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2007-07-13Small improvement to the STUN support so it can be used byrizzo1-25/+109
sockets other than RTP ones. The main change is a new API function in main/rtp.c (see there for a description) int ast_stun_request(int s, struct sockaddr_in *dst, const char *username, struct sockaddr_in *answer) which can be used to send an STUN request on a socket, and optionally wait for a reply and store the STUN_MAPPED_ADDRESS into the 'answer' argument (obviously, the version that waits for a reply is blocking, but this is no different from DNS resolutions). Internally there are minor modifications to let stun_handle_packet() be somewhat configurable on how to parse the body of responses. At the moment i am not committing any change to the clients, but adding STUN client support is extremely simple, e.g. chan_sip.c could do something like this: + add a variable to store the stun server address; static struct sockaddr_in stunaddr = { 0, }; /*!< stun server address */ + add code to parse a config file of the form "stunaddr=my.stun.server.org:3478" (not shown for brevity); + right after binding the main sip socket, talk to the stun server to determine the externally visible address if (stunaddr.sin_addr.s_addr != 0) ast_stun_request(sipsock, &stunaddr, NULL, &externip); so now 'externip' is set with the externally visible address. so it is really trivial. Similarly ast_stun_request could be called when creating the RTP socket (possibly adding a struct sockaddr_in field in the struct ast_rtp to store the externalip). git-svn-id: http://svn.digium.com/svn/asterisk/trunk@75034 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-12more cleanup, this time to stun_handle_packet(). Among other things:rizzo1-24/+44
+ mark a potentially dangerous write-past-end-of-buffer + localize some variables in the block generating stun replies. As before, not ready yet for a merge to 1.4 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@74850 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-12a little bit of code cleanup to rtp.c, mostly to function rizzo1-54/+64
ast_rtp_new_with_bindaddr(): 1. add comments to the logic of the main loop; 2. use a common exit point on failure so the cleanup is done only in one place; 3. handle failures in rtp_socket() in the main loop of the function; No functional changes except for #3 above, so it is not yet worthwhile merging this and other changes to 1.4 Once the cleanup work on this file will be complete (which among other things should include some extensions to the stun support) it might be a good thing to push all the changes to 1.4 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@74813 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-11add a bit of documentation on what the stun code in rtp.c doesrizzo1-1/+60
(which is very little, at the moment). Eventually, when the functionality is extended, the changes can be merged back to 1.4. At the moment this is pointless. Note, this change is whitespace only. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@74571 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-27Merged revisions 72112 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r72112 | russell | 2007-06-27 11:34:24 -0500 (Wed, 27 Jun 2007) | 3 lines Only output debug information related to RTCP timestamps when RTCP debug is turned on (issue #10066, patch by me) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@72113 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-26Merged revisions 71915 via svnmerge from qwell1-0/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r71915 | qwell | 2007-06-26 15:36:09 -0500 (Tue, 26 Jun 2007) | 4 lines Don't dereference a pointer that may be NULL here. Issue 10017. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@71916 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-25Convert so more logging to ast_debug (issue #10045, dimas)russell1-13/+12
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@71557 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-24Conversions to ast_debug()russell1-82/+47
(issue #9984, patches from eliel and dimas) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@71338 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-22Behold the magic of casting!file1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@71146 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-22Merged revisions 71063 via svnmerge from murf1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r71063 | murf | 2007-06-22 08:10:24 -0600 (Fri, 22 Jun 2007) | 1 line My conditions for merging amaflags info was naive; DOCUMENTATION is the default, although null is possible; theft of user-settable fields is not good. Just copy them, leave them alone. This is for bug 10016. (plus a small fix to rtp, to elim a compiler warning (dev mode)) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@71093 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-21Add manager events for RTCP statistics.qwell1-0/+98
Also adds a new "reporting" permission for manager, since it can be incredibly spammy. This permission was discussed on the -dev mailing list some months back. Issue 8613, patch by johann8384, with some minor changes by me. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@70961 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-21Merged revisions 70727 via svnmerge from file1-0/+13
