summaryrefslogtreecommitdiff
path: root/channels
AgeCommit message (Collapse)Author
2009-07-20reg->username is parsed only once on sip reloadDavid Vossel
The registration string can contain an expanded user portion of the form user@domain. This expanded user portion was stored in reg->username and parsed each time there is a registration refresh. Now, the domain portion of the user is parsed and stored separately in the regdomain field. (closes issue #14331) Reported by: Nick_Lewis Patches: chan_sip.c.domainparse3.patch uploaded by Nick (license 657) Tested by: Nick_Lewis, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207484 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-20Merged revisions 207423 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207423 | mmichelson | 2009-07-20 14:39:59 -0500 (Mon, 20 Jul 2009) | 33 lines Answer video SDP offers properly when videosupport is not enabled. Copied from Review board: In issue 12434, the reporter describes a situation in which audio and video is offered on the call, but because videosupport is disabled in sip.conf, Asterisk gives no response at all to the video offer. According to RFC 3264, all media offers should have a corresponding answer. For offers we do not intend to actually reply to with meaningful values, we should still reply with the port for the media stream set to 0. In this patch, we take note of what types of media have been offered and save the information on the sip_pvt. The SDP in the response will take into account whether media was offered. If we are not otherwise going to answer a media offer, we will insert an appropriate m= line with the port set to 0. It is important to note that this patch is pretty much a bandage being applied to a broken bone. The patch *only* helps for situations where video is offered but videosupport is disabled and when udptl_pt is disabled but T.38 is offered. Asterisk is not guaranteed to respond to every media offer. Notable cases are when multiple streams of the same type are offered. The 2 media stream limit is still present with this patch, too. In trunk and the 1.6.X branches, things will be a bit different since Asterisk also supports text in SDPs as well. (closes issue #12434) Reported by: mnnojd Review: https://reviewboard.asterisk.org/r/311 Review: https://reviewboard.asterisk.org/r/313 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207424 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-18Merged 207316 fromRichard Mudgett
https://origsvn.digium.com/svn/asterisk/be/branches/C.2-... .......... r207316 | rmudgett | 2009-07-17 23:05:05 -0500 (Fri, 17 Jul 2009) | 20 lines Fixed incoming calls being matched to MSNs without type-of-number prefix added. For an incoming ISDN call the dialed.number is incorrectly matched against the configured MSNs in misdn.conf. The numbers passed to the dialplan include the configured prefix for the dialed.number_type, whereas the check against the configured MSNs (to decide if the call is accepted at all), is executed without the configured prefix. e.g., dialed.number = 241168020, TON = national, configured national prefix is "0". (This is the TON which is used by ISDN providers in the Netherlands.) In chan_misdn.c:cb_events() in case EVENT_SETUP the call to misdn_cfg_is_msn_valid() uses the unnormalized number 241168020, but 57 lines later the call to read_config() adds the prefix, and the dialed.number is now 0241168020, which is then used in the dialplan. misdn_cfg_is_msn_valid() must use the normalized number, too. JIRA ABE-1912 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207318 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-17fixes an error in r203638 CEL commitDavid Vossel
(closes issue #15525) Reported by: elguero Patches: iax2-double-unlock.patch uploaded by elguero (license 37) 15525.diff uploaded by dvossel (license 671) Tested by: dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207225 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-17Merged revisions 207155 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207155 | jpeeler | 2009-07-17 14:36:19 -0500 (Fri, 17 Jul 2009) | 7 lines Fix format specifier to print out an unsigned long long. Yep, it's even ifdefed out code. But it made it to the RR list... (closes issue #14726) Reported by: lmadsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207156 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-17sip option flags handled incorrectlyDavid Vossel
(closes issue #15376) Reported by: Takehiko Ooshima Tested by: dvossel, Takehiko_Ooshima git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207029 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-17Fix segfault in sig_analog when using callwaiting, respect callwaiting optionsJeff Peeler
Sig_analog handles allocating the sub channel for callwaiting, so no longer try to do it in chan_dahdi. Modified analog_alloc_sub to only mark the sub as allocated upon success of the alloc_sub callback, which was responsible for the segfault. Also, the callwaiting and callwaitingcallerid options were being unconditionally set to true. Now, the options are properly set from chan_dahdi.conf. (closes issue #15508) Reported by: elguero Tested by: elguero git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206998 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-17Merged revisions 206938 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206938 | dvossel | 2009-07-17 11:05:06 -0500 (Fri, 17 Jul 2009) | 14 lines SIP incorrect From: header information when callpres is prohib Some ITSP make use of the "Anonymous" display name to detect a requirement to withhold caller id across the PSTN. This does not work if the display name is "Unknown". (closes issue #14465) Reported by: Nick_Lewis Patches: chan_sip.c-callerpres.patch uploaded by Nick (license 657) chan_sip.c-callerpres_trunk.patch uploaded by dvossel (license 671) Tested by: Nick_Lewis, dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-15Session timer were not activated if Supported header field in INVITE had ↵David Vossel
