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Change subject: vty info: MS power levels in dBm are not negative
......................................................................
Patch Set 1: Code-Review+1
(1 comment)
Patchset:
PS1:
oh my, what an embarrassing bug...
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Change subject: ipa: Use pseudo-random number for SLS in IPA->M3UA direction
......................................................................
ipa: Use pseudo-random number for SLS in IPA->M3UA direction
In Change-Id Ice7bab997b84cfed00c7d6d780c70f4e9fac6002 we introduced
code that would make the LSB of the file descriptor be used as SLS
when passing packets from IPA in M3UA direction.
This did however not achieve sufficient entropy in real-world use cases.
In this change, we change over to allocating a pseudo-random SLS to each
IPA connection at the time it is established; We then assign that SLS
to each packet received on that IPA connection.
Change-Id: Ia4e66d660b6057338f66a47fffc8a0d32759f733
Related: SYS#6543
Closes: SYS#6802
---
M src/osmo_ss7_asp.c
M stp/stp_main.c
2 files changed, 32 insertions(+), 3 deletions(-)
Approvals:
pespin: Looks good to me, but someone else must approve
Jenkins Builder: Verified
dexter: Looks good to me, but someone else must approve
fixeria: Looks good to me, approved
diff --git a/src/osmo_ss7_asp.c b/src/osmo_ss7_asp.c
index 1f45749..f290fdb 100644
--- a/src/osmo_ss7_asp.c
+++ b/src/osmo_ss7_asp.c
@@ -803,9 +803,9 @@
msg->dst = asp;
rate_ctr_inc2(asp->ctrg, SS7_ASP_CTR_PKT_RX_TOTAL);
- /* we can use the 'fd' return value of osmo_stream_srv_get_fd() here unverified as all we do
- * is 'roll the dice' to obtain a 4-bit SLS value. */
- return ipa_rx_msg(asp, msg, fd & 0xf);
+ /* we simply use the lower 4 bits of the asp_id, which is initialized to a pseudo-random value upon
+ * connect */
+ return ipa_rx_msg(asp, msg, asp->asp_id & 0xf);
}
/* netif code tells us we can read something from the socket */
@@ -916,6 +916,12 @@
if (asp->cfg.trans_proto == IPPROTO_SCTP) {
rc = ss7_asp_apply_peer_primary_address(asp);
rc = ss7_asp_apply_primary_address(asp);
+ } else {
+ if (asp->cfg.proto == OSMO_SS7_ASP_PROT_IPA) {
+ /* we use the lower 4 bits of the asp_id feld as SLS; let's initialize it here from a
+ * pseudo-random value */
+ asp->asp_id = rand() & 0xf;
+ }
}
if (asp->lm && asp->lm->prim_cb) {
diff --git a/stp/stp_main.c b/stp/stp_main.c
index 01d1865..d630032 100644
--- a/stp/stp_main.c
+++ b/stp/stp_main.c
@@ -213,6 +213,8 @@
{
int rc;
+ srand(time(NULL));
+
tall_stp_ctx = talloc_named_const(NULL, 1, "osmo-stp");
msgb_talloc_ctx_init(tall_stp_ctx, 0);
osmo_init_logging2(tall_stp_ctx, &log_info);
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Change subject: jobs/ttcn3: …-pablo-gtp: don't wipe workspace
......................................................................
Patch Set 1:
(1 comment)
Commit Message:
https://gerrit.osmocom.org/c/osmo-ci/+/36486/comment/a74e3e80_b10fe884
PS1, Line 10: cloning the kernel repositories takes a long time and lots of traffic
maybe it's worth doing a shallow clone, avoiding 90% of the traffic and time if all we want is one specific [base] version, and nothing earlier than that?
Also: if we know there are [likely] other kernel trees around somewhere in [other] workspaces, we can specify those with --reference-if-able to get any objects locally, if available/present.
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Change subject: msc: f_expect_paging(): fix by_tmsi arg
......................................................................
Patch Set 1:
(1 comment)
File msc/BSC_ConnectionHandler.ttcn:
https://gerrit.osmocom.org/c/osmo-ttcn3-hacks/+/36455/comment/7df0e0f3_f4f5…
PS1, Line 1360: boolean by_tmsi := true
> You already spent more time saying how much time you'd spend changing it, than actually applying the […]
I think we should in general try to write idiomatic code, whether it's python or TTCN3. If it's a huge effort/detour, ok, then we can deviate from it. Or if we simply are not knowledgeable enough in a given language. But here it really seems relatively trivial (unless I'm missing something) to do things "right".
