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Sprocket to call sprocket: 403 Forbidden

I have 2 servers with Asterisks on them: 192.168.241.98 and 192.168.243.112.

There is a valid registration for the first:

register => wagateway: qwerty@192.168.243.112 :5060

CLI Output:

 CLI> sip show registry Host dnsmgr Username Refresh State Reg.Time 192.168.243.112:5060 N wagateway 105 Registered Wed, 26 Jun 2013 16:42:42

And peers at 243.112 are just beautiful:

 CLI> sip show peers Name/username Host Dyn Forcerport ACL Port Status Description wacaller/wacaller 192.168.242.235 D a 5062 OK (13 ms) wagateway/s 192.168.241.98 D a 5060 OK (1 ms)

extensions.conf on 243.112:

 [watest] exten => 123123123,1,NoOp() exten => 123123123,n,Dial(SIP/wagateway) exten => 123123123,n,Hangup()

sip.conf on 243.112:

 [wacaller] type=friend secret=qwerty host=dynamic context=watest qualify=yes allow=ulaw allow=alaw [wagateway] type=friend secret=qwerty fromuser=wagateway host=dynamic context=watest qualify=yes allow=ulaw allow=alaw

Now I'm trying to call 123123123 from the wacaller Grandstream phone.

243.112 CLI says:

 [Jun 27 09:27:54] WARNING[20447][C-0000000b]: chan_sip.c:23213 handle_response_invite: Received response: "Forbidden" from '"WACaller" <sip: wagateway@192.168.243.112 >;tag=as30b27eae'

Debugging sip on 243.112:

