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2010-07-09Kill some startup warnings and errors and make some messages more helpful in ↵tilghman5-14/+14
tracking down the source. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275105 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-09Return logic of sip_debug_test_addr() to its original functionality.mmichelson1-1/+18
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275104 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-09Fix sip_uri_parse test comparison.mmichelson1-2/+2
Part of the change with the IPv6 changes is to treat a host:port as a single 'domain' entity. This test was not updated to have the correct expectation after calling parse_uri(). git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274984 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-09Copy the address into the peer structure after we set the default portsimon.perreault1-1/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274947 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-08Fix calls of ast_sockaddr_from_sin() from IPv6 integration.rmudgett3-9/+9
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274828 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-08Add IPv6 to Asterisk.mmichelson15-1081/+1372
This adds a generic API for accommodating IPv6 and IPv4 addresses within Asterisk. While many files have been updated to make use of the API, chan_sip and the RTP code are the files which actually support IPv6 addresses at the time of this commit. The way has been paved for easier upgrading for other files in the near future, though. Big thanks go to Simon Perrault, Marc Blanchet, and Jean-Philippe Dionne for their hard work on this. (closes issue #17565) Reported by: russell Patches: asteriskv6-test-report.pdf uploaded by russell (license 2) Review: https://reviewboard.asterisk.org/r/743 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274783 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-08Implement AstData API data providers as part of the GSOC 2010 project,eliel4-2/+890
midterm evaluation. Review: https://reviewboard.asterisk.org/r/757/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274727 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-07Fixes some ref count issues introduced by r274539dvossel1-2/+8
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274686 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-07Add missing conditional around chan_dahdi mfcr2_skip_category config parameter.rmudgett1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274639 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-07Merged revisions 274579 via svnmerge from rmudgett1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r274579 | rmudgett | 2010-07-07 13:12:41 -0500 (Wed, 07 Jul 2010) | 1 line Close the DAHDI FD on error when processing chan_dahdi toneduration config parameter. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274595 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-07Use the relatedpeer field of a sip_pvt during INVITE processing.mmichelson1-49/+35
Review: https://reviewboard.asterisk.org/r/629 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274539 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-06Merged revisions 274280 via svnmerge from twilson1-10/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r274280 | twilson | 2010-07-06 17:08:20 -0500 (Tue, 06 Jul 2010) | 9 lines Add option to not do a call forward on 482 Loop Detected Asterisk has always set up a forwarded call when receiving a 482 Loop Detected. This prevents handling the call failure by just continuing on in the dialplan. Since this would be a change in behavior, the new option to disable this behavior is forwardloopdetected which defaults to 'yes'. Review: https://reviewboard.asterisk.org/r/764/ ........ (no option for trunk, just changing the behavior) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274284 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-06Status shows all non-CRC4 lines as "yellow", even if "yellow" was not in the ↵tilghman1-1/+2
bitfield. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274281 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-03Merged revisions 273793 via svnmerge from tilghman3-3/+12
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273793 | tilghman | 2010-07-02 16:36:39 -0500 (Fri, 02 Jul 2010) | 9 lines Have the DEADLOCK_AVOIDANCE macro warn when an unlock fails, to help catch potentially large software bugs. (closes issue #17407) Reported by: pdf Patches: 20100527__issue17407.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/751/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@273830 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-02Fix various typos reported by Lintiantzafrir6-24/+24
(Also fix the typos in the comments) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@273641 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-01correct handling of get_destination return valuesdvossel2-29/+55
A failure when calling the get_destination can mean multiple things. If the extension is not found, a 404 error is appropriate, but if the URI scheme is incorrect, a 404 is not approperiate. This patch adds the get_destination_result enum to differentiate between these and other failure types. The only logical difference in this patch is that we now send a "416 Unsupported URI scheme" response instead of a "404" when the scheme is not recognized. This indicates to the initiator of the INVITE to retry the request with a correct URI. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@273427 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-29Merged revisions 273060 via svnmerge from tilghman1-0/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273060 | tilghman | 2010-06-29 18:15:28 -0500 (Tue, 29 Jun 2010) | 10 lines Allow the "useragent" value to be restored into memory from the realtime backend. This value is purely informational. It does not alter configuration at all. (closes issue #16029) Reported by: Guggemand Patches: realtime-useragent.patch uploaded by Guggemand (license 897) Tested by: Guggemand ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@273078 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-29send a 400 Bad Request on malformed sip requestdvossel1-1/+7