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Change subject: SIP_Templates: Rework templates to make them more extensible
......................................................................
Patch Set 2: Code-Review+2
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Change subject: Convert RTP/RTCP/OSMUX I/O from osmo_fd to osmo_io
......................................................................
Patch Set 4:
(5 comments)
File src/libosmo-mgcp/mgcp_network.c:
https://gerrit.osmocom.org/c/osmo-mgw/+/36363/comment/d7843810_68ce3add
PS4, Line 1047: * \param[in] msg message buffer that holds the data to be send.
> Please document here: […]
Done
https://gerrit.osmocom.org/c/osmo-mgw/+/36363/comment/b60c709c_b5cddd7c
PS4, Line 1520: .from_addr = (struct osmo_sockaddr *) saddr,
> I odn't see why the cast is needed here? saddr is already that type.
but not const... we need to de-constify it, as 'struct osmo_rtp_msg_ctx' has a non-const from_addr member.
https://gerrit.osmocom.org/c/osmo-mgw/+/36363/comment/0568d840_72e84b46
PS4, Line 1696: if ((end->rtp && osmo_iofd_get_fd(end->rtp) != -1) ||
> You can probably skip the "osmo_iofd_get_fd(end->rtp) != -1" check here, or otherwise make sure you […]
I originally did not have that check in place and it created problems (I think a crash). I don't recall the details. In any case, I'm just translating the old code with its -1 check to osmo_io. Anything else would be outside of the scope of this patch. The smaller such a patch is, and the less it changes the existing code, the better.
File src/libosmo-mgcp/mgcp_osmux.c:
https://gerrit.osmocom.org/c/osmo-mgw/+/36363/comment/e178a3da_f902f6c6
PS4, Line 526: goto out_free_v4;
> you may be missing an "osmo_iofd_unregister()" in the section you are going with goto here. […]
osmo_iofd_free() includes both closing and unregistering. This is documented in the omsmo_iofd documentation, and the code even has a comment below at the goto label that explicitly explains/states this.
File tests/mgcp/mgcp_test.c:
https://gerrit.osmocom.org/c/osmo-mgw/+/36363/comment/385b76a2_6a1da5ee
PS4, Line 661: uint8_t *buf = msgb_data(msg);
> const
Done
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Hello Jenkins Builder, dexter, fixeria, neels, pespin,
I'd like you to reexamine a change. Please visit
https://gerrit.osmocom.org/c/osmo-mgw/+/36363?usp=email
to look at the new patch set (#5).
The following approvals got outdated and were removed:
Code-Review+1 by fixeria, Verified+1 by Jenkins Builder
Change subject: Convert RTP/RTCP/OSMUX I/O from osmo_fd to osmo_io
......................................................................
Convert RTP/RTCP/OSMUX I/O from osmo_fd to osmo_io
Converting from osmo_fd to osmo_io allows us to switch to the new
io_uring backend and benefit from related performance benefits.
In a benchmark running 200 concurrent bi-directional voice calls with
GSM-EFR codec, I am observing:
* the code before this patch uses 40..42% of a single core on a
Ryzen 5950X at 200 calls (=> 200 endpoints with each two connections)
* no increase in CPU utilization before/after this patch, i.e. the
osmo_io overhead for the osmo_fd backend is insignificant compared
to the direct osmo_fd mode before
* an almost exactly 50% reduction of CPU utilization when running the
same osmo-mgw build with LIBOSMO_IO_BACKEND=IO_URING - top shows
19..21% for the same workload instead of 40..42% with the OSMO_FD
default backend.
* An increase of about 4 Megabytes in both RSS and VIRT size when
enabling the OSMO_IO backend. This is likely the memory-mapped rings.
No memory leakage is observed when using either of the backends.
Change-Id: I8471960d5d8088a70cf105f2f40dfa5d5458169a
---
M include/osmocom/mgcp/mgcp.h
M include/osmocom/mgcp/mgcp_network.h
M src/libosmo-mgcp/mgcp_conn.c
M src/libosmo-mgcp/mgcp_iuup.c
M src/libosmo-mgcp/mgcp_network.c
M src/libosmo-mgcp/mgcp_osmux.c
M tests/mgcp/mgcp_test.c
7 files changed, 234 insertions(+), 169 deletions(-)
git pull ssh://gerrit.osmocom.org:29418/osmo-mgw refs/changes/63/36363/5
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Change subject: hnbgw: Introduce tests about UE registration
......................................................................
