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2007-06-05briding is a bool, fixed copy and paste issue.crichter1-3/+3
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@67307 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-05simplified the EVENT_SETUP handling in the cb_events function a lot. ↵crichter1-175/+155
Commented the different possibilities a bit and made functions of shared code. When the dialed extension does not exist in the extensions.conf we'll jump into the 'i' extension if this does exist, else we disconnect the call with the cause:1 = No Route to Destination. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@67306 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-05Backport of the overlap_dial functionality from asterisk-1.4's chan_misdn.nadi3-6/+197
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@67239 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-05added possibility to deactivate bridging per portcrichter3-0/+13
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@67209 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-31It is now possible for this path of execution to have the frame pointer be ↵file1-1/+1
NULL, therefore we need to check for it before trying to access it. (issue #9836 reported by barthpbx) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@66764 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-29Issue #9802 - Change inuse counter on CANCELoej1-0/+4
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@66349 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-24Allow RFC2833 to be negotiated when an INVITE comes in without SDP and is ↵file1-0/+1
not matched to a user or peer. (issue #9546 reported by mcrawford) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@65837 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-24we should only activate the generator in chan_misdn, when asterisk hask not ↵crichter1-0/+2
yet taken the call (WAITING4DIGS state). Alerting audio will be generated fomr asterisk for example. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@65767 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-23ensure that variables are set on a newly created channel before we start a ↵kpfleming1-4/+6
PBX on it git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@65682 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-23if we are going to set variables on a newly created channel, it should be ↵kpfleming1-3/+5
done *before* we start the PBX on it git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@65676 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-23Revert revision 62417 as someone reported problems with it to Mark. This wasrussell1-16/+31
related to issue #9588. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@65588 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-22we stop the tones only when we're in the pre-call phase, otherwise e.g. when ↵crichter1-1/+2
in CONNECTED state we should not stop tones when we receive an Information Message git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@65328 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-18Not getting an ACK to a 200 OK in the initial invite is critical to the call.oej1-1/+4
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@65122 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-18Issue 9235 - part of the problem, maybe not all. Please retry with this ↵oej1-2/+7
patch (and no other patch) if you have problems with hanging SIP channels. Thank you. A special Thank You to WeBRainstorm that gave me access to his system. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@65075 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-18fixed a warning regarding Keypad encoding. encode the IE sending_complete at ↵crichter2-4/+4
the right position. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@65007 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-18we *need* to send a PROCEEDING when sending_complete is set, even if ↵crichter1-1/+2
need_more_infos is requested. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@64902 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-16Fixing possible bug in auth of BYEoej1-1/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@64603 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-16Support SIP uri's starting with SIP: and sip: (reported by Tony Mountfield ↵oej1-15/+15
on the mailing list. Thanks!) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@64535 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-16Issue #9726 - rlister - Better logging for ACL denialsoej1-11/+27
While at it, also added better logging and handling of peers that are not supposed to register. My patch, stole the issue report from Russell. My apologies, Russell :-) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@64514 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-16in the case immediate=yes, we directly jump into the dialplan, where people ↵crichter1-10/+12
can use PlayTones to indicate a Dialtone, so we don't need to to that by ourself. also we should not do a dialtone_indicate for incoming calls on a TE port in overlapdialmode. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@64513 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-10Fix an issue with trying to kill a thread before it gets created.qwell1-1/+2
Issue 9709, patch by nic_bellamy. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@63828 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-10Do not allocate SIP pvt's for PEERs we can not reach. oej1-0/+4
This was seen as a lot of dialogs being created then immediately destroyed at reload/restart of the SIP channel. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@63748 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-09Make sure we only create a DSP if it's requested on SUB_REALmattf1-18/+20
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@63653 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-09Properly handle hints that point to multiple devices in chan_sip. Why ↵file1-5/+8
chan_sip is even doing this I have no idea but I would rather not go into a rant. (issue #9536 reported by rlister) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@63610 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-09release_chan frees ch, so we should never touch ch after release_chan, this ↵crichter1-1/+3
may cause segfaults. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@63519 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-08added application misdn_check_l2l1 which tries to pull up the L1/L2 on all ↵crichter2-12/+127
ports that have the layers down in a group. It waits then for a timeout. This helps for scenarios where multiple PMP BRIs are grouped together, or where a provider has a faulty PTP Implementation, that looses the L2 after a while. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@63402 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-03When a peer is seeded or built tell the devicestate core to update it's ↵file1-0/+2
