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Author SHA1 Message Date
marian
6627caaa08 Override channel dtmf methods in chan.dtmf only if explicitly requested. 
git-svn-id: http://voip.null.ro/svn/yate@5276 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-09-19 14:11:11 +00:00
marian
5893033961 Changed info allowed debug message level. Show it when compiled with xdebug. 
git-svn-id: http://voip.null.ro/svn/yate@5266 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-09-18 11:14:58 +00:00
marian
de2b64b26b Allow send dtmf method(s) to be configurable. Detect remote party INFO support from 'Allow' header. 
git-svn-id: http://voip.null.ro/svn/yate@5264 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-09-18 08:40:29 +00:00
marian
4682d3a3da Fixed bug: don't change party when processing inactive events to avoid overriding it with a wrong one. Moved party changed debug message to DebugInfo level. 
git-svn-id: http://voip.null.ro/svn/yate@5251 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-09-05 10:19:37 +00:00
marian
48dbd2df8b Added configurable option (defaults to disable) used to change remote party's ip/port when a channel receives a response or a new transaction from a different address. 
git-svn-id: http://voip.null.ro/svn/yate@5250 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-09-03 09:41:46 +00:00
oana
c14f9e2e31 Added separate setting for activating the SIP flood protection mechanism. 
git-svn-id: http://voip.null.ro/svn/yate@5234 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-08-20 11:06:06 +00:00
oana
afc7680016 Set allowed number of simultaneous calls from configuration files. If not set, default to the maxchans setting in yate.conf. 
git-svn-id: http://voip.null.ro/svn/yate@5227 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-08-17 13:29:31 +00:00
oana
7a778bc084 Added mechanism to drop INVITE/REGISTER/SUBSCRIBE/OPTIONS messages when detecting a flood. 
Other messages, as well as reINVITEs are still allowed.
Note: the meaning of the floodevents setting from ysipchan.conf has changed: now it sets the threshold for dropping messages.


git-svn-id: http://voip.null.ro/svn/yate@5226 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-08-17 12:31:07 +00:00
oana
72ba0c6911 Fix bug: first check if a message is a re-INVITE and handle it if required, otherwise check if the INVITE can be accepted. 
git-svn-id: http://voip.null.ro/svn/yate@5223 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-08-15 12:44:53 +00:00
paulc
6225ee6901 Added support for updating remote dialog and party on reINVITE. 
git-svn-id: http://voip.null.ro/svn/yate@5216 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-08-03 21:20:10 +00:00
paulc
e121386c7d Use setMedia(0) instead of clearEndpoint() so chan.rtp with terminate=true is emitted when changing media in SIP. 
git-svn-id: http://voip.null.ro/svn/yate@5189 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-07-09 13:41:14 +00:00
paulc
3f0c364a23 Don't copy the "handlers" parameter from user.auth 
git-svn-id: http://voip.null.ro/svn/yate@5118 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-06-14 17:58:58 +00:00
paulc
7e77fd2cfb Added possibility to track message progress through handlers. 
git-svn-id: http://voip.null.ro/svn/yate@5107 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-06-12 23:47:01 +00:00
paulc
17f591b08e Protect the generic methods list with the global mutex. 
git-svn-id: http://voip.null.ro/svn/yate@5106 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-06-12 08:45:39 +00:00
paulc
8094964bc3 Properly map privacy=full back to SIP Remote-Party-ID and Privacy headers. 
Got rid of a few useless String allocations.


git-svn-id: http://voip.null.ro/svn/yate@5090 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-06-04 12:06:33 +00:00
paulc
57577ceb53 Allow setting the external address of a NAT in the SDP sent by SIP. 
git-svn-id: http://voip.null.ro/svn/yate@5058 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-05-11 21:33:21 +00:00
paulc
69ebd3ebfa Add SIP call parameters before performing authentication. 
This allows user.auth to overwrite rather than duplicate parameters.


git-svn-id: http://voip.null.ro/svn/yate@5055 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-05-11 12:57:16 +00:00
paulc
e7c6d16ea8 Execute a hangup immediately if received a CANCEL message. 
This fixes the total duration of a cancelled call in slow routing scenarios.


