VoIP Mailing List Archives
Mailing list archives for the VoIP community |
|
View previous topic :: View next topic |
Author |
Message |
eric at fnords.org Guest
|
Posted: Mon May 05, 2008 3:31 pm Post subject: [asterisk-users] DTMF |
|
|
Remove the T/t/w/W option from the Dial line.
Jason Wolfe wrote:
Quote: | Ok, ever had one of those issues that you're sure is quite simple to solve
but you can't seem to get anything useful from Google or anywhere else and
so you're ready to throw your computer out the window? Well, I'm there!
I am using a simple Zyxel VoIP phone to dial outbound calls to a PSTN
termination provider, so my extension file is one command. Dial()
Anywhere I call I probably need to enter an extension, but as it should,
asterisk tries to respond to these key presses. How do I pass the DTMF tones
through so that I can navigate the IVR of the system I'm calling???
| --
Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS,
T-1, PRI, Frame Relay, Linux, and network design. Based near
Birmingham, AL. Now accepting clients worldwide. |
|
Back to top |
|
|
jason at clickforacall... Guest
|
Posted: Mon May 05, 2008 3:47 pm Post subject: [asterisk-users] DTMF |
|
|
Ok, I removed the T/t/w/W options but unfortunately it is still responding
the same way.
Ps. I have no options set on the dial() function now.
jason
::::::::-----Original Message-----
::::::::From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-
::::::::users-bounces at lists.digium.com] On Behalf Of Eric Wieling
::::::::Sent: Monday, May 05, 2008 4:31 PM
::::::::To: Asterisk Users Mailing List - Non-Commercial Discussion
::::::::Subject: Re: [asterisk-users] DTMF
::::::::
::::::::Remove the T/t/w/W option from the Dial line.
::::::::
::::::::Jason Wolfe wrote:
::::::::> Ok, ever had one of those issues that you're sure is quite
::::::::simple to solve
::::::::> but you can't seem to get anything useful from Google or
::::::::anywhere else and
::::::::> so you're ready to throw your computer out the window? Well,
::::::::I'm there!
::::::::>
::::::::> I am using a simple Zyxel VoIP phone to dial outbound calls to
::::::::a PSTN
::::::::> termination provider, so my extension file is one command.
::::::::Dial()
::::::::>
::::::::> Anywhere I call I probably need to enter an extension, but as
::::::::it should,
::::::::> asterisk tries to respond to these key presses. How do I pass
::::::::the DTMF tones
::::::::> through so that I can navigate the IVR of the system I'm
::::::::calling???
::::::::
::::::::
::::::::--
::::::::Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN,
::::::::WAN, QoS,
::::::::T-1, PRI, Frame Relay, Linux, and network design. Based near
::::::::Birmingham, AL. Now accepting clients worldwide.
::::::::
::::::::_______________________________________________
::::::::-- 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 |
|
Back to top |
|
|
jason at clickforacall... Guest
|
Posted: Mon May 05, 2008 3:48 pm Post subject: [asterisk-users] DTMF |
|
|
Also, I'm using IAX to dial()
Jason Wolfe
::::::::-----Original Message-----
::::::::From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-
::::::::users-bounces at lists.digium.com] On Behalf Of Eric Wieling
::::::::Sent: Monday, May 05, 2008 4:31 PM
::::::::To: Asterisk Users Mailing List - Non-Commercial Discussion
::::::::Subject: Re: [asterisk-users] DTMF
::::::::
::::::::Remove the T/t/w/W option from the Dial line.
::::::::
::::::::Jason Wolfe wrote:
::::::::> Ok, ever had one of those issues that you're sure is quite
::::::::simple to solve
::::::::> but you can't seem to get anything useful from Google or
::::::::anywhere else and
::::::::> so you're ready to throw your computer out the window? Well,
::::::::I'm there!
::::::::>
::::::::> I am using a simple Zyxel VoIP phone to dial outbound calls to
::::::::a PSTN
::::::::> termination provider, so my extension file is one command.
