Ian Pattison
2005-Apr-06 10:34 UTC
[Asterisk-Users] Keypad disabled on AriaVoice SIP phone
Hi All, I've got a couple of AriaVoice C-301P SIP phones connecting to * and have a bit of a problem. I can dial out with no problem but once * bridges the call to another channel (such as a Zap channel to the PSTN or an internal Analog phone) it appears that the keypad is then disabled which keeps me from navigating other peoples IVR trees. I've seen some ramblings on DTMF relay (usually when applied to Cisco products) but don't know if that's my issue at all. I've tried both RFC2833 and Inband Audio for DTMF handling with identical results. It literally appears that the keypad is being disabled after dialling. I've attempted to contact the vendor of the phones several times and have been unsuccessful in reaching anyone. Anyone had a similar experience? A snapshot of the phone's config can be seen at http://www.technologyassociates.ca/phone.jpg Thanks, Ian Ian Pattison, Senior Analyst Technology Associates Inc. Tel: 905-459-2100 ext. 204 Mobile: 416-568-6548 E-mail: ianp@technologyassociates.ca -------------- next part -------------- BEGIN:VCARD VERSION:2.1 N:Pattison;Ian FN:Ian Pattison ORG:Technology Associates Inc. ADR:;;9052 Creditview Rd.;Brampton;Ontario;L6V 1A1;Canada TEL;WORK:416-657-2464 TEL;WORK:905-459-2100 TEL;CELL:416-568-6548 EMAIL:ianp@technologyassociates.ca END:VCARD
I had a simliar problem with my C-302P SIP phones until I added "dtmfmode=rfc2833" to my sip.conf Ian Pattison wrote:>Hi All, > >I've got a couple of AriaVoice C-301P SIP phones connecting to * and have a bit of a problem. I can dial out with no problem but once * bridges the call to another channel (such as a Zap channel to the PSTN or an internal Analog phone) it appears that the keypad is then disabled which keeps me from navigating other peoples IVR trees. I've seen some ramblings on DTMF relay (usually when applied to Cisco products) but don't know if that's my issue at all. I've tried both RFC2833 and Inband Audio for DTMF handling with identical results. It literally appears that the keypad is being disabled after dialling. > >I've attempted to contact the vendor of the phones several times and have been unsuccessful in reaching anyone. > >Anyone had a similar experience? > >A snapshot of the phone's config can be seen at http://www.technologyassociates.ca/phone.jpg > >Thanks, > >Ian > >Ian Pattison, Senior Analyst >Technology Associates Inc. >Tel: 905-459-2100 ext. 204 >Mobile: 416-568-6548 >E-mail: ianp@technologyassociates.ca > > > >------------------------------------------------------------------------ > >BEGIN:VCARD >VERSION:2.1 >N:Pattison;Ian >FN:Ian Pattison >ORG:Technology Associates Inc. >ADR:;;9052 Creditview Rd.;Brampton;Ontario;L6V 1A1;Canada >TEL;WORK:416-657-2464 >TEL;WORK:905-459-2100 >TEL;CELL:416-568-6548 >EMAIL:ianp@technologyassociates.ca >END:VCARD > > > >------------------------------------------------------------------------ > >_______________________________________________ >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 >
Kanuri, Seshu (Company IT)
2005-Apr-06 10:57 UTC
[Asterisk-Users] Keypad disabled on AriaVoice SIP phone
Are these the same as YUXIN phones sold by eezeePhone.com? Do they have PA1688 Chipset? This does not look like a problem of the phones, but something to do with Asterisk Dial plan. Are you using 'Answer' or 'Dial' command? 1)If you are usind Dial command, do not use T or t flags 2)DTMF mode Inband works only for Ulaw. If You use any other codecs, use RFC2833 Seshu -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Ian Pattison Sent: Wednesday, April 06, 2005 1:34 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Keypad disabled on AriaVoice SIP phone Hi All, I've got a couple of AriaVoice C-301P SIP phones connecting to * and have a bit of a problem. I can dial out with no problem but once * bridges the call to another channel (such as a Zap channel to the PSTN or an internal Analog phone) it appears that the keypad is then disabled which keeps me from navigating other peoples IVR trees. I've seen some ramblings on DTMF relay (usually when applied to Cisco products) but don't know if that's my issue at all. I've tried both RFC2833 and Inband Audio for DTMF handling with identical results. It literally appears that the keypad is being disabled after dialling. I've attempted to contact the vendor of the phones several times and have been unsuccessful in reaching anyone. Anyone had a similar experience? A snapshot of the phone's config can be seen at http://www.technologyassociates.ca/phone.jpg Thanks, Ian Ian Pattison, Senior Analyst Technology Associates Inc. Tel: 905-459-2100 ext. 204 Mobile: 416-568-6548 E-mail: ianp@technologyassociates.ca -------------------------------------------------------- NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited.