both "timer" and other options. (closes issue #15403) Reported by: makoto Patches: sip-session-timer.patch uploaded by makoto (license 38) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206768 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-15The dialing flag was mistakingly removed from sig_pri.Jeff Peeler
This readds the proper setting of the flag and is really a continuation of r205731. The flag was being set properly in sig_analog, but use of the newly added set_dialing callback allowed for some simplification in chan_dahdi. (closes issue #15486) Reported by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206767 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-15Merged revisions 206706 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r206706 | rmudgett | 2009-07-15 15:44:55 -0500 (Wed, 15 Jul 2009) | 26 lines Merged revision 206700 from https://origsvn.digium.com/svn/asterisk/be/branches/C.2-... .......... Fixed chan_misdn crash because mISDNuser library is not thread safe. With Asterisk the mISDNuser library is driven by two threads concurrently: 1. channels/misdn/isdn_lib.c::manager_event_handler() 2. channels/misdn/isdn_lib.c::misdn_lib_isdn_event_catcher() Calls into the library are done concurrently and recursively from isdn_lib.c. Both threads can fiddle with the master/child layer3_proc_t lists. One thread may traverse the list when the other interrupts it and then removes the list element which the first thread was currently handling. This is exactly what caused the crash. About 60 calls were needed to a Gigaset CX475 before it occurred once. This patch adds locking when calling into the mISDNuser library. This also fixes some cb_log calls with wrong port parameter. JIRA ABE-1913 Patches: misdn-locking.patch (Modified with mostly cosmetic changes) .......... ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206707 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-15callerid(num) is wrong when username is missing David Vossel
A domain only sip uri <sip:123.123.123.123> would return 123.123.123.123 as callid num. Now, if the username is missing from a uri, the callerid num field is left empty. (closes issue #15476) Reported by: viraptor git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206702 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-14Restore some missing functionality to sig_analog.Jeff Peeler
The main purpose of this commit is to restore missing functionality present in the ss_thread before all the sig related work was done. Two of the biggest missing things were distinctive ring detection and cid handling for V23. fxsoffhookstate and associated mwi variables have been moved inside sig_analog as they were not being set properly as well. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206566 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-14Merged revisions 206487 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206487 | rmudgett | 2009-07-14 11:44:47 -0500 (Tue, 14 Jul 2009) | 28 lines Fixes several call transfer issues with chan_misdn. * issue #14355 - Crash if attempt to transfer a call to an application. Masquerade the other pair of the four asterisk channels involved in the two calls. The held call already must be a bridged call (not an applicaton) or it would have been rejected. * issue #14692 - Held calls are not automatically cleared after transfer. Allow the core to initate disconnect of held calls to the ISDN port. This also fixes a similar case where the party on hold hangs up before being transferred or taken off hold. * JIRA ABE-1903 - Orphaned held calls left in music-on-hold. Do not simply block passing the hangup event on held calls to asterisk core. * Fixed to allow held calls to be transferred to ringing calls. Previously, held calls could only be transferred to connected calls. * Eliminated unused call states to simplify hangup code. * Eliminated most uses of "holded" because it is not a word. (closes issue #14355) (closes issue #14692) Reported by: sodom Patches: misdn_xfer_v14_r205839.patch uploaded by rmudgett (license 664) Tested by: rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206489 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-14Merged revisions 206385 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r206385 | russell | 2009-07-14 09:48:00 -0500 (Tue, 14 Jul 2009) | 13 lines Merged revisions 206384 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r206384 | russell | 2009-07-14 09:45:47 -0500 (Tue, 14 Jul 2009) | 6 lines Ensure apathetic replies are sent out on the proper socket. chan_iax2 supports multiple address bindings. The send_apathetic_reply() function did not attempt to send its response on the same socket that the incoming message came in on. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206386 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-14Merged revisions 206284 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206284 | rmudgett | 2009-07-13 19:17:28 -0500 (Mon, 13 Jul 2009) | 4 lines Fix some memory leaks in chan_misdn. JIRA ABE-1911 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206341 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-13dns lookup of peername rather than peer's host in transmit_register()David Vossel