::::::::Dial()
::::::::>
::::::::> Anywhere I call I probably need to enter an extension, but as
::::::::it should,
::::::::> asterisk tries to respond to these key presses. How do I pass
::::::::the DTMF tones
::::::::> through so that I can navigate the IVR of the system I'm
::::::::calling???
::::::::
::::::::
::::::::--
::::::::Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN,
::::::::WAN, QoS,
::::::::T-1, PRI, Frame Relay, Linux, and network design. Based near
::::::::Birmingham, AL. Now accepting clients worldwide.
::::::::
::::::::_______________________________________________
::::::::-- 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 |
|
Back to top |
|
|
eric at fnords.org Guest
|
Posted: Mon May 05, 2008 3:58 pm Post subject: [asterisk-users] DTMF |
|
|
Did you remember to do a "reload" in the Asterisk CLI?
Jason Wolfe wrote:
Quote: | Ok, I removed the T/t/w/W options but unfortunately it is still responding
the same way.
Ps. I have no options set on the dial() function now.
jason
::::::::-----Original Message-----
::::::::From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-
::::::::users-bounces at lists.digium.com] On Behalf Of Eric Wieling
::::::::Sent: Monday, May 05, 2008 4:31 PM
::::::::To: Asterisk Users Mailing List - Non-Commercial Discussion
::::::::Subject: Re: [asterisk-users] DTMF
::::::::
::::::::Remove the T/t/w/W option from the Dial line.
::::::::
::::::::Jason Wolfe wrote:
::::::::> Ok, ever had one of those issues that you're sure is quite
::::::::simple to solve
::::::::> but you can't seem to get anything useful from Google or
::::::::anywhere else and
::::::::> so you're ready to throw your computer out the window? Well,
::::::::I'm there!
::::::::>
::::::::> I am using a simple Zyxel VoIP phone to dial outbound calls to
::::::::a PSTN
::::::::> termination provider, so my extension file is one command.
::::::::Dial()
::::::::>
::::::::> Anywhere I call I probably need to enter an extension, but as
::::::::it should,
::::::::> asterisk tries to respond to these key presses. How do I pass
::::::::the DTMF tones
::::::::> through so that I can navigate the IVR of the system I'm
::::::::calling???
::::::::
::::::::
::::::::--
::::::::Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN,
::::::::WAN, QoS,
::::::::T-1, PRI, Frame Relay, Linux, and network design. Based near
::::::::Birmingham, AL. Now accepting clients worldwide.
::::::::
::::::::_______________________________________________
::::::::-- 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
_______________________________________________
-- 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
|
--
Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS,
T-1, PRI, Frame Relay, Linux, and network design. Based near
Birmingham, AL. Now accepting clients worldwide. |
|
Back to top |
|
|
jason at clickforacall... Guest
|
Posted: Mon May 05, 2008 4:31 pm Post subject: [asterisk-users] DTMF |
|
|
Yes, and I verified watching the output that it was reading the new .conf
file.
jason
::::::::-----Original Message-----
::::::::From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-
::::::::users-bounces at lists.digium.com] On Behalf Of Eric Wieling
::::::::Sent: Monday, May 05, 2008 4:58 PM
::::::::To: Asterisk Users Mailing List - Non-Commercial Discussion
::::::::Subject: Re: [asterisk-users] DTMF
::::::::
::::::::Did you remember to do a "reload" in the Asterisk CLI?
::::::::
::::::::Jason Wolfe wrote:
::::::::> Ok, I removed the T/t/w/W options but unfortunately it is still
::::::::responding
::::::::> the same way.
::::::::>
::::::::> Ps. I have no options set on the dial() function now.
