Multiphone and outgoing calls without register=true, is it possible?

Question

Telephony

Good evening.
Only outgoing calls are needed, it works only if the parameter is set in the gateway settings
<param name="register" value="true"/>,
if set to false, then I get Originate Resulted in Error Cause: 55 [UNKNOWN]

There is a proxy and outbound-proxy parameter in the gateway settings.
Is it possible to make multiphone ring with register=false?
Here is the log with siptrace enabled

INVITE sip:[email protected];transport=tcp SIP/2.0
   Via: SIP/2.0/TCP YYYYYYYYY:5080;rport;branch=z9hG4bKZ0943DvBc1a1a
   Max-Forwards: 69
   From: "NNNNNNNN" <sip:[email protected]>;tag=Zc04tBKj1rcaj
   To: <sip:[email protected]>
   Call-ID: 3d51c04d-07e7-1234-a483-525400107afc
   CSeq: 83551966 INVITE
   Contact: <sip:[email protected]:5080;tport=tcp;transport=tcp;gw=861>
   User-Agent: FreeSWITCH-mod_sofia/1.4.20+git~20150724T013613Z~bf08a378cb~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
   Supported: timer, path, replaces
   Allow-Events: talk, hold, conference, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 346
   X-FS-Support: update_display,send_info
   Remote-Party-ID: "NNNNNNNN" <sip:[email protected]>;party=calling;screen=yes;privacy=off

   v=0
   o=FreeSWITCH 1447749897 1447749898 IN IP4 XXXXXXXXXXXXXX
   s=FreeSWITCH
   c=IN IP4 XXXXXXXXXXXXXXXXX
   t=0 0
   m=audio 26164 RTP/AVP 111 9 0 8 101 13
   a=rtpmap:111 opus/48000/2
   a=fmtp:111 minptime=10; useinbandfec=1
   a=rtpmap:9 G722/8000
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   ------------------------------------------------------------------------
recv 317 bytes from tcp/[XXXXXXXXXXXX]:5060 at 19:01:01.246924:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/TCP XXXXXXXXXXXXXX:5080;received=XXXXXXXXXXXXX;branch=z9hG4bKZ0943DvBc1a1a;rport=PPPPP
   From: "NNNNNNNN" <sip:[email protected]>;tag=Zc04tBKj1rcaj
   To: <sip:[email protected]>
   Call-ID: 3d51c04d-07e7-1234-a483-525400107afc
   CSeq: 83551966 INVITE
   Content-Length: 0

   ------------------------------------------------------------------------
recv 395 bytes from tcp/[193.201.229.35]:5060 at 19:01:01.250707:
   ------------------------------------------------------------------------
   SIP/2.0 403 Forbidden
   Via: SIP/2.0/TCP 194.177.22.196:5080;received=1XXXXXXXXXXXXX;branch=z9hG4bKZ0943DvBc1a1a;rport=PPPPP
   From: "NNNNNNNN" <sip:[email protected]>;tag=Zc04tBKj1rcaj
   To: <sip:[email protected]>;tag=aprqngfrt-spa5iui0qnptd
   Call-ID: 3d51c04d-07e7-1234-a483-525400107afc
   CSeq: 83551966 INVITE
   Reason: Q.850;cause=55;text="Call Terminated"
   Content-Length: 0

   ------------------------------------------------------------------------
send 369 bytes to tcp/[XXXXXXXXXXXXX]:5060 at 19:01:01.250917:
   ------------------------------------------------------------------------
   ACK sip:[email protected];transport=tcp SIP/2.0
   Via: SIP/2.0/TCP XXXXXXXXXXXXXX:5080;rport;branch=z9hG4bKZ0943DvBc1a1a
   Max-Forwards: 69
   From: "NNNNNNNNNNNN" <sip:[email protected]>;tag=Zc04tBKj1rcaj
   To: <sip:[email protected]>;tag=aprqngfrt-spa5iui0qnptd
   Call-ID: 3d51c04d-07e7-1234-a483-525400107afc
   CSeq: 83551966 ACK
   Content-Length: 0

Answer 1

Perhaps the multiphone does not pass without prior registration.
Each operator has its own cockroaches ...
Take more white IPs, create a separate sofia profile for each and "smear" gateways across profiles.
If the addresses are tense, try hanging profiles on different ports (I haven’t tried it with some of them).