Patch Set 2:
(1 comment)
File hnbgw/HNBGW_Tests.ttcn:
https://gerrit.osmocom.org/c/osmo-ttcn3-hacks/+/36484/comment/759910e0_367e…
PS1, Line 1011: testcase TC_ue_register_before_hnb_register() runs on test_CT {
> You are missing adding these tests to control()?
Done
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Hello Jenkins Builder,
I'd like you to reexamine a change. Please visit
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Verified+1 by Jenkins Builder
Change subject: hnbgw: Introduce tests about UE registration
......................................................................
hnbgw: Introduce tests about UE registration
We didn't have any test coverage for HNBAP UE registration so far.
Let's start with two basic tests:
* normal / successful case
* abnormal case: UE Register without prior HNB Register
Change-Id: Ice2743d376ab8041646259fa25117d6fd0e8c2fd
---
M hnbgw/HNBGW_Tests.ttcn
M library/hnbap/HNBAP_Templates.ttcn
2 files changed, 234 insertions(+), 0 deletions(-)
git pull ssh://gerrit.osmocom.org:29418/osmo-ttcn3-hacks refs/changes/84/36484/2
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Change subject: HNBAP: Support IMSI identity type in hnbgw_tx_ue_register_rej()
......................................................................
Patch Set 4:
(1 comment)
File src/osmo-hnbgw/hnbgw_hnbap.c:
https://gerrit.osmocom.org/c/osmo-hnbgw/+/36481/comment/003819c1_d73da41f
PS3, Line 213: char imsi[16];
> we probably have defined for this 16?
Done
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Hello Jenkins Builder,
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Change subject: HNBAP: Support IMSI identity type in hnbgw_tx_ue_register_rej()
......................................................................
HNBAP: Support IMSI identity type in hnbgw_tx_ue_register_rej()
Change-Id: I2e00968cbf686f78f5c9655e899963f2b84dd78b
---
M src/osmo-hnbgw/hnbgw_hnbap.c
1 file changed, 21 insertions(+), 0 deletions(-)
git pull ssh://gerrit.osmocom.org:29418/osmo-hnbgw refs/changes/81/36481/4
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Change subject: Introduce asterisk-master docker image
......................................................................
Patch Set 1: Code-Review+1
(1 comment)
File asterisk-master/Dockerfile:
https://gerrit.osmocom.org/c/docker-playground/+/36491/comment/7836af88_0c1…
PS1, Line 42: RUN git
btw, particularly with large git repos it might be sufficient to do a shallow clone here, as we don't really care about the history during build. Not saying it must be done here, just sharing an idea.
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Change subject: use json to decode counters from nftables
......................................................................
Patch Set 4:
(1 comment)
Patchset:
PS4:
from the build errors and looking at the patch it seems like it's missing using the LIBJANSSON_LIBS / LIBJANSSON_CFLAGS in the Makefile.am ?
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laforge has posted comments on this change. ( https://gerrit.osmocom.org/c/osmo-ttcn3-hacks/+/36489?usp=email )
Change subject: SIP_Templates: Rework templates to make them more extensible
......................................................................
Patch Set 2: Code-Review+1
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Change subject: SIP: generate seq_nr within allowed range
......................................................................
Patch Set 1: Code-Review+1
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Change subject: HNBAP: Make sure to respond with correct "reject"
......................................................................
Patch Set 3: Code-Review+2
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Change subject: HNBAP: Use proper cause values during HNB-REGISTER-REQ processing
......................................................................
Patch Set 2: Code-Review+2
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Hello Jenkins Builder,
I'd like you to reexamine a change. Please visit
https://gerrit.osmocom.org/c/osmo-ttcn3-hacks/+/36490?usp=email
to look at the new patch set (#2).
The following approvals got outdated and were removed:
Verified-1 by Jenkins Builder
Change subject: Introduce Asterisk_Tests testsuite
......................................................................
Introduce Asterisk_Tests testsuite
Add initial infrastructure to run tests against an Asterisk process.
An not-yet-finished draft test doing registration is submitted to
validate communication towards Asterisk works.
The testsuite will be improved in follow-up commits, but this way other
people can already start using it and we can set up the dockerized setup
+ jenkins jobs to run it nightly.
