8162 Commits (07583c288828a496cd7730b55112128fea31eaef)

Author	SHA1	Message	Date
Steve Davies	07583c2888	Further fixes to improper usage of scheduler ... When ASTERISK-25449 was closed, a number of scheduler issues mentioned in the comments were missed. These have since beed raised in ASTERISK-25476 and elsewhere. This patch attempts to collect all of the scheduler issues discovered so far and address them sensibly. ASTERISK-25476 #close Change-Id: I87a77d581e2e0d91d33b4b2fbff80f64a566d05b	10 years ago
Alexander Traud	1bff400df7	ast_format_cap_get_names: To display all formats, the buffer was increased. ... ASTERISK-25533 #close Change-Id: Ie1a9d1a6511b3f1a56b93d04475fbf8a4e40010a	10 years ago
Corey Farrell	0393bd6bed	chan_sip: Allow websockets to be disabled. ... This patch adds a new setting "websockets_enabled" to sip.conf. Setting this to false allows chan_sip to be used without causing conflicts with res_pjsip_transport_websocket. ASTERISK-24106 #close Reported by: Andrew Nagy Change-Id: I04fe8c4f2d57b2d7375e0e25826c91a72e93bea7	10 years ago
Alexander Traud	1256aedf66	chan_sip: Do not send all codecs on INVITE. ... Since version 13, Asterisk sent all allowed codecs as callee, even when the caller did not request/support them. In case of dynamic RTP payloads, this led to the same ID for different codecs, which is not allowed by SIP/SDP. Now, the intersection between the requested and the supported codecs is send again. ASTERISK-24543 #close Change-Id: Ie90cb8bf893b0895f8d505e77343de3ba152a287	10 years ago
George Joseph	5f593e7c38	build: GCC 5.1.x catches some new const, array bounds and missing paren issues ... Fixed 1 issue in each of the affected files. ASTERISK-25494 #close Reported-by: George Joseph Tested-by: George Joseph Change-Id: I818f149cd66a93b062df421e1c73c7942f5a4a77	10 years ago
Alexander Traud	869ef2a8ee	chan_sip: Fix autoframing=yes. ... With Asterisk 13, the structures ast_format and ast_codec changed. Because of that, the paketization timing (framing) of the RTP channel moved away from the formats/codecs. In the course of that change, the ptime of the callee was not honored anymore, when the optional autoframing was enabled. ASTERISK-25484 #close Change-Id: Ic600ccaa125e705922f89c72212c698215d239b4	10 years ago
Matt Jordan	f8707ae9a5	channels/chan_sip: Set cause code to 44 on RTP timeout ... To quote Olle: "When issuing a hangup due to RTP timeouts the cause code is not set. I have selected 44 based on Cisco's implementation..." ASTERISK-25135 #close Reported by: Olle Johansson patches: rtp-timeout-cause-1.8.diff uploaded by Olle Johansson (License 5267) Change-Id: Ia62100c55077d77901caee0bcae299f8dc7375fc	10 years ago
Joshua Colp	1c799b1f1e	Merge "chan_pjsip: Fix crash on reINVITE before initial INVITE completes." into 13	10 years ago
Florian Sauerteig	f939e2bd48	chan_sip: Fix port parsing for IPv6 addresses in SIP Via headers. ... If a Via header containes an IPv6 address and a port number is ommitted, as it is the standard port, we now leave the port empty and to not set it to the value after the first colon of the IPv6 address. ASTERISK-25443 #close Change-Id: Ie3c2f05471cd006bf04ed15598589c09577b1e70	10 years ago
Richard Mudgett	426263a64d	chan_pjsip: Fix crash on reINVITE before initial INVITE completes. ... Apparently some endpoints attempt to send a reINVITE before completing the initial INVITE transaction. In this case PJSIP responds appropriately to the reINVITE with a 491 INVITE request pending. Unfortunately chan_pjsip is using the initial INVITE transaction state to determine if an INVITE is the initial INVITE or a reINVITE. Since the initial INVITE transaction has not been confirmed yet chan_pjsip thinks the reINVITE is an initial INVITE and starts another PBX thread on the channel. The extra PBX thread ensures that hilarity ensues. * Fix checks for a reINVITE on incoming requests to look for the presence of a to-tag instead of the initial INVITE transaction state. * Made caller_id_incoming_request() determine what to do if there is a channel on the session or not. After a channel is created it is too late to just store the new party id on the session because the session's party id has already been copied to the channel's caller id. ASTERISK-25404 #close Reported by: Chet Stevens Change-Id: Ie78201c304a2b13226f3a4ce59908beecc2c68be	10 years ago
