Hi support,
It is possible to send a call to a yealink device (example T20P) autoanswer the call with SIPAddHeader(Alert-Info: <http://localhost/>;info=alert-autoanswer);
and if the device is in an active call interrupts this call and plays the send call in the speaker ?
I already send the call and autoanswer it, but if the device is busy the calls end without being play
Thanks in advance
Hello,
Please enable intercom barge under Feature-> Intercom and try again.
Regards,
Thanks,
Yes I have this function enable
If the line is busy in the destination the generate calls is missing
I am testing on a T20P with 9.73.0.40 firmware
Could be a firmware issue ?
I update this issue saying that I am doing a page call,
That is because maybe intercom and page have a different behaviour
As long as phone receive a same call-info header or alert-info header, there should be no difference in this case.
I attach a pcap trace sample for your reference. 101 is talking with 104, then 102 do a page/intercom to 104, 104 automatically answer the call and hold 101.
Thanks a lot for the pcap file
I am sending the following sip header
SIPAddHeader(Alert-Info: <http://localhost/>;info=alert-autoanswer);
Which is the header to set the current call on hold ?
Do you have a documentation of the different headers that yealink supports ?
Thanks in advance
We support the three auto-answer header. When phon answer the second call the current call will be automatically put in hold. Can you sene the pcap trace in your side?
Alert-Info:info=alert-autoanswer
Alert-Info:answer-after=0
Call-info:answer-after=0
Regards,
Yes I have check the file and I see intercom=true but I did not see the autoanswer option
I am using
SIPAddHeader(Alert-Info: <http://localhost/>;info=alert-autoanswer)
SIPAddHeader(Alert-Info: <http://localhost/>;answer-after=0);
But the second call is not answered
The goal of this is to implement like an evacuation protocol in which when the page call is generated an announcement is played in the phone despite the device is ringing or busy
I found in internet this header
P-Auto-Answer: <http://localhost/>; info=normal
Could be the info value something different like normal ?
I leave the header as I is en the pcap file for example
SIPAddHeader(Call-Info: <sip:192.168.1.73>;answer-after=0);
But if the phone is busy it does not auto answer the call
-- Executing [s@macro-page:2] SIPAddHeader("Local/756@page1-c8ea,2", "Call-Info: <sip:192.168.1.73>;answer-after=0") in new stack
-- Executing [s@macro-page:3] Dial("Local/756@page1-c8ea,2", "SIP/756") in new stack
-- Called 756
-- Got SIP response 486 "Busy Here" back from 192.168.1.176
-- SIP/756-0000010d is busy