We lose the fact that there is a swap channel if there is one. We
currently wind up rejoining the stasis bridge as a normal join after the
swap channel has already been kicked from the bridge.
This patch preserves the swap channel so the AMI/ARI events can note that
the channel joining the bridge is swapping with another channel. Another
benefit to swaqpping in one operation is if there are any channels that
get lonely (MOH, bridge playback, and bridge record channels). The lonely
channels won't leave before the joining channel has a chance to come back
in under stasis if the swap channel is the only reason the lonely channels
are staying in the bridge.
ASTERISK-25947 #close
Reported by: Richard Mudgett
ASTERISK-24649
Reported by: John Bigelow
ASTERISK-24782
Reported by: John Bigelow
Change-Id: If37ea508831d1fed6dbfac2f191c638fc0a850ee
When create_new_id_hdr creates a new RPID or PAI header, it starts by cloning
the From header, then it overwrites the display name and uri from the channel's
connected.id. If the connected.id.name wasn't valid, create_new_id_hdr was
leaving the display name from the From header in the new RPID or PAI header.
On an attended transfer where the originator had a caller id number set but not
a display name, the re-INVITE to the final transferee had the number of the
originator but the display name of the transferer.
Added a check to clear out the display name in the new header if
connected.id.name was invalid.
ASTERISK-25942 #close
Change-Id: I60b4bf7a7ece9b7425eba74151c0b4969cd2738b
The PJSIP parsing functions provide a nice concise way to check the
length of a hostname in a SIP URI. The problem is that in order to use
those parsing functions, it's required to use them from a thread that
has registered with PJLib.
On startup, when parsing AOR configuration, the permanent URI handler
may not be run from a PJLib-registered thread. Specifically, this could
happen when Asterisk was started in daemon mode rather than
console-mode. If PJProject were compiled with assertions enabled, then
this would cause Asterisk to crash on startup.
The solution presented here is to do our own parsing of the contact URI
in order to ensure that the hostname in the URI is not too long. The
parsing does not attempt to perform a full SIP URI parse/validation,
since the hostname in the URI is what is important.
ASTERISK-25928 #close
Reported by Joshua Colp
Change-Id: Ic3d6c20ff3502507c17244a8b7e2ca761dc7fb60
Recent changes to the PJSIP registrar resulted in tests failing due to
missing AOR_CONTACT_ADDED test events. The reason for this was that the
user_agent string had junk values in it, resulting in being unable to
generate the event.
I'm going to be honest here, I have no idea why this was happening. Here
are the steps needed for the user_agent variable to get messed up:
* REGISTER is received
* First contact in the REGISTER results in a contact being removed
* Second contact in the REGISTER results in a contact being added
* The contact, AOR, expiration, and user agent all have to be passed as
format parameters to the creation of a string. Any subset of those
parameters would not be enough to cause the problem.
Looking into what was happening, the thing that struck me as odd was
that the user_agent variable was meant to be set to the value of the
User-Agent SIP header in the incoming REGISTER. However, when removing a
contact, the user_agent variable would be set (via ast_strdupa inside a
loop) to the stored contact's user_agent. This means that the
user_agent's value would be incorrect when attempting to process further
contacts in the incoming REGISTER.
The fix here is to use a different variable for the stored user agent
when removing a contact. Correcting the behavior to be correct also
means the memory usage is less weird, and the issue no longer occurs.
ASTERISK-25929 #close
Reported by Joshua Colp
Change-Id: I7cd24c86a38dec69ebcc94150614bc25f46b8c08
At shutdown it is possible for modules to be unloaded that wouldn't
normally be unloaded. This allows the environment to be cleaned up.
The res_pjsip_transport_management module did not have the unload
logic in it to clean itself up causing the res_pjsip module to not
get unloaded. As a result the res_pjsip monitor thread kept going
processing traffic and timers when it shouldn't.