Related: SYS#6782
Change-Id: I66f776d5df6fb5dc488d9e589b84a6b2385406e8
---
M Makefile
A asterisk/Asterisk_Tests.cfg
A asterisk/Asterisk_Tests.default
A asterisk/Asterisk_Tests.ttcn
A asterisk/README.md
A asterisk/expected-results.xml
A asterisk/gen_links.sh
A asterisk/regen_makefile.sh
M library/SIP_Emulation.ttcn
M library/SIP_Templates.ttcn
10 files changed, 414 insertions(+), 1 deletion(-)
git pull ssh://gerrit.osmocom.org:29418/osmo-ttcn3-hacks refs/changes/90/36490/2
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pespin has uploaded this change for review. ( https://gerrit.osmocom.org/c/docker-playground/+/36492?usp=email )
Change subject: Introduce ttcn3-asterisk-ims-ue-test
......................................................................
Introduce ttcn3-asterisk-ims-ue-test
This dockerized setup allows running the Asterisk_Tests ttcn3 testsuite
against Asterisk from the asterisk-master docker image, in order to test
the IMS UE feature which will be developed soon.
Related: SYS#6782
Depends: osmo-ttcn3-hacks.git Change-Id I66f776d5df6fb5dc488d9e589b84a6b2385406e8
Change-Id: Idd262b9224ea879a6e941792944fe0b0db3e68ba
---
A ttcn3-asterisk-ims-ue-test/Asterisk_Tests.cfg
A ttcn3-asterisk-ims-ue-test/Dockerfile
A ttcn3-asterisk-ims-ue-test/Makefile
A ttcn3-asterisk-ims-ue-test/asterisk/asterisk.conf
A ttcn3-asterisk-ims-ue-test/asterisk/asterisk.sh
A ttcn3-asterisk-ims-ue-test/asterisk/extensions.conf
A ttcn3-asterisk-ims-ue-test/asterisk/pjsip.conf
A ttcn3-asterisk-ims-ue-test/jenkins.sh
8 files changed, 361 insertions(+), 0 deletions(-)
git pull ssh://gerrit.osmocom.org:29418/docker-playground refs/changes/92/36492/1
diff --git a/ttcn3-asterisk-ims-ue-test/Asterisk_Tests.cfg b/ttcn3-asterisk-ims-ue-test/Asterisk_Tests.cfg
new file mode 100644
index 0000000..c8744bb
--- /dev/null
+++ b/ttcn3-asterisk-ims-ue-test/Asterisk_Tests.cfg
@@ -0,0 +1,22 @@
+[ORDERED_INCLUDE]
+"/osmo-ttcn3-hacks/Common.cfg"
+"/osmo-ttcn3-hacks/asterisk/Asterisk_Tests.default"
+
+[LOGGING]
+
+[TESTPORT_PARAMETERS]
+*.SIP.default_local_address := "172.18.11.103"
+*.SIP.local_sip_port := "5060"
+*.SIP.default_dest_address := "172.18.11.10"
+*.SIP.default_dest_port := "5094"
+
+[MODULE_PARAMETERS]
+Asterisk_Tests.mp_local_sip_host := "172.18.11.103"
+Asterisk_Tests.mp_remote_sip_host := "172.18.11.10"
+Asterisk_Tests.mp_local_sip_port := 5060
+Asterisk_Tests.mp_remote_sip_port := 5094
+
+[MAIN_CONTROLLER]
+
+[EXECUTE]
+Asterisk_Tests.control
diff --git a/ttcn3-asterisk-ims-ue-test/Dockerfile b/ttcn3-asterisk-ims-ue-test/Dockerfile
new file mode 100644
index 0000000..1a2f8aa
--- /dev/null
+++ b/ttcn3-asterisk-ims-ue-test/Dockerfile
@@ -0,0 +1,13 @@
+ARG REGISTRY
+ARG USER
+FROM $REGISTRY/$USER/debian-bookworm-titan
+ARG OSMO_TTCN3_BRANCH="master"
+
+ADD https://gerrit.osmocom.org/plugins/gitiles/osmo-ttcn3-hacks/+/$OSMO_TTCN3_B… /tmp/commit
+RUN ttcn3-docker-prepare "$OSMO_TTCN3_BRANCH" asterisk
+
+VOLUME /data
+
+COPY Asterisk_Tests.cfg /data/Asterisk_Tests.cfg
+
+CMD ttcn3-docker-run asterisk Asterisk_Tests
diff --git a/ttcn3-asterisk-ims-ue-test/Makefile b/ttcn3-asterisk-ims-ue-test/Makefile
new file mode 100644
index 0000000..8d0e10b
--- /dev/null
+++ b/ttcn3-asterisk-ims-ue-test/Makefile
@@ -0,0 +1 @@
+include ../make/Makefile
diff --git a/ttcn3-asterisk-ims-ue-test/asterisk/asterisk.conf b/ttcn3-asterisk-ims-ue-test/asterisk/asterisk.conf
new file mode 100644
index 0000000..a17d606
--- /dev/null
+++ b/ttcn3-asterisk-ims-ue-test/asterisk/asterisk.conf
@@ -0,0 +1,142 @@
+[directories](!)
