Commit Graph

1559 Commits

Author SHA1 Message Date
Anthony Minessale d3e320ef56 FS-11346: [core] add api to pass pre-parsed values instead of dial strings to switch_ivr_originate
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_`
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.
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
* 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
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
* 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
* 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
* 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
* 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
* 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
* 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
* 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
* 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