 <--- SIP read from UDP:192.168.242.235:5062 ---> INVITE sip: 123123123@192.168.243.112 SIP/2.0 Via: SIP/2.0/UDP 192.168.242.235:5062;branch=z9hG4bK97733114;rport From: "WACaller" <sip: wacaller@192.168.243.112 >;tag=1014197566 To: <sip: 123123123@192.168.243.112 > Call-ID: 298833112-5062-25@BJC.BGI.CEC.CDF CSeq: 240 INVITE Contact: "WACaller" <sip: wacaller@192.168.242.235 :5062> Max-Forwards: 70 User-Agent: Grandstream GXP1400 1.0.4.13 Privacy: none P-Preferred-Identity: "WACaller" <sip: wacaller@192.168.243.112 > Supported: replaces, path, timer Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: application/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 412 v=0 o=wacaller 8000 8000 IN IP4 192.168.242.235 s=SIP Call c=IN IP4 192.168.242.235 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 9 97 2 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (16 headers 19 lines) --- Sending to 192.168.242.235:5062 (no NAT) Sending to 192.168.242.235:5062 (no NAT) Using INVITE request as basis request - 298833112-5062-25@BJC.BGI.CEC.CDF Found peer 'wacaller' for 'wacaller' from 192.168.242.235:5062 <--- Reliably Transmitting (no NAT) to 192.168.242.235:5062 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.242.235:5062;branch=z9hG4bK97733114;received=192.168.242.235;rport=5062 From: "WACaller" <sip: wacaller@192.168.243.112 >;tag=1014197566 To: <sip: 123123123@192.168.243.112 >;tag=as5a3de236 Call-ID: 298833112-5062-25@BJC.BGI.CEC.CDF CSeq: 240 INVITE Server: Asterisk PBX SVN-trunk-r385782 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4f84bef0" Content-Length: 0 <------------> Scheduling destruction of SIP dialog ' 298833112-5062-25@BJC.BGI.CEC.CDF ' in 6400 ms (Method: INVITE) <--- SIP read from UDP:192.168.242.235:5062 ---> ACK sip: 123123123@192.168.243.112 SIP/2.0 Via: SIP/2.0/UDP 192.168.242.235:5062;branch=z9hG4bK97733114;rport From: "WACaller" <sip: wacaller@192.168.243.112 >;tag=1014197566 To: <sip: 123123123@192.168.243.112 >;tag=as5a3de236 Call-ID: 298833112-5062-25@BJC.BGI.CEC.CDF CSeq: 240 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from UDP:192.168.242.235:5062 ---> INVITE sip: 123123123@192.168.243.112 SIP/2.0 Via: SIP/2.0/UDP 192.168.242.235:5062;branch=z9hG4bK1881861609;rport From: "WACaller" <sip: wacaller@192.168.243.112 >;tag=1014197566 To: <sip: 123123123@192.168.243.112 > Call-ID: 298833112-5062-25@BJC.BGI.CEC.CDF CSeq: 241 INVITE Contact: "WACaller" <sip: wacaller@192.168.242.235 :5062> Authorization: Digest username="wacaller", realm="asterisk", nonce="4f84bef0", uri="sip: 123123123@192.168.243.112 ", response="53cdb5b8c1822c80870faab15a6dc6d2", algorithm=MD5 Max-Forwards: 70 User-Agent: Grandstream GXP1400 1.0.4.13 Privacy: none P-Preferred-Identity: "WACaller" <sip: wacaller@192.168.243.112 > Supported: replaces, path, timer Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: application/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 412 v=0 o=wacaller 8000 8000 IN IP4 192.168.242.235 s=SIP Call c=IN IP4 192.168.242.235 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 9 97 2 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (17 headers 19 lines) --- Sending to 192.168.242.235:5062 (no NAT) Using INVITE request as basis request - 298833112-5062-25@BJC.BGI.CEC.CDF Found peer 'wacaller' for 'wacaller' from 192.168.242.235:5062 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 4 Found RTP audio format 18 Found RTP audio format 9 Found RTP audio format 97 Found RTP audio format 2 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G723 for ID 4 Found audio description format G729 for ID 18 Found audio description format G722 for ID 9 Found audio description format iLBC for ID 97 Found audio description format G726-32 for ID 2 Found audio description format telephone-event for ID 101 Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(g723|ulaw|alaw|g726|g729|ilbc|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.242.235:5004 Looking for 123123123 in watest (domain 192.168.243.112) list_route: route/path hop: <sip: wacaller@192.168.242.235 :5062> <--- Transmitting (no NAT) to 192.168.242.235:5062 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.242.235:5062;branch=z9hG4bK1881861609;received=192.168.242.235;rport=5062 From: "WACaller" <sip: wacaller@192.168.243.112 >;tag=1014197566 To: <sip: 123123123@192.168.243.112 > Call-ID: 298833112-5062-25@BJC.BGI.CEC.CDF CSeq: 241 INVITE Server: Asterisk PBX SVN-trunk-r385782 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip: 123123123@192.168.243.112 :5060> Content-Length: 0 <------------> Audio is at 17372 Adding codec 100003 (ulaw) to SDP Adding codec 100004 (alaw) to SDP Adding codec 100002 (gsm) to SDP Adding codec 100017 (testlaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 192.168.241.98:5060: INVITE sip: s@192.168.241.98 :5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK67b16b32;rport Max-Forwards: 70 From: "WACaller" <sip: wagateway@192.168.243.112 >;tag=as3f5f372a To: <sip: s@192.168.241.98 :5060> Contact: <sip: wagateway@192.168.243.112 :5060> Call-ID: 758899861bee35980dadd87912ef805a@192.168.243.112 :5060 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-trunk-r385782 Date: Wed, 26 Jun 2013 08:31:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 326 v=0 o=root 2059284449 2059284449 IN IP4 192.168.243.112 s=Asterisk PBX SVN-trunk-r385782 c=IN IP4 192.168.243.112 t=0 0 m=audio 17372 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from UDP:192.168.241.98:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK67b16b32;received=192.168.243.112;rport=5060 From: "WACaller" <sip: wagateway@192.168.243.112 >;tag=as3f5f372a To: <sip: s@192.168.241.98 :5060>;tag=as22eeeac0 Call-ID: 758899861bee35980dadd87912ef805a@192.168.243.112 :5060 CSeq: 102 INVITE Server: Asterisk PBX 1.8.12.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="603b4bbf" Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Transmitting (NAT) to 192.168.241.98:5060: ACK sip: s@192.168.241.98 :5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK67b16b32;rport Max-Forwards: 70 From: "WACaller" <sip: wagateway@192.168.243.112 >;tag=as3f5f372a To: <sip: s@192.168.241.98 :5060>;tag=as22eeeac0 Contact: <sip: wagateway@192.168.243.112 :5060> Call-ID: 758899861bee35980dadd87912ef805a@192.168.243.112 :5060 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-trunk-r385782 Content-Length: 0 --- Audio is at 17372 Adding codec 100003 (ulaw) to SDP Adding codec 100004 (alaw) to SDP Adding codec 100002 (gsm) to SDP Adding codec 100017 (testlaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 192.168.241.98:5060: INVITE sip: s@192.168.241.98 :5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK6cd34725;rport Max-Forwards: 70 From: "WACaller" <sip: wagateway@192.168.243.112 >;tag=as3f5f372a To: <sip: s@192.168.241.98 :5060> Contact: <sip: wagateway@192.168.243.112 :5060> Call-ID: 758899861bee35980dadd87912ef805a@192.168.243.112 :5060 CSeq: 103 INVITE User-Agent: Asterisk PBX SVN-trunk-r385782 Authorization: Digest username="s", realm="asterisk", algorithm=MD5, uri="sip: s@192.168.241.98 :5060", nonce="603b4bbf", response="059cae207fb81fb76ea9061f71258895" Date: Wed, 26 Jun 2013 08:31:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 326 v=0 o=root 2059284449 2059284450 IN IP4 192.168.243.112 s=Asterisk PBX SVN-trunk-r385782 c=IN IP4 192.168.243.112 t=0 0 m=audio 17372 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from UDP:192.168.241.98:5060 ---> SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK6cd34725;received=192.168.243.112;rport=5060 From: "WACaller" <sip: wagateway@192.168.243.112 >;tag=as3f5f372a To: <sip: s@192.168.241.98 :5060>;tag=as22eeeac0 Call-ID: 758899861bee35980dadd87912ef805a@192.168.243.112 :5060 CSeq: 103 INVITE Server: Asterisk PBX 1.8.12.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Transmitting (NAT) to 192.168.241.98:5060: ACK sip: s@192.168.241.98 :5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK6cd34725;rport Max-Forwards: 70 From: "WACaller" <sip: wagateway@192.168.243.112 >;tag=as3f5f372a To: <sip: s@192.168.241.98 :5060>;tag=as22eeeac0 Contact: <sip: wagateway@192.168.243.112 :5060> Call-ID: 758899861bee35980dadd87912ef805a@192.168.243.112 :5060 CSeq: 103 ACK User-Agent: Asterisk PBX SVN-trunk-r385782 Content-Length: 0 --- [Jun 26 16:31:48] WARNING[20447][C-0000000a]: chan_sip.c:23213 handle_response_invite: Received response: "Forbidden" from '"WACaller" <sip: wagateway@192.168.243.112 >;tag=as3f5f372a' Scheduling destruction of SIP dialog ' 758899861bee35980dadd87912ef805a@192.168.243.112 :5060' in 6400 ms (Method: INVITE) Scheduling destruction of SIP dialog ' 298833112-5062-25@BJC.BGI.CEC.CDF ' in 6400 ms (Method: INVITE)