Change-Id: Ic8cadee131e3b2c436a81d3ae8bb5775999ae00a
The scheduler thread that kills idle TCP connections was not registering
with PJProject properly and causing assertions if PJProject was built in
debug mode.
This change registers the thread with PJProject the first time that the
scheduler callback executes.
AST-2016-005
Change-Id: I5f7a37e2c80726a99afe9dc2a4a69bdedf661283
"Idle" here means that someone connects to us and does not send a SIP
request. PJProject will not automatically time out such connections, so
it's up to Asterisk to do it instead.
When we receive an incoming TCP connection, we will start a timer
(equivalent to transaction timer D) waiting to receive an incoming
request. If we do not receive a request in that timeframe, then we will
shut down the TCP connection.
ASTERISK-25796 #close
Reported by George Joseph
AST-2016-005
Change-Id: I7b0d303e5d140d0ccaf2f7af562071e3d1130ac6
Due to some ignored return values, Asterisk could crash if processing an
incoming REGISTER whose contact URI was above a certain length.
ASTERISK-25707 #close
Reported by George Joseph
Patches:
0001-res_pjsip-Validate-that-URIs-don-t-exceed-pjproject-.patch
AST-2016-004
Change-Id: I3ea7cee16f29c8088794de3085ca7523c1c4833d
* Added Useragent and RegExpire headers to AMI Event
ContactStatusDetail with associated documentation.
ASTERISK-25903 #close
Change-Id: If3d121e943e588d016ba51d4eb9c6a421a562239
The first available transport of the appropriate type is used now.
This patch adds new config option 'transport' for outbound-publish.
If transport is set then outbound PUBLISH requests will use this transport.
ASTERISK-25901 #close
Change-Id: Ib389130489b70e36795b0003fa5fd386e2680151
Contact expiration can occur in several places: res_pjsip_registrar,
res_pjsip_registrar_expire, and automatically when anyone calls
ast_sip_location_retrieve_aor_contact. At the same time, res_pjsip_registrar
may also be attempting to renew or add a contact. Since none of this was locked
it was possible for one thread to be renewing a contact and another thread to
expire it immediately because it was working off of stale data. This was the
casue of intermittent registration/inbound/nominal/multiple_contacts test
failures.
Now, the new named lock functionality is used to lock the aor during contact
expire and add operations and res_pjsip_registrar_expire now checks the
expiration with the lock held before deleting the contact.
ASTERISK-25885 #close
Reported-by: Josh Colp
Change-Id: I83d413c46a47796f3ab052ca3b349f21cca47059
BLF pickup isn't working on Cisco SPA and Snom phones
if the direction="recipient" attribute is missing in 'dialog' tag.
This patch adds direction="recipient" if extension state is
Ringing.
ASTERISK-24601 #close
Change-Id: I5b2c097ca29fd59e92ba237ca5d397cb1b0bcd8c
This eliminates some casts that I made a note saying v10 and above
would no longer need them.
Better late than never :)
Change-Id: I346cdb3032b6478ceb40eb6fe732978b54035572
When shutting down, the PJSIP sorcery is destroyed. The registrar
expiration module queries the PJSIP sorcery to determine what
to expire. As there was no synchronization between termination
of the expiration thread and the unloading of the module it was
possible for the thread to try to access the PJSIP sorcery after
it had been destroyed.
This change ensures that the thread is shut down before allowing
the module to be considered unloaded.
Change-Id: I69fd239edbaaf160c2d37ae00d3ac06e5596fe8b
This adds a new ARI method that allows for you to dial a channel that
you previously created in ARI.
By combining this with the create method for channels, it allows for a
workflow where a channel can be created, manipulated, and then dialed.
The channel is under control of the ARI application during all stages of
the Dial and can even be manipulated based on channel state changes
observed within an ARI application.
The overarching goal for this is to eventually be able to add a dialed
channel to a Stasis bridge earlier than the "Up" state. However, at the
moment more work is needed in the Dial and Bridge APIs in order to
facilitate that.
