Anthony Minessale
d704fea95b
FS-10022: [mod_sofia] verify-profile 'none' not parsed for TLS #resolve
2017-02-09 11:25:29 -06:00
Seven Du
08603c7e89
FS-9904 #resolve cleanup and refactor msrp
2017-02-07 20:41:46 +08:00
Brian West
52728c253c
FS-10006: [core] Allow adding parameters to P-Asserted-Identity #resolve
2017-02-03 17:14:32 -06:00
Anthony Minessale
b81d6990ee
FS-9997: [mod_verto] Invalid JSON-RPC Response to an incorrect JSON-RPC Request. #resolve
2017-02-03 10:57:08 -06:00
Mike Jerris
0c99d5062f
Merge pull request #1166 in FS/freeswitch from ~ANTONIO/freeswitch:bugfix/FS-9966-invalid-contact-header-witn-private to master
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* commit '8673e0177c310f6fa5b2ae42dd5562968ce00df9':
[mod_sofia] FS-9966 fix private ip in contact header when invite w/ nosdp
2017-02-02 18:45:48 -06:00
Mike Jerris
c8256837b2
FS-9975: [mod_sofia] add contact params to request uri of outbound recovery reinvite for originally inbound calls
2017-01-27 17:02:01 -06:00
Mike Jerris
722feefd56
FS-9970: [mod_sofia] don't detect nat in cases when the contact is in the acl, but the packet actually came from a proxy. We need to check where we got the packet from as being a natted address instead of the contact in order to properly handle nat to our next hop
2017-01-27 15:13:18 -06:00
Antonio
8673e0177c
[mod_sofia] FS-9966 fix private ip in contact header when invite w/ nosdp
2017-01-24 15:11:01 +01:00
Mike Jerris
5ef273b4b3
Merge pull request #1146 in FS/freeswitch from bugfix/FS-9206-proxy-media-with-enable-3pcc-proxy to master
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* commit 'a597e216bc699567ddb77d1765cf095c3bb31183':
FS-9206: [core] endable proxy media auto-adjust on re-invite for text and video every time as the streams may be being added on re-invite
2017-01-17 13:11:30 -06:00
Mike Jerris
a597e216bc
FS-9206: [core] endable proxy media auto-adjust on re-invite for text and video every time as the streams may be being added on re-invite
2017-01-17 13:10:06 -06:00
Luis Azedo
52e1785d94
[mod_sofia] FS-9940 fix finding a-leg parameter
2017-01-12 08:37:18 -06:00
Mike Jerris
5d5b815e42
FS-9931: [mod_sofia] don't send display updates to endpoints who don't have UPDATE in their Allow header
2017-01-10 16:26:43 -06:00
Mike Jerris
f2f89f28f5
Merge pull request #1142 in FS/freeswitch from bugfix/FS-9844-sip_full_route-variable-doesnt-show to master
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* commit 'f418baf7c85c91b79ecb1cd593b570f99a7c0e2d':
FS-9844: [mod_sofia] populate sip_full_route var with all of the route headers, not just the first one
2017-01-09 10:44:08 -06:00
Mike Jerris
ad183fdea4
FS-9809: [mod_sofia] url encode caller id number before sticking it in the from header in case we have non url safe chars in the cid number in the caller profile
2017-01-06 16:16:16 -06:00
Brian West
c0423c5877
FS-9855: [mod_spandsp] Refused T38 reinvite on b-leg breaks T38 negotiation on a-leg when using T38 gateway mode #resolve
2017-01-05 16:34:11 -06:00
Mike Jerris
f418baf7c8
FS-9844: [mod_sofia] populate sip_full_route var with all of the route headers, not just the first one
2017-01-05 16:02:17 -06:00
Brian West
f54c7f9f34
FS-9855: [mod_spandsp] Refused T38 reinvite on b-leg breaks T38 negotiation on a-leg when using T38 gateway mode #resolve
2017-01-05 15:51:52 -06:00
Mike Jerris
62e2928889
FS-9915: [mod_sofia] fix non null terminated parsed sip body being passed in when sending to sip messages in a row on tcp in a single packet
2017-01-05 15:06:42 -06:00
Brian West
4ead2fcffc
FS-9916: [mod_spandsp] OB fax calls go zombie #resolve
2017-01-04 16:14:59 -06:00
Mike Jerris
dbd1c8684d
Merge pull request #1124 in FS/freeswitch from ~TCULJAGA/freeswitch_tc:bugfix/FS-9873-a1-hash_for_mod_verto to master
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* commit '1f7a7e336f0d2d0d4c67ee45478bf9b759e0dda8':
FS-9873 mod_verto a1-hash - squashed
2017-01-04 11:37:32 -06:00
Tihomir Culjaga
1f7a7e336f
FS-9873 mod_verto a1-hash - squashed
2017-01-04 06:48:35 -06:00
Anthony Minessale
e313b6ea3f
FS-9206: [mod_sofia] proxy media with enable-3pcc=proxy does not properly pass audio after 3pcc re-invite #resolve
2017-01-03 18:32:32 -06:00
Josh Allmann
7248a4f3eb
FS-9910 [mod_sofia]: Set SIP reason header for BYE events.
