b5ea89e780
Add SIP CANCEL Cause processing
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SIP CANCEL Causes are now processed and forwarded to OSMO-CC.
- Q.850 cause 127 = "internetwork, unspecified" will be ignored as if no cause was given.
- SIP cause 200 with description "call completed elsewhere" will be translated to ISDN cause 26 = "non-selected user clearing"
This translation should avoid ISDN terminals logging a missed call, if another ringing SIP extension answers an incoming call.
2024-02-03 19:11:41 +01:00
c28cdd4b43
Updated libs
2024-01-25 20:13:41 +01:00
76f0466af5
Move from local to external osmo* libraries
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src/libdebug -> libosmocore
src/libselect -> libosmocore
src/libtimer -> libosmocore
src/libosmocc -> libosmo-cc
src/libg711 -> libosmo-cc
2024-01-25 20:13:31 +01:00
33b65d687f
Fix use-after-free buf
2024-01-04 16:09:06 +01:00
96eb0c480d
show that "NOTIFY" is not supported
2024-01-04 15:59:55 +01:00
7fd0d87aa0
show created callref when remote invites us to a call
2024-01-04 15:59:55 +01:00
b54069541f
minor fix on configure.ac
2024-01-04 15:59:55 +01:00
842d941d06
In release_and_destroy() show correct cc_callref
2024-01-04 15:59:55 +01:00
2638a11e6b
Add option to specify allowed remote user agent
2023-11-07 16:30:13 +01:00
0f1e234c91
Add feature to block calls/registrations with wrong credentials
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A wrong credential causes a command to be executed. The command will
receive the source IP. It can then use a shell command to add blocking
to the firewall.
2023-11-07 16:30:11 +01:00
acb35e3cb5
Add timer to expire registration (from remote)
2023-11-07 16:30:10 +01:00
2d20ce6f71
Send/receive no ringing after progress (default for SIP)
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Instead of having no extra ringing message after progress message, we
send/expect no ringing message after progress message with SDP. (early
audio). This is the default for SIP. Optionally there can be extra
messages, but both sides must agree with that.
Option changes: receive-ner -> receive-rap
Option changes: send-ner -> send-rap
2023-11-07 16:30:09 +01:00
94534cccaf
Fix call release issues
2023-11-07 16:30:08 +01:00
da61c674b4
Use option timer to release call, if options request fails.
2023-03-28 19:14:05 +02:00
172dee960a
Send Contract when receving a call
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Contact with correct peer address must be included, so the call proceeds
correctly.
Only the host/port is used, no user. This seems to work.
2023-03-28 19:10:49 +02:00
18af4626db
If the remote unregisters, remove remote_peer
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Subsequent calls will be rejected with "out of order", as they should be.
2023-03-26 17:44:56 +02:00
ee9ea365d8
After receiving an IP via STUN, do not trigger registration, if not used
2023-03-25 16:51:32 +01:00
bab1b391c2
After failing to resolve via STUN, set timer to try again
2023-03-25 16:51:05 +01:00
7fc8bb5954
Fixed SDP response on every 200 INVITE response
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Calls will fail, if the SDP is not included, even if it was included in
session progress message. If the remote SIP peer ignores it, it must be
included in the final 200 OK message.
2023-01-01 17:12:33 +01:00
f071f0e738
Updated libs
2022-10-30 17:07:54 +01:00
2bb1f893a9
Fixed broken Contact line
2022-02-25 07:00:35 +01:00
0ad29dcc63
Updated Libs
2022-02-25 07:00:35 +01:00
aed1210513
Updated libs
2022-02-25 07:00:18 +01:00
2a203482d1
Fixed bug in handling public IP string
2021-12-27 19:09:07 +01:00
a3d3f884cd
Fix and cleanup of from/to uri generation for INVITE
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If a remote peer registers to us, we can still use dialed number of an
outgoing call (INVITE) to replace the user of the to-line.
E.g. To: <user>@remote.host (if no dialed number is given)
To: <number>@remote.host (if dialed number is given)
2021-09-17 15:42:33 +02:00
755bdb8505
Fixed bug that stopped registration towards remote registrar
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If someone tries to register to us, which does not make sense, caused
to stop the registration process to the remote registrar. This is now
fixed.
2021-09-17 15:39:35 +02:00
4cdc6437a6
Add --remote-nat-sip --remote-nat-rtp
2021-09-14 15:06:42 +02:00
95fdc4d6da
Updated libs
2021-09-14 15:06:30 +02:00
4801128069
Updated libs
2021-05-24 12:55:52 +02:00
603b437a80
Fixed compiler warning
2021-03-28 12:55:42 +02:00
91fd188a92
Fixed incoming (remote-to-local) registration procedure
2021-03-21 13:33:01 +01:00
ada147349f
Add option to change User-Agent tag
2021-03-20 16:11:06 +01:00
d8977d4fd4
Re-register as soon as public IP changes when using STUN
2021-03-20 15:35:05 +01:00
e15e2810a6
Minor fixes
2021-03-20 11:36:31 +01:00
6083ccaa09
If SDP has been received earlier, do not require it on answer of call
2021-03-20 11:35:45 +01:00
be7a07309e
Updated libs
2021-03-20 11:34:48 +01:00
660b673958
Fix to configure.ac
2021-03-20 07:18:39 +01:00
8c5c775de1
Add missing m4 directory
2021-03-20 06:51:49 +01:00
4cc4703908
Add option to change config file path
2021-03-14 11:15:06 +01:00
6152ea8349
Updated libs
2021-03-14 11:14:23 +01:00
eb48c349c6
Fixed memory leak, caused by not freeing saved events
2021-01-25 08:51:48 +01:00
Martin Hauke
345d953ab6
Fix typos
2021-01-03 10:12:01 +01:00
cc13b6f321
Updated libs
2021-01-03 10:11:28 +01:00
a156278f90
Add LICENSE file
2021-01-03 09:46:21 +01:00
9569b2591e
Bug: Fixed shutdown of Sofia stack
2021-01-03 09:43:22 +01:00
ab8fef45d6
Add libraries from Osmocom-Analog
2020-12-29 11:18:33 +01:00
ec0042ae70
Inital GIT import
2020-12-29 11:18:32 +01:00
Dennis Grunert
Martin Hauke