The SIP transaction state was reset when emitting an UPDATE or a re-INVITE
related to a COLP, preventing RTP packets to be emitted.
ASTERISK-28647
Change-Id: Ie7a30fa7a97f711e7ba6cc17f221a0993d48bd8b
The db_init() function ultimately calls db_sync() which signals the
condition before it is initialized.
Change-Id: Id4a4e025b637bc4ac7d90557fcb71d56598892ab
The use of '?' is a fairly new addition to jansson and is not
supported in the version of jansson that can be used by 13.
This change returns to previous supported behavior and removes
usage of '?'.
ASTERISK-28663
Change-Id: I6d596007ae85e8724d928865d99968f679be1142
Temporary channel lifespan is very short and CDR deactivation request
through ast_cdr_set_property() may happen when CDR is not available
yet. Use CDR_PROP() dialplan function instead, it will first wait
for pending CDR insertion requests to be processed.
ASTERISK-28636
Change-Id: I1cbe09e8d2169c0962c1195133ff260d291f2074
Instead of trying to use the defined MySQL client version from the
header use a configure check to determine whether the bool or my_bool
type should be used for defining a boolean.
ASTERISK-28604
Change-Id: Id2225b3785115de074c50c123ff1a68005b4a9c7
ConfBridge has the ability to move between different sample
rates for mixing the conference bridge. Up until now there has
only been the ability to set the conference bridge to mix at
a specific sample rate, or to let it move between sample rates
as necessary. This change adds the ability to configure a
conference bridge with a maximum sample rate so it can move
between sample rates but only up to the configured maximum.
ASTERISK-28658
Change-Id: Idff80896ccfb8a58a816e4ce9ac4ebde785963ee
add missing argument "rtt" and "status" to the documentation
The change to the dtd file allow an enumlist to contain one or many
configOptionToEnum or enum.
This is different from the previous patch I submitted when you could have a
configOptionToEnum or (a configOptionToEnum followed by one or manu enums) or
(one or many enums)
ASTERISK-28626
Change-Id: Ia71743ee7ec813f40297b0ddefeee7909db63b6d
This reverts commit 3778e1abcd.
Reason for revert: Regression in XML validation.
validity error : Content model of enumlist is not determinist:
(configOptionToEnum | (configOptionToEnum , enum+) | enum+)
As we are preparing to do releases and this is not critical
I am reverting this for now until resolved.
Change-Id: I07b539d0406e8af08934a91223e850444bde1b75
chan_sip.c:3782 __sip_xmit: sip_xmit of 0x7f1478069230 (len 600) to
213.150.203.60:1492 returned -2: Interrupted system call
returned -2 implies this wasn't actually an OS error, so errno makes no
sense either. Internal error was already logged higher up, and -2
generally means that either there isn't a valid connection available, or
the pipe notification failed, and that is already correctly logged.
ASTERISK-28651 #close
Change-Id: I46eb82924beeff9dfd86fa6c7eb87d2651b950f2
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
RFC3261 Section 10 "Registrations", specifically paragraph
"10.2.4: Refreshing Bindings", states that a user agent compares
each contact address (in a 200 REGISTER response) to see if it
created the contact. If the Asterisk endpoint has the
rewrite_contact option set however, the contact host and port sent
back in the 200 response will be the rewritten one and not the
one sent by the user agent. This prevents the user agent from
matching its own contact. Some user agents get very upset when
this happens and will not consider the registration successful.
While this is rare, it is acceptable behavior especially if more
than 1 user agent is allowed to register to a single endpoint/aor.
This commit updates res_pjsip_nat (where rewrite_contact is
implemented) to store the original incoming Contact header in
a new "x-ast-orig-host" URI parameter before rewriting it, and to
restore the original host and port to the Contact headers in the
outgoing response.