+astcachedir => /var/cache/asterisk
+astetcdir => /etc/asterisk
+astmoddir => /usr/lib/asterisk/modules
+astvarlibdir => /var/lib/asterisk
+astdbdir => /var/lib/asterisk
+astkeydir => /var/lib/asterisk
+astdatadir => /var/lib/asterisk
+astagidir => /var/lib/asterisk/agi-bin
+astspooldir => /var/spool/asterisk
+astrundir => /var/run/asterisk
+astlogdir => /data
+astsbindir => /usr/sbin
+
+[options]
+;verbose = 3
+;debug = 3
+;trace = 0 ; Set the trace level.
+;refdebug = yes ; Enable reference count debug logging.
+;alwaysfork = yes ; Same as -F at startup.
+;nofork = yes ; Same as -f at startup.
+;quiet = yes ; Same as -q at startup.
+;timestamp = yes ; Same as -T at startup.
+;execincludes = yes ; Support #exec in config files.
+;console = yes ; Run as console (same as -c at startup).
+;highpriority = yes ; Run realtime priority (same as -p at
+ ; startup).
+;initcrypto = yes ; Initialize crypto keys (same as -i at
+ ; startup).
+;nocolor = yes ; Disable console colors.
+;dontwarn = yes ; Disable some warnings.
+;dumpcore = yes ; Dump core on crash (same as -g at startup).
+;languageprefix = yes ; Use the new sound prefix path syntax.
+;systemname = my_system_name ; Prefix uniqueid with a system name for
+ ; Global uniqueness issues.
+;autosystemname = yes ; Automatically set systemname to hostname,
+ ; uses 'localhost' on failure, or systemname if
+ ; set.
+;mindtmfduration = 80 ; Set minimum DTMF duration in ms (default 80 ms)
+ ; If we get shorter DTMF messages, these will be
+ ; changed to the minimum duration
+;maxcalls = 10 ; Maximum amount of calls allowed.
+;maxload = 0.9 ; Asterisk stops accepting new calls if the
+ ; load average exceed this limit.
+;maxfiles = 1000 ; Maximum amount of openfiles.
+;minmemfree = 1 ; In MBs, Asterisk stops accepting new calls if
+ ; the amount of free memory falls below this
+ ; watermark.
+;cache_media_frames = yes ; Cache media frames for performance
+ ; Disable this option to help track down media frame
+ ; mismanagement when using valgrind or MALLOC_DEBUG.
+ ; The cache gets in the way of determining if the
+ ; frame is used after being freed and who freed it.
+ ; NOTE: This option has no effect when Asterisk is
+ ; compiled with the LOW_MEMORY compile time option
+ ; enabled because the cache code does not exist.
+ ; Default yes
+;cache_record_files = yes ; Cache recorded sound files to another
+ ; directory during recording.
+;record_cache_dir = /tmp ; Specify cache directory (used in conjunction
+ ; with cache_record_files).
+;transmit_silence = yes ; Transmit silence while a channel is in a
+ ; waiting state, a recording only state, or
+ ; when DTMF is being generated. Note that the
+ ; silence internally is generated in raw signed
+ ; linear format. This means that it must be
+ ; transcoded into the native format of the
+ ; channel before it can be sent to the device.
+ ; It is for this reason that this is optional,
+ ; as it may result in requiring a temporary
+ ; codec translation path for a channel that may
+ ; not otherwise require one.
+;transcode_via_sln = yes ; Build transcode paths via SLINEAR, instead of
+ ; directly.
+;runuser = asterisk ; The user to run as.
+;rungroup = asterisk ; The group to run as.
+;lightbackground = yes ; If your terminal is set for a light-colored
+ ; background.