Matt Jordan	41b0e516e2	Merge "Fix improper usage of scheduler exposed by 5c713fdf18f" into 13	10 years ago
Matt Jordan	50fa9ff997	Fix improper usage of scheduler exposed by 5c713fdf18 ... When 5c713fdf18 was merged, it allowed for scheduled items to have an ID of '0' returned. While this was valid per the documentation for the API, it was apparently never returned previously. As a result, several users of the scheduler API viewed the result as being invalid, causing them to reschedule already scheduled items or otherwise fail in interesting ways. This patch corrects the users such that they view '0' as valid, and a returned ID of -1 as being invalid. Note that the failing HEP RTCP tests now pass with this patch. These tests failed due to a duplicate scheduling of the RTCP transmissions. ASTERISK-25449 #close Change-Id: I019a9aa8b6997584f66876331675981ac9e07e39	10 years ago
Debian Amtelco	8f777ab584	chan_pjsip: Add Referred-By header to the PJSIP REFER packet. ... Some systems require the REFER packet to include a Referred-By header. If the channel variable SIPREFERREDBYHDR is set, it passes that value as the Referred-By header value. Otherwise, it adds the current dialog’s local info. Reported by: Dan Cropp Tested by: Dan Cropp Change-Id: I3d17912ce548667edf53cb549e88a25475eda245	10 years ago
Joshua Colp	c02f07b8c4	Merge "chan_sip: Fix From header truncation for extremely long CALLERID(name)." into 13	10 years ago
Walter Doekes	b59c4d82b5	chan_sip: Fix From header truncation for extremely long CALLERID(name). ... The CALLERID(num) and CALLERID(name) and other info are placed into the `char from[256]` in initreqprep. If the name was too long, the addr-spec and params wouldn't fit. Code is moved around so the addr-spec with params is placed there first, and then fitting in as much of the display-name as possible. ASTERISK-25396 #close Change-Id: I33632baf024f01b6a00f8c7f35c91e5f68c40260	10 years ago
Scott Griepentrog	fb6b5c684b	PJSIP: avoid crash when getting rtp peer ... Although unlikely, if the tech private is returned as a NULL, chan_pjsip_get_rtp_peer() would crash. ASTERISK-25323 Change-Id: Ie231369bfa7da926fb2b9fdaac228261a3152e6a	10 years ago
Rodrigo Ramírez Norambuena	865377fc38	chan_sip.c: Validation on module reload ... Change validation on reload module because now used the cli function for reload. The sip_reload() function never fail and ever return NULL for this reason on reload() now use the call the sip_reload() and return AST_MODULE_LOAD_SUCCESS. This problem is dectected on reload by PUT method on ARI, getting always 404 http code when the module is reloaded. ASTERISK-25325 #close Reporte by: Rodrigo Ramírez Norambuena Change-Id: I41215877fb2cfc589e0d4d464000cf6825f4d7fb	10 years ago
Matt Jordan	319fe8224f	Merge "channels/pjsip/dialplan_functions: Add an option for extracting the SIP call-id" into 13	10 years ago
Matt Jordan	78d0b9d97e	channels/pjsip/dialplan_functions: Add an option for extracting the SIP call-id ... This patch adds a new option to the CHANNEL function that allows for the extraction of the SIP call-id. It is used in conjunction with the 'pjsip' option, and will return the Call-ID of the INVITE request that established the PJSIP channel. ASTERISK-25352 Change-Id: I278d1f8bcfe3a53c5aa1dadebc14e92b0abd476a	10 years ago
Scott Griepentrog	f1cd636658	Chaos: make hangup NULL tolerant ... In chan_pjsip_new, if allocation of the pvt structure fails, ast_hangup is called. But it was written to assume pvt was valid, and this change corrects that. ASTERISK-25323 Reported by: Scott Griepentrog Change-Id: I5f47860fe9cee4cd56abd3f79b108678ab72cc87	10 years ago
Joshua Colp	c01111223f	chan_sip: Allow call pickup to set the hangup cause. ... The call pickup implementation in chan_sip currently sets the channel hangup cause to "normal clearing" if call pickup is successfully performed. This action overwrites the "answered elsewhere" hangup cause set by the call pickup code and can result in the SIP device in question showing a missed call when it should not. This change sets the hangup cause to "normal clearing" as a default initially but allows the call pickup to change it as needed. ASTERISK-25346 #close Change-Id: I00ac2c269cee9e29586ee2c65e83c70e52a02cff	10 years ago
