Re: [asterisk-users] Auto dialing Polycoms and other SIP phones
Hello Mike, It is possible for Polycom phones to auto-answer an incoming call with speakerphone. I don’t have the details available right now, but it requires changing the phone’s configuration and sending a custom sip header with the INVITE. Great care should be taken when implementing this, as it could allow anyone to listen in or spy on your customer and/or their employees without their knowledge. No need to physically put bugs in the office anymore. The phones will already be there and they’ll be configured to allow monitoring of the room. Autoanswer after one ring of a “beep” (or similar) ringtone may be a better solution for you and the customer. Hope this helps. With kind regards, Pan From: Mike Sent: Thursday, May 05, 2011 3:30 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Auto dialing Polycoms and other SIP phones Hi, Is there a reliable way to auto-dial SIP phones (specifically Polycom) with some sort of TAPI driver in Windows? I am aware of SIPTAPI, which makes the user’s phone ring, and when picked up dials the desired number, but I (and more to the point, many of my customers) find this annoying. I’d like the phone to autodial on speakerphone (or headset if there is one), without any human intervention. This doesn’t have to be a free solution. If anyone knows one, I’d appreciate. Regards, 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-- _ -- 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] Auto dialing Polycoms and other SIP phones
Pan, Thank you, that makes sense. I’ll investigate further. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pan B. Christensen Sent: Thursday, May 05, 2011 11:47 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Auto dialing Polycoms and other SIP phones Hello Mike, It is possible for Polycom phones to auto-answer an incoming call with speakerphone. I don’t have the details available right now, but it requires changing the phone’s configuration and sending a custom sip header with the INVITE. Great care should be taken when implementing this, as it could allow anyone to listen in or spy on your customer and/or their employees without their knowledge. No need to physically put bugs in the office anymore. The phones will already be there and they’ll be configured to allow monitoring of the room. Autoanswer after one ring of a “beep” (or similar) ringtone may be a better solution for you and the customer. Hope this helps. With kind regards, Pan From: Mike mailto:l...@net-wall.com Sent: Thursday, May 05, 2011 3:30 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' mailto:asterisk-users@lists.digium.com Subject: [asterisk-users] Auto dialing Polycoms and other SIP phones Hi, Is there a reliable way to auto-dial SIP phones (specifically Polycom) with some sort of TAPI driver in Windows? I am aware of SIPTAPI, which makes the user’s phone ring, and when picked up dials the desired number, but I (and more to the point, many of my customers) find this annoying. I’d like the phone to autodial on speakerphone (or headset if there is one), without any human intervention. This doesn’t have to be a free solution. If anyone knows one, I’d appreciate. Regards, 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 -- _ -- 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] Auto dialing: .call file vs. manager interface
The manager interface expects Exten NOT Extension argument header. Well honk my hooter! I had been using 'Extension' but since I always used the 's' extension I never noticed anything goofy until I tried a numeric extension. Thanks for the heads up. -MC ___ --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] Auto dialing: .call file vs. manager interface
The manager interface expects Exten NOT Extension argument header. On 12/5/06, Michael Collins [EMAIL PROTECTED] wrote: Question: I'm using a .call file to make some test calls. The call file works great. When I try the same thing with the manager 'originate' action I get something weird - the originate action looks for the 's' extension in my context, regardless of what I supply as the 'extension' argument. The .call file does what I expect - it finds exten _9.,1,Noop(Looks good). The error I get in the log is as follows: Dec 5 16:44:25 VERBOSE[19670] logger.c: == Starting Zap/1-1 at autodial_start,s,1 failed so falling back to exten 's' Dec 5 16:44:25 VERBOSE[19670] logger.c: == Starting Zap/1-1 at autodial_start,s,1 still failed so falling back to context 'default' The autodial_start context looks like this: [autodial_start] exten = _9.,1,Noop(Looks good) exten = _9.,n,Goto(dialout,s,1) The dialout context just has the call handling stuff, AMD, etc. It works when the Goto works, but the Goto only seems to work when using a .call file and not the manager interface. The .call file looks like this: Channel: Zap/g0/5596221408 Callerid: 5597337550 MaxRetries: 0 RetryTime: 30 WaitTime: 30 Context: autodial_start Extension: 95596221408 Priority: 1 Account: 5898832 Has anyone experienced this issue and/or found a way around it? Thanks, MC ___ --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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --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] Auto dialing
On Wed, 2006-02-01 at 21:58 -0800, Abhishek wrote: Channel: SIP/[EMAIL PROTECTED] even when i can dial out manually through the same context(sip_proxy-out) in sip.conf. as used in the call file sip_proxy-out shouldnt be a context, it should instead be an account defined in sip.conf. Not sure if you misspoke or not ... If you want to dial via a context try: Channel: Local/[EMAIL PROTECTED] For others looking at this, keep in mind though that it will connect to the Channel first then once connected it will connect to the context or application specified elsewhere in the call file. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --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] Auto Dialing
You can do a couple of things. If you are using a SIP device, or a device that
uses a PLAR code (Private Line Automatic Ringdown) then you can put that
number that you want to dial when the phone goes off-hook in that section.
That way, when you pick up the phone it will dial that number first which
will then prompt you via some sort of IVR. Or, you could create an AGI script
(observe below) to Dial the number, wait ten seconds, then Dial the dialed
digits when you receive confirmation.
--script--
extensions.conf
[default]
exten = _.,1,agi,savings.agi|${EXTEN}
exten = _.,2,Hangup
/var/lib/asterisk/agi-bin/savings.agi # Set chmod 755 in order to execute
#!/usr/bin/perl
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
$AGI-setcallback(\mycallback);
print STDERR AGI Environment Dump:\n;
foreach $i (sort keys %input) {
print STDERR -- $i = $input{$i}\n;
}
my $userid = $input{'calleridname'};
my $exten = $input{'extension'};
my $savings_number = 74949000;
if($exten eq 'h') { exit; }
$AGI-exec(Dial, Zap/g1/$savings_number); # Replace with way that you are
dialing out to PSTN
sleep(1);
--end script--
On Thursday 18 November 2004 06:16 am, Simon wrote:
In my house i am using an autodialer to dial 74949000 to access to
gateway and then i dial my mobile or local number to benefit from the
saving Can we do that in asterisk to autodial to the gateway 74949000 and
wait 10 before i eneter my destination number
Please advise me on that how to make a effective dialplan...suing the
above information
Thanking you in advance
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Brian Wilkins
Software Engineer
[EMAIL PROTECTED]
Heritage Communications Corporation
Melbourne, FL USA 32935
321.308.4000 x33
http://www.hcc.net
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