::::::::>
::::::::> jason
::::::::>
::::::::> ::::::::-----Original Message-----
::::::::> ::::::::From: asterisk-users-bounces at lists.digium.com
::::::::[mailto:asterisk-
::::::::> ::::::::users-bounces at lists.digium.com] On Behalf Of Eric
::::::::Wieling
::::::::> ::::::::Sent: Monday, May 05, 2008 4:31 PM
::::::::> ::::::::To: Asterisk Users Mailing List - Non-Commercial
::::::::Discussion
::::::::> ::::::::Subject: Re: [asterisk-users] DTMF
::::::::> ::::::::
::::::::> ::::::::Remove the T/t/w/W option from the Dial line.
::::::::> ::::::::
::::::::> ::::::::Jason Wolfe wrote:
::::::::> ::::::::> Ok, ever had one of those issues that you're sure is
::::::::quite
::::::::> ::::::::simple to solve
::::::::> ::::::::> but you can't seem to get anything useful from Google
:::::::r
::::::::> ::::::::anywhere else and
::::::::> ::::::::> so you're ready to throw your computer out the
::::::::window? Well,
::::::::> ::::::::I'm there!
::::::::> ::::::::>
::::::::> ::::::::> I am using a simple Zyxel VoIP phone to dial outbound
::::::::calls to
::::::::> ::::::::a PSTN
::::::::> ::::::::> termination provider, so my extension file is one
::::::::command.
::::::::> ::::::::Dial()
::::::::> ::::::::>
::::::::> ::::::::> Anywhere I call I probably need to enter an
::::::::extension, but as
::::::::> ::::::::it should,
::::::::> ::::::::> asterisk tries to respond to these key presses. How
::::::::do I pass
::::::::> ::::::::the DTMF tones
::::::::> ::::::::> through so that I can navigate the IVR of the system
::::::::I'm
::::::::> ::::::::calling???
::::::::> ::::::::
::::::::> ::::::::
::::::::> ::::::::--
::::::::> ::::::::Consulting for Asterisk, Polycom, Sangoma, Digium,
::::::::Cisco, LAN,
::::::::> ::::::::WAN, QoS,
::::::::> ::::::::T-1, PRI, Frame Relay, Linux, and network design.
::::::::Based near
::::::::> ::::::::Birmingham, AL. Now accepting clients worldwide.
::::::::> ::::::::
::::::::> ::::::::_______________________________________________
::::::::> ::::::::-- 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
::::::::>
::::::::>
::::::::> _______________________________________________
::::::::> -- 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
::::::::>
::::::::>
::::::::
::::::::--
::::::::Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN,
::::::::WAN, QoS,
::::::::T-1, PRI, Frame Relay, Linux, and network design. Based near
::::::::Birmingham, AL. Now accepting clients worldwide.
::::::::
::::::::_______________________________________________
::::::::-- 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 |
|
Back to top |
|
|
jjones at danrj.com Guest
|
Posted: Mon May 05, 2008 5:15 pm Post subject: [asterisk-users] DTMF |
|
|
And you are using g.711 so the sounds are passing correctly and not
being distorted? Try calling a person and pressing digits to verify
they are inband during call?
On May 5, 2008, at 4:31 PM, Jason Wolfe wrote:
Quote: | Yes, and I verified watching the output that it was reading the
new .conf
file.
jason
::::::::-----Original Message-----
::::::::From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-
::::::::users-bounces at lists.digium.com] On Behalf Of Eric Wieling
::::::::Sent: Monday, May 05, 2008 4:58 PM
::::::::To: Asterisk Users Mailing List - Non-Commercial Discussion
::::::::Subject: Re: [asterisk-users] DTMF
::::::::
::::::::Did you remember to do a "reload" in the Asterisk CLI?
::::::::
::::::::Jason Wolfe wrote:
::::::::> Ok, I removed the T/t/w/W options but unfortunately it is
still
::::::::responding
::::::::> the same way.
::::::::>
::::::::> Ps. I have no options set on the dial() function now.