git-svn-id: http://voip.null.ro/svn/yate@5017 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-04-21 11:12:35 +00:00
paulc
3c4d068f0d Make sure we disconnect a SIP call on all branches of hangup. 
Fixes BYE without CANCEL not being propagated.


git-svn-id: http://voip.null.ro/svn/yate@4994 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-04-04 17:36:34 +00:00
paulc
852a2874fb Map received disconnect cause "noanswer" to 480 Temporarily Unavailable. 
Code 487 should be used only when a transaction is CANCELed.


git-svn-id: http://voip.null.ro/svn/yate@4984 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-03-29 16:34:32 +00:00
paulc
dd6479f1c0 Allow the "number" parameter to be processed in the user.auth message. 
git-svn-id: http://voip.null.ro/svn/yate@4983 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-03-27 15:16:42 +00:00
paulc
cfe4edbc26 Changed the way SIP authentication is performed by UAS. 
This allows implementing custom authentication schemes.


git-svn-id: http://voip.null.ro/svn/yate@4975 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-03-26 00:25:00 +00:00
paulc
f1a062da01 Added possibility to return custom SIP failure from user.auth message. 
git-svn-id: http://voip.null.ro/svn/yate@4971 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-03-19 17:37:21 +00:00
paulc
747cf8bef1 Added the SIP user agent as "device" parameter of the user.auth message. 
git-svn-id: http://voip.null.ro/svn/yate@4970 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-03-18 16:57:28 +00:00
paulc
9b8f77f172 Clear any local SIP media channels before populating the 200 answer to BYE. 
This allows returning P-RTP-Stat when the other party initiates the hangup.


git-svn-id: http://voip.null.ro/svn/yate@4860 acf43c95-373e-0410-b603-e72c3f656dc1
 2012-02-08 15:56:00 +00:00
paulc
4c00b8682b Set the custom 3xx redirect code in the channel's reason too, not only in SIP message. 
git-svn-id: http://voip.null.ro/svn/yate@4762 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-12-15 10:06:40 +00:00
paulc
5f0a31cfd7 Forward the SIP audio change information to the peer channel. 
Change SIP media format if the peer changed.
This allows switching to T.38 when RTP is proxied.


git-svn-id: http://voip.null.ro/svn/yate@4712 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-11-18 13:01:34 +00:00
marian
5b718f2810 Add login/logout failure error to user.notify message. 
git-svn-id: http://voip.null.ro/svn/yate@4677 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-11-04 15:30:55 +00:00
paulc
4d74b5138f Added new static mutex to protect channel disconnect parameters during access. 
git-svn-id: http://voip.null.ro/svn/yate@4662 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-10-28 18:23:26 +00:00
paulc
e48bdcbefe Fixed SIP status command broken by SVN commit 3776. 
git-svn-id: http://voip.null.ro/svn/yate@4661 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-10-28 17:30:45 +00:00
paulc
2dc0283ae6 Added hack to preserve the RTP session when just the address has changed. 
Can be used together with the hack for ignoring port change.


git-svn-id: http://voip.null.ro/svn/yate@4639 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-10-05 20:43:25 +00:00
paulc
005c873bce Allow global calls limits to be reloaded. 
git-svn-id: http://voip.null.ro/svn/yate@4636 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-09-26 14:55:42 +00:00
paulc
9a85d0facd Fixed a few errors exposed by -Wunused-but-set-variable. 
Removed some leftover variables exposed by same new compiler check.


git-svn-id: http://voip.null.ro/svn/yate@4589 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-09-08 10:36:32 +00:00
paulc
ec22b3c4ca At end of routing check if the INVITE transaction still exists. 
This allows proper processing of an early CANCEL while routing is in progress.