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r70727 | file | 2007-06-21 11:22:39 -0400 (Thu, 21 Jun 2007) | 2 lines Do not Packet2Packet bridge if packetization settings do not allow it. (issue #9117 reported by phsultan) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@70729 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-20Merged revisions 70360 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r70360 | file | 2007-06-20 13:52:57 -0400 (Wed, 20 Jun 2007) | 2 lines Put the speex packetization values back in but disable it when setting up the smoother. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@70361 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-19Merged revisions 70003 via svnmerge from file1-12/+17
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r70003 | file | 2007-06-19 13:07:40 -0400 (Tue, 19 Jun 2007) | 10 lines Merged revisions 69992 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r69992 | file | 2007-06-19 13:00:58 -0400 (Tue, 19 Jun 2007) | 2 lines Handle the CC field in the RTP header. (issue #9384 reported by DoodleHu) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@70006 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-12Merged revisions 68922 via svnmerge from file1-0/+5
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r68922 | file | 2007-06-12 10:23:11 -0400 (Tue, 12 Jun 2007) | 10 lines Merged revisions 68921 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r68921 | file | 2007-06-12 10:18:57 -0400 (Tue, 12 Jun 2007) | 2 lines Bring RTP back to Asterisk at the end of a native bridge no matter what. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@68923 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-07Fix a bunch of doxygen errors and document more thingsrussell1-0/+1
(issue #9842, snuffy) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@68339 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-06Issue 9869 - replace malloc and memset with ast_calloc, and other coding ↵tilghman1-8/+8
guidelines changes git-svn-id: http://svn.digium.com/svn/asterisk/trunk@67864 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-06Merged revisions 67650 via svnmerge from file1-1/+6
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r67650 | file | 2007-06-06 09:30:25 -0400 (Wed, 06 Jun 2007) | 10 lines Merged revisions 67649 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r67649 | file | 2007-06-06 09:28:34 -0400 (Wed, 06 Jun 2007) | 2 lines Reinvite the RTP back to the Asterisk machine when the timeout happens. (issue #9888 reported by gasparz) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@67651 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-04Merged revisions 67071 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r67071 | russell | 2007-06-04 16:47:36 -0500 (Mon, 04 Jun 2007) | 2 lines Add a missing \n. (pointed out by jcmoore on IRC) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@67072 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-29Merged revisions 66437 via svnmerge from file1-2/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r66437 | file | 2007-05-29 12:44:34 -0400 (Tue, 29 May 2007) | 2 lines Handle cases where a frame may have no data. (issue #9519 reported by dmb) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@66438 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-24Make this build on *my* machine again, and hopefully not break others.russell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@65964 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-24Merged revisions 65863 via svnmerge from file1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r65863 | file | 2007-05-24 11:08:17 -0400 (Thu, 24 May 2007) | 2 lines I like it when the RTP stack compiles myself... ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@65869 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-24Merged revisions 65842 via svnmerge from russell1-19/+37
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r65842 | russell | 2007-05-24 09:49:05 -0500 (Thu, 24 May 2007) | 5 lines Fix the calculation of the RTT for RTCP. The previous code would result in oscillating and incorrect data. Additionally, the RTT would sometimes report negative values due to incorrect calculations. (issue #9601, patch from davetroy) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@65843 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-30Add support for setting the CoS for VLAN traffic (802.1p) in Linux. Therussell1-6/+3
file doc/qos.tex has been updated to document the new functionality. (issue #9540, patch submitted by IgorG) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62457 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-20Merged revisions 61707 via svnmerge from qwell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61707 | qwell | 2007-04-20 16:35:27 -0500 (Fri, 20 Apr 2007) | 8 lines Avoid invalid seqno cycling detection. Per comment from Dave Troy: This adds back in some simple typecasting I had in an earlier version which I realize now may be breaking things. Issue #9554. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@61708 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-20Merged revisions 61697 via svnmerge from russell1-3/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61697 | russell | 2007-04-20 15:42:02 -0500 (Fri, 20 Apr 2007) | 2 lines Remove a stray debug message introduced by a recent commit. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@61698 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-18Merged revisions 61676 via svnmerge from oej1-18/+18