2017-01-03 16:21:43 -05:00
Mike Jerris
20697cf0bd
Merge pull request #1118 in FS/freeswitch from ~ANTONIO/freeswitch:bugfix/FS-9877-mod_loopback-no-audio to master
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* commit '5dfc63c126f664cff09ddc6d2a67e05c2426f940':
FS-9877 [mod_loopback] fix no audio
2017-01-03 14:39:40 -06:00
Mike Jerris
f273ecda3f
FS-9903: [msrp] fix namespacing and visibility of some structs
2017-01-03 13:51:30 -06:00
Anthony Minessale
ded506f611
FS-9898: [mod_sofia] Call hanging in FS if HOLD not successful #resolve
2017-01-03 12:01:48 -06:00
Seven Du
7e24a79580
FS-9903 WIP MSRP client mode support
2017-01-02 10:34:41 +08:00
Anthony Minessale
57f5932f01
FS-9206: [mod_sofia] proxy media with enable-3pcc=proxy does not properly pass audio after 3pcc re-invite #resolve
2016-12-30 17:36:29 -06:00
Brian West
d28f29594f
FS-9776: [mod_sofia] SIP Transfer generates high CPU #resolve
2016-12-28 12:40:06 -06:00
Mike Jerris
15632a0bd8
Merge pull request #1084 in FS/freeswitch from ~MOCHOUINARD/freeswitch:FS-9792 to master
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* commit '8c1ed38d5eef031e4f471fe5f69ad052a9711997':
FS-9792: Set channel variable based on the sip phone Accept Language SIP message
2016-12-27 13:30:40 -06:00
Mike Jerris
6e2764776f
Merge pull request #1110 in FS/freeswitch from ~SEBASTIAN/freeswitch:bugfix/FS-9840-fix-some-warnings-V2 to master
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* commit '8c94e6f57fd9adee5e6a12535811fff8e4d9ae46':
FS-9840 mod_avmd: Fix implicit declaration warning
FS-9840 sofia-sip: fix implicit declaration warning
FS-9840 mod-verto: fix implicit declaration warning
FS-9840 mod_sofia: fix redefine warning
2016-12-27 12:36:58 -06:00
Sebastian Kemper
8063ad658b
FS-9840 mod-verto: fix implicit declaration warning
...
This fixes the following compile-time warning:
making all mod_verto
make[7]: Entering directory '/home/sk/tmp/lede/build_dir/target-mips_24kc_musl-1.1.15/freeswitch-1.8.0/src/mod/endpoints/mod_verto'
CC mod_verto_la-mod_verto.lo
CC mod_verto_la-ws.lo
ws.c: In function 'hton64':
ws.c:730:14: error: implicit declaration of function '__bswap_64' [-Werror=implicit-function-declaration]
else return __bswap_64(val);
^
cc1: all warnings being treated as errors
Fix by including byteswap.h, which is available on Linux and also
everywhere glibc is used (wpa_supplicant includes this header the same
way).
