Hook flash

Asterisk Asterisk Support
1

Hi people
I need help
my scheme is collected so: analog lines come to external port CO1 of my panasonic kx-tem824 (on it in jack2, jack3, jack4, jack5 are inserted internal subscribers with numbers 102, 103, 104, 105), (jack1 with number 101) this panasonic the line leaves the first internal port in voip dlink dvg-6004s lock, it is registered on asterisk by means of trunk SIP:
host=dynamic
username=7001
secret=7001
type=friend
dtmfmode=info
qualify=yes
context=from-trunk
insecure=very
canreinvite=no
nat=no

On dvg-6004s hotline on number 07001 is registered, the group of 5 SIP subscribers of asterisk is specified asterisk in this number, which are connected to the server through linksys pap2t lock (1 lock on 2 subscribers) after which subscribers have analog phones. The scheme of communication works so: the call came to panasonic, further according to the routing plan the call gets on jack1 and leaves on dvg-6004s, and dvg-6004s according to hotline transfers a call to sip of subscribers of asterisk. The subscriber lifts a tube and talks. Suddenly the subscriber of SIP needed to transfer a call through dvg-6004s back to panasonic to the subscriber 101.
Question how to realize return on one line through jack1. I assume that it is necessary to do it with hook-flash use, but something is impossible, without asterisk the scheme was checked by everything works, but it is necessary to me through it. In ravines on flash ® button pressing on analog phone I see: WARNING[1939]: chan_sip.c:19620 handle_request_info: Unable to parse INFO message from 4a250ea060eabc657bea1d917136e268@XXX.XXX.XXX.XXX:5060. Content and respectively the flash ® button doesn’t work. I tried to connect sip phone, but in reply to me: all-circuits-busy-now&pls-try-call-later
Thanks

2

Could you repeat question in your native language and/or provide a diagram, as I’m finding the machine translation very difficult to understand. In particular, I’ve no idea what the correct word should be where the translator has used “lock”.

RFC 2833 implementations don’t have to support hook flash. After making sure that you have enabled RFC 2833, check the SDP (sip set debug on) to make sure that the gateway device is offering to support events 0-16 and not just the default of 0-15. Do the same for the phones, as they will generally not offer to accept events that they are not prepared to send, although it is possible that they will not offer an event that they can send but not usefully receive.

Even then, I’m not too sure how well Asterisk forwards event 16 across the PABX.

Also note that, if insecure=very works, you are using an obsolete version of Asterisk. If you are using a current version, you must be using the same password in both directions, in which case you should probably not have insecure at all. Also, if you have static addresses, it is better to configure everything as static.