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61676 | oej | 2007-04-18 22:46:23 +0200 (Wed, 18 Apr 2007) | 2 lines Clean upp formatting, add some doxygen stuff while we're in cleaning mode... Thanks Kevin! ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@61677 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-18Merged revisions 61674 via svnmerge from oej1-7/+5
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61674 | oej | 2007-04-18 22:28:53 +0200 (Wed, 18 Apr 2007) | 2 lines Issue #9554 - Improve RTCP (Dave Troy) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@61675 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-29Merged revisions 59358 via svnmerge from russell1-4/+8
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r59358 | russell | 2007-03-29 12:17:41 -0500 (Thu, 29 Mar 2007) | 13 lines Merged revisions 59357 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59357 | russell | 2007-03-29 12:14:33 -0500 (Thu, 29 Mar 2007) | 5 lines If an error occurs when reading from an RTP socket, and the error code does not indicate that we should try again, then return NULL instead of a "null frame". This will prevent Asterisk from trying over and over again, and eventually causing the system to crash. (issue #8285, john) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@59359 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-26Merged revisions 59207 via svnmerge from russell1-2/+14
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59207 | russell | 2007-03-26 12:45:55 -0500 (Mon, 26 Mar 2007) | 7 lines The AUDIORTPQOS and VIDEORTPQOS variables are not fully functional in some because they get set in sip_hangup. So, there are common situations where the variables will not be available in the dialplan at all. So, this patch provides an alternate method for getting to this information by introducing AUDIORTPQOS and VIDEORTPQOS dialplan functions. (issue #9370, patch by Corydon76, with some testing by blitzrage) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@59208 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-12Merged revisions 58783 via svnmerge from file1-20/+22
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58783 | file | 2007-03-11 21:21:12 -0400 (Sun, 11 Mar 2007) | 2 lines Allow RFC2833 compensation to compensate for even stupider implementations by queueing up the end frame at the start, not the actual end. (issue #8963 reported by AndrewZ) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@58784 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-08Merged revisions 58436 via svnmerge from file1-6/+12
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58436 | file | 2007-03-08 13:01:00 -0500 (Thu, 08 Mar 2007) | 2 lines Make early SDP seeding even smarter! We have to check codecs in the make_compatible function too. (issue #9221 reported by marcelbarbulescu) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@58437 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-07Merged revisions 58240 via svnmerge from file1-1/+12
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58240 | file | 2007-03-07 12:52:58 -0500 (Wed, 07 Mar 2007) | 2 lines Ensure we have (or should have) at least one matching codec before attempting early bridge SDP seeding. (issue #9221 reported by marcelbarbulescu) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@58241 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-05Merged revisions 57768 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r57768 | file | 2007-03-04 22:22:17 -0500 (Sun, 04 Mar 2007) | 2 lines Preserve marker bit when P2P bridging. (issue #9198 reported by edgreenberg) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@57769 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-24Doxygen additions, correctionsoej1-4/+6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@56665 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-16Adding Realtime Text support (T.140) to Asteriskoej1-32/+86
T.140/RFC 2793 is a live communication channel, originally created for IP based text phones for hearing impaired. Feels very much like the old Unix talk application. This code is developed and disclaimed by John Martin of Aupix, UK. Tested for interoperability by myself and Omnitor in Sweden, the company that wrote most of the specifications. A big thank you to everyone involved in this. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@54838 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-07Merged revisions 53434 via svnmerge from file1-0/+8
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53434 | file | 2007-02-07 12:53:03 -0500 (Wed, 07 Feb 2007) | 2 lines We can not reliably do P2P bridging with DTMF passing back with compensation if we need to listen for DTMF frames. (issue #8962 reported by caio1982) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@53435 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-07Merged revisions 53429 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53429 | russell | 2007-02-07 11:39:31 -0600 (Wed, 07 Feb 2007) | 7 lines When parsing the NTP timestamp in a sender report message, you are supposed to take the low 16 bits of the integer part, and the high 16 bits of the fractional part. However, the code here was erroneously taking the low 16 bits of the fractional part. It then shifted the result 16 bits down, so the result was always zero. This fix makes it grab the appropriate high 16 bits, instead. (issue #8991, pointed out by andre_abrantes) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@53431 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-02Merged revisions 53120 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53120 | file | 2007-02-02 11:15:22 -0600 (Fri, 02 Feb 2007) | 2 lines Correct a copy/pasted error message line for RTCP. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@53122 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-01Merged revisions 53052 via svnmerge from file1-1/+14