(closes issue #15052) Reported by: fsantulli Patches: chan_sip_bug_15052_[20090626204511].patch uploaded by fsantulli (license 818) Tested by: fsantulli git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206280 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-10SIP register not using peer's outbound proxyDavid Vossel
If callbackextension is defined for a peer it successfully causes a registration to occur, but the registration ignores the outboundproxy settings for the peer. This patch allows the peer to be passed to obproxy_get() in transmit_register(). (closes issue #14344) Reported by: Nick_Lewis Patches: callbackextension_peer_trunk.diff uploaded by dvossel (license 671) Tested by: dvossel Review: https://reviewboard.asterisk.org/r/294/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205985 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-10Merged revisions 205877 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r205877 | mmichelson | 2009-07-10 12:39:13 -0500 (Fri, 10 Jul 2009) | 23 lines Merged revisions 205776 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205776 | mmichelson | 2009-07-10 10:56:45 -0500 (Fri, 10 Jul 2009) | 16 lines Merged revisions 205775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul 2009) | 10 lines Ensure that outbound NOTIFY requests are properly routed through stateful proxies. With this change, we make note of Record-Route headers present in any SUBSCRIBE request that we receive so that our outbound NOTIFY requests will have the proper Route headers in them. (closes issue #14725) Reported by: ibc ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205878 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-10Merged revisions 205804 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205804 | dvossel | 2009-07-10 11:23:59 -0500 (Fri, 10 Jul 2009) | 31 lines SIP registration auth loop caused by stale nonce If an endpoint sends two registration requests in a very short period of time with the same nonce, both receive 401 responses from Asterisk, each with a different nonce (the second 401 containing the current nonce and the first one being stale). If the endpoint responds to the first 401, it does not match the current nonce so Asterisk sends a third 401 with a newly generated nonce (which updates the current nonce)... Now if the endpoint responds to the second 401, it does not match the current nonce either and Asterisk sends a fourth 401 with a newly generated nonce... This loop goes on and on. There appears to be a simple fix for this. If the nonce from the request does not match our nonce, but is a good response to a previous nonce, instead of sending a 401 with a newly generated nonce, use the current one instead. This breaks the loop as the nonce is not updated until a response is received. Additional logic has been added to make sure no nonce can be responded to twice though. (closes issue #15102) Reported by: Jamuel Patches: patch-bug_0015102 uploaded by Jamuel (license 809) nonce_sip.diff uploaded by dvossel (license 671) Tested by: Jamuel Review: https://reviewboard.asterisk.org/r/289/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205840 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-10Merged revisions 205775 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul 2009) | 10 lines Ensure that outbound NOTIFY requests are properly routed through stateful proxies. With this change, we make note of Record-Route headers present in any SUBSCRIBE request that we receive so that our outbound NOTIFY requests will have the proper Route headers in them. (closes issue #14725) Reported by: ibc ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205776 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-09Repair ability of SendFAX/ReceiveFAX to respond to T.38 switchover.Kevin P. Fleming
Recent changes in T.38 negotiation in Asterisk caused these applications to not respond when the other endpoint initiated a switchover to T.38; this resulted in the T.38 switchover failing, and the FAX attempt to be made using an audio connection, instead of T.38 (which would usually cause the FAX to fail completely). This patch corrects this problem, and the applications will now correctly respond to the T.38 switchover request. In addition, the response will include the appopriate T.38 session parameters based on what the other end offered and what our end is capable of. (closes issue #14849) Reported by: afosorio git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205696 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-08Merged revisions 205471 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205471 | dvossel | 2009-07-08 18:15:54 -0500 (Wed, 08 Jul 2009) | 10 lines Fixes 8khz assumptions Many calculations assume 8khz is the codec rate. This is not always the case. This patch only addresses chan_iax.c and res_rtp_asterisk.c, but I am sure there are other areas that make this assumption as well. Review: https://reviewboard.asterisk.org/r/306/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205479 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-07Permit setting custom headers from the peer definition.Tilghman Lesher
(closes issue #14059) Reported by: fnordian git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205086 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-07Fix a deadlock in sig_analogMatthew Nicholson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205047 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-06Add CEL transfer events to analog (chan_dahdi) transfers.Matthew Nicholson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205014 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-03Add a configure check for Reverse Charging Indication support in LibPRI.Sean Bright