Kevin Harwell	25af2d71c8	chan_sip.c: wrong peer searched in sip_report_security_event ... In chan_sip, after handling an incoming invite a security event is raised describing authorization (success, failure, etc...). However, it was doing a lookup of the peer by extension. This is fine for register messages, but in the case of an invite it may search and find the wrong peer, or a non existent one (for instance, in the case of call pickup). Also, if the peers are configured through realtime this may cause an unnecessary database lookup when caching is enabled. This patch makes it so that sip_report_security_event searches by IP address when looking for a peer instead of by extension after an invite is processed. ASTERISK-25320 #close Change-Id: I9b3f11549efb475b6561c64f0e6da1a481d98bc4	10 years ago
Mark Michelson	d37488c7c3	Merge topic 'ASTERISK-25315' into 13 ... * changes: chan_dahdi.c: Flush the DAHDI write buffer after starting DTMF. chan_dahdi.c: Lock private struct for ast_write().	10 years ago
Richard Mudgett	256bc52b66	chan_dahdi.c: Flush the DAHDI write buffer after starting DTMF. ... Pressing DTMF digits on a phone to go out on a DAHDI channel can result in the digit not being recognized or even heard by the peer. Phone -> Asterisk -> DAHDI/channel Turns out the DAHDI behavior with DTMF generation (and any other generated tones) is exposed by the "buffers=" setting in chan_dahdi.conf. When Asterisk requests to start sending DTMF then DAHDI waits until its write buffer is empty before generating any samples for the DTMF tones. When Asterisk subsequently requests DAHDI to stop sending DTMF then DAHDI immediately stops generating the DTMF samples. As a result, the more samples there are in the DAHDI write buffer the shorter the time DTMF actually gets sent on the wire. If there are more samples in the write buffer than the time DTMF is supposed to be sent then no DTMF gets sent on the wire. With the "buffers=12,half" setting and each buffer representing 20 ms of samples then the DAHDI write buffer is going to contain around 120 ms of samples. For DTMF to be recognized by the peer the actual sent DTMF duration needs to be a minimum of 40 ms. Therefore, the intended duration needs to be a minimum of 160 ms for the peer to receive the minimum DTMF digit duration to recognize it. A simple and effective solution to work around the DAHDI behavior is for Asterisk to flush the DAHDI write buffer when sending DTMF so the full duration of DTMF is actually sent on the wire. When someone is going to send DTMF they are not likely to be talking before sending the tones so the flushed write samples are expected to just contain silence. * Made dahdi_digit_begin() flush the DAHDI write buffer after requesting to send a DTMF digit. ASTERISK-25315 #close Reported by John Hardin Change-Id: Ib56262c708cb7858082156bfc70ebd0a220efa6a	10 years ago
Richard Mudgett	800e0ea48d	chan_dahdi.c: Lock private struct for ast_write(). ... There is a window of opportunity for DTMF to not go out if an audio frame is in the process of being written to DAHDI while another thread starts sending DTMF. The thread sending the audio frame could be past the currently dialing check before being preempted by another thread starting a DTMF generation request. When the thread sending the audio frame resumes it will then cause DAHDI to stop the DTMF tone generation. The result is no DTMF goes out. * Made dahdi_write() lock the private struct before writing to the DAHDI file descriptor. ASTERISK-25315 Reported by John Hardin Change-Id: Ib4e0264cf63305ed5da701188447668e72ec9abb	10 years ago
Alexander Traud	f68c995bc9	chan_sip: Fix negotiation of iLBC 30. ... iLBC 20 was advertised in a SIP/SDP negotiation. However, only iLBC 30 is supported. Removes "a=fmtp:x mode=y" from SDP. Because of RFC 3952 section 5, only iLBC 30 is negotiated now. ASTERISK-25309 #close Change-Id: I92d724600a183eec3114da0ac607b994b1a793da	10 years ago
