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10.3.0
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10.5.0
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${ noResults }
4952 Commits (b2a7d6ccfb2de296afd49baeb04b890bd4c674eb)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
3cc8a1bb84 |
Merge "websocket: Consider pending SSL data when waiting for socket input" into 17
|
6 years ago |
|
|
7038c8676e |
res_agi: Improve GET FULL VARIABLE documentation
ASTERISK-28673 #close Reported by: Jonathan Harris Change-Id: I591afdec669622bfa19243aabec31b579652c92f |
6 years ago |
|
|
f344356290 |
websocket: Consider pending SSL data when waiting for socket input
When TLS is in use, checking the readiness of the underlying FD is insufficient for determining if there is data available to be read. So before polling the FD, check if there is any buffered data in the TLS layer and use that first. ASTERISK-28562 #close Reported by: Robert Sutton Change-Id: I95fcb3e2004700d5cf8e5ee04943f0115b15e10d |
6 years ago |
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b401891036 |
Merge "res_fax: wrap v21 detected Asterisk initiated negotiation with config option" into 17
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6 years ago |
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2634e63f4d |
Merge "res_rtp_asterisk: Add frame list cleanups to ast_rtp_read" into 17
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6 years ago |
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58a6d4ecf5 |
Merge "res_pjsip_session: Set stream state on created streams for incoming SDP." into 17
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6 years ago |
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c7966269f5 |
Merge "res_pjsip_nat: Restore original contact for REGISTER responses" into 17
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6 years ago |
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c9147a759e |
res_pjsip_session: Set stream state on created streams for incoming SDP.
A previous review, 13174, made a change whereby on an incoming offer SDP the pending topology was initialized to the configured. This caused a problem for bundle with WebRTC where bundle could reference a stream that did not actually exist if the configuration had both audio and video but the offer SDP only contained audio. This change undoes that review and instead fixes the original problem it sought to solve by setting the state of created streams based on the contents of the offer SDP. This way the stream state is not inactive until negotiation later completes. ASTERISK-28659 Change-Id: Ic5ae5a86437d3e686ac5afd91d133cc916198355 |
6 years ago |
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68724425e4 |
res_fax: wrap v21 detected Asterisk initiated negotiation with config option
A previous patch: Gerrit Change-Id: I73bb24799bfe1a48adae9c034a2edbae54cc2a39 made it so a T.38 Gateway tries to negotiate with both sides by sending T.38 negotiation request to both endpoints supported T.38 versus the previous behavior of forwarding negotiation to the "other" channel once a preamble was detected. This had the unfortunate side effect of breaking some setups. Specifically ones that set the max datagram option on an endpoint configuration (configured max datagram was not propagated since Asterisk now initiates negotiations). This patch adds a configuration option, "negotiate_both", that when enabled makes it so Asterisk initiates the negotiation requests to both endpoints vs. the previous behavior of waiting, and forwarding the request. The default is disabled keeping with the old behavior. ASTERISK-28660 Change-Id: I5deb875f3485e20bc75119ec743090655d864a1a |
6 years ago |
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86822722b6 |
res_rtp_asterisk: Add frame list cleanups to ast_rtp_read
In Asterisk 16+, there are a few places in ast_rtp_read where we've allocated a frame list but return a null frame instead of the list. In these cases, any frames left in the list won't be freed. In the vast majority of the cases, the list is empty when we return so there's nothing to free but there have been leaks reported in the wild that can be traced back to frames left in the list before returning. The escape paths now all have logic to free frames left in the list. ASTERISK-28609 Reported by: Ted G Change-Id: Ia1d7075857ebd26b47183c44b1aebb0d8f985f7a |
6 years ago |
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5fb0d1d562 |
Merge "res_pjsip_registrar.c: Prevent potential double free if AOR is not found" into 17
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6 years ago |
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fd14d7302f |
res_pjsip_nat: Restore original contact for REGISTER responses
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 |
6 years ago |
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af079f5085 |
Merge "res_pjsip_outbound_registration: add support for SRV failover" into 17
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6 years ago |
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6685a1862c |
Merge "res_pjsip_registrar.c: Prevent possible buffer overflow with domain aliases" into 17
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6 years ago |
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478a602e70 |
Merge "res_pjsip_session.c: Prevent use-after-free with TEST_FRAMEWORK enabled" into 17
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6 years ago |
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77b5ec3e7a |
Merge "parking: Fall back to parker channel name even if it matches parkee." into 17
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6 years ago |
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4e057eb9d2 |
res_pjsip_registrar.c: Prevent potential double free if AOR is not found
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 |
6 years ago |
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f26e5bacc0 |
res_pjsip_registrar.c: Prevent possible buffer overflow with domain aliases
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 |
6 years ago |
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4d56adf8fb |
res_pjsip_session.c: Prevent use-after-free with TEST_FRAMEWORK enabled
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 |
6 years ago |
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41d58a4ce2 |
parking: Fall back to parker channel name even if it matches parkee.
