asterisk

Commit Graph

Author	SHA1	Message	Date
David Brooks	a2933fefef	Fix incorrect parsing in chan_gtalk when xmpp contains extra whitespaces To drill into the xmpp to find the capabilities between channels, chan_gtalk calls iks_child() and iks_next(). iks_child() and iks_next() are functions in the iksemel xml parsing library that traverse xml nodes. The bug here is that both iks_child() and iks_next() will return the next iks_struct node regardless of type. chan_gtalk expects the next node to be of type IKS_TAG, which in most cases, it is, but in this case (a call being made from the Empathy IM client), there exists iks_struct nodes which are not IKS_TAG data (they are extraneous whitespaces), and chan_gtalk doesn't handle that case, so capabilities don't match, and a call cannot be made. iks_first_tag() and iks_next_tag(), on the other hand, will not return the very next iks_struct, but will check to see if the next iks_struct is of type IKS_TAG. If it isn't, it will be skipped, and the next struct of type IKS_TAG it finds will be returned. This assures that chan_gtalk will find the iks_struct it is looking for. This fix simply changes all calls to iks_child() and iks_next() to become calls to iks_first_tag() and iks_next_tag(), which resolves the capability matching. The following is a payload listing from Empathy, which, due to the extraneous whitespace, will not be parsed correctly by iksemel: <iq from='dbrooksjab@235-22-24-10/Telepathy' to='astjab@235-22-24-10/asterisk' type='set' id='542757715704'> <session xmlns='http://www.google.com/session' initiator='dbrooksjab@235-22-24-10/Telepathy' type='initiate' id='1837267342'> <description xmlns='http://www.google.com/session/phone'> <payload-type clockrate='16000' name='speex' id='96'/> <payload-type clockrate='8000' name='PCMA' id='8'/> <payload-type clockrate='8000' name='PCMU' id='0'/> <payload-type clockrate='90000' name='MPA' id='97'/> <payload-type clockrate='16000' name='SIREN' id='98'/> <payload-type clockrate='8000' name='telephone-event' id='99'/> </description> </session> </iq> git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@185362 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Philippe Sultan	d045d36561	Set the initiator attribute to lowercase in our replies when receiving calls. This attribute contains a JID that identifies the initiator of the GoogleTalk voice session. The GoogleTalk client discards Asterisk's replies if the initiator attribute contains uppercase characters. (closes issue #13984) Reported by: jcovert Patches: chan_gtalk.2.patch uploaded by jcovert (license 551) Tested by: jcovert git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175029 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Philippe Sultan	ab1b8017e2	Apply fix for issue #13310 to branch 1.4, too. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@139283 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Philippe Sultan	59410a3f01	Do not link the guest account with any configured XMPP client (in jabber.conf). The actual connection is made when a call comes in Asterisk. Fix the ast_aji_get_client function that was not able to retrieve an XMPP client from its JID. (closes issue #12085) Reported by: junky Tested by: phsultan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@119740 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	a4732cfb3c	When modules are embedded, they take on a different name, without the ".so" extension. Specifically check for this name, when we're checking if a module is loaded. (Closes issue #12534) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114708 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Philippe Sultan	fbf0f7107e	Free newly allocated channel before returning git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@112820 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Philippe Sultan	5e5094f89e	Prevent call connections when codecs don't match. (closes issue #10604) Reported by: keepitcool Patches: branch-1.4-10604-2.diff uploaded by phsultan (license 73) Tested by: phsultan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@112766 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jason Parker	ea47c2d0b7	Copy voicemail dependency logic for res_adsi to chan_gtalk (for jabber). (closes issue #12014) Reported by: junky git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@107714 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Philippe Sultan	863ef8d7d2	Set the caller id within the gtalk_alloc function. As underlined in issue #10437 by Josh, we need to prevent a possible memory leak. We only set the name part of the caller id, the number part is not relevant when dealing with JIDs. Closes issue #11549. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@97489 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	40cd95fc27	Don't try to allocate memory that we're just going to re-allocate later anyways. Issue 11130, patch by eliel. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@87906 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Philippe Sultan	ab36dd5843	Fix the output for this channel help CLI command git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@85800 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Philippe Sultan	b558ac3acf	Closes issue #9401 , reported and patched by irrot, with slight modifications by me. Handle DTMF sent by Asterisk properly. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@82309 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Philippe Sultan	8e10babcbd	Various string length fixes. Removed an unused variable in aji_client structure (context) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@81743 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Philippe Sultan	1da5446f1a	Make the 'gtalk show channels' CLI command available. Closes issue 10548, reported by keepitcool. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@81410 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Philippe Sultan	0518339c30	Closes issue #10509 Googletalk calls are answered too early, which results in CDRs wrongly stating that a call was ANSWERED when the calling party cancelled a call before before being established. We must not answer the call upon reception of a 'transport-accept' iq packet, but this packet still needs to be acknowledged, otherwise the remote peer would close the call (like in #8970). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@80661 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	9a35428295	(closes issue #10437 ) Reported by: haklin Don't set the callerid name and number a second time on a newly created channel. ast_channel_alloc itself already sets it and setting it twice would cause a memory leak. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@79174 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	5055793871	Make payload IDs for iLBC/Speex match to our list. Since these are dynamic payloads the other side shouldn't care. (issue #9426 reported by irroot) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@72331 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	7508fce25b	Don't modify a variable that we don't want modified. Make a copy of it instead. Issue 10029, patch by phsultan with slight modifications by me (to remove needless casts). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@72125 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	a0a33670bd	Only attempt to queue a hangup on the owner channel if it actually exists. (issue #9795, patch from zandbelt) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@70084 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	5f9897ba56	handle the GNUTLS library properly in the configure script and build system don't build in OSP support unless we have found and are allowed to use SSL support git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66157 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	0b14e93d00	oops, use #ifdef instead of #if git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65967 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	e49f14e6cb	don't reference GnuTLS headers and functions unless the configure script found it git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65966 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	4eab565b25	Issue 7672 - fix by zandbelt - Asterisk core dump since the GnuTLS interface did not support multithreading correctly. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65901 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	e8addde3f1	Issue 8193 - NAT issues with gtalk/STUN. Patch by phsultan. Thanks! git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65892 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	abc8683ce7	Issue 7686, fix by phsultan, NAT issues when calling from gtalk to SIP over nat. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65857 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	e4f9cc15e6	Issue #8536 - Caller ID not set in CDR for jingle git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65841 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	7d5a79a0b9	This is a big improvement over the current CDR fixes. It may still need refinement, but this won't have as many folks bothered. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@60989 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	b937d7e8bc	Fix locking issue, and accept "transport-accept" as a valid accept message. This should solve issues 8970 and 8503. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55954 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Jason Parker	67421802de	Fix segfault when buddy couldn't be found. Issue 7764, patch by sailer git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55799 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Jason Parker	e0b88aba12	No need to cast nor free with strdupa (thanks file) 55555! git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55555 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Kevin P. Fleming	ff17d402de	another dependency git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53781 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	8f7ddbef0d	Update channel drivers to use module referencing so that unloading them while in use will not result in crashes. (issue #8897 reported by junky) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51788 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	0e37d936da	Fix VLDTMF support in chan_gtalk. AST_FRAME_DTMF and AST_FRAME_DTMF_END are actually the same thing. So, a digit would have been interpreted incorrectly here. Since the channel driver will always have the begin and end callbacks called for a digit, only support the button-down and button-up messages. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51328 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	33235b40d6	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: https://origsvn.digium.com/svn/asterisk/branches/1.4@51311 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	b2b70adede	Do not do a partial bridge for Google Talk since we need to handle STUN. (issue #8448 reported by phsultan) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48168 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	487d4512d3	Make this module build again git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47329 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Steve Murphy	517978fd5f	These mods are to solve the problem in bug 7506. It's a lot of rework to solve a fairly small problem... such is life. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47303 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Matt O'Gorman	6a10570614	bind address support from bug 8164 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46822 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Matt O'Gorman	0c4d8edca6	fix for bug 7764. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44982 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Matt O'Gorman	592cbc0452	fix issue with dialing client without resource. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44312 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Matt O'Gorman	ff11e64e67	updates for better compontent support git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43466 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Matt O'Gorman	ec4bf7a849	seperate jingle and gtalk so it will be easier to track changes in both of the moving specs. Currently chan_gtalk is compatible with the latest gtalk/libjingle version, and chan_jingle needs a lot of work. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43185 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago

42 Commits (1c72a8ab495aa9eb658b0afc0041dcb9babd56e7)