Changes

asterisk: pjsip.conf: Use TCP when forwarding INVITE to IMS core

Change-Id: I67bc01e4d53d47eb805e81736378f292b2834fd2

asterisk: pjsip.conf: Disable remote bridging between local SIP and IMS

While implementing a first ttcn3 test validating the MO call scenario
(SIP-UA -> Asterisk -> IMS-CORE) [1] I was running into the scenario
where, after the first SIP INVITE + 200 OK + ACK, Asterisk was sending
a RE-INVITE to both parties to attempt to remotely bridge them (RTP
traffic flowing directly between them without passing through Asterisk).

This happened in part because I'm so far configure A-LAW on both sides
so asterisk figures out it can do so. I still need to change IMS-core
to EVS only.

In any case, regardless of the codecs used, my understanding is that
asterisk should never attempt remote bridging when using the "volte_ims"
endpoint, since that network segment is separate from the local network
where the local SIP UAs are located.

The "direct_media=no" option just addresses the issue; Asterisk no longer
tries to re-invite after the call is established

Related: SYS#6782
Change-Id: I4edea96151b31f02bf292b43b757922389375429

asterisk-master: Install AMR enc/dec dependencies

Change-Id: Ic5de8ae6ac8ade375b9692427e5865bee49dc01f

asterisk: pjsip.conf: Fix msisdn not used in From field during IMS INVITE

Related: SYS#6950
Change-Id: I972e75b36457c532bd7528ffb5a7b80544abba4a

Introduce dnsmasq docker container

This container will be used by the ttcn3-asterisk testsuite setup, so
that asterisk can resolve NAPTR nameservers for the IMS 3gpp DNS.

Change-Id: Ic1490334375c2a9b11b923a6757e844faf524634

ttcn3-asterisk: Use dnsmasq to let asterisk resolve IMS DNS

Depends: osmo-ttcn3-hacks.git Change-Id I23afdc8b70132dbd947c280f6e5aadbc84849f01
Change-Id: I2da10c80a81ef5603082e4ee1445a39745c0c24f

asterisk: pjsip.conf: Configure Contact field

Change-Id: I0b52407f9e7d1ca7f186c77db8f9e1a0ca272ff7

asterisk: pjsip.conf: identify incoming register started by local SIP UAs

This config section is needed so that Asterisk can figure out the endpoint
when receiving a "REGISTER sip:172.18.248.10". Without it, it will answer with
a 404 Not Found.

Change-Id: I39568c3cebb89fc5a62a8144f448b50c48d79f10

asterisk: pjsip.conf: Update IMS port asterisk config

The config names changes recently during development.
Adapt to the new range style fields.

Change-Id: I4aceb3b62ce7f446abd02b1a53993a3baecf2f20

asterisk: pjsip.conf: Set default p_access_network_info

The value is overwritten if set through AMI PJSIPAccessNetworkInfo
action.

Change-Id: I2bb0b89ec36137bbae613962ff5f5aad56721f34