+;forceblackbackground = yes ; Force the background of the terminal to be
+ ; black, in order for terminal colors to show
+ ; up properly.
+;defaultlanguage = en ; Default language
+documentation_language = en_US ; Set the language you want documentation
+ ; displayed in. Value is in the same format as
+ ; locale names.
+;hideconnect = yes ; Hide messages displayed when a remote console
+ ; connects and disconnects.
+;lockconfdir = no ; Protect the directory containing the
+ ; configuration files (/etc/asterisk) with a
+ ; lock.
+;stdexten = gosub ; How to invoke the extensions.conf stdexten.
+ ; macro - Invoke the stdexten using a macro as
+ ; done by legacy Asterisk versions.
+ ; gosub - Invoke the stdexten using a gosub as
+ ; documented in extensions.conf.sample.
+ ; Default gosub.
+;live_dangerously = no ; Enable the execution of 'dangerous' dialplan
+ ; functions and configuration file access from
+ ; external sources (AMI, etc.) These functions
+ ; (such as SHELL) are considered dangerous
+ ; because they can allow privilege escalation.
+ ; Configuration files are considered dangerous
+ ; if they exist outside of the Asterisk
+ ; configuration directory.
+ ; Default no
+;entityid=00:11:22:33:44:55 ; Entity ID.
+ ; This is in the form of a MAC address.
+ ; It should be universally unique.
+ ; It must be unique between servers communicating
+ ; with a protocol that uses this value.
+ ; This is currently is used by DUNDi and
+ ; Exchanging Device and Mailbox State
+ ; using protocols: XMPP, Corosync and PJSIP.
+;rtp_use_dynamic = yes ; When set to "yes" RTP dynamic payload types
+ ; are assigned dynamically per RTP instance vs.
+ ; allowing Asterisk to globally initialize them
+ ; to pre-designated numbers (defaults to "yes").
+;rtp_pt_dynamic = 35 ; Normally the Dynamic RTP Payload Type numbers
+ ; are 96-127, which allow just 32 formats. The
+ ; starting point 35 enables the range 35-63 and
+ ; allows 29 additional formats. When you use
+ ; more than 32 formats in the dynamic range and
+ ; calls are not accepted by a remote
+ ; implementation, please report this and go
+ ; back to value 96.
+;hide_messaging_ami_events = no; This option, if enabled, will
+ ; suppress all of the Message/ast_msg_queue channel's
+ ; housekeeping AMI and ARI channel events. This can
+ ; reduce the load on the manager and ARI applications
+ ; when the Digium Phone Module for Asterisk is in use.
+;sounds_search_custom_dir = no; This option, if enabled, will
+ ; cause Asterisk to search for sounds files in
+ ; AST_DATA_DIR/sounds/custom before searching the
+ ; normal directories like AST_DATA_DIR/sounds/<lang>.
+
+; Changing the following lines may compromise your security.
+;[files]
+;astctlpermissions = 0660
+;astctlowner = root
+;astctlgroup = apache
+;astctl = asterisk.ctl
+
diff --git a/ttcn3-asterisk-ims-ue-test/asterisk/asterisk.sh b/ttcn3-asterisk-ims-ue-test/asterisk/asterisk.sh
new file mode 100755
index 0000000..24a20db
--- /dev/null
+++ b/ttcn3-asterisk-ims-ue-test/asterisk/asterisk.sh
@@ -0,0 +1,16 @@
+#!/bin/bash
+set +e
+set -x
+
+ASTERISK_CFG_PATH="/etc/asterisk"
+
+#rm -rf "${ASTERISK_CFG_PATH}"
+#mkdir -p "${ASTERISK_CFG_PATH}"
+
+#cp -r /etc/asterisk/* "${ASTERISK_CFG_PATH}/"
+cp /data/asterisk.conf "${ASTERISK_CFG_PATH}/"
+#sed -i "s#/etc/asterisk#${ASTERISK_CFG_PATH}#" "${ASTERISK_CFG_PATH}/asterisk.conf"
+cp /data/pjsip.conf "${ASTERISK_CFG_PATH}/"
+cat /data/extensions.conf >>"${ASTERISK_CFG_PATH}/extensions.conf"
+
+/usr/sbin/asterisk -C "${ASTERISK_CFG_PATH}/asterisk.conf" -f -g -vvvvv -ddddd
diff --git a/ttcn3-asterisk-ims-ue-test/asterisk/extensions.conf b/ttcn3-asterisk-ims-ue-test/asterisk/extensions.conf
new file mode 100644
index 0000000..3c78a2d
--- /dev/null
+++ b/ttcn3-asterisk-ims-ue-test/asterisk/extensions.conf
@@ -0,0 +1,37 @@
+[get-valid-endpoints]
+
+exten => s,1,Verbose(5, Entering get-valid-endpoints gosub.)