RFC 2361 section 24.4.1 send a 400 Bad Request if the request can not be understood due to malformed syntax. Currently we simply ignore a packet with a missing callid, to, from, or via header. Instead of ignoring we now send the 400 Bad request. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272981 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-28rfc compliant sip option parsing + new unit testdvossel4-113/+308
RFC 3261 section 8.2.2.3 states that if any unsupported options are found in the Require header field, a "420 (Bad Extension)" response should be sent with an Unsupported header field containing only the unsupported options. This is not currently being done correctly. Right now, if Asterisk detects any unsupported sip options in a Require header the entire list of options are returned in the Unsupported header even if some of those options are in fact supported. This patch fixes that by building an unsupported options character buffer when parsing the options that can be sent with the 420 response. A unit test verifying this functionality has been created. Some code refactoring was required. Review: https://reviewboard.asterisk.org/r/680/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272880 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-28Merged revisions 272804 via svnmerge from mmichelson1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r272804 | mmichelson | 2010-06-28 12:31:40 -0500 (Mon, 28 Jun 2010) | 5 lines Decode URI in contact header of 302 response. ABE-2352 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272805 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-28code guidelines cleanup for retrans_pkt() functiondvossel1-33/+42
I am doing work in this function. I noticed a large number of coding guidline fixes that needed to be made. Rather than have those changes distract from my functional changes I decided to separate these into a separate patch. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272652 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-25chan_sip: more accurate retransmissionsdvossel1-0/+5
RFC3261 states that Timer A should start at 500ms (T1) by default. In chan_sip this value initially started at 1000ms and I changed it to 500ms recently. After doing that I noticed in my packet captures that it still occasionally retransmitted starting at 1000ms instead of 500ms like I told it to. This occurs because the scheduler runs in the do_monitor thread. If a new retransmission is added while the do_monitor thread is sleeping then it may not detect that retransmission for nearly 1000ms. To fix this I just poke the do_monitor thread to wake up when a new packet is sent reliably requiring retransmits. The thread then detects the new scheduler entry and adjusts its sleep time to account for it. Review: https://reviewboard.asterisk.org/r/747 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272557 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-24Merged revisions 272446 via svnmerge from rmudgett1-0/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r272446 | rmudgett | 2010-06-24 16:58:49 -0500 (Thu, 24 Jun 2010) | 10 lines ss_thread calls pri_grab without lock during overlap dial Recent changes to chan_dahdi with relation to overlap dialing call pri_grab without first obtaining a lock. (closes issue #17414) Reported by: pdf Patches: bug17414.patch uploaded by jpeeler (license 325) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272447 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-23Resolve some errors produced during module unload of chan_iax2.russell1-41/+62
The external test suite stops Asterisk using the "core stop gracefully" command. The logs from the tests show that there are a number of problems with Asterisk trying to cleanly shut down. This patch addresses the following type of error that comes from chan_iax2: [Jun 22 16:58:11] ERROR[29884]: lock.c:129 __ast_pthread_mutex_destroy: chan_iax2.c line 11371 (iax2_process_thread_cleanup): Error destroying mutex &thread->lock: Device or resource busy For an example in the context of a build, see: http://bamboo.asterisk.org/browse/AST-TRUNK-739/log The primary purpose of this patch is to change the thread pool shutdown procedure to be more explicit to ensure that the thread exits from a point where it is not holding a lock. While testing that, I encountered various crashes due to the order of operations in unload_module() being problematic. I reordered some things there, as well. Review: https://reviewboard.asterisk.org/r/736/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272370 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-23Add new AMI command LocalOptimizeAway.tringenbach1-0/+73
This command lets you request a "/n" local channel optimize itself out of the way anyway. Review: https://reviewboard.asterisk.org/r/732/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272218 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-23D'oh! Defaultenabled FTL.tilghman1-2/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272150 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-23Load all lines from realtime, not just the first one.tilghman1-26/+35
