: Log attached. --- SIP read from UDP:10.10.10.16:5060 --- SIP/2.0 401 Unauthorized context from sip.conf: [ShoreTel] type=peer qualify=yes port=5060 host=10.10.10.16 context=incoming canreinvite=no Your context is not setup properly for outbound, you have no credentials defined. None needed on the ShoreTel side and as I mentioned before regular SIP users can dial out through the Asterisk box using this trunk. Keep in mind, this is a development system on a tightly-controlled network, and I'm trying to start with the simplest case possible, which includes no digest auth for the trunk connection. Maybe I missed something here? SIP users configured within Asterisk can dial out just fine through the trunk. It's just when I try to use AMI that it fails. -- _ -- 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
[asterisk-users] Can't dial out through AMI
SIP user = Asterisk 1.6 server = SIP Trunk = external destination:
works
AMI script = Asterisk 1.6 server = SIP Trunk = external destination:
Failed to authenticate on INVITE to 'asterisk
sip:asterisk@(ipaddr);tag=alphanumeric'
I¹ve tried doing things like ³include = contextwithtrunk in various
places, googling, re-reading relevant portions of the largish O'Reilly
Asterisk book, no avail.
The call will go through to a registered SIP user just fine, but won't seem
to go out off the trunk.
Here's the basic set of commands being sent through AMI:
Action: Originate
Channel: SIP/ShoreTel
Variable: Data=teletubbie-murder
Context: accept
priority: 1
Number: (external number reachable from regular SIP user account)
Here's the accept context:
[accept]
include = incoming
include = outbound-pbx
exten = s,1,Answer
exten = s,n,Playback(custom/msg1)
exten = s,n,Background(custom/how-to-ack)
exten = s,n,WaitExten(5,m)
exten = 1,1,ForkCDR(v,s(fullcmd=${Data}))
exten = 1,n,Background(${Data})
exten = 1,n,Background(discon-or-out-of-service)
exten = 1,n,WaitExten(5,m)
exten = 1,n,Hangup
exten = 2,1,Background(de-activated)
exten = 2,n,ForkCDR(v,s(reject=${Data}))
exten = 2,n,Hangup
exten = 3,1,Goto(accept,1,2)
exten = *,1,Goto(accept,s,1)
exten = i,1,Goto(accept,s,1)
exten = t,1,Goto(accept,s,1)
Obviously, I'm playing around with the context a bit but for now just want
to get the outbound call working.
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Re: [asterisk-users] Can't dial out through AMI
Log attached. It looks like the call is trying to do an invite to the sip trunk and fails there - it never actually tries to send the destination to the ShoreTel system on the other end of the trunk. Here's the ShoreTel context from sip.conf: [ShoreTel] type=peer qualify=yes port=5060 host=10.10.10.16 context=incoming canreinvite=no On 7/6/10 4:21 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Tue, Jul 6, 2010 at 4:10 PM, Mike Ely mike...@amyskitchen.net wrote: Obviously, I'm playing around with the context a bit but for now just want to get the outbound call working. debug log would be helpful: http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.t xt nosiptrunk.txt Description: Binary data -- _ -- 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't dial out through AMI
-Original Message- From: asterisk-users-boun...@lists.digium.com on behalf of Paul Belanger Sent: Tue 7/6/2010 5:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject:Re: [asterisk-users] Can't dial out through AMI On Tue, Jul 6, 2010 at 8:00 PM, Mike Ely mike...@amyskitchen.net wrote: Log attached. --- SIP read from UDP:10.10.10.16:5060 --- SIP/2.0 401 Unauthorized context from sip.conf: [ShoreTel] type=peer qualify=yes port=5060 host=10.10.10.16 context=incoming canreinvite=no Your context is not setup properly for outbound, you have no credentials defined. None needed on the ShoreTel side and as I mentioned before regular SIP users can dial out through the Asterisk box using this trunk. Keep in mind, this is a development system on a tightly-controlled network, and I'm trying to start with the simplest case possible, which includes no digest auth for the trunk connection. -- _ -- 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] Remote Party ID issue
As an interesting aside, every email I get on this list coming from Tilghman Lesher is marked with a To Do flag by my email client. Every single one. I don't have any inbound filter that would explain the behavior either. On 7/1/10 1:15 PM, Tilghman Lesher tles...@digium.com wrote: On Thursday 01 July 2010 15:01:45 Danny Nicholas wrote: On Thursday, July 01, 2010 2:37 PM, Adam Moffett wrote: Steve Howes wrote: DON'T reply to people off list. And stop bloody top posting. Is bottom posting your personal preference or is that a rule on this list? I have personally always found top posting easier to follow because the newer content is at the top. It's a rule on this list, although it's frequently ignored. BP is not a RULE and I wish people would STOP BITCHING about it. If you use MS Outlook to reply to this list, TOP POSTING is the default behavior. If someone wants to write a nice how-to on Bottom posting in MS Outlook, I'll be happy to read it. http://mailformat.dan.info/config/outlook.html http://mailformat.dan.info/quoting/bottom-posting.html http://home.in.tum.de/~jain/software/outlook-quotefix/ -- _ -- 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] Remote Party ID issue
