George Yang (IOPSYS) authored 10 months ago and Erik Karlsson committed 1 month ago

No logic changes

(cherry picked from commit 2d00db39)

George Yang (IOPSYS) authored 10 months ago and Erik Karlsson committed 1 month ago

No logic changes

(cherry picked from commit 8fe717c2)

George Yang authored 1 year ago and Erik Karlsson committed 1 month ago

In case multiple server addresses exist, once client received
SIP regisration response 408/503/500 from one address, it will
then move to next address. If all addresses are tried and failed,
the client will schedule a new retry with certain interval.

Following options are added to UCI asterisk sip_service_provider
section for configuring SIP failover.

   - asterisk.sip0.failover_max_retries
     Max number of failover retries.
     By default, '-1', which means unlimited.

   - asterisk.sip0.failover_interval1
     The failover begin with the retry interval.
     By default, 10 seconds.

   - asterisk.sip0.failover_interval2
     After every 3 continuous failed retries (one RETRY trys all server addresses)
     the interval is used to schedule the next retry.
     By default, 60 seconds.

(cherry picked from commit 927915ea)

Adapted from:

iopsys@d5eb8f63

Lwithwma Brahma authored 2 years ago and Erik Karlsson committed 1 month ago

The outbound_proxy was added as per the asterisk doc.
https://wiki.asterisk.org/wiki/display/AST/PJSIP+with+Proxies

Change-Id: Id942ae1bae9f5e8242b7381d6e1cfd77601d3229
(cherry picked from commit 0e189382)

George Yang authored 2 years ago and Erik Karlsson committed 1 month ago

When applying asterisk server URI, asterisk.sip_service_provider.host
should take precedence over asterisk.sip_service_provider.domain.

TR104 parameters ...SIP.Network.RegistrarServer and .ProxyServer
are all mapped to UCI asterisk.sip_service_provider.host.

Change-Id: I685480308740b9ced922274f6837d0471bdf574c
(cherry picked from commit 53014fcf)

(cherry picked from commit f8380c4d)

George Yang authored 1 year ago and Erik Karlsson committed 1 month ago

(cherry picked from commit f57639f7)

Suru Dissanaike authored 1 year ago and Erik Karlsson committed 1 month ago

(cherry picked from commit 45d9fc14)

asterisk-chan-voicemngr_2.1.14: Fix a bug that call clearing tones are not started when two other call parties hangup

- Fixed a bug in CDR file rename
- Increase debug logging for RTP telephone event

asterisk-chan-voicemngr-2.1.12: Show more information of channels and sub-channels in CLI "telephony show status"

Set connection as -1 in chan_voicemngr_close_connection() if there is
other call on the line.

This config option has become obsolete afterv the below commit in asterisk.
voice/asterisk@bbc11efe

Queue DTMF frames to the channel if the call is terminated by Asterisk core or application

SIP provider like sip0, sip1 other than user (phone number) shall be used in extensions.conf
for incoming calls.

Fix a crash when an internal call between two DECT handsets is terminated

Forward inbound RTP telephone event packets to voicemngr transparently.

- Remove uci option asterisk.sipX.dtmf_mode and use asterisk.sip_options.dtmf_mode for all SIP
  clients
- Remove dtmfrelay from chan_telephony.conf since it is not used any longer. Now this config is
  got from the peer SIP channel

Reduce source jitters by chaning the scheduling policy and priority
of the RTP handling threads.

Fix a deadlock issue triggered by R1

Support dynamic DTMF payload type

No logic changes

Fix a regression that is introduced by RTP events generated by DSP

If multiple arguments are allowed to be passed to echo, it is possible
to inject newlines using something like '-e aaa\nbbb\nccc" in UCI.

Also fix missing quotation and missing escape_sed

And also with few improvements.

No logic changes