ASTERISK-28631 Change-Id: Ia74d084799fbb9bee3403e30d2391aacd46243cc |
6 years ago |
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4ac0299bfb |
res_pjsip_t38: T.38 error correction mode selection at 200 ok received
if asterisk offer T38 SDP with none error correction scheme and the endpoint respond with redundancy EC scheme, asterisk switch to that mode. Since we configure the endpoint as none EC mode we should not switch to any other mode except none. following logic implemented in code. 1. If asterisk offer none, and anything except none in answer will be ignored. 2. If asterisk offer fec, answer with fec, redundancy and none will be accepted. 3. If asterisk offer redundancy, answer with redundancy and none will be accepted. ASTERISK-28621 Change-Id: I343c62253ea4c8b7ee17abbfb377a4d484a14b19 |
6 years ago |
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8c99930375 |
res_pjsip_outbound_registration: add support for SRV failover
ASTERISK-28624 Change-Id: I8da7c300dd985ab7b10dbd5194aff2f737808561 |
6 years ago |
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76ef36fafc |
res_pjsip_registrar: Fix uninitlized variable warning
Fixes: error: ‘domain_name’ may be used uninitialized in this function Found with gcc (Ubuntu 9.2.1-9ubuntu2) 9.2.1 20191008 Change-Id: I44413b49ea1205aa25538142161deb73883c79e8 |
6 years ago |
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f1f28aa9e3 |
Merge "parking: Fix case where we can't get the parker." into 17
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6 years ago |
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16066ce5fc |
Merge "various files - fix some alerts raised by lgtm code analysis" into 17
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6 years ago |
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0c486e7edf |
res_rtp_asterisk: Always return provided DTLS packet length.
OpenSSL can not tolerate if the packet sent out does not match the length that it provided to the sender. This change lies and says that each time the full packet was sent. If a problem does occur then a retransmission will occur as appropriate. ASTERISK-28576 Change-Id: Id42455b15c9dc4eb987c8c023ece6fbf3c22a449 |
6 years ago |
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8bc6fa0fbd |
various files - fix some alerts raised by lgtm code analysis
This patch fixes several issues reported by the lgtm code analysis tool: https://lgtm.com/projects/g/asterisk/asterisk Not all reported issues were addressed in this patch. This patch mostly fixes confirmed reported errors, potential problematic code points, and a few other "low hanging" warnings or recommendations found in core supported modules. These include, but are not limited to the following: * innapropriate stack allocation in loops * buffer overflows * variable declaration "hiding" another variable declaration * comparisons results that are always the same * ambiguously signed bit-field members * missing header guards Change-Id: Id4a881686605d26c94ab5409bc70fcc21efacc25 |
6 years ago |
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de433cdcaf |
parking: Fix case where we can't get the parker.
ASTERISK-28616 Change-Id: Iabe31ae38d01604284fcc5c2438d44e29a32ea4d |
6 years ago |
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d638d9c6c6 |
parking: Use channel snapshot instead of channel.