Debugging debugging on the target server:

 <--- SIP read from UDP:192.168.243.112:5060 ---> INVITE sip: s@192.168.241.98 :5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK3159e4b1;rport Max-Forwards: 70 From: "WACaller" <sip: wagateway@192.168.243.112 >;tag=as30b27eae To: <sip: s@192.168.241.98 :5060> Contact: <sip: wagateway@192.168.243.112 :5060> Call-ID: 5dc37059030845ca3d974c513993876d@192.168.243.112 :5060 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-trunk-r385782 Date: Thu, 27 Jun 2013 01:27:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 326 v=0 o=root 1301894386 1301894386 IN IP4 192.168.243.112 s=Asterisk PBX SVN-trunk-r385782 c=IN IP4 192.168.243.112 t=0 0 m=audio 15838 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (14 headers 14 lines) --- Sending to 192.168.243.112:5060 (NAT) Using INVITE request as basis request - 5dc37059030845ca3d974c513993876d@192.168.243.112 :5060 Found peer 'wagateway' for 'wagateway' from 192.168.243.112:5060 <--- Reliably Transmitting (no NAT) to 192.168.243.112:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK3159e4b1;received=192.168.243.112;rport=5060 From: "WACaller" <sip: wagateway@192.168.243.112 >;tag=as30b27eae To: <sip: s@192.168.241.98 :5060>;tag=as671c0824 Call-ID: 5dc37059030845ca3d974c513993876d@192.168.243.112 :5060 CSeq: 102 INVITE Server: Asterisk PBX 1.8.12.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0b63a660" Content-Length: 0 <------------> Scheduling destruction of SIP dialog ' 5dc37059030845ca3d974c513993876d@192.168.243.112 :5060' in 32000 ms (Method: INVITE) <--- SIP read from UDP:192.168.243.112:5060 ---> ACK sip: s@192.168.241.98 :5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK3159e4b1;rport Max-Forwards: 70 From: "WACaller" <sip: wagateway@192.168.243.112 >;tag=as30b27eae To: <sip: s@192.168.241.98 :5060>;tag=as671c0824 Contact: <sip: wagateway@192.168.243.112 :5060> Call-ID: 5dc37059030845ca3d974c513993876d@192.168.243.112 :5060 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-trunk-r385782 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from UDP:192.168.243.112:5060 ---> INVITE sip: s@192.168.241.98 :5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK40e56655;rport Max-Forwards: 70 From: "WACaller" <sip: wagateway@192.168.243.112 >;tag=as30b27eae To: <sip: s@192.168.241.98 :5060> Contact: <sip: wagateway@192.168.243.112 :5060> Call-ID: 5dc37059030845ca3d974c513993876d@192.168.243.112 :5060 CSeq: 103 INVITE User-Agent: Asterisk PBX SVN-trunk-r385782 Authorization: Digest username="s", realm="asterisk", algorithm=MD5, uri="sip: s@192.168.241.98 :5060", nonce="0b63a660", response="537f37fe2fb8d0fd40733cb190ea70c8" Date: Thu, 27 Jun 2013 01:27:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 326 v=0 o=root 1301894386 1301894387 IN IP4 192.168.243.112 s=Asterisk PBX SVN-trunk-r385782 c=IN IP4 192.168.243.112 t=0 0 m=audio 15838 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (15 headers 14 lines) --- Sending to 192.168.243.112:5060 (no NAT) Using INVITE request as basis request - 5dc37059030845ca3d974c513993876d@192.168.243.112 :5060 Found peer 'wagateway' for 'wagateway' from 192.168.243.112:5060 <--- Reliably Transmitting (no NAT) to 192.168.243.112:5060 ---> SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK40e56655;received=192.168.243.112;rport=5060 From: "WACaller" <sip: wagateway@192.168.243.112 >;tag=as30b27eae To: <sip: s@192.168.241.98 :5060>;tag=as671c0824 Call-ID: 5dc37059030845ca3d974c513993876d@192.168.243.112 :5060 CSeq: 103 INVITE Server: Asterisk PBX 1.8.12.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> Scheduling destruction of SIP dialog ' 5dc37059030845ca3d974c513993876d@192.168.243.112 :5060' in 32000 ms (Method: INVITE) <--- SIP read from UDP:192.168.243.112:5060 ---> ACK sip: s@192.168.241.98 :5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK40e56655;rport Max-Forwards: 70 From: "WACaller" <sip: wagateway@192.168.243.112 >;tag=as30b27eae To: <sip: s@192.168.241.98 :5060>;tag=as671c0824 Contact: <sip: wagateway@192.168.243.112 :5060> Call-ID: 5dc37059030845ca3d974c513993876d@192.168.243.112 :5060 CSeq: 103 ACK User-Agent: Asterisk PBX SVN-trunk-r385782 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- dev-ast*CLI> sip set debug off SIP Debugging Disabled