status. This is easier then having chan_sip load before pbx_config. (issue #9658 reported by dlynes) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@62987 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-03when we're in state WAITING4DIGS, we use the asterisk tone-generator which ↵crichter2-6/+23
prods us, so we can't just return -1 in misdn_write in this case. Added a MISDN_KEYPAD channel variable, and fixed the sending of keypad. this enables us to modify the call forward parameters in the switch. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@62945 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-03fixed the problem that misdn_write did not return -1 when called with 0 ↵crichter2-1/+5
samples in a frame this resultet in a deadlock in some circumstances, when the call ended because of a busy extension. added encoding of keypad. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@62885 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-02Issue 9638 - if a text frame is sent with no terminating NULL through a bridgedtilghman1-0/+7
IAX connection, the remote end will receive garbage characters tacked onto the end. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@62691 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-30This patch fixes an issue where depending on the cause code, when the networkrussell1-31/+16
sends a PRI disconnect, the call may not be properly hung up. (issue #9588, reported and patched by softins) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@62417 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-27Issue #7351 - SIP Cancel fails due to the wrong contact uri. Reported by ↵oej1-0/+2
PPYY, failed to fix by OEJ final fix by wojtekka - THANKS!!!! THis was a hard one to catch. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@62126 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-26Revert previous fix for when the IAX2 channel goes funky (that's the ↵file1-2/+6
technical term). This is causing legit calls to be prematurely hung up. (issue #9600 reported by justdave) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@62037 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-25handle a very bizarre race condition with channels being redirected before a ↵kpfleming1-0/+10
simple switch can be started on them (issue #9286) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@61913 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-25If the callerid= option is specified, but empty, clear any previous data.russell1-0/+8
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@61866 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-25Ensure that callerid settings are reset on a reload.russell1-6/+16
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@61862 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-25Fix a typo where cid_num got copied instead of cid_ani. russell1-1/+1
(issue #9587, reported and patched by xrg) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@61798 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-24removed #if 0 block from chan_phone, chan_zap, and chan_modem restart_monitor()dhubbard3-13/+3
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@61777 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-24removed pthread_join in restart_monitor() to make it like chan_zap and other ↵dhubbard1-0/+2
channel drivers git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@61776 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-24Allow RFC2833 to be sent in the response SDP when an INVITE comes in without ↵file1-0/+2
SDP. (issue #9546 reported by mcrawford) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@61771 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-24added lock for sending messages to avoid double sending. shuffled some ↵crichter3-39/+66
empty_chans after the cb_event calls, this avoids that a release_complete from a quite different call releases a fresh created setup by accident. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@61770 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-19Send NOTIFY to Contact: in SUBSCRIBE oej1-0/+4
- as reported by Intertex and Citel. Fixed during SIPit 20 in Antwerp. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@61685 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-16Don't stop RTP on errors on INFO messages.oej1-8/+11
Disclaimer: This patch was needed for Edvina AstHoloApp and was meant to be included in 1.2, but never made it in time so I felt I could add it now. No, just joking, patching error found while testing T.140 with Omnitor earlier this spring. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@61663 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-11If someone sets the "useragent" option in sip.conf to be empty, then don't addrussell1-4/+8
the User-Agent header at all. It is an optional header, anyway. Also, the bug report says that some of Japan's SIP providers don't allow it for some weird reason. (issue #9488, reported by makoto, fixed by me) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@61476 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-11Fix a bug with switching between host=dynamic and using specific hosts forrussell1-5/+5
peers. The code would only reset the peer's address when it is dynamic if it was a new peer structure. Now, it will also reset the address if it was already in the peer list, but before the reload, it was not dynamic. (issue #9515, reported by caio1982, fixed by me) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@61426 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-11Remove the attempt at reporting configuration errors in sip.conf. This canrussell1-7/+4
cause a bunch of improper messages when using realtime. I give up. As oej tried to convince me when I put this in, there is just no easy way to do it. (inspired by a message on the -dev list) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@61376 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-11some fixes for PMP Hold/Retrieve, it should work now, when briding=nocrichter2-7/+13
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@61357 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-10msns config parameter defaults to '*'nadi1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@61170 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-09- Don't send ActionID before Response: header. oej1-6/+7
- Don't use a blank in an AMI header git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@61038 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-09Bug 9505 - If the return value for local_queue_frame is set, then p->lock is ↵tilghman1-11/+14
no longer valid. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@60846 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-24 04:11:41 +0000