There exists a scenario where a thread can hold a lock on the channels container while trying to lock a bridge. At the same time another thread can hold the lock for said bridge while attempting to retrieve a channel. This causes a deadlock. This change fixes this scenario by retrieving a channel snapshot instead of a channel, as information present in the snapshot is all that is needed. ASTERISK-28616 Change-Id: I68ceb1d62c7378addcd286e21be08a660a7cecf2 |
6 years ago |
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3084a6c617 |
Merge "res_pjsip_session: initialize pending's topology to endpoint's" into 17
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6 years ago |
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ea3daa94c8 |
res_pjsip_session: initialize pending's topology to endpoint's
Found during some testing, there is a race condition between selecting an appropriate bridge type for a call versus the applying of media on the callee's session. In some instances a native bridge type would have been chosen, but due to the callee's media not yet being established at bridge compatibility check time the simple bridge type is picked instead. When using chan_pjsip this initiates a topology change event. The topologies are then compared for the two sessions. However, when the topology was created for the caller its streams are initialized to "inactive". This topology is then used as a base when creating the callee's topology, and streams. Soon after the caller's topology's stream(s) get updated based on the sdp (get set to sendrecv in the failing scenario). Now when the topology change event is raised, and the two topologies are compared, the comparison fails due to a stream state mismatch (sendrecv vs inactive). And since they differ a reinvite is sent out (to the caller in this case). This patch makes it such that when the caller's topology is initially created it gets created based on its configured endpoint's media topology. When the endpoint's topology is created its stream's state(s) are initialized to sendrecv instead of inactive. Subsequently, now when the callee's topology is created its topology streams are now initialized to sendrecv. Thus when the topology change event occurs due to the mentioned scenario the stream states match for the given sessions, and the reinvite is not sent unless due to some other valid mismatch. Note, this patch only changes one pending media state's creation point. It's possible other places *could* be changed, however for now it was deemed best to only alter what's here. Change-Id: I6ba3a6a75f64824a1b963044c37acbe951c389c7 |
6 years ago |
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7202624b3b |
stasis: Don't hold app_registry and session locks unnecessarily
resource_events:stasis_app_message_handler() was locking the session, then attempting to determine if the app had debug enabled which locked the app_registry container. res_stasis:__stasis_app_register was locking the app_registry container then calling app_update which caused app_handler (which locks the session) to run. The result was a deadlock. * Updated resource_events:stasis_app_message_handler() to determine if debug was set (which locks the app_registry) before obtaining the session lock. * Updated res_stasis:__stasis_app_register to release the app_registry container lock before calling app_update (which locks the sesison). ASTERISK-28423 Reported by Ross Beer Change-Id: I58c69d08cb372852a63933608e4d6c3e456247b4 |
6 years ago |
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eea2d499f4 |
res_pjsip_outbound_registration: Extend documentation for "max_retries".
If the "max_retries" option is set to 0 then upon failure no further attemps are made, so explicitly document the behavior. ASTERISK-28602 Change-Id: I1e30daae9dd6c49ce18744164214d3def505acbf |
6 years ago |
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43ea6e21db |
Merge "res_calendar: Resolve memory leak on calendar destruction" into 17
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6 years ago |
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b3792e1288 |
res_calendar: Resolve memory leak on calendar destruction
Calling ne_uri_parse allocates memory that needs to be freed with a corresponding call to ne_uri_free. ASTERISK-28572 #close Change-Id: I8a6834da27000a6807d89cb7a157b2a88fcb5e61 |
6 years ago |
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e37d546109 |
res_ari_events: Add module reference when a WebSocket is open.
This change ensures that the module isn't unloaded when a WebSocket is open. Previously it was possible to unload the module manually or during shutdown which could cause a crash when any active WebSockets were terminated. ASTERISK-28585 Change-Id: I85c71ab112f99875b586419a34c08c8b34c14c5c |
6 years ago |
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1b4502ec5d |
Merge "ExternalMedia: Change return object from ExternalMedia to Channel" into 17
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6 years ago |
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2d665091a3 |
ExternalMedia: Change return object from ExternalMedia to Channel
When we created the External Media addition to ARI we created an ExternalMedia object to be returned from the channels/externalMedia REST endpoint. This object contained the channel object that was created plus local_address and local_port attributes (which are also in the Channel variables). At the time, we thought that creating an ExternalMedia object would give us more flexibility in the future but as we created the sample speech to text application, we discovered that it doesn't work so well with ARI client libraries that a) don't have the ExternalMedia object defined and/or b) can't promote the embedded channel structure to a first-class Channel object. This change causes the channels/externalMedia REST endpoint to return a Channel object (like channels/create and channels/originate) instead of the ExternalMedia object. Change-Id: If280094debd35102cf21e0a31a5e0846fec14af9 |
6 years ago |
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b8ae799ca9 |
res_rtp_asterisk: Remove a log message that slipped in.