Also go back and wrap all of the places that use the specific reverse charge APIs with preprocessor conditionals. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204919 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-02Merged revisions 204834 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204834 | rmudgett | 2009-07-02 16:59:43 -0500 (Thu, 02 Jul 2009) | 10 lines Removed confusing warning message "Got Busy in Connected State" If an incoming mISDN call is answered with the Answer application and a subsequent Dial gets a busy endpoint then it is valid for that already connected channel to get the busy indication. Asterisk will play the busy tones until the dialplan plays something else or hangs up the call. (closes issue #11974) Reported by: fvdb ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204835 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-02Support setting and receiving Reverse Charging Indication over ISDN PRI.Sean Bright
This is a continuation of revision 885 to LibPRI (Capture and expose the Reverse Charging Indication IE on ISDN PRI) which added the ability to get/set Reverse Charging Indication in LibPRI. This patch adds the ability to specify RCI on the outbound leg of a PRI call from within Asterisk, by prefixing the dialed number with a capital 'C' like: ...,Dial(DAHDI/g1/C4445556666) And to read it off an inbound channel: exten => s,1,Set(RCI=${CHANNEL(reversecharge)}) Thanks again to rmudgett for the thorough review. (closes issue #13760) Reported by: mrgabu Review: https://reviewboard.asterisk.org/r/303/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204749 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-30Move the masquerade in local_attended_transfer to a point where we hold the ↵Mark Michelson
channel lock. Masquerading without the channel's lock held is a *horrible* idea. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204532 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-30Remove some bogus deadlock avoidance code from local_attended_transfer.Mark Michelson
First of all, the code was unnecessary. The goal was to lock a channel which was already locked. Second, the assumption of the deadlock avoidance loop was that the sip_pvt was already locked and we were trying to get the channel lock. The problem is that the sip_pvt was unlocked a few lines above. Basically, I'm removing 5 lines of no-op. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204530 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-29Merged revisions 204300 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204300 | mmichelson | 2009-06-29 17:45:34 -0500 (Mon, 29 Jun 2009) | 9 lines Add error message so that it is clear why a SIP peer was not processed when a DNS lookup fails on a host or outboundproxy. (closes issue #13432) Reported by: p_lindheimer Patches: outboundproxy.patch uploaded by p (license 558) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204301 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-29Merged revisions 204243,204246 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204243 | mmichelson | 2009-06-29 16:23:43 -0500 (Mon, 29 Jun 2009) | 22 lines Fix a problem where chan_sip would ignore "old" but valid responses. chan_sip has had a problem for quite a long time that would manifest when Asterisk would send multiple SIP responses on the same dialog before receiving a response. The problem occurred because chan_sip only kept track of the highest outgoing sequence number used on the dialog. If Asterisk sent two requests out, and a response arrived for the first request sent, then Asterisk would ignore the response. The result was that Asterisk would continue retransmitting the requests and ignoring the responses until the maximum number of retransmissions had been reached. The fix here is to rearrange the code a bit so that instead of simply comparing the sequence number of the response to our latest outgoing sequence number, we walk our list of outstanding packets and determine if there is a match. If there is, we continue. If not, then we ignore the response. In doing this, I found a few completely useless variables that I have now removed. (closes issue #11231) Reported by: flefoll Review: https://reviewboard.asterisk.org/r/298 ........ r204246 | mmichelson | 2009-06-29 16:37:05 -0500 (Mon, 29 Jun 2009) | 3 lines Fix build oops. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204247 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-27Merged revisions 203908 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203908 | rmudgett | 2009-06-26 19:55:12 -0500 (Fri, 26 Jun 2009) | 16 lines The ISDN CPE side should not exclusively pick B channels normally. Before this patch, Asterisk unconditionally picked B channels exclusively on the CPE side and normally allowed alternative B channels on the network side. Now Asterisk does the opposite. Reasons for the CPE side to normally not pick B channels exclusively: * For CPE point-to-multipoint mode (i.e. phone side), the CPE side does not have enough information to exclusively pick B channels. (There may be other devices on the line.) * Q.931 gives preference to the network side picking B channels. * Some telcos require the CPE side to not pick B channels exclusively. (closes issue #14383) Reported by: mbrancaleoni ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203909 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-26Merged revisions 203848 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203848 | jpeeler | 2009-06-26 17:09:19 -0500 (Fri, 26 Jun 2009) | 5 lines Make sure to recreate the dahdi pseudo channel after dahdi restart (closes issue #14477) Reported by: timking ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203853 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-26Ensure the TCP read buffer is fully initialized before handling each packet.Russell Bryant