::::::::>
::::::::> jason
::::::::>
::::::::> ::::::::-----Original Message-----
::::::::> ::::::::From: asterisk-users-bounces at lists.digium.com
::::::::[mailto:asterisk-
::::::::> ::::::::users-bounces at lists.digium.com] On Behalf Of Eric
::::::::Wieling
::::::::> ::::::::Sent: Monday, May 05, 2008 4:31 PM
::::::::> ::::::::To: Asterisk Users Mailing List - Non-Commercial
::::::::Discussion
::::::::> ::::::::Subject: Re: [asterisk-users] DTMF
::::::::> ::::::::
::::::::> ::::::::Remove the T/t/w/W option from the Dial line.
::::::::> ::::::::
::::::::> ::::::::Jason Wolfe wrote:
::::::::> ::::::::> Ok, ever had one of those issues that you're
sure is
::::::::quite
::::::::> ::::::::simple to solve
::::::::> ::::::::> but you can't seem to get anything useful from
Google
:::::::r
::::::::> ::::::::anywhere else and
::::::::> ::::::::> so you're ready to throw your computer out the
::::::::window? Well,
::::::::> ::::::::I'm there!
::::::::> ::::::::>
::::::::> ::::::::> I am using a simple Zyxel VoIP phone to dial
outbound
::::::::calls to
::::::::> ::::::::a PSTN
::::::::> ::::::::> termination provider, so my extension file is one
::::::::command.
::::::::> ::::::::Dial()
::::::::> ::::::::>
::::::::> ::::::::> Anywhere I call I probably need to enter an
::::::::extension, but as
::::::::> ::::::::it should,
::::::::> ::::::::> asterisk tries to respond to these key presses.
How
::::::::do I pass
::::::::> ::::::::the DTMF tones
::::::::> ::::::::> through so that I can navigate the IVR of the
system
::::::::I'm
::::::::> ::::::::calling???
::::::::> ::::::::
::::::::> ::::::::
::::::::> ::::::::--
::::::::> ::::::::Consulting for Asterisk, Polycom, Sangoma, Digium,
::::::::Cisco, LAN,
::::::::> ::::::::WAN, QoS,
::::::::> ::::::::T-1, PRI, Frame Relay, Linux, and network design.
::::::::Based near
::::::::> ::::::::Birmingham, AL. Now accepting clients worldwide.
::::::::> ::::::::
::::::::> ::::::::_______________________________________________
::::::::> ::::::::-- 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
::::::::>
::::::::>
::::::::> _______________________________________________
::::::::> -- 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
::::::::>
::::::::>
::::::::
::::::::--
::::::::Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN,
::::::::WAN, QoS,
::::::::T-1, PRI, Frame Relay, Linux, and network design. Based near
::::::::Birmingham, AL. Now accepting clients worldwide.
::::::::
::::::::_______________________________________________
::::::::-- 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
_______________________________________________
-- 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 |
|
|
Back to top |
|
|
jason at clickforacall... Guest
|
Posted: Mon May 05, 2008 6:45 pm Post subject: [asterisk-users] DTMF |
|
|
Ok, I just called myself and tried the key presses and didn't hear the DTMF
tone on the called end of the line. Is g.711 the only codec that can be used
for this? I heard a clicking type noise which is suspect is the distorted
key press not making it through. You've given me something to go on here so
I'll look into the codec. Any further thoughts are much appreciated.
jason
::::::::-----Original Message-----
::::::::From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-
::::::::users-bounces at lists.digium.com] On Behalf Of Jerry Jones
::::::::Sent: Monday, May 05, 2008 6:16 PM
::::::::To: Asterisk Users Mailing List - Non-Commercial Discussion
::::::::Subject: Re: [asterisk-users] DTMF
::::::::
::::::::And you are using g.711 so the sounds are passing correctly and
::::::::not
::::::::being distorted? Try calling a person and pressing digits to
::::::::verify
::::::::they are inband during call?