+ ; usage: no arguments, returns DIALGROUP object of registered and valid 050x endpoints
+ same => n,Set(ENDPOINT_1_STATE=${DEVICE_STATE(PJSIP/0501)})
+ same => n,Set(ENDPOINT_2_STATE=${DEVICE_STATE(PJSIP/0502)})
+ same => n,Set(ENDPOINT_3_STATE=${DEVICE_STATE(PJSIP/0503)})
+ same => n,Set(ENDPOINT_4_STATE=${DEVICE_STATE(PJSIP/0504)})
+ same => n,Set(ENDPOINT_1_EXPR=$[$["${ENDPOINT_1_STATE}" = "UNAVAILABLE"] | $["${ENDPOINT_1_STATE}" = "UNKNOWN"] | $["${ENDPOINT_1_STATE}" = "INVALID"]])
+ same => n,Set(ENDPOINT_2_EXPR=$[$["${ENDPOINT_2_STATE}" = "UNAVAILABLE"] | $["${ENDPOINT_2_STATE}" = "UNKNOWN"] | $["${ENDPOINT_2_STATE}" = "INVALID"]])
+ same => n,Set(ENDPOINT_3_EXPR=$[$["${ENDPOINT_3_STATE}" = "UNAVAILABLE"] | $["${ENDPOINT_3_STATE}" = "UNKNOWN"] | $["${ENDPOINT_3_STATE}" = "INVALID"]])
+ same => n,Set(ENDPOINT_4_EXPR=$[$["${ENDPOINT_4_STATE}" = "UNAVAILABLE"] | $["${ENDPOINT_4_STATE}" = "UNKNOWN"] | $["${ENDPOINT_4_STATE}" = "INVALID"]])
+ same => n,Set(DIALGROUP(CALL_VALID_LIST)=) ; clear list
+ same => n,ExecIf($[${ENDPOINT_1_EXPR} = 0]?Set(DIALGROUP(CALL_VALID_LIST,add)=PJSIP/0501))
+ same => n,ExecIf($[${ENDPOINT_2_EXPR} = 0]?Set(DIALGROUP(CALL_VALID_LIST,add)=PJSIP/0502))
+ same => n,ExecIf($[${ENDPOINT_3_EXPR} = 0]?Set(DIALGROUP(CALL_VALID_LIST,add)=PJSIP/0503))
+ same => n,ExecIf($[${ENDPOINT_4_EXPR} = 0]?Set(DIALGROUP(CALL_VALID_LIST,add)=PJSIP/0504))
+ same => n,Return(${DIALGROUP(CALL_VALID_LIST)})
+
+[from-phone]
+
+exten => 0500,1,Verbose(5,${EXTEN}: Call all registered pjsips from ${CALLERID(num)})
+ same => n,Gosub(get-valid-endpoints,s,1())
+ same => n,Set(DIALGROUP(CALL_EVERYONE_LIST)=${GOSUB_RETVAL})
+ same => n,Set(DIALGROUP(CALL_EVERYONE_LIST,del)=PJSIP/${CALLERID(num)}) ; remove the caller
+ same => n,Dial(${DIALGROUP(CALL_EVERYONE_LIST)})
+ same => n,Hangup(16)
+
+exten => _050X,1,Verbose(5,${EXTEN}: Call pjsip endpoint from ${CALLERID(num)})
+ same => n,Dial(PJSIP/${EXTEN})
+ same => n,Hangup(16)
+
+exten => _X,1,Verbose(5,${EXTEN}: Call external number from ${CALLERID(num)})
+ same => n,Dial(PJSIP/${EXTEN}@973171)
+ same => n,Hangup(16)
+
+
diff --git a/ttcn3-asterisk-ims-ue-test/asterisk/pjsip.conf b/ttcn3-asterisk-ims-ue-test/asterisk/pjsip.conf
new file mode 100644
index 0000000..39cf9bb
--- /dev/null
+++ b/ttcn3-asterisk-ims-ue-test/asterisk/pjsip.conf
@@ -0,0 +1,71 @@
+
+[transport-udp]
+type=transport
+protocol=udp
+bind=172.18.11.10:5094
+local_net=172.18.11.0/24
+;external_media_address=[obfuscated]
+;external_signaling_address=[obfuscated]
+
+
+[endpoint_phone_template]
+type=endpoint
+transport=transport-udp
+context=from-phone
+disallow=all
+allow=alaw
+allow=ulaw
+
+[auth_phone_template]
+type=auth
+auth_type=userpass
+
+[aor_phone_template]
+type=aor
+max_contacts=1
+remove_existing=yes
+
+[0501](endpoint_phone_template)
+auth=0501
