Sayson IP Phone Auto Answer

How To Configure Asterisk Auto Answer with Aastra IP Phones


The ALERT_INFO variable works for 480i, 480i CT, 9133i, 9112i firmware 1.2.x or later

Set it like in the example below:

Asterisk 1.4.x and higher:
exten=_*55XXX,1,SIPAddHeader(Alert-Info: info=alert-autoanswer)
exten=_*55XXX,2,Dial(SIP/${EXTEN:3},12,Ttr)

Asterisk 1.2.x:
exten=_*55XXX,1,SetVar(ALERT_INFO=info=alert-autoanswer)
exten=_*55XXX,2,Dial(SIP/${EXTEN:3},12,Ttr)

(*55 will be the prefix for the normal phone number; if a single digit is used --or anything of a different length-- adjust the slicing of the ${EXTEN}, like ${EXTEN:1} for a single digit)

Note: It's possible to make Sayson analog ADSI phones AutoAnswer as well. This has been done on other PBX systems. The trick is to create an ADSI script that will go off-hook when a certain sequence of events occur like caller-id followed by message waiting. For more information, email cheryl@sayson.com

January 30 2007
by KMorley

The following recipe works for Asterisk 1.2.13, FreePBX 2.2.0 and Aastra 480i firmware 1.4.1.1077:

There are at least two ways to get the intercom splash tone to work using the stock extensions_additional.conf file auto-generated by FreePBX. The low-tech way is to modify the DIAL statement to play a sound file (beep.gsm in this example):

exten => _*80.,n,Dial(Local/${dialnumber}@from-internal/n,12,TtrA(beep))

My preference is to send a SIP packet to the phone requesting that the phone internally generate the splash tone. This method avoids the timing problems listed in the methods below. Simply replace:

exten => _*80.,n,Set(__ALERT_INFO=Ring Answer)

with:

exten => _*80.,n,Set(__SIPADDHEADER=Alert-Info: \;info=alert-autoanswer)
(two underscores preceed SIPADDHEADER)

Note that FreePBX will overwrite these modifications the next time it auto-generates extensions_additional.conf unless you modify the FreePBX module that actually does the auto-generation.

My aastra.cfg file contains the following intercom-related settings:

sip intercom type: 2
sip intercom line: 1
sip intercom prefix code: *80
sip intercom mute mic: 0
sip allow auto answer: 1

Regardless of documentation, the following settings don't seem to have any impact on intercom functionality:

directed pickup: 1
play a ring splash: 1
priority alerting enabled: 1

Note that the SetVar methods listed below did not work for my combination of Asterisk/FreePBX/AAstra 480i firmware. Further, I've read that the SetVar method is or will be deprecated in future versions of Asterisk. Also, there may be other Alert-Info messages that these phones will respond to, but *none* are publicly documented. I discovered this alert strictly through trial and error.
EDIT: why not simply do a "strings" on the firmware binary of the phone to check for possibly supported commands?
Why go for the easy way when there is a much more challenging way? ;-)

Jun19/06
by Mustardman
Tested with Asterisk 1.2 and Aastra v1.4 firmware. Added _ in front of ALERT_INFO as pointed out by Robert. No longer a delay problem and confirmed that my 9133i beeps on it's own. Both must have been fixed in the newer firmware.
EXTEN:3 matches the number of digits in the extension. Can also use XXX for 3 digits as others have show. I prefer the wildcard "."

exten=_*55.,1,SetVar(_ALERT_INFO=info=alert-autoanswer)
exten=_*55.,2,Dial(SIP/${EXTEN:3},12,Tt)

Aug15/05
by Robertm1771

The following worked for me note the _ is needed at the start of ALERT_INFO to work with CVS-HEAD
XXX matches your phones extension number. I might be mistaken but I think my 480i beeps on pickup.
Using Firmware 1.2.0.162

exten=*5XXX,1,SetVar(_ALERT_INFO=info=alert-autoanswer)
exten=*5XXX,2,Dial(SIP/${EXTEN:2},12,Tt)

Aug10/05
by Mustardman
use wildcard "."instead of "X". Add beep to announce autoanswer to called party.
exten=_*55.,1,SetVar(ALERT_INFO=info=alert-autoanswer)
exten=_*55.,2,Dial(SIP/${EXTEN:3},12,TtrA(beep))

Note: it takes approximately 500ms for phone to produce sound so an audio file with 500ms of silence at the beginning is preferred.


