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Hello,

Is there a sip notify message that's handled by Yealink's phone that's auto-answering a call ?
The aim of this is to auto-answer an incoming call when the user has an headphone with microphone, without touching to the phone.
He just has to click a button on a webpage and the call is answered. But I don't want to use the URL request :
Code:
http://admin:admin@192.168.1.1/cgi-bin/ConfigManApp.com?Id=34&Command=1&Number=100&Account=101

Thanks.
Hi,

This is a notify message sample,

***************************************************************
NOTIFY sip:00156542C479@192.168.2.137:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.145:5060;branch=z9hG4bK14d54da8;rport
Max-Forwards: 70
From: "7039" <sip:7039@192.168.2.145>;tag=as2a0fd9d9
To: <sip:00156542C479@192.168.2.137:5062>;tag=389713952
Contact: <sip:7039@192.168.2.145>
Call-ID: 0db88cab3867e0d378c1de2d3ac2a05b@192.168.2.145
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX 1.6.0.28-samy-r115
Event: talk
Subscription-State: active
Max-Forwards: 20
Content-Length: 0
***************************************************************

But using an action URI will be easier, you just need to send below action URI to phone.

http://10.2.5.134/servlet?key=OK (repace the 10.2.5.134 with yours)

Regards,
James
I've sent a SIP NOTIFY to the phone, but it did not seems to accept it.

Here is the NOTIFY event :
Code:
[event-talk]
Event=>talk
Content-Length=>0

And the sip trace :
Code:
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '4a36b4ec12317b4a49ae4aa02f6ae9ee@192.168                                                                                                                                                                    .1.123:14310' Method: NOTIFY
ComDirector*CLI> sip notify autoanswer-yealink SIP50-1_10
Sending NOTIFY of type 'autoanswer-yealink' to 'SIP50-1_10'
Scheduling destruction of SIP dialog '7764cfd24e4bc8400cfc070c0c92a936                                                                                                                                                                    @192.168.1.123:14310' in 32000 ms (Method: NOTIFY)
Reliably Transmitting (NAT) to 192.168.1.131:5062:
NOTIFY sip:SIP50-1_10@192.168.1.131:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.123:14310;branch=z9hG4bK4ed3b039;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.123:14310>;tag=as36ae03ec
To: <sip:SIP50-1_10@192.168.1.131:5062>
Contact: <sip:Unknown@192.168.1.123:14310>
Call-ID: 7764cfd24e4bc8400cfc070c0c92a936@192.168.1.123:14310
CSeq: 102 NOTIFY
User-Agent: AmiritelV4
Date: Wed, 21 Jan 2015 10:19:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, IN                                                                                                                                                                    FO, PUBLISH, MESSAGE
Supported: replaces, timer
Subscription-State: terminated
Event: talk
Content-Length: 0

Maybe is it because the "From" is unknown when sent from Asterisk CLI. Any idea how to make it work ?
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