ASTERISK-25889 #close
Change-Id: Ic6c399c791e66c4aa52454222fe4f8b02483a205
This adds a new ARI method to the channels resource that allows for the
creation of a new channel. The channel is created and then placed into
the specified Stasis application.
This is different from the existing originate method that creates a
channel, dials it, and then places the answered channel into the
dialplan or a Stasis application. This method does not attempt to call
the channel at all. Dialing is left as a later step after channel
creation. This allows for pre-dialing channel manipulation if desired.
ASTERISK-25889
Change-Id: I3c96a0aba914b08e39f6256371a5bd4c92cbded8
Some SIP devices indicate hold/unhold using deferred SDP reinvites. In
other words, they provide no SDP in the reinvite.
A typical transaction that starts hold might look something like this:
* Device sends reinvite with no SDP
* Asterisk sends 200 OK with SDP indicating sendrecv on streams.
* Device sends ACK with SDP indicating sendonly on streams.
At this point, PJMedia's SDP negotiator saves Asterisk's local state as
being recvonly.
Now, when the device attempts to unhold, it again uses a deferred SDP
reinvite, so we end up doing the following:
* Device sends reinvite with no SDP
* Asterisk sends 200 OK with SDP indicating recvonly on streams
* Device sends ACK with SDP indicating sendonly on streams
The problem here is that Asterisk offered recvonly, and by RFC 3264's
rules, if an offer is recvonly, the answer has to be sendonly. The
result is that the device is not taken off hold.
What is supposed to happen is that Asterisk should indicate sendrecv in
the 200 OK that it sends. This way, the device has the freedom to
indicate sendrecv if it wants the stream taken off hold, or it can
continue to respond with sendonly if the purpose of the reinvite was
something else (like a session timer refresher).
The fix here is to alter the SDP negotiator's state when we receive a
reinvite with no SDP. If the negotiator's state is currently in the
recvonly or inactive state, then we alter our local state to be
sendrecv. This way, we allow the device to indicate the stream state as
desired.
ASTERISK-25854 #close
Reported by Robert McGilvray
Change-Id: I7615737276165eef3a593038413d936247dcc6ed
I forgot the new voicemail_extension wasn't a stringfield and didn't check
for NULL where I should have.
Change-Id: I029482d5c2ab72474838750461bd46b0809c90fb
The stasis_app_playback and stasis_app_recording structs need to have a
struct stasis_app_control ref. Other threads can get a reference to the
playback and recording structs from their respective global container.
These other threads can then use the control pointer they contain after
the control struct has gone.
* Add control ref to stasis_app_playback and stasis_app_recording structs.
With the refs added, the control command queue can now have a circular
control reference which will cause the control struct to never get
released if the control's command queue is not flushed when the channel
leaves the Stasis application. Also the command queue needs better
protection from adding commands if the control->is_done flag is set.
* Flush the control command queue on exit.
ASTERISK-25882 #close
Change-Id: I3cf1fb59cbe6f50f20d9e35a2c07ac07d7f4320d
* Give the struct stasis_app_control ao2 object a ref to the channel held
in the object. Now the channel will still be around if a thread needs to
post a stasis message instead of crash because the topic was destroyed.
* Moved stopping any lingering silence generator out of the struct
stasis_app_control destructor and made it a part of exiting the Stasis
application. Who knows which thread the destructor will be called under
so it cannot affect the channel's silence generator. Not only was the
channel unprotected when the silence generator was stopped, stasis may no
longer even control the channel.
ASTERISK-25882
Change-Id: I21728161b5fe638cef7976fa36a605043a7497e4
The only caller of ari_bridges_play_found() has this note:
If ari_bridges_play_found fails because the channel is unavailable for
playback, The channel will be removed from the playback list soon. We can
keep trying to get channels from the list until we either get one that
will work or else there isn't a channel for this bridge anymore, in which
case we'll revert to ari_bridges_play_new.