Mark Michelson	39cc28f6ea	res_http_websocket: Avoid passing strlen() to ast_websocket_write(). ... We have seen a rash of test failures on a 32-bit build agent. Commit 48698a5e21 solved an obvious problem where we were not encoding a 64-bit value correctly over the wire. This commit, however, did not solve the test failures. In the failing tests, ARI is attempting to send a 537 byte text frame over a websocket. When sending a frame this small, 16 bits are all that is required in order to encode the payload length on the websocket frame. However, ast_websocket_write() thinks that the payload length is greater than 65535 and therefore writes out a 64 bit payload length. Inspecting this payload length, the lower 32 bits are exactly what we would expect it to be, 537 in hex. The upper 32 bits, are junk values that are not expected to be there. In the failure, we are passing the result of strlen() to a function that expects a uint64_t parameter to be passed in. strlen() returns a size_t, which on this 32-bit machine is 32 bits wide. Normally, passing a 32-bit unsigned value to somewhere where a 64-bit unsigned value is expected would cause no problems. In fact, in manual runs of failing tests, this works just fine. However, ast_websocket_write() uses the Asterisk optional API, which means that rather than a simple function call, there are a series of macros that are used for its declaration and implementation. These macros may be causing some sort of error to occur when converting from a 32 bit quantity to a 64 bit quantity. This commit changes the logic by making existing ast_websocket_write() calls use ast_websocket_write_string() instead. Within ast_websocket_write_string(), the 64-bit converted strlen is saved in a local variable, and that variable is passed to ast_websocket_write() instead. Note that this commit message is full of speculation rather than certainty. This is because the observed test failures, while always present in automated test runs, never occur when tests are manually attempted on the same test agent. The idea behind this commit is to fix a theoretical issue by performing changes that should, at the least, cause no harm. If it turns out that this change does not fix the failing tests, then this commit should be reverted. Change-Id: I4458dd87d785ca322b89c152b223a540a3d23e67	10 years ago
Richard Mudgett	bc5d7f9c37	chan_sip.c: Tweak glue->update_peer() parameter nil value. ... Change glue->update_peer() parameter from 0 to NULL to better indicate it is a pointer. Change-Id: I8ff2e5087f0e19f6998e3488a712a2470cc823bd	10 years ago
Joshua Colp	2749721791	pjsip: Add rtp_timeout and rtp_timeout_hold endpoint options. ... This change adds support for the 'rtp_timeout' and 'rtp_timeout_hold' endpoint options. These allow the channel to be hung up if RTP is not received from the remote endpoint for a specified number of seconds. ASTERISK-25259 #close Change-Id: I3f39daaa7da2596b5022737b77799d16204175b9	10 years ago
Matt Jordan	34207887e6	Merge "chan_pjsip: Don't change formats when frame of unsupported format is received." into 13	10 years ago
Joshua Colp	2c626ceb64	chan_pjsip: Don't change formats when frame of unsupported format is received. ... Receipt of an RTP packet currently causes the formats on an PJSIP channel to change to the format of the RTP packet. In some off-nominal cases it's possible for this to be a format that has not been configured or negotiated. This change makes it so only formats explicitly configured on the endpoint are allowed. ASTERISK-25258 #close Change-Id: If93d641fb6418a285928839300d7854cab8c1020	10 years ago
Patric Marschall	abb14ac5b8	sig_pri.h: force_restart_unavailable_chans in wrong scope ... In channels/sig_pri.h, struct sig_pri_span, the field force_restart_unavailable_chans is only defined if #if defined(HAVE_PRI_MCID) is true. All other occurences of force_restart_unavailable_chans are outside of the #if defined(HAVE_PRI_MCID) endif scope. ASTERISK-25257 #close Reported by: Patric Marschall Change-Id: I071de89cc2cd0d85927a013036e235851f672549	10 years ago
Joshua Colp	cba550df7a	Merge "chan_sip: Fix early call pickup channel leak." into 13	10 years ago
Matt Jordan	49f81ddb85	Makefile: Remove coverage files on 'make clean' ... This patch updates a variety of Makefiles in Asterisk's build system to remove .gcda and .gcno files when 'make clean' is executed. These files are generated when '--enable-coverage' is passed to the Asterisk configure script. Change-Id: Ib70b41eea2ee2908885bff02e80faf9f40c84602	10 years ago
Joshua Colp	a9f721303d	Merge "chan_vpb.cc: Fix compiler warning Jenkins found." into 13	10 years ago
Walter Doekes	e0f565663b	chan_sip: Fix early call pickup channel leak. ... When handle_invite_replaces() was called, and either ast_bridge_impart() failed or there was no bridge (because the channel we're picking up was still ringing), chan_sip would leak a channel. Thanks Matt and Corey for checking the bridge path. ASTERISK-25226 #close Change-Id: Ie736bb182170a73eef5bcef0ab0376f645c260c8	10 years ago