(closes issue #17144) Reported by: nahuelgreco Patches: 20100513__issue17144__trunk.diff.txt uploaded by tilghman (license 14) Tested by: tilghman git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272145 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-23Add extra protection for reinvite glare scenario.mmichelson1-1/+1
Testing proved that if Asterisk sent a connected line reinvite, and the endpoint to which the reinvite were being sent sent a reinvite, Asterisk would not properly respond with a 491 response. The reason is that on connected line reinvites, we set the dialog's invitestate to INV_CALLING to prevent Asterisk from sending a rapid flurry of connected line reinvites. For other reinvites we do not do this. Because of the current invitestate, when Asterisk received the reinvite, we interpreted this as a spiraled INVITE, and thus did not behave properly. The fix for this is to not enter the loop detection or spiral logic in handle_request_invite if the channel state is currently up. This way, no mid-call reinvites will be misinterpreted, no matter what the nature of the reinvite may have been. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272090 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-22Don't try to lock/unlock an uninitialized lock on a dahdi_pri.russell1-0/+3
This small changes prevents destroy_all_channels() from accessing a lock on an unused dahdi_pri struct, resolving a ton of ERRORs that get spewed out when shutting Asterisk down gracefully. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272052 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-22ignore CANCEL request after having already received final response to INVITEdvossel1-1/+8
RFC 3261 section 9 states that a CANCEL has no effect on a request to a UAS that has already given a final response. This patch checks to make sure there is a pending invite before allowing a CANCEL request to be processed, otherwise it responds to the CANCEL with a "481 Call/Transaction Does Not Exist". Review: https://reviewboard.asterisk.org/r/697/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@271977 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-22Merged revisions 271902 via svnmerge from mnicholson1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r271902 | mnicholson | 2010-06-22 12:31:57 -0500 (Tue, 22 Jun 2010) | 8 lines Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is set. This is necessary to keep the ref count correct. (closes issue #16815) Reported by: rain Patches: chan_sip-unref-fix.diff uploaded by rain (license 327) (modified) Tested by: rain ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@271903 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-22Merged revisions 271689 via svnmerge from mnicholson2-145/+129
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r271689 | mnicholson | 2010-06-22 07:52:27 -0500 (Tue, 22 Jun 2010) | 8 lines Modify chan_sip's packet generation api to automatically calculate the Content-Length. This is done by storing packet content in a buffer until it is actually time to send the packet, at which time the size of the packet is calculated. This change was made to ensure that the Content-Length is always correct. (closes issue #17326) Reported by: kenner Tested by: mnicholson, kenner Review: https://reviewboard.asterisk.org/r/693/ ........ This change also adds an ast_str_copy_string() function (similar to ast_copy_string), that copies one ast_str into another, properly handling embedded nulls. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@271690 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-21fixes crash when From header URI is missing "sip:"dvossel2-2/+24
(closes issue #17437) Reported by: klaus3000 Patches: sip_crash uploaded by dvossel (license 671) Tested by: klaus3000 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@271553 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-17fixes some coding guideline issuedvossel1-6/+6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@271300 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-17retransmit response to BYE requests until timer J expiresdvossel3-7/+34
According to RFC 3261 section 17.2.2, which describes non-INVITE server transaction, when a dialog enters the Completed state it must destroy the dialog after Timer J (T1*64) fires. For a BYE transaction Asterisk terminates the dialog immediately during sip_hangup() when it should be waiting T1*64 ms. This results in some odd behavior. For instance if Asterisk receives a BYE and transmits a 200ok in response, if the endpoint never receives the 200ok it will retransmit the BYE to which Asterisk responds with a "481 Call leg/transaction does not exist" because the dialog is already gone. To resolve this I made a function called sip_scheddestroy_final(). This differs slightly from sip_schedestroy() in that it enables a flag that will prevent the destruction from ever being rescheduled or canceled afterwards. It also prevents the pvt's needdestroy flag from being set which triggers the destruction of the dialog within the do_monitor thread(). By using this function we are guaranteed destruction will not occur until the scheduled time. This allows Asterisk to respond to any possible retransmits for a dialog after we process the initial BYE request for T1*64 ms. Other changes: I removed two instances where sip_cancel_destroy is used right before calling sip_scheddestroy. sip_scheddestroy always calls sip_cancel_destroy before scheduling the new destruction so it is completely unnecessary. Review: https://reviewboard.asterisk.org/r/694/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@271262 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-17Change expected operation from error to debug messagejpeeler1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@271192 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-16addition of more parse_uri test casesdvossel1-0/+54