git-svn-id: http://voip.null.ro/svn/yate@4573 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-08-29 09:42:30 +00:00
marian
b3102149c5 Added extra check to avoid building a sip party with invalid remote address. 
git-svn-id: http://voip.null.ro/svn/yate@4566 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-08-24 15:03:23 +00:00
marian
4fc28f755c String to integer conversion can now check the result against allowed min/max values. Use it in sip module. 
git-svn-id: http://voip.null.ro/svn/yate@4562 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-08-23 15:14:28 +00:00
marian
94117426bd Double the timeout period before using it. Decreased retransmission counters default values to keep the same overall timeout. 
git-svn-id: http://voip.null.ro/svn/yate@4559 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-08-22 15:16:26 +00:00
marian
fa56ec84d8 Made configurable the number of times to transmit a request or a final response when retransmission is required. 
git-svn-id: http://voip.null.ro/svn/yate@4556 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-08-22 14:11:51 +00:00
marian
3fe340c3ee Fixed bug: return proxy address when a line's domain is requested and there is no domain or registrar configured. This bug was introduced in SVN Rev. 4493. 
git-svn-id: http://voip.null.ro/svn/yate@4546 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-08-18 11:29:54 +00:00
marian
dc20a43184 Added configurable certificate file to present on outgoing TLS connections. 
git-svn-id: http://voip.null.ro/svn/yate@4545 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-08-18 10:49:27 +00:00
marian
d8ac36e914 Always reset a line's party when logout is required in user.login message. 
git-svn-id: http://voip.null.ro/svn/yate@4541 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-08-11 14:52:32 +00:00
marian
11ea7cbf94 Properly handle host and port in sip generate message handler. Use default when missing. 
git-svn-id: http://voip.null.ro/svn/yate@4540 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-08-10 11:22:18 +00:00
marian
56c62825b1 Listener type now defaults to udp if invalid. Handle listener type change in config. 
git-svn-id: http://voip.null.ro/svn/yate@4531 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-07-29 11:03:19 +00:00
marian
8afd45a4f3 Build the tls listener even if context is empty. Added debug. 
git-svn-id: http://voip.null.ro/svn/yate@4526 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-07-27 09:27:01 +00:00
marian
2e3c53bb1e Return line's resolved party address/port to match it for incoming requests. 
git-svn-id: http://voip.null.ro/svn/yate@4523 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-07-25 13:45:43 +00:00
paulc
90a16a73c8 Added missing ISUP content to disconnection SIP messages (BYE, CANCEL, 4xx/5xx). 
git-svn-id: http://voip.null.ro/svn/yate@4521 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-07-21 17:27:29 +00:00
marian
0aa1a6bc83 Made virtual base class destructor. 
git-svn-id: http://voip.null.ro/svn/yate@4496 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-07-12 15:21:39 +00:00
marian
f9f3241df5 Added support for tcp/tls sip transport. The sip module can now use (bind on) more then one address for udp. 
git-svn-id: http://voip.null.ro/svn/yate@4493 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-07-12 14:55:02 +00:00
paulc
0e7849e338 Emit new message "user.authfail" for paswword mismatch for existing users. 
Use that message in the ban script.


git-svn-id: http://voip.null.ro/svn/yate@4451 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-06-10 13:39:24 +00:00
paulc
e217640446 Optimized const String usage in the SIP module. 
git-svn-id: http://voip.null.ro/svn/yate@4431 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-06-03 15:14:38 +00:00
paulc
84a3ef7eea Pick up any new formats (even if not negotiated before) in RTP forward mode. 
git-svn-id: http://voip.null.ro/svn/yate@4417 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-05-28 12:43:15 +00:00
paulc
ffded5390e Add the billid parameter to SIP generated user.auth messages. 
git-svn-id: http://voip.null.ro/svn/yate@4396 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-05-19 11:21:30 +00:00
paulc
2ae8c1764d Added support for altering the reason and error code of SIP Registrar answers. 
git-svn-id: http://voip.null.ro/svn/yate@4382 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-05-17 11:12:37 +00:00
paulc
0a393ea086 Copy UAS SIP headers between user.register and REGISTER messages and answers. 
In UAC mode copy returned headers from REGISTER answer to user.notify message.


