Re: [Asterisk-Users] Cisco 7960 (SIP) with Asterisk: how to get # towork during a call
Andrew, I presume you mean in the Cisco 7940/7960 SIP Phone Administrator's Guide? When you say "mapped", dou mean that it needs an explicit entry in the dialplan.xml like: Mike - Original Message - From: "Andrew Latham" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Thursday, June 16, 2005 2:53 PM Subject: Re: [Asterisk-Users] Cisco 7960 (SIP) with Asterisk: how to get # towork during a call # and * are mapped later in the SIP(Default/MAC).cnf it has a section in the manual if you want to see why. On 6/16/05, Michael J. Tubby B.Sc (Hons) G8TIC <[EMAIL PROTECTED]> wrote: Gents, I've built an Asterisk system to replace our PBX at work and have Cisco 7960 phones (SIP 7.4) running with Asterisk 1.0.7. How to I get Asterisk to recognise the '#' being pressed during a call? In sip.conf I have entries likle this: [2001] type=friend context=local-phone auth=md5 username=2001 secret=xyzzy callerid=Jack Tubby <2001> host=dynamic nat=no canreinvite=no dtmfmode=rfc2833 incominglimit=2 [EMAIL PROTECTED] disallow=all allow=alaw allow=ulaw callgroup=2 pickupgroup=2 and in the SIPDefault.cnf for the phones I have: # Inband DTMF Settings (0-disable, 1-enable (default)) dtmf_inband: 1 # Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt ) dtmf_outofband: avt # DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up) dtmf_db_level: 3 DTMF works for voicemail and for remote services over both analogue Zap channels and digital (ISDN) channels. Asterisk doesn't appear to be 'monitoring' the audio so I can't get to Asterisk features like Asterisk's transfer, parked calls and one-tuch-record... Am I missing something? Mike ___ 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 -- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! ___ 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
Re: [Asterisk-Users] Cisco 7960 (SIP) with Asterisk: how to get # towork during a call
Mike, Were you able to get this working? Even after with a entry in the dialplan.xml does not work for me. Thanks, Ken On 6/20/05, Michael J. Tubby B.Sc (Hons) G8TIC <[EMAIL PROTECTED]> wrote: Andrew,I presume you mean in the Cisco 7940/7960 SIP Phone Administrator's Guide?When you say "mapped", dou mean that it needs an explicit entry in the dialplan.xml like: Mike- Original Message - From: "Andrew Latham" <[EMAIL PROTECTED]>To: "Asterisk Users Mailing List - Non-Commercial Discussion"< asterisk-users@lists.digium.com>Sent: Thursday, June 16, 2005 2:53 PMSubject: Re: [Asterisk-Users] Cisco 7960 (SIP) with Asterisk: how to get #towork during a call# and * are mapped later in the SIP(Default/MAC).cnf it has a section in the manual if you want to see why.On 6/16/05, Michael J. Tubby B.Sc (Hons) G8TIC <[EMAIL PROTECTED]>wrote:>> Gents,>> I've built an Asterisk system to replace our PBX at work and have Cisco > 7960 phones (SIP 7.4) running with Asterisk 1.0.7.>> How to I get Asterisk to recognise the '#' being pressed during a call?>> In sip.conf I have entries likle this:>> [2001] > type=friend> context=local-phone> auth=md5> username=2001> secret=xyzzy> callerid=Jack Tubby <2001>> host=dynamic> nat=no > canreinvite=no> dtmfmode=rfc2833> incominglimit=2> [EMAIL PROTECTED]> disallow=all> allow=alaw> allow=ulaw> callgroup=2> pickupgroup=2 >> and in the SIPDefault.cnf for the phones I have:>> # Inband DTMF Settings (0-disable, 1-enable (default))> dtmf_inband: 1>> # Out of band DTMF Settings (none-disable, avt-avt enable (default), > avt_always - always avt )> dtmf_outofband: avt>> # DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default),> 4-3db up, 5-6dB up)> dtmf_db_level: 3> > DTMF works for voicemail and for remote services over both analogue Zap> channels and digital (ISDN) channels.>> Asterisk doesn't appear to be 'monitoring' the audio so I can't get to> Asterisk > features like Asterisk's transfer, parked calls and one-tuch-record...>> Am I missing something?>>> Mike>>> ___ > 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>> --Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)WWW: http://lathama.comEmail: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED]If any of the above are down we have bigger problems than my email!___ Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Cisco 7960 (SIP) with Asterisk: how to get # towork during a call
Mahilal Silva wrote: Mike, Were you able to get this working? Even after with a entry in the dialplan.xml does not work for me. Thanks, This is what I have in my dialplan.xml Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ --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