Ian Pattison
2005-Apr-06 11:09 UTC
[Asterisk-Users] Keypad disabled on AriaVoice SIP phone
I've had that in there for a while now... no result for me. Ian>>> jon@silverstarcabinets.com 06/04/2005 13:46 >>>I had a simliar problem with my C-302P SIP phones until I added "dtmfmode=rfc2833" to my sip.conf Ian Pattison wrote:>Hi All, > >I've got a couple of AriaVoice C-301P SIP phones connecting to * and have a bit of a problem. I can dial out with no problem but once * bridges the call to another channel (such as a Zap channel to the PSTN or an internal Analog phone) it appears that the keypad is then disabled which keeps me from navigating other peoples IVR trees. I've seen some ramblings on DTMF relay (usually when applied to Cisco products) but don't know if that's my issue at all. I've tried both RFC2833 and Inband Audio for DTMF handling with identical results. It literally appears that the keypad is being disabled after dialling. > >I've attempted to contact the vendor of the phones several times and have been unsuccessful in reaching anyone. > >Anyone had a similar experience? > >A snapshot of the phone's config can be seen at http://www.technologyassociates.ca/phone.jpg > >Thanks, > >Ian > >Ian Pattison, Senior Analyst >Technology Associates Inc. >Tel: 905-459-2100 ext. 204 >Mobile: 416-568-6548 >E-mail: ianp@technologyassociates.ca > > > >------------------------------------------------------------------------ > >BEGIN:VCARD >VERSION:2.1 >N:Pattison;Ian >FN:Ian Pattison >ORG:Technology Associates Inc. >ADR:;;9052 Creditview Rd.;Brampton;Ontario;L6V 1A1;Canada >TEL;WORK:416-657-2464 >TEL;WORK:905-459-2100 >TEL;CELL:416-568-6548 >EMAIL:ianp@technologyassociates.ca >END:VCARD > > > >------------------------------------------------------------------------ > >_______________________________________________ >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 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
Ian Pattison
2005-Apr-06 11:38 UTC
[Asterisk-Users] Keypad disabled on AriaVoice SIP phone
I'm not sure about the chipset but they do appear to be similar... with the
exception of IAX2 support which mine do not have, SIP, H.323 and MGCP only.
A bit more detail that I've uncovered... if I call into asterisk from an
outside line and connect to my SIP phone I get DTMF tones passed in both
directions. If I call from my SIP phone to an internal Zap extension DTMF is
passed properly. But if I call from my SIP phone to an external phone (via a ZAP
FXO channel) the keypad shuts down and nothing is passed. Here's the
important part of my extensions.conf:
exten => _9NXXNXXXXXX,1,Dial(Zap/2/${EXTEN:1})
Thanks,
Ian
>>> Seshu.Kanuri@morganstanley.com 06/04/2005 13:57 >>>
Are these the same as YUXIN phones sold by eezeePhone.com? Do they have
PA1688 Chipset?