Any help?

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3 answers

Another problem you are facing is a loop, you send a call to your gateway, and when a call arrives at your gateway, you send it back to the gateway, which is why you get barred when you dial SIP / wagateway (on wagateway) you do not have extensions, your way of calling is the client ---> gateway ---> gateway, try changing the extension to look at something like below

 [watest] exten => 123123123,1,NoOp(Call comming from ${CALLERID(all)}) exten => 123123123,n,Answer() exten => 123123123,n,PlayBack(tt-monkeys) exten => 123123123,n,Hangup()
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Have you tried with:

 exten => 123123123,n,Dial(SIP/wagateway/${EXTEN})

INVITATION sip: s@192.168.241.98 : 5060

You send a call to s in the [watest] context (which is the default if you do not specify an extension), but s does not exist, only 123123123.

edit1: Ok, don't add the [wacaller] add modification:

 type=peer ;instead of friend insecure=invite,port nat=yes

let me know if this worked, thanks.

edit2: try deleting / commenting

 ;fromuser=wagateway

Check the Grandstream forum , this is most likely a problem with the phone.

Edit3: The 99% problem is that you are registering on one server (192.168.243.112), and invitations are sent to wagateway / s (192.168.241.98) on another server or IP address The registry string is not the same as the invitation, and there you will receive a prohibited message. this should help: Unsure = invite, port
on the gateway for the trunk if you want to save this network setting.

Hello

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Comparison with one of my Asterisk-Asterisk SIP buses ...

It looks like I'm using the defaultuser= parameter in my sip.conf and not fromuser=

From the original, sip.conf , which comes with make samples - defaultuser , is described as an "Authentication user for outgoing proxies." Although this is not a proxy server in this case, I believe that this is the parameter that will be used when executing this SIP request.

However, you can also use the iax protocol if you have the convenience of configuring a trunk between two asterisk servers. These are the standards for Inter-Asterisk eXchange, and I find it easier to use. And it’s especially simpler, it seems that the same problems as SIP do not suffer from NAT traversal.

Here is an example of a SIP trunk between my two sprockets.

Box A, New York:

 register => newyork: VERYSECRET@192.168.1.21 [tokyo] nat=yes type=friend context=insidecaller host=192.168.1.21 defaultuser=newyork secret=VERYSECRET disallow=all allow=ulaw

And on box B, Tokyo:

 [newyork] directmedia=no type=friend secret=VERYSECRET context=outsidecaller host=dynamic disallow=all allow=ulaw

Note how the defaultuser on Box A configuration for communicating with tokyo (aka Box B) matches the device name [newyork] in field B sip.conf

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Source: https://habr.com/ru/post/1488402/

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