git-svn-id: http://voip.null.ro/svn/yate@4380 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-05-17 10:20:16 +00:00
paulc
bac2e503df Use the proxy address if domain is not set for outbound calls. 
git-svn-id: http://voip.null.ro/svn/yate@4372 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-05-12 20:55:26 +00:00
paulc
97dae7ffa3 Allow setting arbitrary SIP headers in rejected calls. 
git-svn-id: http://voip.null.ro/svn/yate@4371 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-05-12 20:29:27 +00:00
paulc
8985bd2e55 Allow copying parameters starting with authfail_ from failed user.auth messages. 
git-svn-id: http://voip.null.ro/svn/yate@4370 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-05-12 19:26:15 +00:00
paulc
ed8f5fca8e Added more mappings for SIP responses to improve mapping to ISUP and ISDN. 
git-svn-id: http://voip.null.ro/svn/yate@4360 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-05-05 12:43:40 +00:00
paulc
05025b8d22 Base64 decode and encode binary bodies in generic SIP messages. 
git-svn-id: http://voip.null.ro/svn/yate@4356 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-05-04 11:07:05 +00:00
marian
5aa5883660 Changed the name of the functions encoding and decoding isup bodies to fix scope related compile errors in VC++. 
git-svn-id: http://voip.null.ro/svn/yate@4355 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-05-04 10:47:17 +00:00
paulc
9837c35be8 Renamed SIP parameter "transport" to "ip_transport" because of collision with audio media transport. 
Fixed lack of media bug introduced by Rev 4349.


git-svn-id: http://voip.null.ro/svn/yate@4353 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-05-04 09:01:43 +00:00
paulc
59a857ef3e Detect generic SIP messages coming from registered accounts, set in_line and skip authentication if so. 
Decode more info from the incoming generic messages.


git-svn-id: http://voip.null.ro/svn/yate@4349 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-04-28 10:36:48 +00:00
paulc
80e58d6914 Fix the hangup of incoming SIP calls that didn't had the dialog tag updated. 
This could happen if a call was dropped immediately after being answered.


git-svn-id: http://voip.null.ro/svn/yate@4333 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-04-20 18:41:14 +00:00
paulc
1350c52677 Fixed a number of concurrency issues regarding reINVITE processing. 
git-svn-id: http://voip.null.ro/svn/yate@4332 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-04-20 18:38:30 +00:00
paulc
23bee28ded Terminate a SIP call for which we had a timeout on a reINVITE as media state becomes uncertain. 
Bug report and patch provided by Matthew.


git-svn-id: http://voip.null.ro/svn/yate@4310 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-04-18 11:17:50 +00:00
paulc
74178c73fc Clear the initiated reINVITE transaction on completion, allow further reINVITEs. 
Bug report and patch provided by Matthew.


git-svn-id: http://voip.null.ro/svn/yate@4309 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-04-18 11:03:23 +00:00
paulc
3736466398 The default RFC 2833 payload can be configured and overriden per call. 
git-svn-id: http://voip.null.ro/svn/yate@4287 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-04-12 15:41:49 +00:00
paulc
7d6644c96e Decode MIME type message/sipfrag as lines of text. 
Allow generic processing of SIP INFO messages that are not used for DTMFs.
Properly add lines of text bodies to generic SIP messages.


git-svn-id: http://voip.null.ro/svn/yate@4273 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-04-07 12:16:12 +00:00
paulc
5bcf9bba3e Honor an earlymedia=false parameter even when using RTP forwarding. 
git-svn-id: http://voip.null.ro/svn/yate@4267 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-04-06 09:13:16 +00:00
oana
7a374f6c37 Fixed checking if a engine.status module is indeed intended for that module. Fixed bug: don't print for a second time the status message for IAX module. 
git-svn-id: http://voip.null.ro/svn/yate@4221 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-03-28 12:40:44 +00:00
paulc
95e866418e Sanitize the custom SIP cause codes, must be >= 300. 
Allow altering the SIP code in the chan.disconnected message.


git-svn-id: http://voip.null.ro/svn/yate@4195 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-03-18 14:03:42 +00:00
paulc
3228166268 Leave the tel: URIs untouched when sent on an outbound SIP call. 
Also don't add sip: in front of sip:number URIs (we turned to sip:sip:number).