(closes issue #14452) Reported by: umberto71 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203779 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-26Fix the 'nat' option to actually do RFC3581 as expected and extend the ↵Joshua Colp
configurable values for finer control. (closes issue #8855) Reported by: mikma Tested by: klaus3000, file git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203735 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-26moving debug message from level 0 to 1.David Vossel
(closes issue #15404) Reported by: leobrown Patches: iax_codec_debug.patch uploaded by leobrown (license 541) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203710 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-26Improve T.38 negotiation by exchanging session parameters between ↵Joshua Colp
application and channel. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203699 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-26Check if polarityonanswerdelay has elapsed before setting a channel as answeredJeff Peeler
after a polarity reversal. Previously on a polarity switch event chan_dahdi would set the channel immediately as answered. This would cause problems if a polarity reversal occurred when the line was picked up as the dial would not have yet occurred. Now if the polarity reversal occurs before delay has elapsed after coming off hook or an answer, it is ignored. Also, some refactoring was done in _handle_event. (closes issue #13917) Reported by: alecdavis Patches: chan_dahdi.bug13917.feb09.diff2.txt uploaded by alecdavis (license 585) Tested by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203672 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-26Merge the new Channel Event Logging (CEL) subsystem.Russell Bryant
CEL is the new system for logging channel events. This was inspired after facing many problems trying to represent what is possible to happen to a call in Asterisk using CDR records. For more information on CEL, see the built in HTML or PDF documentation generated from the files in doc/tex/. Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard work developing this code. Also, thanks to Matt Nicholson (mnicholson) and Sean Bright (seanbright) for their assistance in the final push to get this code ready for Asterisk trunk. Review: https://reviewboard.asterisk.org/r/239/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203638 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-25make sure chan_dahdi compiles with only libss7 and not libpri installedJeff Peeler
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203479 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-25Picking nitsRichard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203443 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-25Remove some unnecessary code and update sample config file with respect to ↵Jeff Peeler
GR-303. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203402 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-25New signaling module to handle PRI/BRI operations in chan_dahdiJeff Peeler
This merge splits the PRI/BRI signaling logic out of chan_dahdi.c into sig_pri.c. Functionality in theory should not change (mostly). A few trivial changes were made in sig_analog with verbose messages and commenting. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203304 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-25Unmute when we get a dtmfup (we muted on dtmfdown) event.Jason Parker
This would occasionally cause one-way audio when using hardware DTMF detection. (closes issue #14761) Reported by: tzafrir Patches: v1-14761.patch uploaded by dimas (license 88) Tested by: tzafrir, dimas git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203258 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-25Add support for multicast RTP paging.Joshua Colp
(closes issue #11797) Reported by: macbrody Review: https://reviewboard.asterisk.org/r/270/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-25Insure ring cadence is set for fxs portsDoug Bailey
Moved SETCADENCE ioctl call to before call into new analog signal module to insure that it gets set. (closes issue #15381) Reported by: alecdavis Patches: fix15381.diff uploaded by dbailey (license 819) Tested by: dbailey git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203126 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-25Merged revisions 203115 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203115 | russell | 2009-06-25 11:02:16 -0500 (Thu, 25 Jun 2009) | 11 lines Resolve a crash related to a T.38 reinvite race condition. This change resolves a crash observed locally during some T.38 testing. A call was set up using a call file, and when the T.38 reinvite came in, the channel state was still AST_STATE_DOWN. The reason is explained by a comment in the code that previously lived in the handling of AST_STATE_RINGING. This change modifies the logic to handle the same race condition for any channel state that is not UP. (closes ABE-1895) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203116 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-24Merged revisions 203036 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203036 | rmudgett | 2009-06-24 16:01:43 -0500 (Wed, 24 Jun 2009) | 8 lines Improved chan_dahdi.conf pritimer error checking. Valid format is: pritimer=timer_name,timer_value * Fixed segfault if the ',' is missing. * Completely check the range returned by pri_timer2idx() to prevent possible access outside array bounds. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203037 65c4cc65-6c06-0410-ace0-fbb531ad65f3
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-04 09:22:24 +0000