See Also:

How To Configure Asterisk Auto Answer with Aastra IP Phones


The ALERT_INFO variable works for 480i, 480i CT, 9133i, 9112i firmware 1.2.x or later

Set it like in the example below:

Asterisk 1.4.x and higher:
exten=_*55XXX,1,SIPAddHeader(Alert-Info: info=alert-autoanswer)
exten=_*55XXX,2,Dial(SIP/${EXTEN:3},12,Ttr)

Asterisk 1.2.x:
exten=_*55XXX,1,SetVar(ALERT_INFO=info=alert-autoanswer)
exten=_*55XXX,2,Dial(SIP/${EXTEN:3},12,Ttr)

(*55 will be the prefix for the normal phone number; if a single digit is used --or anything of a different length-- adjust the slicing of the ${EXTEN}, like ${EXTEN:1} for a single digit)

Note: It's possible to make Sayson analog ADSI phones AutoAnswer as well. This has been done on other PBX systems. The trick is to create an ADSI script that will go off-hook when a certain sequence of events occur like caller-id followed by message waiting. For more information, email cheryl@sayson.com

January 30 2007
by KMorley

The following recipe works for Asterisk 1.2.13, FreePBX 2.2.0 and Aastra 480i firmware 1.4.1.1077:

There are at least two ways to get the intercom splash tone to work using the stock extensions_additional.conf file auto-generated by FreePBX. The low-tech way is to modify the DIAL statement to play a sound file (beep.gsm in this example):

exten => _*80.,n,Dial(Local/${dialnumber}@from-internal/n,12,TtrA(beep))

My preference is to send a SIP packet to the phone requesting that the phone internally generate the splash tone. This method avoids the timing problems listed in the methods below. Simply replace:

exten => _*80.,n,Set(__ALERT_INFO=Ring Answer)

with:

exten => _*80.,n,Set(__SIPADDHEADER=Alert-Info: \;info=alert-autoanswer)
(two underscores preceed SIPADDHEADER)

Note that FreePBX will overwrite these modifications the next time it auto-generates extensions_additional.conf unless you modify the FreePBX module that actually does the auto-generation.

My aastra.cfg file contains the following intercom-related settings:

sip intercom type: 2
sip intercom line: 1
sip intercom prefix code: *80
sip intercom mute mic: 0
sip allow auto answer: 1

Regardless of documentation, the following settings don't seem to have any impact on intercom functionality:

directed pickup: 1
play a ring splash: 1
priority alerting enabled: 1

Note that the SetVar methods listed below did not work for my combination of Asterisk/FreePBX/AAstra 480i firmware. Further, I've read that the SetVar method is or will be deprecated in future versions of Asterisk. Also, there may be other Alert-Info messages that these phones will respond to, but *none* are publicly documented. I discovered this alert strictly through trial and error.
EDIT: why not simply do a "strings" on the firmware binary of the phone to check for possibly supported commands?
Why go for the easy way when there is a much more challenging way? ;-)

Jun19/06
by Mustardman
Tested with Asterisk 1.2 and Aastra v1.4 firmware. Added _ in front of ALERT_INFO as pointed out by Robert. No longer a delay problem and confirmed that my 9133i beeps on it's own. Both must have been fixed in the newer firmware.
EXTEN:3 matches the number of digits in the extension. Can also use XXX for 3 digits as others have show. I prefer the wildcard "."

exten=_*55.,1,SetVar(_ALERT_INFO=info=alert-autoanswer)
exten=_*55.,2,Dial(SIP/${EXTEN:3},12,Tt)

Aug15/05
by Robertm1771

The following worked for me note the _ is needed at the start of ALERT_INFO to work with CVS-HEAD
XXX matches your phones extension number. I might be mistaken but I think my 480i beeps on pickup.
Using Firmware 1.2.0.162

exten=*5XXX,1,SetVar(_ALERT_INFO=info=alert-autoanswer)
exten=*5XXX,2,Dial(SIP/${EXTEN:2},12,Tt)

Aug10/05
by Mustardman
use wildcard "."instead of "X". Add beep to announce autoanswer to called party.
exten=_*55.,1,SetVar(ALERT_INFO=info=alert-autoanswer)
exten=_*55.,2,Dial(SIP/${EXTEN:3},12,TtrA(beep))

Note: it takes approximately 500ms for phone to produce sound so an audio file with 500ms of silence at the beginning is preferred.


See Also:

Created by: frisketdog, Last modification: Thu 04 of Nov, 2010 (06:37 UTC) by admin
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