Walter Doekes	a5a262be78	chan_mgcp: Don't call close on fd -1. ... ASTERISK-25220 #close Change-Id: Ic48f3a82f51ada87f2fb0e016c9efe0ad56f1ee3	10 years ago
Richard Mudgett	78a1f4aa46	chan_vpb.cc: Fix compiler warning Jenkins found. ... Change-Id: I0ec7fd10d56d90d5a60b12b5a7d6807f265ac5e0	10 years ago
Alexander Traud	a419c69def	chan_sip: Reload peer without its old capabilities. ... On reload, previously allowed codecs were not removed. Therefore, it was not possible to remove codecs while Asterisk was running. Furthermore, newly added codecs got appended behind the previous codecs. Therefore, it was not possible to add a codec with a priority of #1. This change removes the old capabilities before the current ones are added. ASTERISK-25182 #close Reported by: Alexander Traud patches: asterisk_13_allow_codec_reload.patch uploaded by Alexander Traud (License 6520) Change-Id: I62a06bcf15e08e8c54a35612195f97179ebe5802	10 years ago
Joshua Colp	74616ae43d	chan_sip: Destroy peers without holding peers container lock. ... Due to the use of stasis_unsubscribe_and_join in the peer destructor it is possible for a deadlock to occur when an event callback is occurring at the same time. This happens because the peer may be destroyed while holding the peers container lock. If this occurs the event callback will never be able to acquire the container lock and the unsubscribe will never complete. This change makes it so the peers that have been removed from the peers container are not destroyed with the container lock held. ASTERISK-25163 #close Change-Id: Ic6bf1d9da4310142a4d196c45ddefb99317d9a33	10 years ago
Mark Michelson	2618d1e638	Merge "chan_sip.c: Update dialog fromtag after request with auth" into 13	10 years ago
Damian Ivereigh	3f57f3f8ec	chan_sip.c: Update dialog fromtag after request with auth ... If a client sends and INVITE which is 401 rejected, then subsequently sends a new INVITE with the auth info and uses a different fromtag from the first INVITE, Asterisk will accept the new INVITE as part of the original dialog - match_req_to_dialog() specifically ignores the fromtag. However it does not update the stored dialog with the new fromtag. This results in Asterisk being unable to match future packets that are part of this dialog (such as the ACK to the OK or the OK to the BYE), and the call is dropped. This problem was originally found when using an NEC-i SV8100-GE (NEC SIP Card). * After a successful match of a packet to the dialog, if the packet is not a SIP_RESPONSE, authentication is present and the fromtags are different, the stored fromtag is updated with the one from the recent INVITE. ASTERISK-25154 #close Reported by: Damian Ivereigh Tested by: Damian Ivereigh Change-Id: I5c16cf3b409e5ef9f2b2fe974b6bd2a45a6aa17e	10 years ago
Matt Jordan	30a0f2d9ac	chan_pjsip: Set the context and extension on the channel when created ... Prior to this patch, chan_pjsip was failing to pass the endpoint's context and the desired extension to the ast_channel_alloc_* routine. This caused a new channel snapshot to be issued without a context and extension, which can cause some reporting issues for users of AMI, CEL, and other APIs. The channel driver would later set the context and extension on the channel such that the channel would start in the correct location in the dialplan, but the information reported in the initial event would be incorrect. This patch modifies the channel driver such that it now passes the context and extension directly into the allocation routine. This provides the information in the new channel snapshot published over Stasis. ASTERISK-25156 #close Reported by: cloos Change-Id: Ic6f8542836e596db8f662071d118e8f934fdf25e	10 years ago
Yousf Ateya	cf98c744d5	chan_iax2: Prevent deadlock between hangup and sending lagrq/ping ... channels/chan_iax.c: Prevent the deadlock between iax2_hangup and send_lagrq/ send_ping. This deadlock happens because the scheduled task send_lagrq(or send_ping) starts execution after the call hangup procedure starts but before it deletes the tasks in the scheduler. The solution is to delete scheduled lagrq (and ping) task asynchronously (i.e. schedule AST_SCHED_DEL for these tasks); By this, AST_SCHED_DEL will be called in a new context (doesn't have callno locked). This commit also cleans up the procedure of sending LAGRQ and PING. main/sched.c: Do not assert when deleting non existant entry from scheduler. This assert seems to be the reason for a lot of awkward code to avoid it. ASTERISK-24983 #close Reported by: Y Ateya Change-Id: I03bec1fc8faacb89630269e935fa667c6d6c080c	10 years ago