Signed-off-by: Sebastian Kemper <sebastian_ml@gmx.net>
2016-12-23 21:44:25 +01:00
Mike Jerris
ddf48b8602
Merge pull request #1105 in FS/freeswitch from bugfix/FS-9832-start-a-single-gateway to master
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* commit '50e0f0195e67208889f15a439ea6ccb567b862e7':
FS-9832 start a single gateway or _all_ gateways
2016-12-23 14:15:53 -06:00
Mike Jerris
d1ccc77d4f
FS-9854: [mod_sofia] SDP O/A fails to put sdp in messages after certain kinds of sip traffic
2016-12-22 11:32:13 -05:00
Antonio
5dfc63c126
FS-9877 [mod_loopback] fix no audio
2016-12-22 14:55:51 +01:00
Anthony Minessale
914a6e764f
FS-9866: [freeswitch-core] 3pcc=proxy for FS client and local SDP #resolve
2016-12-20 18:28:15 -06:00
Brian West
eef2313a40
FS-9846: [mod_sofia] Bugs related with Hold and Proxy Hold option added in FS-9192 after merges in 1.6.11 #resolve
2016-12-20 16:19:30 -06:00
Anthony Minessale
ade6e856a2
FS-9745: [mod_sofia] Call to FS WebRTC Gateway fails when no SDP on invite #resolve
2016-12-20 12:20:41 -06:00
Anthony Minessale
634490495f
FS-9806: [mod_loopback] mod_loopback: bowout is always done even if set to false loopback_bowout_on_execute=false,loopback_bowout=false #resolve
2016-12-20 11:08:17 -06:00
Brian West
3387b90705
FS-9829 #resolve [FreeSWITCH 200ok to second reINVITE on a dialog doesn't contain an SDP.]
2016-12-13 16:39:57 -06:00
Seven Du
50e0f0195e
FS-9832 start a single gateway or _all_ gateways
2016-12-08 20:47:22 +08:00
Brian West
df8f7f4639
FS-9823 free subclass properly
2016-12-07 08:41:45 -06:00
Marc Olivier Chouinard
8c1ed38d5e
FS-9792: Set channel variable based on the sip phone Accept Language SIP message
2016-12-06 17:17:39 -05:00
Seven Du
6528ae00b5
FS-9817 #resolve fix regression from 828d6eaf0177caff9b60052be3c83b2008f85416
2016-12-06 11:33:01 +08:00
Seven Du
b26fa6e17c
FS-9810 fix ws write fail on slow network
2016-12-04 12:49:46 +08:00
Brian West
89063a1a4c
FS-9765 one tweak from submitted patch to use switch_channel_var_true instead of switch_channel_get_variable no need to allocate on every hold/unhold just to check if this is enabled.
2016-12-02 11:51:49 -06:00
Brian West
c9a05d7e60
Merge pull request #1077 in FS/freeswitch from ~STEPHALNET/freeswitch:FS-9777 to master
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* commit '86bcee03518ff5ecbb7bae8e78f3821b4027ad09':
remove redundant `if (rep)` statement
2016-12-02 11:44:04 -06:00
Brian West
dac1b67c20
Merge pull request #888 in FS/freeswitch from ~MZAKA/freeswitch:bugfix/FS-9277-sip-info-record to master
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* commit 'addf7555bff15889d73e48bf70445d6d27d79fce':
FS-9277: sip info with record: on and off doesn't start and stop call recording sessions
2016-12-01 20:22:57 -06:00
Mike Jerris
b338bb559b
FS-9782: [mod_sofia] on recovery, flip the order of the record route on inbound calls only, use the record route in the same order on inbound calls and in reverse order on outbound calls as the initial route set when doing the recover invite. Account for the call direction based on how sip considers it, not based on freeswitch direction so inbound calls after recovery are treated as outbound in this logic
2016-11-30 15:32:03 -07:00
Mike Jerris
d498e8a8b3
FS-9782: [mod_sofia] on recovery, don't flip the order of the record route ever, on outbound calls use the record route in the reverse order as the initial route set when doing the recover invite
2016-11-29 15:04:17 -07:00