Change-Id: Ib068141b367ccaa17be0dab4181c98e26c5127d6
Background:
If your extension is 1000 and the voicemail access extension is 1571 and you
dial 1571, usually a dialplan rule calls voicemailmain with your extension and
you are placed directly in your mailbox. Therefore most admins program the
voicemail (or other speed dial) button on their phones to the access extension.
Some phones (Snom at least) use whatever is programmed there to also subscribe
for MWI and so can't dial one number and subscribe to another. This works fine
in chan_sip because chan_sip completely ignores the user portion of the
SUBSCRIBE message request URI. If it can match the peer, is subscribes to the
peer's mailbox. The user could be set to anything or nothing and you'd still
get subscribed to your mailbox.
Issue:
chan_pjsip actually uses the user portion of the URI to find an aor and its
mailboxes. Therefore a subscribe to 1571 results in a 404. Sure, you can
create an aor for 1571 but you certainly can't add your entire voicemail
system's mailboxes to it and everyone would get notified of every MWI.
Solution:
When an MWI subscribe comes in and an aor can't be found that matches the
resource directly, check the resource against the endpoint's aors. If an aor
is found that has a voicemail_extension that matches the resource, use it.
ASTERISK-25865
Reported-by: Ross Beer
Change-Id: I770ea185f751f1ada888fafb4b452115f1c06e9e
res_pjsip_mwi was missing the chan_sip "vmexten" functionality which adds
the Message-Account header to the MWI NOTIFY. Also, specifying mailboxes
on endpoints for unsolicited mwi and on aors for subscriptions required
that the admin know in advance which the client wanted. If you specified
mailboxes on the endpoint, subscriptions were rejected even if you also
specified mailboxes on the aor.
Voicemail extension:
* Added a global default_voicemail_extension which defaults to "".
* Added voicemail_extension to both endpoint and aor.
* Added ast_sip_subscription_get_dialog for support.
* Added ast_sip_subscription_get_sip_uri for support.
When an unsolicited NOTIFY is constructed, the From header is parsed, the
voicemail extension from the endpoint is substituted for the user, and the
result placed in the Message-Account field in the body.
When a subscribed NOTIFY is constructed, the subscription dialog local uri
is parsed, the voicemail_extension from the aor (looked up from the
subscription resource name) is substituted for the user, and the result
placed in the Message-Account field in the body.
If no voicemail extension was defined, the Message-Account field is not added
to the NOTIFY body.
mwi_subscribe_replaces_unsolicited:
* Added mwi_subscribe_replaces_unsolicited to endpoint.
The previous behavior was to reject a subscribe if a previous internal
subscription for unsolicited MWI was found for the mailbox. That remains the
default. However, if there are mailboxes also set on the aor and the client
subscribes and mwi_subscribe_replaces_unsolicited is set, the existing internal
subscription is removed and replaced with the external subscription. This
allows an admin to configure mailboxes on both the endpoint and aor and allows
the client to select which to use.
ASTERISK-25865 #close
Reported-by: Ross Beer
Change-Id: Ic15a9415091760539c7134a5ba3dc4a6a1217cea
Commit 1bce690ccb was incrementing txcount
for rtcp packets as well as rtp packets and that was causing sender reports
to be generated instead of receiver reports in cases where no rtp was actually
being sent.
Moved the txcount increment from __rtp_sento, which handles both rtp and rtcp,
to rtp_sento which only handles rtp packets.
Discovered by the hep/rtcp-receiver test.
Change-Id: Ie442e4bb947a68847a676497021ba10ffaf376d5
Added the ability to show channel statistics to chan_pjsip (cli_functions.c)
Moved the existing 'pjsip show channel(s)' functionality from
pjsip_configuration to cli_functions.c. The stats needed chan_pjsip's
private header so it made sense to move the existing channel commands as well.
Now using stasis_cache_dump to get the channel snapshots rather than retrieving
all endpoints, then getting each one's channel snapshots. Much more efficient.
Change-Id: I03b114522126d27434030b285bf6d531ddd79869