This does not look like a problem of the phones, but something to do
with Asterisk Dial plan. Are you using 'Answer' or 'Dial'
command?
1)If you are usind Dial command, do not use T or t flags
2)DTMF mode Inband works only for Ulaw. If You use any other codecs, use
RFC2833
Seshu
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Ian
Pattison
Sent: Wednesday, April 06, 2005 1:34 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Keypad disabled on AriaVoice SIP phone
Hi All,
I've got a couple of AriaVoice C-301P SIP phones connecting to * and
have a bit of a problem. I can dial out with no problem but once *
bridges the call to another channel (such as a Zap channel to the PSTN
or an internal Analog phone) it appears that the keypad is then disabled
which keeps me from navigating other peoples IVR trees. I've seen some
ramblings on DTMF relay (usually when applied to Cisco products) but
don't know if that's my issue at all. I've tried both RFC2833 and
Inband
Audio for DTMF handling with identical results. It literally appears
that the keypad is being disabled after dialling.
I've attempted to contact the vendor of the phones several times and
have been unsuccessful in reaching anyone.
Anyone had a similar experience?
A snapshot of the phone's config can be seen at
http://www.technologyassociates.ca/phone.jpg
Thanks,
Ian
Ian Pattison, Senior Analyst
Technology Associates Inc.
Tel: 905-459-2100 ext. 204
Mobile: 416-568-6548
E-mail: ianp@technologyassociates.ca
--------------------------------------------------------
NOTICE: If received in error, please destroy and notify sender. Sender does not
waive confidentiality or privilege, and use is prohibited.
_______________________________________________
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
Kanuri, Seshu (Company IT)
2005-Apr-06 12:41 UTC
[Asterisk-Users] Keypad disabled on AriaVoice SIP phone
Let us try the opposite of what I have suggested and see what it does.
Change the Dial command as under and see how that goes.
exten => _9NXXNXXXXXX,1,Dial(Zap/2/${EXTEN:1},30, Tt)
Seshu
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Ian
Pattison
Sent: Wednesday, April 06, 2005 2:38 PM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Keypad disabled on AriaVoice SIP phone
I'm not sure about the chipset but they do appear to be similar... with
the exception of IAX2 support which mine do not have, SIP, H.323 and
MGCP only.
A bit more detail that I've uncovered... if I call into asterisk from an
outside line and connect to my SIP phone I get DTMF tones passed in both
directions. If I call from my SIP phone to an internal Zap extension
DTMF is passed properly. But if I call from my SIP phone to an external
phone (via a ZAP FXO channel) the keypad shuts down and nothing is
passed. Here's the important part of my extensions.conf:
exten => _9NXXNXXXXXX,1,Dial(Zap/2/${EXTEN:1})
Thanks,
Ian
>>> Seshu.Kanuri@morganstanley.com 06/04/2005 13:57 >>>
Are these the same as YUXIN phones sold by eezeePhone.com? Do they have
PA1688 Chipset?
This does not look like a problem of the phones, but something to do
with Asterisk Dial plan. Are you using 'Answer' or 'Dial'
command?
1)If you are usind Dial command, do not use T or t flags 2)DTMF mode
Inband works only for Ulaw. If You use any other codecs, use
RFC2833
Seshu
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Ian
Pattison
Sent: Wednesday, April 06, 2005 1:34 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Keypad disabled on AriaVoice SIP phone
Hi All,
I've got a couple of AriaVoice C-301P SIP phones connecting to * and
have a bit of a problem. I can dial out with no problem but once *
bridges the call to another channel (such as a Zap channel to the PSTN
or an internal Analog phone) it appears that the keypad is then disabled
which keeps me from navigating other peoples IVR trees. I've seen some
ramblings on DTMF relay (usually when applied to Cisco products) but
don't know if that's my issue at all. I've tried both RFC2833 and
Inband
Audio for DTMF handling with identical results. It literally appears
that the keypad is being disabled after dialling.