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@271056 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-16Fix the actual place that was pointed out, for previous commit.qwell1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270983 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-16Merged revisions 270980 via svnmerge from qwell1-0/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r270980 | qwell | 2010-06-16 16:10:09 -0500 (Wed, 16 Jun 2010) | 4 lines Need to lock the agent chan before access its internal bits. Pointed out by russellb on asterisk-dev mailing list. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270981 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-16addition of G.719 pass-through supportdvossel2-0/+14
(closes issue #16293) Reported by: malcolmd Patches: g719.passthrough.patch.7 uploaded by malcolmd (license 924) format_g719.c uploaded by malcolmd (license 924) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270940 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-16Merged revisions 270866 via svnmerge from dvossel1-6/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r270866 | dvossel | 2010-06-16 12:35:29 -0500 (Wed, 16 Jun 2010) | 22 lines fixes chan_iax2 race condition There is code in chan_iax2.c that attempts to guarantee that only a single active thread will handle a call number at a time. This code works once the thread is added to an active_list of threads, but we are not currently guaranteed that a newly activated thread will enter the active_list immediately because it is left up to the thread to add itself after frames have been queued to it. This means that if two frames come in for the same call number at the same time, it is possible for them to grab two separate threads because the first thread did not add itself to the active_list fast enough. This causes some pretty complex problems. This patch resolves this race condition by immediately adding an activated thread to the active_list within the network thread and only depending on the thread to remove itself once it is done processing the frames queued to it. By doing this we are guaranteed that if another frame for the same call number comes in at the same time, that this thread will immediately be found in the active_list of threads. Review: https://reviewboard.asterisk.org/r/720/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270867 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-16Fix no call waiting caller IDjpeeler1-1/+0
Clearing the callwaitcas flag in analog_call was causing the incoming D digit to be ignored which triggers sending the caller ID. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270836 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-15Don't blow up if an ast_channel doesn't get allocated.russell1-1/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270726 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-15Make contactdeny apply to src ip when nat=yestwilson1-21/+22
chan_sip's "contactdeny" feature screens the "to be registered contact". In case of nat=yes it should not use the address information from the Contact header (which is not used at all for routing), but the source IP address of the request. Thus, if nat=yes and a client sends a request from a denied IP address (e.g. by spoofing the src-IP address) it can bypass the screening. This commit makes contactdeny apply to the src ip when nat=yes instead. (closes issue #17276) Reported by: klaus3000 Patches: patch-asterisk-trunk-contactdeny.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270658 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-14Extract sig_ss7_init_linkset() to sig_ss7.rmudgett3-16/+28
Also found a place where sig_pri_init_pri() was inlined and called it instead. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270298 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-14Add option to get untruncated channel name from AGENT function.qwell1-0/+7
The "channel" option would chop the channel name at the last '-', which made it useless for something like a channel transfer from the dialplan. The "fullchannel" option will return the channel name as-is. ABE-2218 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270260 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-14Add digit manipulation tag support to chan_dahdi/sig_pri like chan_misdn.rmudgett3-1/+77
Add the append_msn_to_cid_tag option to chan_dahdi like chan_misdn. Review: https://reviewboard.asterisk.org/r/696/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270219 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-10Attempt to fix a FreeBSD build error by including sys/stat.h.russell1-0/+1
http://bamboo.asterisk.org/download/AST-TRUNKFREEBSD/build_logs/AST-TRUNKFREEBSD-187.log git-svn-id: http://svn.digium.com/svn/asterisk/trunk@269602 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-09Merged revisions 269495 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r269495 | russell | 2010-06-09 17:18:37 -0500 (Wed, 09 Jun 2010) | 2 lines Don't stop Asterisk if chan_oss fails to register 'Console' (due to another channel driver already claiming it). ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@269497 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-09Add missing API function to sig_ss7: sig_ss7_fixup().rmudgett3-7/+32
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@269308 f38db490-d61c-443f-a65b-d21fe96a405b
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