::::::::
::::::::
::::::::On May 5, 2008, at 4:31 PM, Jason Wolfe wrote:
::::::::
::::::::> Yes, and I verified watching the output that it was reading the
::::::::> new .conf
::::::::> file.
::::::::>
::::::::> jason
::::::::>
::::::::>
::::::::> ::::::::-----Original Message-----
::::::::> ::::::::From: asterisk-users-bounces at lists.digium.com
::::::::> [mailto:asterisk-
::::::::> ::::::::users-bounces at lists.digium.com] On Behalf Of Eric
::::::::Wieling
::::::::> ::::::::Sent: Monday, May 05, 2008 4:58 PM
::::::::> ::::::::To: Asterisk Users Mailing List - Non-Commercial
::::::::Discussion
::::::::> ::::::::Subject: Re: [asterisk-users] DTMF
::::::::> ::::::::
::::::::> ::::::::Did you remember to do a "reload" in the Asterisk CLI?
::::::::> ::::::::
::::::::> ::::::::Jason Wolfe wrote:
::::::::> ::::::::> Ok, I removed the T/t/w/W options but unfortunately
::::::::it is
::::::::> still
::::::::> ::::::::responding
::::::::> ::::::::> the same way.
::::::::> ::::::::>
::::::::> ::::::::> Ps. I have no options set on the dial() function now.
::::::::> ::::::::>
::::::::> ::::::::> jason
::::::::> ::::::::>
::::::::> ::::::::> ::::::::-----Original Message-----
::::::::> ::::::::> ::::::::From: asterisk-users-bounces at lists.digium.com
::::::::> ::::::::[mailto:asterisk-
::::::::> ::::::::> ::::::::users-bounces at lists.digium.com] On Behalf Of
::::::::Eric
::::::::> ::::::::Wieling
::::::::> ::::::::> ::::::::Sent: Monday, May 05, 2008 4:31 PM
::::::::> ::::::::> ::::::::To: Asterisk Users Mailing List - Non-
::::::::Commercial
::::::::> ::::::::Discussion
::::::::> ::::::::> ::::::::Subject: Re: [asterisk-users] DTMF
::::::::> ::::::::> ::::::::
::::::::> ::::::::> ::::::::Remove the T/t/w/W option from the Dial line.
::::::::> ::::::::> ::::::::
::::::::> ::::::::> ::::::::Jason Wolfe wrote:
::::::::> ::::::::> ::::::::> Ok, ever had one of those issues that
::::::::you're
::::::::> sure is
::::::::> ::::::::quite
::::::::> ::::::::> ::::::::simple to solve
::::::::> ::::::::> ::::::::> but you can't seem to get anything useful
::::::::from
::::::::> Google
::::::::> :::::::r
::::::::> ::::::::> ::::::::anywhere else and
::::::::> ::::::::> ::::::::> so you're ready to throw your computer out
::::::::the
::::::::> ::::::::window? Well,
::::::::> ::::::::> ::::::::I'm there!
::::::::> ::::::::> ::::::::>
::::::::> ::::::::> ::::::::> I am using a simple Zyxel VoIP phone to
::::::::dial
::::::::> outbound
::::::::> ::::::::calls to
::::::::> ::::::::> ::::::::a PSTN
::::::::> ::::::::> ::::::::> termination provider, so my extension file
::::::::is one
::::::::> ::::::::command.
::::::::> ::::::::> ::::::::Dial()
::::::::> ::::::::> ::::::::>
::::::::> ::::::::> ::::::::> Anywhere I call I probably need to enter an
::::::::> ::::::::extension, but as
::::::::> ::::::::> ::::::::it should,
::::::::> ::::::::> ::::::::> asterisk tries to respond to these key
::::::::presses.
::::::::> How
::::::::> ::::::::do I pass
::::::::> ::::::::> ::::::::the DTMF tones
::::::::> ::::::::> ::::::::> through so that I can navigate the IVR of
::::::::the
::::::::> system
::::::::> ::::::::I'm
::::::::> ::::::::> ::::::::calling???
