Anthony Minessale
d3e320ef56
FS-11346: [core] add api to pass pre-parsed values instead of dial strings to switch_ivr_originate
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SWITCH_DECLARE(switch_status_t) switch_dial_handle_create(switch_dial_handle_t **handle);
SWITCH_DECLARE(void) switch_dial_handle_destroy(switch_dial_handle_t **handle);
SWITCH_DECLARE(void) switch_dial_handle_add_leg_list(switch_dial_handle_t *handle, switch_dial_leg_list_t **leg_listP);
SWITCH_DECLARE(void) switch_dial_leg_list_add_leg(switch_dial_leg_list_t *parent, const char *dial_string, switch_dial_leg_t **legP);
SWITCH_DECLARE(void) switch_dial_handle_add_global_var(switch_dial_handle_t *handle, const char *var, const char *val);
SWITCH_DECLARE(void) switch_dial_handle_add_global_var_printf(switch_dial_handle_t *handle, const char *var, const char *fmt, ...);
SWITCH_DECLARE(switch_status_t) switch_dial_handle_add_leg_var(switch_dial_leg_t *leg, const char *var, const char *val);
SWITCH_DECLARE(switch_status_t) switch_dial_handle_add_leg_var_printf(switch_dial_leg_t *leg, const char *var, const char *fmt, ...);
SWITCH_DECLARE(int) switch_dial_handle_get_peers(switch_dial_handle_t *handle, int idx, char **array, int max);
SWITCH_DECLARE(int) switch_dial_handle_get_vars(switch_dial_handle_t *handle, int idx, switch_event_t **array, int max);
SWITCH_DECLARE(switch_event_t *) switch_dial_handle_get_global_vars(switch_dial_handle_t *handle);
SWITCH_DECLARE(switch_event_t *) switch_dial_leg_get_vars(switch_dial_leg_t *leg);
SWITCH_DECLARE(int) switch_dial_handle_get_total(switch_dial_handle_t *handle);
SWITCH_DECLARE(void) switch_ivr_orig_and_bridge(switch_core_session_t *session, const char *data, switch_dial_handle_t *dh);
add switch_dial_handle_t *dh to end of args for switch_ivr_originate
2018-08-22 18:20:13 +00:00
Brian West
a14dcfef3d
FS-10913: [mod_sofia] ignore_early_media=ring_ready not transitioning #resolve
2018-07-24 07:21:40 +00:00
Brian West
027ae79516
FS-10913: [mod_sofia] ignore_early_media=ring_ready not transitioning #resolve
2018-07-24 07:21:40 +00:00
Stephane Alnet
159c4ce95d
FS-6816 [mod_sofia] Set empty callee id if `_undef_`
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In some scenarios (e.g. MetaSwitch interop) the `display` field of callee-id should be left empty instead of being overwritten with the number.
As is done in other places, we allow for `_undef_` to mean "leave the field empty".
2018-02-02 10:19:03 +01:00
Anthony Minessale
4fabca25ef
FS-10792: [mod_sofia] when behind 1-to-1 NAT, console logs invalid handle #resolve
2017-11-15 13:04:15 -06:00
Hristo Trendev
9d4c26825f
FS-10617: [mod_sofia] Nightmare transfer: expose remote server's channel UUIDs as variables.
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Currently there is no easy way for ESL and dial plan users to easily correlate
the channel UUID of the call legs involved in a nightmare transfer. This patch
adds two new channel variables, which are set to the remote server call leg
UUIDs (transfer_refer_from, transfer_refer_for).
The UUIDs are passed from the remote server in custom headers (X-FS-Refer-From
and X-FS-Refer-For).
2017-08-28 13:38:45 +02:00
lazedo
da96699c1e
FS-10592: [mod_sofia] add sofia_profile_url to channel vars
2017-08-15 22:33:37 +01:00
Mike Jerris
719937ff8f
Merge pull request #1306 in FS/freeswitch from ~HRISTO/freeswitch:FS-10407-set-some-redirect-channel-variables to master
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* commit '1d15e411f9f5b6c8883cda47534cc1e9c3a77f95':
FS-10407: [mod_sofia] Set redirect variables when outbound_redirect_fatal is true
2017-07-06 21:15:07 +00:00
Mike Jerris
1729eb108f
FS-10439: [mod_sofia] fix small leak when receiving REFER message
2017-06-29 13:23:03 -05:00
Hristo Trendev
1d15e411f9
FS-10407: [mod_sofia] Set redirect variables when outbound_redirect_fatal is true
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In case of outbound_redirect_fatal=true none of the redirect variables are
set. This makes it impossible for ESL applications to extract any information
related to the "302 Moved Temporarily" reply.
2017-06-21 09:25:23 +02:00
Mike Jerris
8afac73cd6
FS-10338: [mod_sofia] add sip_invite_stamp variable of the time we received initial invite on an inbound call leg
2017-05-23 17:07:08 -04:00
Brian West
48f6978d55
FS-10059: [sofia-sip] handle re-invites during t.38 call
2017-04-05 16:31:38 -05:00
Brian West
f859a404f8
FS-10183 [mod_sofia] Broadsoft Shared line pickup would fall if a-leg is PCMU and your pickup device has G722 as its first codec.
2017-03-27 11:43:41 -05:00
Brian West
3dccd0a82f
FS-10149 [freeswitch-core] ZRTP encrypted calls drop on reinvite
2017-03-21 10:09:36 -05:00
Mike Jerris
2f4c9b363e
FS-10067: [mod_sofia] add update-refresher profile param and sip_update_refresher channel var to use update for session timers
2017-03-03 11:01:26 -06:00
Anthony Minessale
40bfe0fff5
FS-9154: [freeswitch-core] Add & remove video on re-invites #resolve
2017-02-15 13:56:35 -06:00
Mike Jerris
e8f6ed3d93
Merge pull request #1183 in FS/freeswitch from ~SAFAROV/freeswitch2:FS-10009 to master
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* commit '6c12f69e0c893646eda0bb010873583040aa136b':
FS-10009: mod_fail2ban - Added logging of network_ip for abandoned calls
2017-02-14 13:32:05 -06:00
Armen Babikyan
6ed86abf9f
FS-9300: Add support for disabling sofia's 100 Trying via configuration, and sending 100 Trying from dialplan
2017-02-13 14:49:05 -08:00
Sergey Safarov
6c12f69e0c
FS-10009: mod_fail2ban - Added logging of network_ip for abandoned calls
2017-02-10 13:17:45 -05:00
Sergey Safarov
df1ab07ca4
FS-9924: Removed extra space in source files
2017-02-09 23:59:49 -05:00
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
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
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
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
Anthony Minessale
ded506f611
FS-9898: [mod_sofia] Call hanging in FS if HOLD not successful #resolve
2017-01-03 12:01:48 -06:00
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
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
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
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
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
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
Stephane Alnet
86bcee0351
remove redundant `if (rep)` statement
2016-11-29 12:02:26 +01:00
Mike Jerris
cf0308b4e1
Merge pull request #1058 in FS/freeswitch from ~LAZEDO/freeswitch:feature/FS-9735 to master
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* commit '498ce4fc83953ed53c74f054163c829a439737df':
FS-9735 - send unknown headers to switch_ivr_set_user
2016-11-21 15:12:20 -06:00
Mike Jerris
08adab1918
FS-9699 regression
2016-11-15 19:25:37 -06:00