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53052 | file | 2007-01-31 18:24:20 -0600 (Wed, 31 Jan 2007) | 2 lines When going on hold have the side that was put on hold reinvite back to Asterisk. When going off hold have the side that was taken off hold reinvited back to the other party. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@53053 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-01Merged revisions 53050 via svnmerge from file1-5/+15
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53050 | file | 2007-01-31 18:19:48 -0600 (Wed, 31 Jan 2007) | 2 lines Add more frame types to forward in the RTP bridge loops. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@53051 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-31Merged revisions 53040 via svnmerge from russell1-9/+9
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r53040 | russell | 2007-01-31 11:45:05 -0600 (Wed, 31 Jan 2007) | 11 lines Merged revisions 53039 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r53039 | russell | 2007-01-31 11:41:51 -0600 (Wed, 31 Jan 2007) | 3 lines Use the proper format string to print unsigned values in the rtp debug output. (issue #8954, wmis) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@53041 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-29Merged revisions 52645 via svnmerge from russell1-1/+9
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r52645 | russell | 2007-01-29 15:26:27 -0600 (Mon, 29 Jan 2007) | 6 lines Fix a problem with packet-to-packet bridging and DTMF mode translation. P2P bridging can only be used when the DTMF modes don't match if the core is monitoring DTMF in both directions. Then, the core will handle the translation. Otherwise, this bridging method can not be used. (issue #8936) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@52646 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-29Only use locking for bridge information if intense P2P bridging is enabled.file1-28/+48
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@52635 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-22Change RTP protos list to be read/write. Most of the time it's only going to ↵file1-12/+12
be read so making it use mutex locks was a waste. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51467 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-22Make the RTP stack better conform to coding guidelines. (issue #8679 ↵file1-32/+32
reported by johann8384) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51463 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-19Merged revisions 51311 via svnmerge from russell1-15/+31
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines Merge the changes from the /team/group/vldtmf_fixup branch. The main bug being addressed here is a problem introduced when two SIP channels using SIP INFO dtmf have their media directly bridged. So, when a DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk would try to emulate a digit of some length by first sending a DTMF BEGIN frame and sending a DTMF END later timed off of incoming audio. However, since there was no audio coming in, the DTMF_END was never generated. This caused DTMF based features to no longer work. To fix this, the core now knows when a channel doesn't care about DTMF BEGIN frames (such as a SIP channel sending INFO dtmf). If this is the case, then Asterisk will not emulate a digit of some length, and will instead just pass through the single DTMF END event. Channel drivers also now get passed the length of the digit to their digit_end callback. This improves SIP INFO support even further by enabling us to put the real digit duration in the INFO message instead of a hard coded 250ms. Also, for an incoming INFO message, the duration is read from the frame and passed into the core instead of just getting ignored. (issue #8597, maybe others...) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51314 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-19in the interest of portability, avoid using %zd when allrizzo1-4/+4
we need is to print is an integer that fits in 16 bits. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51310 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-18Merged revisions 51211 via svnmerge from file1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51211 | file | 2007-01-17 19:18:44 -0500 (Wed, 17 Jan 2007) | 2 lines Pass data as well for hold/unhold/vidupdate frames. (issue #8840 reported by mdu113) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51212 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-17Merged revisions 51182 via svnmerge from file1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51182 | file | 2007-01-17 01:36:41 -0500 (Wed, 17 Jan 2007) | 2 lines Return the correct result when directly writing out a packet so that the core doesn't then decide to handle it the regular way again. (issue #8833 reported by rcourtna) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51183 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-05 08:19:09 +0000