This is only done if the request is a REGISTER and rewrite_contact
is enabled.
pjsip_message_filter was also updated to ensure that if a request
comes in with any existing x-ast-* URI parameters, we remove them
so they don't conflict. Asterisk will never send a request
with those headers in it but someone might just decide to add them
to a request they craft and send to Asterisk.
NOTE: If a device changes its contact address and registers again,
it's a NEW registration. If the device didn't unregister the
original registration then all existing behavior based
on aor/remove_existing and aor/max_contacts apply.
ASTERISK-28502
Reported-by: Ross Beer
Change-Id: Idc263ad2d2d7bd8faa047e5804d96a5fe1cd282e
Due to use in res_rtp_asterisk there is a need to be able to apply an
ACL without logging any invalid/denies. It's probably sensible to at
least validate the ACL once directly after load and report invalid ACLs.
Change-Id: I256169229d945ca7c1bbf228fc492d91df345843
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
The simple fix here is simply to NULL out username and password after we call
ast_free on them. Unfortunately, I noticed that we weren't checking for
allocation failures for username and password, and adding those checks made
things noisy and cumbersome.
So instead we partially rollback the recent LGTM patch, and move the alloca
calls into find_aor_name().
ASTERISK-28641 #close
Reported by: Ross Beer
Change-Id: Ic9d01624e717a020be0b0aee31f0814e7f1ffbe2
We're appropriately sizing the id_domain_alias buffer, but then copying the data
into the id_domain one. We were then using the uninitialized id_domain_alias
buffer we just allocated.
This is ASTERISK~28641 adjacent, but significant enough to warrant its own
patch.
Change-Id: I81c38724d18deab8c6573153e2b99dbb6e2f33d9
During capabilities selection (joint capabilities of us and peer,
configured capability for this peer, or general configured
capabilities), if sip_new() does not keep framing information,
then directmedia activation will fail for any framing different
from default framing.
ASTERISK-28637
Change-Id: I99257502788653c2816fc991cac7946453082466
We need to copy the endpoint name before we call ao2_cleanup() on it,
otherwise we might try to access memory that has been reclaimed.
ASTERISK-28445 #close
Reported by: Bernhard Schmidt
Change-Id: I404b952608aa606e0babd3c4108346721fb726b3
The problem is essentially the same as in ASTERISK~28245. Besides
the direct media scenario we have an additional scenario where a
special client is involved. This device mutes audio by default in
transmit direction (no rtp frames) and activates audio only by a
foot switch. In this situation dtmf input (pin for conferences,
transfer features codes , etc) using SIP INFO mode is not
understood properly especially when SIP INFO messages are sent
quickly.
This patch ensures that SIP INFO frames are properly queued and
processed in the above scenario. The patch also corrects situations
where successive dtmf events are received quicker than the
signalled event duration (plus minimum gap/pause) allows, i.e. DTMF
events have to be buffered in the ast channel read queue and
emulation has to be processed asynchronously at slower speed.
Reported by: Thomas Arimont
patches:
trigger_dtmf_emulation.patch submitted by Thomas Arimont (license 5525)
Change-Id: I309bf61dd065c9978c8e48f5b9a936ab47de64c2
If the name of a peer is known and a SIP request is sent using that
peer's name, the address of the peer will change even if the request
fails the authentication challenge. This means that an endpoint can
be altered and even rendered unusuable, even if it was in a working
state previously. This can only occur when the nat option is set to the
default, or auto_force_rport.
This change checks the result of authentication first to ensure it is
successful before setting the address and the nat option.
ASTERISK-28589 #close
Change-Id: I581c5ed1da60ca89f590bd70872de2b660de02df
Boris P. Korzun
George Joseph
Jean Aunis
Sean Bright
Friendly Automation
Joshua C. Colp
Pascal Cadotte Michaud
Frederic LE FOLL
Stanislav
Asterisk Development Team
Joshua Colp
Jaco Kroon
Walter Doekes
Benjamin Keith Ford