Sorry to answer my own question here - had a look at the headers of Tilghman's last email and it contained this: X-message-flag: Major security vulnerability detected! You should shutdown your computer immediately and upgrade to Ubuntu Linux 8.04 or later. Cute. Leaving aside the fact that I don't use Outlook, Ubuntu 8.04 isn't exactly going to be a security improvement over what I already use. On 7/1/10 1:19 PM, Mike Ely mike...@amyskitchen.net wrote: As an interesting aside, every email I get on this list coming from Tilghman Lesher is marked with a To Do flag by my email client. Every single one. I don't have any inbound filter that would explain the behavior either. On 7/1/10 1:15 PM, Tilghman Lesher tles...@digium.com wrote: On Thursday 01 July 2010 15:01:45 Danny Nicholas wrote: On Thursday, July 01, 2010 2:37 PM, Adam Moffett wrote: Steve Howes wrote: DON'T reply to people off list. And stop bloody top posting. Is bottom posting your personal preference or is that a rule on this list? I have personally always found top posting easier to follow because the newer content is at the top. It's a rule on this list, although it's frequently ignored. BP is not a RULE and I wish people would STOP BITCHING about it. If you use MS Outlook to reply to this list, TOP POSTING is the default behavior. If someone wants to write a nice how-to on Bottom posting in MS Outlook, I'll be happy to read it. http://mailformat.dan.info/config/outlook.html http://mailformat.dan.info/quoting/bottom-posting.html http://home.in.tum.de/~jain/software/outlook-quotefix/ -- _ -- 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] Call file structure and syntax
Yep, I saw that and it's just not the case. I was having it dial my desk extension, which was decidedly not busy at the time... On 6/28/10 5:30 PM, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: Well, I¹ve tried this, and something just isn¹t right. Look here: Event: Hangup Channel: SIP/ShoreTel-1-0004 Cause: 17 Cause-txt: User busy -- _ -- 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] Call file structure and syntax
Of Mike Ely Sent: Tuesday, June 22, 2010 12:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Call file structure and syntax Hi there, I¹ve been looking to do an outbound dialer for systems alerting, etc. and have in large part followed the recipe here: http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out That and the associated pages at voip-info give a basic set of recipes for callfiles, but nowhere there or in my copy of the O¹Reilly book by Meggelen, Madsen, Smith can I find a detailed discussion of what goes into a callfile, how to get it to do things like interact with the shell (I¹d like ³Press 2² in my outbound call to do something of value), etc. I¹ve googled around but haven¹t found what I¹m looking for, just other people¹s ³Hello World² callfiles. As of now, I can make outbound calls well enough, but want more... Can someone point me in the right direction for this? Thanks, Mike -- _ -- 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
[asterisk-users] Call file structure and syntax
Hi there, I¹ve been looking to do an outbound dialer for systems alerting, etc. and have in large part followed the recipe here: http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out That and the associated pages at voip-info give a basic set of recipes for callfiles, but nowhere there or in my copy of the O¹Reilly book by Meggelen, Madsen, Smith can I find a detailed discussion of what goes into a callfile, how to get it to do things like interact with the shell (I¹d like ³Press 2² in my outbound call to do something of value), etc. I¹ve googled around but haven¹t found what I¹m looking for, just other people¹s ³Hello World² callfiles. As of now, I can make outbound calls well enough, but want more... Can someone point me in the right direction for this? Thanks, Mike -- _ -- 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] Call file structure and syntax
That¹s a good start. In my case, I want it to dial a round-robin queue (set
up separately) and if the user presses 2, stop dialing the queue and log
which user acknowledged the alarm. If the user presses 1, repeat the
message, if no key is pressed before a timeout, hang up and dial the next
user in the queue. Or something like that. One of the things I also want
to be able to do with this is echo out something to the shell, either a
textfile or an actual command so that I can trigger some other actions not
necessarily related to Asterisk.
It¹s a fun project except for the knowledge that successful completion is
going to mean it wakes me up some night at 3am.
On 6/22/10 10:31 AM, Danny Nicholas da...@debsinc.com wrote:
#1 once you¹ve got to this point, AMI would be a better option than a call
file
#2 - using AMI or a call file, you are going to want to use the context-based
method instead of application to get the most ³bang for your buck²
I use a bigger instance of this to play a message and accept 1 or 2 from the
user
; this context is used by AMI to play a message
[accept]
exten = s,1,Answer
exten = s,n,Background(important)
exten = s,n,WaitExten(5,m)
exten = 1,1,ForkCDR(v,s(fullcmd=${Data}))
exten = 1,n,Background(${Data})
exten = 1,n,Background(repeatmsg)
exten = 1,n,WaitExten(5,m)
exten = 1,n,Hangup
exten = 2,1,Background(calllater)
exten = 2,n,ForkCDR(v,s(reject=${Data}))
exten = 2,n,Hangup
exten = 3,1,Goto(accept|1|2)
exten = *,1,Goto(accept|s|1)
exten = i,1,Goto(accept|s|1)
exten = t,1,Goto(accept|s|1)
here¹s the call file
Action = 'Originate',
Channel = DAHDI/1,
Variable = Data=/tmp/test.gsm²,
Exten = 'SIP/170',
Context = 'accept',
priority = 1,
Number = 5551212
Using the accept context, 5551212 is called on DAHDI/1 and user hears
important.gsm. then they press 1 to hear test.gsm or 2 to hear it later.
Hope this is helpful
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Ely
Sent: Tuesday, June 22, 2010 12:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Call file structure and syntax
Hi there,
I¹ve been looking to do an outbound dialer for systems alerting, etc. and have
in large part followed the recipe here:
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
That and the associated pages at voip-info give a basic set of recipes for
callfiles, but nowhere there or in my copy of the O¹Reilly book by Meggelen,
Madsen, Smith can I find a detailed discussion of what goes into a callfile,
how to get it to do things like interact with the shell (I¹d like ³Press 2² in
my outbound call to do something of value), etc. I¹ve googled around but
haven¹t found what I¹m looking for, just other people¹s ³Hello World²
callfiles. As of now, I can make outbound calls well enough, but want more...
Can someone point me in the right direction for this?
Thanks,
Mike
--
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