Corey Farrell	55c8daf88b	Fix unsafe uses of ast_context pointers. ... Although ast_context_find, ast_context_find_or_create and ast_context_destroy perform locking of the contexts table, any context pointer can become invalid at any time that the contexts table is unlocked. This change adds locking around all complete operations involving these functions. Places where ast_context_find was followed by ast_context_destroy have been replaced with calls ast_context_destroy_by_name. ASTERISK-25094 #close Reported by: Corey Farrell Change-Id: I1866b6787730c9c4f3f836b6133ffe9c820734fa	10 years ago
Corey Farrell	0d266cbe02	Stasis: Fix unsafe use of stasis_unsubscribe in modules. ... Many uses of stasis_unsubscribe in modules can be reached through unload. These have been switched to stasis_unsubscribe_and_join. Some subscription callbacks do nothing, for these I've created a noop callback function in stasis.c. This is used by some modules that monitor MWI topics in order to enable cache, since the callback does not become invalid after dlclose it is safe to use stasis_unsubscribe on these, even during module unload. ASTERISK-25121 #close Change-Id: Ifc2549fbd8eef7d703c222978e8f452e2972189c	10 years ago
snuffy	f9114179e6	chan_pjsip: Fix crash during off-nominal when no endpoint specified. ... Add missing return -1 when no endpoint name is specified. ASTERISK-25086 #close Reported by: snuffy Change-Id: I9de76c2935a1f4e3f0cffe97a670106f5605e89e	10 years ago
Richard Mudgett	c780b6e431	chan_dahdi/sig_pri: Fix crash on ISDN call hangup collision. ... If an ISDN call is hungup by both sides at the same time a crash could happen. * Added missing NULL checks for the owner channel after calling pri_queue_pvt_cause_data() in two places. Code after those calls need to check the owner channel pointer for NULL before use because pri_queue_pvt_cause_data() needs to do deadlock avoidance to lock the owner and the owner may get hung up. ASTERISK-21893 #close Reported by: Alexandr Gordeev Change-Id: Ica3e266ebc7a894b41d762326f08653e1904bb9a	10 years ago
Corey Farrell	ad6ea29697	Remove unneeded uses of optional_api providers. ... A few cases exist where headers of optional_api provders are included but not needed. This causes unneeded calls to ast_optional_api_use. * Don't include optional_api.h from sip_api.h. * Move 'struct ast_channel_monitor' to channel.h. * Don't include monitor.h from chan_sip.c, channel.c or features.c. The move of struct ast_channel_monitor is needed since channel.c depends on it. This has no effect on users of monitor.h since channel.h is included from monitor.h. ASTERISK-25051 #close Reported by: Corey Farrell Change-Id: I53ea65a9fc9693c89f8bcfd6120649bfcfbc3478	10 years ago
Richard Mudgett	d3c310a28c	chan_dahdi: Add the chan_dahdi.conf force_restart_unavailable_chans option. ... Some telco switches occasionally ignore ISDN RESTART requests. The fix for ASTERISK-19608 added an escape clause for B channels in the restarting state if the telco ignores a RESTART request. If the telco fails to acknowledge the RESTART then Asterisk will assume the telco acknowledged the RESTART on the second call attempt requesting the B channel by the telco. The escape clause is good for dealing with RESTART requests in general but it does cause the next call for the restarting B channel to be rejected if the telco insists the call must go on that B channel. chan_dahdi doesn't really need to issue a RESTART request in response to receiving a cause 44 (Requested channel not available) code. Sending the RESTART in such a situation is not required (nor prohibited) by the standards. I think chan_dahdi does this for historical reasons to deal with buggy peers to get channels unstuck in a similar fashion as the chan_dahdi.conf resetinterval option. * Add the chan_dahdi.conf force_restart_unavailable_chans compatability option that when disabled will prevent chan_dahdi from trying to RESTART the channel in response to a cause 44 code. ASTERISK-25034 #close Reported by: Richard Mudgett Change-Id: Ib8b17a438799920f4a2038826ff99a1884042f65	10 years ago