Lwithwma Brahma authored 2 years ago and George Yang committed 1 year ago

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

Change-Id: Id942ae1bae9f5e8242b7381d6e1cfd77601d3229

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

Wenpeng Song authored 1 year ago and Yalu Zhang committed 1 year ago

Skip to use the combination of `sed` and `escape_sed_substitution` in this case. 
This fixes the incoming issue after the openwrt lift, due to the syntax and return
value for the combination usage of `sed` and `escape_sed_substitution`.

Option 'fac' in 'tel_options' is not used.
We have list of feature codes in prefixinfo settings.
Add prefixrange value if prefix is enabled.

Lukasz Kotasa authored 1 year ago and Yalu Zhang committed 1 year ago

When the option is '0', no SIP client is functioning.

Wenpeng Song authored 1 year ago and Yalu Zhang committed 1 year ago

- Retrieve a codec list from a sip provider as the default cap for chan_voicemngr.
- For incoming requests, respond with only the first codec in the remote list that
  is also in the local list(remote_first) to increase stability.

Wenpeng Song authored 1 year ago and Yalu Zhang committed 1 year ago

Now SIP signaling messages can be transported via UDP, TCP and TLS.

"uk"   - the default config which support flash-hook (R) only triggering call waiting and 3-way conference.
         R4 and R5 are for attended and unattended call transfer respectively with a timer.
"etsi" - Using R0, R1, R2, to trigger different ways of handling call waiting. R3 for 3-way conference.
         R4 and R5 are for attended and unattended call transfer respectively without a timer.

The bug happened on a newer version of busybox.

According to pjsip specification, media_encryption parameter should have
value of:
no   - res_pjsip will offer no encryption and allow no encryption to be setup.
sdes - res_pjsip will offer standard SRTP setup via in-SDP keys.
       Encrypted SIP transport should be used in conjunction with this option
       to prevent exposure of media encryption keys.
dtls - res_pjsip will offer DTLS-SRTP setup.

We support at the moment only sdes media encryption hence translate
encryption=0 from uci config to media_encryption=no in pjsip config
and encryption=1 to media_encryption=sdes.

The original value was "yes" or "no". config_asterisk.sh is also updated for the adaptation.
(cherry picked from commit 8b7701f1)

Outgoing anonymous call is handled in chan_brcm.c based on the 'anonymouscallenable' value from
chan_telephony.conf. Additional handling it by the dialplan in extension.conf is not needed and
introduced issues.

Wenpeng Song authored 2 years ago and Yalu Zhang committed 2 years ago

The original value was "yes" or "no". config_asterisk.sh is also updated for the adaptation.

Grzegorz Sluja authored 2 years ago and Yalu Zhang committed 2 years ago

Outgoing anonymous call is handled in chan_brcm.c based on the 'anonymouscallenable' value from
chan_telephony.conf. Additional handling it by the dialplan in extension.conf is not needed and
introduced issues.

Also correct some indentations.

Wenpeng Song authored 2 years ago and Yalu Zhang committed 2 years ago

Add Hangup() at the end of the Voice mail extension in order to avoid calling the local phone number
after closing the voice mail application by pressing "#".

Wenpeng Song authored 2 years ago and Yalu Zhang committed 2 years ago

Do some manipulations with users with a UK telephone number, e.g. +441473000281.

Hemlata authored 3 years ago and Yalu Zhang committed 3 years ago

Remove the unused asterisk.extensionX.boxnumber reference

Wenpeng Song authored 3 years ago and Yalu Zhang committed 3 years ago

Valid external callback enable config:
asterisk.set1.ccbs_enable='1'
asterisk.set1.internal_service='0'

/etc/firewall.sip is also removed.

Rules for those SIP and RTP ports shall be configured in firewall directly.

Hemlata authored 3 years ago and Yalu Zhang committed 3 years ago

The purpose is to remove vendor extensions in TR-104 and use PrefixRange and FacilityAction instead.

- Also change the listening TCP port from 5061 to 5060 which is the same as UDP as per RFC3261 by
default.
- Remove firewall rules when stopping asterisk

Also fix an error in /lib/voice/config_asterisk.sh for default RTP port ranges.

Markus Gothe authored 3 years ago and Yalu Zhang committed 3 years ago

The bind directive in pjsip.conf has never been working
correctly. This commit fixes the regexp to it matches
the layout in the template config.

Erik Karlsson authored 3 years ago and Yalu Zhang committed 3 years ago

The tr command does not take regular expressions and should not be
used with ranges such as [a-z] which furthermore can cause issues if
unquoted as they are interpreted by the shell.

Hemlata authored 3 years ago and Yalu Zhang committed 3 years ago

InterDigitTimerOpen, MaximumNumberOfDigits, MinimumNumberOfDigits and other placeholders