Hi,
I've been trying to route incoming calls based on CID to a trunk but the
calls are not getting though. I am trying to use a wild card prefix based
on countries so I can point the call to the appropriate trunk.
I am running Asterisk 1.8 with FreePBX.
Here is a sample of my configuration in extentions_custom.conf
exten => _00336123412xx/44XX.,1,Set(RINGTIME=90,g)
exten => _00336123412xx/44XX.,n,Answer(10) exten =>
_00336123412xx/44XX.,n,Dial(SIP/trunk01/${EXTEN},${RINGTIME})
exten => _00336123412xx/44XX.,n,Hangup()
Any kind suggestions is appreciated. Thanks.
Best Regards,
Geoffrey Yeoh
Try: exten => _00336123412xx/_44XX.,1,Set(RINGTIME=90,g)
Notice the _ on your callerid pattern
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces
at lists.digium.com] On Behalf Of Geoffrey Yeoh
Sent: Thursday, October 11, 2012 1:15 PM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] Call routing based on CID
Hi,
I've been trying to route incoming calls based on CID to a trunk but the
calls are not getting though. I am trying to use a wild card prefix based on
countries so I can point the call to the appropriate trunk.
I am running Asterisk 1.8 with FreePBX.
Here is a sample of my configuration in extentions_custom.conf
exten => _00336123412xx/44XX.,1,Set(RINGTIME=90,g)
exten => _00336123412xx/44XX.,n,Answer(10) exten =>
_00336123412xx/44XX.,n,Dial(SIP/trunk01/${EXTEN},${RINGTIME})
exten => _00336123412xx/44XX.,n,Hangup()
Any kind suggestions is appreciated. Thanks.
Best Regards,
Geoffrey Yeoh
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Thanks Eric. That works.
--------------------------
Try: exten => _00336123412xx/_44XX.,1,Set(RINGTIME=90,g)
Notice the _ on your callerid pattern
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Geoffrey
Yeoh
Sent: Thursday, October 11, 2012 1:15 PM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] Call routing based on CID
Hi,
I've been trying to route incoming calls based on CID to a trunk but the
calls are not getting though. I am trying to use a wild card prefix based
on countries so I can point the call to the appropriate trunk.
I am running Asterisk 1.8 with FreePBX.
Here is a sample of my configuration in extentions_custom.conf
exten => _00336123412xx/44XX.,1,Set(RINGTIME=90,g)
exten => _00336123412xx/44XX.,n,Answer(10) exten =>
_00336123412xx/44XX.,n,Dial(SIP/trunk01/${EXTEN},${RINGTIME})
exten => _00336123412xx/44XX.,n,Hangup()
Any kind suggestions is appreciated. Thanks.
Best Regards,
Geoffrey Yeoh
Hi all, I have an Asterisk PBX under development, that I would like to link to a Skype account if possible. The idea is that people would call a particular Skype username, and be redirected to my SIP and through that to Asterisk. Is this doable? I have looked around and saw the Skype for Asterisk driver, but of course that has been discontinued. Are there any other options? I would prefer not to have to go through the regular PSTN telephone network but directly from Skype to Asterisk via SIP. If you have any tips on how to configure my sip.conf to get this working, this would also be highly appreciated. Thanks in advance for your help! Kind regards, Philip Bennefall
Christopher Harrington
2012-Oct-12 18:54 UTC
[asterisk-users] Connecting Skype to Asterisk
On Fri, Oct 12, 2012 at 1:44 PM, Philip Bennefall <philip at blastbay.com> wrote:> Hi all, > > I have an Asterisk PBX under development, that I would like to link to a > Skype account if possible. The idea is that people would call a particular > Skype username, and be redirected to my SIP and through that to Asterisk. Is > this doable? I have looked around and saw the Skype for Asterisk driver, but > of course that has been discontinued. Are there any other options? I would > prefer not to have to go through the regular PSTN telephone network but > directly from Skype to Asterisk via SIP. If you have any tips on how to > configure my sip.conf to get this working, this would also be highly > appreciated. >It looks like this is what you want: http://www.skype.com/intl/en/business/skype-connect/ -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248
On 13/10/2012, at 7:54 AM, Christopher Harrington <chris at acsdi.com> wrote:> On Fri, Oct 12, 2012 at 1:44 PM, Philip Bennefall <philip at blastbay.com> wrote: >> Hi all, >> >> I have an Asterisk PBX under development, that I would like to link to a >> Skype account if possible. The idea is that people would call a particular >> Skype username, and be redirected to my SIP and through that to Asterisk. Is >> this doable? I have looked around and saw the Skype for Asterisk driver, but >> of course that has been discontinued. Are there any other options? I would >> prefer not to have to go through the regular PSTN telephone network but >> directly from Skype to Asterisk via SIP. If you have any tips on how to >> configure my sip.conf to get this working, this would also be highly >> appreciated. >> > > It looks like this is what you want: > http://www.skype.com/intl/en/business/skype-connect/ >This is pretty straight forward to use for inbound skype business user names and outbound either to pstn, skype numbers are a little more to setup There is a monthly cost but its not much and if you have skype users out there its a good way for them to connect in> > -- > -Chris Harrington > ACSDi Office: 763.559.5800 > Mobile Phone: 612.326.4248 > > -- > _____________________________________________________________________ > -- 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