Grzegorz Sluja authored 3 years ago and Sukru Senli committed 3 years ago

After removing 'from_user' config from  pjsip_endpoint config file we
need to use 'contact_user' which is translated to proper |USER| value,
otherwise default 'asterisk' user is used.

Hemlata authored 3 years ago and Sukru Senli committed 3 years ago

- UBUS object asterisk
- UBUS object voice.asterisk
- UCI /etc/config/asterisk

Grzegorz Sluja authored 3 years ago and Sukru Senli committed 3 years ago

Check the asterisk sounds url availability. If it is not available
(sometimes asterisk server is down and we could not build our asterisk
fork) then use the iopsys mirror url instead to build the asterisk sounds.

Markus Gothe authored 3 years ago and Sukru Senli committed 3 years ago

When we originate a call from an extension to another we
get a NULL pointer issue when going on hook.

Enable CONFIG_PACKAGE_asterisk-res-clioriginate and then
use the command below:

asterisk -rx 'channel originate local/#123457@local_extensions
extension 0000@local_extensions'

After going on hook on the FXS port, which is mapped to extension
0000, there will be a segmentation fault because the other channel
doesn't implement the getRtpStats() function pointer.

Grzegorz Sluja authored 3 years ago and Sukru Senli committed 3 years ago

This commit includes revert of:
commit d178f497
"res_pjsip:  Filter out non SIP(S) requests"

Yalu Zhang authored 3 years ago and Sukru Senli committed 3 years ago

This solves the issue of Unknown state of a device.

Grzegorz Sluja authored 3 years ago and Sukru Senli committed 3 years ago

After 'from_user' config has been removed from pjsip_endpoint config the
user in FROM header was wrong. Fix it with using session->id.number
instead of connected_id.number (which is wrong in this case).

Yalu Zhang authored 3 years ago and Sukru Senli committed 3 years ago

Don't destroy the connection when the peer sub-channel is in dialing or ringback state.

Grzegorz Sluja authored 3 years ago and Sukru Senli committed 3 years ago

There was a possibility that during attended call transfer, deadlock
happened in locking bridge_channel and channel.
bridge_channel_queue_deferred_frames() was waiting for brcm_indicate()
to unlock the channel, while brcm_indicate() which called
ast_bridge_channel_queue_frame() was waiting for
bridge_channel_queue_deferred_frames() to unlock the bridge.

Creating new macro to lock both channel and bridge_channel at the same
time and sched_yield() -> retry if failed and use it in
bridge_channel_queue_deferred_frames() avoiding the deadlock.

Yalu Zhang authored 3 years ago and Sukru Senli committed 3 years ago

Those two lines are included in brcm_signal_dialtone() which is about to invoke.

Grzegorz Sluja authored 3 years ago and Sukru Senli committed 3 years ago

Call Status for each line can be get by:
ubus call asterisk call_status '{ "line" : X }', where X is line:
{0, 1, 2, or 3}

callStatus is set in chan_brcm according to TR-104 requirements:
Idle: no ongoing calls
Dialing: the process of collecting digits for the calling number in an outgoing call, and ringing in outgoing call.
Delivered: N/A
Connected: a call is established
Alerting: ringing when an incoming call is received
Disconnected: the remote party has ended a call. but the local party has not gone on-hook.

Yalu Zhang authored 3 years ago and Sukru Senli committed 3 years ago

- asterisk.telephony -> voice.line, "line" -> "id"
- asterisk.sip -> voice.sip.client, "line" -> "uri"
- asterisk.mwi -> voice.mwi

Yalu Zhang authored 3 years ago and Sukru Senli committed 3 years ago

Also remove unused parameter "data":0 in asterisk.telephony ubus events.