::::::::> ::::::::> ::::::::
::::::::> ::::::::> ::::::::
::::::::> ::::::::> ::::::::--
::::::::> ::::::::> ::::::::Consulting for Asterisk, Polycom, Sangoma,
::::::::Digium,
::::::::> ::::::::Cisco, LAN,
::::::::> ::::::::> ::::::::WAN, QoS,
::::::::> ::::::::> ::::::::T-1, PRI, Frame Relay, Linux, and network
::::::::design.
::::::::> ::::::::Based near
::::::::> ::::::::> ::::::::Birmingham, AL. Now accepting clients
::::::::worldwide.
::::::::> ::::::::> ::::::::
::::::::> ::::::::>
::::::::::::::::_______________________________________________
::::::::> ::::::::> ::::::::-- 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
::::::::> ::::::::>
::::::::> ::::::::>
::::::::> ::::::::> _______________________________________________
::::::::> ::::::::> -- 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
::::::::> ::::::::>
::::::::> ::::::::>
::::::::> ::::::::
::::::::> ::::::::--
::::::::> ::::::::Consulting for Asterisk, Polycom, Sangoma, Digium,
::::::::Cisco, LAN,
::::::::> ::::::::WAN, QoS,
::::::::> ::::::::T-1, PRI, Frame Relay, Linux, and network design.
::::::::Based near
::::::::> ::::::::Birmingham, AL. Now accepting clients worldwide.
::::::::> ::::::::
::::::::> ::::::::_______________________________________________
::::::::> ::::::::-- 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
::::::::>
::::::::>
::::::::> _______________________________________________
::::::::> -- 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
::::::::
::::::::
::::::::_______________________________________________
::::::::-- 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 |
|
Back to top |
|
|
stotaro at totarotechn... Guest
|
Posted: Mon May 05, 2008 7:53 pm Post subject: [asterisk-users] DTMF |
|
|
On Mon, May 5, 2008 at 3:59 PM, Jason Wolfe <jason at clickforacall.com> wrote:
Quote: |
Ok, ever had one of those issues that you're sure is quite simple to solve
but you can't seem to get anything useful from Google or anywhere else and
so you're ready to throw your computer out the window? Well, I'm there!
I am using a simple Zyxel VoIP phone to dial outbound calls to a PSTN
termination provider, so my extension file is one command? Dial()
Anywhere I call I probably need to enter an extension, but as it should,
asterisk tries to respond to these key presses. How do I pass the DTMF tones
through so that I can navigate the IVR of the system I'm calling???
Jason Wolfe
j at sonwolfe.com
|
Have you tried an IVR on your local Asterisk box?
You may need to set your sip.conf to rfc2833 or inband and make sure
it matches on the phone's setup and test it both ways, reloading in
between.
Thanks,
Steve Totaro |
|
Back to top |
|
|
jason at clickforacall... Guest
|
Posted: Mon May 05, 2008 9:27 pm Post subject: [asterisk-users] DTMF |
|
|
I'm using IAX, not SIP. I'm thinking that IAX doesn't support in-band DTMF?
::::::::-----Original Message-----
::::::::From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-
::::::::users-bounces at lists.digium.com] On Behalf Of Steve Totaro
::::::::Sent: Monday, May 05, 2008 8:53 PM
::::::::To: Asterisk Users Mailing List - Non-Commercial Discussion
::::::::Subject: Re: [asterisk-users] DTMF
::::::::
::::::::On Mon, May 5, 2008 at 3:59 PM, Jason Wolfe
::::::::<jason at clickforacall.com> wrote:
::::::::>
::::::::>
::::::::>
::::::::>
::::::::> Ok, ever had one of those issues that you're sure is quite
::::::::simple to solve
::::::::> but you can't seem to get anything useful from Google or
::::::::anywhere else and
::::::::> so you're ready to throw your computer out the window? Well,
::::::::I'm there!