git-svn-id: http://voip.null.ro/svn/yate@4194 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-03-18 13:14:36 +00:00
paulc
5e47c82025 Moved the post-disconnect hooking code from ysigchan to the Channel class. 
Added capability of sending arbitrary SIP headers on call disconnect.


git-svn-id: http://voip.null.ro/svn/yate@4193 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-03-17 20:57:34 +00:00
paulc
b2283a0c2e Added flags that control how SIP message components are automatically completed. 
These flags can be configured per engine and can be overridden in some messages by setting an "xsip_flags" parameter.


git-svn-id: http://voip.null.ro/svn/yate@4161 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-03-07 13:50:59 +00:00
paulc
9cb25fc845 Added capability to set response code and additional parameters when redirecting a SIP call as UAS. 
git-svn-id: http://voip.null.ro/svn/yate@4149 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-03-02 16:58:22 +00:00
paulc
233a5f74a1 Copy SIP headers and decode ISUP (SIP-T/SIP-I) body to disconnect parameters. 
Parameters are put in local chan.hangup and peer's chan.disconnected messages.


git-svn-id: http://voip.null.ro/svn/yate@4142 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-03-01 16:08:52 +00:00
paulc
8b851caa08 Translate between SIP Remote-Party-ID fields "party" and "id-type" and Yate parameters "remote_party" and "remote_id_type". 
git-svn-id: http://voip.null.ro/svn/yate@4083 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-01-28 13:53:30 +00:00
paulc
fa661af327 Automatically copy disconnect parameters to the chan.hangup message. 
Provide those parameters to the peer call endpoint when disconnecting it.
Add numeric cause_q931 to disconnect params of h323chan.


git-svn-id: http://voip.null.ro/svn/yate@4052 acf43c95-373e-0410-b603-e72c3f656dc1
 2011-01-21 10:21:01 +00:00
oana
5abf18900b Changed the format of the accounts, links and interfaces status reported through engine.status. Added accountUsername OID. Modified the monitoring module to deal with the status change and the need to handle the accountUsername OID information. 
git-svn-id: http://voip.null.ro/svn/yate@3938 acf43c95-373e-0410-b603-e72c3f656dc1
 2010-12-09 14:30:47 +00:00
oana
55813dcbe3 Added SNMP support. 
git-svn-id: http://voip.null.ro/svn/yate@3776 acf43c95-373e-0410-b603-e72c3f656dc1
 2010-11-03 16:27:30 +00:00
paulc
e8f806c950 Added thread priority setting for the SIP module. 
git-svn-id: http://voip.null.ro/svn/yate@3753 acf43c95-373e-0410-b603-e72c3f656dc1
 2010-10-25 13:13:35 +00:00
paulc
b19d86b8e0 Made several Regexps static const so they are compiled only once, speeds up processing. 
One instance spotted by Allan Sandfeld Jensen, others by grep.