This was only supposed to be for testing, so now it can be removed. Change-Id: I3dfc2e776e70b3196aeed5688372ea80c0214b59 |
6 years ago |
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3c712391c6 |
Merge "res_pjsip_mwi: potential double unref, and potential unwanted double link" into 17
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6 years ago |
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236ff0af7a |
Merge "cdr_pgsql cel_pgsql res_config_pgsql: compatibility with PostgreSQL 12" into 17
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6 years ago |
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79cc8ae3b8 |
cdr_pgsql cel_pgsql res_config_pgsql: compatibility with PostgreSQL 12
PostgreSQL 12 finally removed column adsrc from table pg_catalog.pg_attrdef (column default values), which has been deprecated since version 8.0. Since then, the official/correct/supported way to retrieve the column default value from the catalog is function pg_catalog.pg_get_expr(). This change breaks compatibility with pre-8.0 PostgreSQL servers, but has reached end-of-support more than a decade ago. cdr_pgsql and res_config_pgsql still have support for pre-7.3 servers, but cleaning that up is perhaps a topic for a major release, not this bugfix. ASTERISK-28571 Change-Id: I834cb3addf1937e19e87ede140bdd16cea531ebe |
6 years ago |
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45c0d99185 |
res_pjsip_mwi: potential double unref, and potential unwanted double link
When creating an unsolicited MWI aggregate subscription it was possible for the subscription object to be double unref'ed. This patch removes the explicit unref as it is not needed since the RAII_VAR will handle it at function end. Less concerning there was also a bug that could potentially allow the aggregate subscription object to be added to the unsolicited container twice. This patch ensures it is added only once. ASTERISK-28575 Change-Id: I9ccfdb5ea788bc0c3618db183aae235e53c12763 |
6 years ago |
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996fc40e2b |
res_pjsip_mwi: use an ao2_global object for mwi containers
On shutdown it's possible for the unsolicited mwi container to be freed before other dependent threads are done using it. This patch ensures this can no longer happen by wrapping the container in an ao2_global object. The solicited container was also changed too. ASTERISK-28552 Change-Id: I8f812286dc19a34916acacd71ce2ec26e1042047 |
6 years ago |
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299ba78b09 |
res_pjsip/res_pjsip_mwi: use centralized serializer pools
Both res_pjsip and res_pjsip_mwi made use of serializer pools. However, they both implemented their own serializer pool functionality that was pretty much identical in each of the source files. This patch removes the duplicated code, and uses the new 'ast_serializer_pool' object instead. Additionally res_pjsip_mwi enables a shutdown group on the pool since if the timing was right the module could be unloaded while taskprocessor threads still needed to execute, thus causing a crash. Change-Id: I959b0805ad024585bbb6276593118be34fbf6e1d |
6 years ago |
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119a18ef08 |
Merge "channel/chan_pjsip: add dialplan function for music on hold" into 17
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6 years ago |
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f7ed688ae8 |
Merge "res_pjsip_pubsub: add endpoint to some warning" into 17
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6 years ago |
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55b760d762 |
channel/chan_pjsip: add dialplan function for music on hold
Add a new dialplan function PJSIP_MOH_PASSTHROUGH that allows the on-hold behavior to be controlled on a per-call basis ASTERISK-28542 #close Change-Id: Iebe905b2ad6dbaa87ab330267147180b05a3c3a8 |
6 years ago |
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4b47d4774d |
res_pjsip_pubsub: add endpoint to some warning
There are some warning messages which are not informative without endpoint: "No registered subscribe handler for event presence.winfo" "No registered publish handler for event presence" This patch adds an endpoint name to these messages. Change-Id: Ia2811ec226d8a12659b4f9d4d224b48289650827 |
6 years ago |
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6527eb8213 |
res_pjsip_transport_websocket: Don't put brackets around local_name if IPv6
ASTERISK-28544 #close Change-Id: I8e62c444d107674c298f472e3545661de8a80dce |
6 years ago |