+aors=0501
+callerid=0501
+
+[0501](auth_phone_template)
+username=0501
+password=secret
+
+[0501](aor_phone_template)
+
+[0502](endpoint_phone_template)
+auth=0502
+aors=0502
+callerid=0502
+
+[0502](auth_phone_template)
+username=0502
+password=secret
+
+[0502](aor_phone_template)
+
+[0503](endpoint_phone_template)
+auth=0503
+aors=0503
+callerid=0503
+
+[0503](auth_phone_template)
+username=0503
+password=secret
+
+[0503](aor_phone_template)
+
+[0504](endpoint_phone_template)
+auth=0504
+aors=0504
+callerid=0504
+
+[0504](auth_phone_template)
+username=0504
+password=secret
+
+[0504](aor_phone_template)
+
diff --git a/ttcn3-asterisk-ims-ue-test/jenkins.sh b/ttcn3-asterisk-ims-ue-test/jenkins.sh
new file mode 100755
index 0000000..7e5361d
--- /dev/null
+++ b/ttcn3-asterisk-ims-ue-test/jenkins.sh
@@ -0,0 +1,44 @@
+#!/bin/sh
+
+. ../jenkins-common.sh
+IMAGE_SUFFIX="${IMAGE_SUFFIX:-master}"
+docker_images_require \
+ "asterisk-$IMAGE_SUFFIX" \
+ "ttcn3-asterisk-ims-ue-test"
+
+set_clean_up_trap
+set -e
+
+mkdir $VOL_BASE_DIR/asterisk-ims-ue-tester
+mkdir $VOL_BASE_DIR/asterisk-ims-ue-tester/unix
+cp Asterisk_Tests.cfg $VOL_BASE_DIR/asterisk-ims-ue-tester/
+write_mp_osmo_repo "$VOL_BASE_DIR/asterisk-ims-ue-tester/Asterisk_Tests.cfg"
+
+mkdir $VOL_BASE_DIR/asterisk
+cp asterisk/* $VOL_BASE_DIR/asterisk/
+
+network_create
+network_replace_subnet_in_configs
+
+echo Starting container with Asterisk
+docker run --rm \
+ $(docker_network_params $SUBNET 10) \
+ --ulimit core=-1 \
+ -v $VOL_BASE_DIR/asterisk:/data \
+ --name ${BUILD_TAG}-asterisk -d \
+ $DOCKER_ARGS \
+ $REPO_USER/asterisk-$IMAGE_SUFFIX \
+ /bin/sh -c "/data/asterisk.sh >/data/asterisk.console.log 2>&1"
+
+# Leave some time for asterisk to start:
+sleep 3
+
+echo Starting container with Asterisk testsuite
+docker run --rm \
+ $(docker_network_params $SUBNET 103) \
+ --ulimit core=-1 \
+ -e "TTCN3_PCAP_PATH=/data" \
+ -v $VOL_BASE_DIR/asterisk-ims-ue-tester:/data \
+ --name ${BUILD_TAG}-ttcn3-asterisk-ims-ue-test \
+ $DOCKER_ARGS \
+ $REPO_USER/ttcn3-asterisk-ims-ue-test
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Gerrit-Project: docker-playground
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Jenkins Builder has posted comments on this change. ( https://gerrit.osmocom.org/c/osmo-ttcn3-hacks/+/36490?usp=email )
Change subject: Introduce Asterisk_Tests testsuite
......................................................................
Patch Set 1:
(1 comment)
File library/SIP_Templates.ttcn:
Robot Comment from checkpatch (run ID jenkins-gerrit-lint-15493):
https://gerrit.osmocom.org/c/osmo-ttcn3-hacks/+/36490/comment/9a182294_fe18…
PS1, Line 528: /* Expect during first REGISTER when authorization is requried: */
'requried' may be misspelled - perhaps 'required'?
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