git-svn-id: http://voip.null.ro/svn/yate@3389 acf43c95-373e-0410-b603-e72c3f656dc1
 2010-06-17 11:08:00 +00:00
paulc
629339b292 Check the line before using it to alter the domain. 
git-svn-id: http://voip.null.ro/svn/yate@3245 acf43c95-373e-0410-b603-e72c3f656dc1
 2010-04-26 21:30:28 +00:00
paulc
cf29c804f9 Added support for detecting the domain of incoming SIP requests, gets copied to the generic "domain" parameter. 
git-svn-id: http://voip.null.ro/svn/yate@3242 acf43c95-373e-0410-b603-e72c3f656dc1
 2010-04-26 13:38:03 +00:00
paulc
9e7255048f Allow setting the outbound SIP proxy as host:port 
git-svn-id: http://voip.null.ro/svn/yate@3209 acf43c95-373e-0410-b603-e72c3f656dc1
 2010-04-19 13:25:08 +00:00
paulc
543d105246 Emit a BYE only if we have a SIP dialog - either early or established. 
git-svn-id: http://voip.null.ro/svn/yate@3175 acf43c95-373e-0410-b603-e72c3f656dc1
 2010-04-08 22:49:58 +00:00
paulc
2fc128e21e Store initially guessed local address, re-register only if it changed. 
git-svn-id: http://voip.null.ro/svn/yate@3172 acf43c95-373e-0410-b603-e72c3f656dc1
 2010-04-07 11:27:30 +00:00
paulc
1c9e786a74 Lock the module while accessing or changing SIP dialog information. 
git-svn-id: http://voip.null.ro/svn/yate@3125 acf43c95-373e-0410-b603-e72c3f656dc1
 2010-03-12 16:25:37 +00:00
paulc
638ffc277c Copy RTP stats in SIP to the CDR and to BYE or 200 message. 
git-svn-id: http://voip.null.ro/svn/yate@3106 acf43c95-373e-0410-b603-e72c3f656dc1
 2010-03-03 19:05:01 +00:00
paulc
98df7e335f Preserve the Call-ID across REGISTER messages in a single login session. 
git-svn-id: http://voip.null.ro/svn/yate@3096 acf43c95-373e-0410-b603-e72c3f656dc1
 2010-02-23 18:22:59 +00:00
paulc
8be9e9991a Add the REGISTER request URI and Call-ID to the user.(un)register messages. 
git-svn-id: http://voip.null.ro/svn/yate@3063 acf43c95-373e-0410-b603-e72c3f656dc1
 2010-02-02 23:59:54 +00:00
paulc
33a98e83aa Added Channel::initChan() method to add the channel to the driver explicitely, after the object is fully constructed. 
git-svn-id: http://voip.null.ro/svn/yate@3033 acf43c95-373e-0410-b603-e72c3f656dc1
 2010-01-26 10:31:32 +00:00
paulc
455b6dbf81 Automatically reset the maxcall timer when entering "answered" status. 
git-svn-id: http://voip.null.ro/svn/yate@2991 acf43c95-373e-0410-b603-e72c3f656dc1
 2009-12-18 19:08:41 +00:00
paulc
6a433cfe25 Copy by default some parameters from call.execute to chan.startup of outgoing calls. 
git-svn-id: http://voip.null.ro/svn/yate@2964 acf43c95-373e-0410-b603-e72c3f656dc1
 2009-11-25 18:34:54 +00:00
paulc
84b7cefe3d Emit call.update only when the SIP dialog tags have changed. 
Avoids flooding the system with useless call.update and call.cdr messages.


git-svn-id: http://voip.null.ro/svn/yate@2874 acf43c95-373e-0410-b603-e72c3f656dc1
 2009-10-21 14:10:47 +00:00
paulc
3624562f55 Added an extra parameter to SDPSession::updateFormats() allowing it to add or remove media. 
Use the SDPSession::updateFormats() method in SIP instead of reimplementing it.


git-svn-id: http://voip.null.ro/svn/yate@2873 acf43c95-373e-0410-b603-e72c3f656dc1
 2009-10-20 22:54:56 +00:00
paulc
6620daff86 Chain the debugging of the SIP engine to the SIP channels plugin. 
git-svn-id: http://voip.null.ro/svn/yate@2867 acf43c95-373e-0410-b603-e72c3f656dc1
 2009-10-16 12:04:32 +00:00
paulc
f9e2c83d85 Fixed the way the SIP transaction answer code is returned to xsip.generate, it wasn't dealing well with timeouts. 
While waiting for the arbitrary transaction to finish call Thread::idle() instead of yield().


git-svn-id: http://voip.null.ro/svn/yate@2845 acf43c95-373e-0410-b603-e72c3f656dc1
 2009-09-18 11:17:50 +00:00
paulc
657c2fef63 Dispatch a chan.rtp with terminate=true when a SIP media is removed or replaced. 
Added several channel parameters to the chan.rtp message.


git-svn-id: http://voip.null.ro/svn/yate@2839 acf43c95-373e-0410-b603-e72c3f656dc1
 2009-09-14 18:52:53 +00:00
paulc
cea5f52b61 Provide the entire SDPMedia to the mediaChanged() mthod, not only the name of the media. 
Add the "rtpid" parameter to the chan.rtp message if the media id() is available.


git-svn-id: http://voip.null.ro/svn/yate@2832 acf43c95-373e-0410-b603-e72c3f656dc1
 2009-09-14 09:06:16 +00:00