I've attempted to contact the vendor of the phones several times and
have been unsuccessful in reaching anyone.
Anyone had a similar experience?
A snapshot of the phone's config can be seen at
http://www.technologyassociates.ca/phone.jpg
Thanks,
Ian
Ian Pattison, Senior Analyst
Technology Associates Inc.
Tel: 905-459-2100 ext. 204
Mobile: 416-568-6548
E-mail: ianp@technologyassociates.ca
--------------------------------------------------------
NOTICE: If received in error, please destroy and notify sender. Sender
does not waive confidentiality or privilege, and use is prohibited.
_______________________________________________
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 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
--------------------------------------------------------
NOTICE: If received in error, please destroy and notify sender. Sender does not
waive confidentiality or privilege, and use is prohibited.
Ian Pattison
2005-Apr-06 13:07 UTC
[Asterisk-Users] Keypad disabled on AriaVoice SIP phone
No change whatsoever. I still think it's something with the phone... none of the keys have any effect after dialling... they do not echo to the display even. It's literally as though the keypad has been disabled. Ian>>> Seshu.Kanuri@morganstanley.com 06/04/2005 15:41 >>>Let us try the opposite of what I have suggested and see what it does. Change the Dial command as under and see how that goes. exten => _9NXXNXXXXXX,1,Dial(Zap/2/${EXTEN:1},30, Tt) Seshu -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Ian Pattison Sent: Wednesday, April 06, 2005 2:38 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Keypad disabled on AriaVoice SIP phone I'm not sure about the chipset but they do appear to be similar... with the exception of IAX2 support which mine do not have, SIP, H.323 and MGCP only. A bit more detail that I've uncovered... if I call into asterisk from an outside line and connect to my SIP phone I get DTMF tones passed in both directions. If I call from my SIP phone to an internal Zap extension DTMF is passed properly. But if I call from my SIP phone to an external phone (via a ZAP FXO channel) the keypad shuts down and nothing is passed. Here's the important part of my extensions.conf: exten => _9NXXNXXXXXX,1,Dial(Zap/2/${EXTEN:1}) Thanks, Ian>>> Seshu.Kanuri@morganstanley.com 06/04/2005 13:57 >>>Are these the same as YUXIN phones sold by eezeePhone.com? Do they have PA1688 Chipset? This does not look like a problem of the phones, but something to do with Asterisk Dial plan. Are you using 'Answer' or 'Dial' command? 1)If you are usind Dial command, do not use T or t flags 2)DTMF mode Inband works only for Ulaw. If You use any other codecs, use RFC2833 Seshu -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Ian Pattison Sent: Wednesday, April 06, 2005 1:34 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Keypad disabled on AriaVoice SIP phone Hi All, I've got a couple of AriaVoice C-301P SIP phones connecting to * and have a bit of a problem. I can dial out with no problem but once * bridges the call to another channel (such as a Zap channel to the PSTN or an internal Analog phone) it appears that the keypad is then disabled which keeps me from navigating other peoples IVR trees. I've seen some ramblings on DTMF relay (usually when applied to Cisco products) but don't know if that's my issue at all. I've tried both RFC2833 and Inband Audio for DTMF handling with identical results. It literally appears that the keypad is being disabled after dialling. I've attempted to contact the vendor of the phones several times and have been unsuccessful in reaching anyone. Anyone had a similar experience? A snapshot of the phone's config can be seen at http://www.technologyassociates.ca/phone.jpg Thanks, Ian Ian Pattison, Senior Analyst Technology Associates Inc. Tel: 905-459-2100 ext. 204 Mobile: 416-568-6548 E-mail: ianp@technologyassociates.ca -------------------------------------------------------- NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. _______________________________________________ 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 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 -------------------------------------------------------- NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. _______________________________________________ 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