::::::::>
::::::::>
::::::::>
::::::::> I am using a simple Zyxel VoIP phone to dial outbound calls to
::::::::a PSTN
::::::::> termination provider, so my extension file is one command.
::::::::Dial()
::::::::>
::::::::>
::::::::>
::::::::> Anywhere I call I probably need to enter an extension, but as
::::::::it should,
::::::::> asterisk tries to respond to these key presses. How do I pass
::::::::the DTMF tones
::::::::> through so that I can navigate the IVR of the system I'm
::::::::calling???
::::::::>
::::::::>
::::::::>
::::::::> Jason Wolfe
::::::::>
::::::::> j at sonwolfe.com
::::::::>
::::::::
::::::::Have you tried an IVR on your local Asterisk box?
::::::::
::::::::You may need to set your sip.conf to rfc2833 or inband and make
::::::::sure
::::::::it matches on the phone's setup and test it both ways, reloading
::::::::in
::::::::between.
::::::::
::::::::Thanks,
::::::::Steve Totaro
::::::::
::::::::_______________________________________________
::::::::-- 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 |
|
Back to top |
|
|
stotaro at totarotechn... Guest
|
Posted: Mon May 05, 2008 10:10 pm Post subject: [asterisk-users] DTMF |
|
|
Oh, I was not aware that the Zyxel VoIP had IAX2 support. Go
figure...............
Thanks,
Steve totaro
On Mon, May 5, 2008 at 10:27 PM, Jason Wolfe <jason at clickforacall.com> wrote:
Quote: | I'm using IAX, not SIP. I'm thinking that IAX doesn't support in-band DTMF?
::::::::-----Original Message-----
::::::::From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-
::::::::users-bounces at lists.digium.com] On Behalf Of Steve Totaro
::::::::Sent: Monday, May 05, 2008 8:53 PM
::::::::To: Asterisk Users Mailing List - Non-Commercial Discussion
::::::::Subject: Re: [asterisk-users] DTMF
::::::::
::::::::On Mon, May 5, 2008 at 3:59 PM, Jason Wolfe
::::::::<jason at clickforacall.com> wrote:
::::::::>
::::::::>
::::::::>
::::::::>
::::::::> Ok, ever had one of those issues that you're sure is quite
::::::::simple to solve
::::::::> but you can't seem to get anything useful from Google or
::::::::anywhere else and
::::::::> so you're ready to throw your computer out the window? Well,
::::::::I'm there!
::::::::>
::::::::>
::::::::>
::::::::> I am using a simple Zyxel VoIP phone to dial outbound calls to
::::::::a PSTN
::::::::> termination provider, so my extension file is one command.
::::::::Dial()
::::::::>
::::::::>
::::::::>
::::::::> Anywhere I call I probably need to enter an extension, but as
::::::::it should,
::::::::> asterisk tries to respond to these key presses. How do I pass
::::::::the DTMF tones
::::::::> through so that I can navigate the IVR of the system I'm
::::::::calling???
::::::::>
::::::::>
::::::::>
::::::::> Jason Wolfe
::::::::>
::::::::> j at sonwolfe.com
::::::::>
::::::::
::::::::Have you tried an IVR on your local Asterisk box?
::::::::
::::::::You may need to set your sip.conf to rfc2833 or inband and make
::::::::sure
::::::::it matches on the phone's setup and test it both ways, reloading
::::::::in
::::::::between.
::::::::
::::::::Thanks,
::::::::Steve Totaro
::::::::
::::::::_______________________________________________
::::::::-- 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
_______________________________________________
-- 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
|
|
|
Back to top |
|
|
|
|
|
You cannot post new topics in this forum You cannot reply to topics in this forum You cannot edit your posts in this forum You cannot delete your posts in this forum You cannot vote in polls in this forum
|
Powered by phpBB © 2001, 2005 phpBB Group
|