проблемма с исходящим маршрутом

В исходящем маршруте указано:

pattern: XXXX. trunk: pstn 

транк к spa3102->pstn

[pstn] host=10.0.0.5 username=100 secret=******** type=friend 

при таком конфиге * говорит что все линии заняты...

если же набрать сначала экстеншен spa3102(100), потом донабрать городской номер то все нормально проходит...

лог при звонке через маршруты

мой ext - 123, звоню на 123456

 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [123456@from-internal:1] Macro("SIP/123-00000146", "user-callerid,LIMIT,") in new stack -- Executing [s@macro-user-callerid:1] Set("SIP/123-00000146", "AMPUSER=123") in new stack -- Executing [s@macro-user-callerid:2] GotoIf("SIP/123-00000146", "0?report") in new stack -- Executing [s@macro-user-callerid:3] ExecIf("SIP/123-00000146", "1?Set(REALCALLERIDNUM=123)") in new stack -- Executing [s@macro-user-callerid:4] Set("SIP/123-00000146", "AMPUSER=123") in new stack -- Executing [s@macro-user-callerid:5] Set("SIP/123-00000146", "AMPUSERCIDNAME=123") in new stack -- Executing [s@macro-user-callerid:6] GotoIf("SIP/123-00000146", "0?report") in new stack -- Executing [s@macro-user-callerid:7] Set("SIP/123-00000146", "AMPUSERCID=123") in new stack -- Executing [s@macro-user-callerid:8] Set("SIP/123-00000146", "CALLERID(all)="123" <123>") in new stack -- Executing [s@macro-user-callerid:9] GotoIf("SIP/123-00000146", "0?limit") in new stack -- Executing [s@macro-user-callerid:10] ExecIf("SIP/123-00000146", "1?Set(GROUP(concurrency_limit)=123)") in new stack -- Executing [s@macro-user-callerid:11] ExecIf("SIP/123-00000146", "0?Set(CHANNEL(language)=)") in new stack -- Executing [s@macro-user-callerid:12] GosubIf("SIP/123-00000146", "0?sub-ccss,s,1(from-internal,123456)") in new stack -- Executing [s@macro-user-callerid:13] GotoIf("SIP/123-00000146", "1?continue") in new stack -- Goto (macro-user-callerid,s,26) -- Executing [s@macro-user-callerid:26] Set("SIP/123-00000146", "CALLERID(number)=123") in new stack -- Executing [s@macro-user-callerid:27] Set("SIP/123-00000146", "CALLERID(name)=123") in new stack -- Executing [s@macro-user-callerid:28] Set("SIP/123-00000146", "CHANNEL(language)=en") in new stack -- Executing [123456@from-internal:2] Set("SIP/123-00000146", "MOHCLASS=default") in new stack -- Executing [123456@from-internal:3] Set("SIP/123-00000146", "_NODEST=") in new stack -- Executing [123456@from-internal:4] Gosub("SIP/123-00000146", "sub-record-check,s,1(out,123456,)") in new stack -- Executing [s@sub-record-check:1] GotoIf("SIP/123-00000146", "1?check") in new stack -- Goto (sub-record-check,s,6) -- Executing [s@sub-record-check:6] Set("SIP/123-00000146", "__MON_FMT=wav") in new stack -- Executing [s@sub-record-check:7] GotoIf("SIP/123-00000146", "1?next") in new stack -- Goto (sub-record-check,s,10) -- Executing [s@sub-record-check:10] ExecIf("SIP/123-00000146", "0?Return()") in new stack -- Executing [s@sub-record-check:11] GotoIf("SIP/123-00000146", "0?out,1") in new stack -- Executing [s@sub-record-check:12] Set("SIP/123-00000146", "__REC_STATUS=INITIALIZED") in new stack -- Executing [s@sub-record-check:13] ExecIf("SIP/123-00000146", "0?Set(__REC_POLICY_MODE=)") in new stack -- Executing [s@sub-record-check:14] Set("SIP/123-00000146", "NOW=1346683778") in new stack -- Executing [s@sub-record-check:15] Set("SIP/123-00000146", "__DAY=03") in new stack -- Executing [s@sub-record-check:16] Set("SIP/123-00000146", "__MONTH=09") in new stack -- Executing [s@sub-record-check:17] Set("SIP/123-00000146", "__YEAR=2012") in new stack -- Executing [s@sub-record-check:18] Set("SIP/123-00000146", "__TIMESTR=20120903-184938") in new stack -- Executing [s@sub-record-check:19] Set("SIP/123-00000146", "__FROMEXTEN=123") in new stack -- Executing [s@sub-record-check:20] Set("SIP/123-00000146", "__CALLFILENAME=out-123456-123-20120903-184938-1346683778.326") in new stack -- Executing [s@sub-record-check:21] Goto("SIP/123-00000146", "out,1") in new stack -- Goto (sub-record-check,out,1) -- Executing [out@sub-record-check:1] ExecIf("SIP/123-00000146", "1?Set(__REC_POLICY_MODE=dontcare)") in new stack -- Executing [out@sub-record-check:2] GosubIf("SIP/123-00000146", "0?record,1(exten,123456,123)") in new stack -- Executing [out@sub-record-check:3] Return("SIP/123-00000146", "") in new stack -- Executing [123456@from-internal:5] Macro("SIP/123-00000146", "dialout-trunk,2,123456,") in new stack -- Executing [s@macro-dialout-trunk:1] Set("SIP/123-00000146", "DIAL_TRUNK=2") in new stack -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/123-00000146", "0?sub-pincheck,s,1()") in new stack -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/123-00000146", "0?disabletrunk,1") in new stack -- Executing [s@macro-dialout-trunk:4] Set("SIP/123-00000146", "DIAL_NUMBER=123456") in new stack -- Executing [s@macro-dialout-trunk:5] Set("SIP/123-00000146", "DIAL_TRUNK_OPTIONS=tr") in new stack -- Executing [s@macro-dialout-trunk:6] Set("SIP/123-00000146", "OUTBOUND_GROUP=OUT_2") in new stack -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/123-00000146", "1?nomax") in new stack -- Goto (macro-dialout-trunk,s,9) -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/123-00000146", "0?skipoutcid") in new stack -- Executing [s@macro-dialout-trunk:10] Set("SIP/123-00000146", "DIAL_TRUNK_OPTIONS=") in new stack -- Executing [s@macro-dialout-trunk:11] Macro("SIP/123-00000146", "outbound-callerid,2") in new stack -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/123-00000146", "0?Set(CALLERPRES()=)") in new stack -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/123-00000146", "0?Set(REALCALLERIDNUM=123)") in new stack -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/123-00000146", "1?normcid") in new stack -- Goto (macro-outbound-callerid,s,6) -- Executing [s@macro-outbound-callerid:6] Set("SIP/123-00000146", "USEROUTCID=") in new stack -- Executing [s@macro-outbound-callerid:7] Set("SIP/123-00000146", "EMERGENCYCID=") in new stack -- Executing [s@macro-outbound-callerid:8] Set("SIP/123-00000146", "TRUNKOUTCID=") in new stack -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/123-00000146", "1?trunkcid") in new stack -- Goto (macro-outbound-callerid,s,12) -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/123-00000146", "0?Set(CALLERID(all)=)") in new stack -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/123-00000146", "0?Set(CALLERID(all)=)") in new stack -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/123-00000146", "0?Set(CALLERID(all)=)") in new stack -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/123-00000146", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/123-00000146", "0?sub-flp-2,s,1()") in new stack -- Executing [s@macro-dialout-trunk:13] Set("SIP/123-00000146", "OUTNUM=123456") in new stack -- Executing [s@macro-dialout-trunk:14] Set("SIP/123-00000146", "custom=SIP/pstn") in new stack -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/123-00000146", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack -- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/123-00000146", "0?Set(DIAL_TRUNK_OPTIONS=M(confirm))") in new stack -- Executing [s@macro-dialout-trunk:17] Macro("SIP/123-00000146", "dialout-trunk-predial-hook,") in new stack -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/123-00000146", "") in new stack -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/123-00000146", "0?bypass,1") in new stack -- Executing [s@macro-dialout-trunk:19] ExecIf("SIP/123-00000146", "1?Set(CONNECTEDLINE(num,i)=123456)") in new stack -- Executing [s@macro-dialout-trunk:20] ExecIf("SIP/123-00000146", "1?Set(CONNECTEDLINE(name,i)=CID:123)") in new stack -- Executing [s@macro-dialout-trunk:21] GotoIf("SIP/123-00000146", "0?customtrunk") in new stack -- Executing [s@macro-dialout-trunk:22] Dial("SIP/123-00000146", "SIP/pstn/123456,300,") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/pstn/123456 == Everyone is busy/congested at this time (1:0/0/1) -- Executing [s@macro-dialout-trunk:23] NoOp("SIP/123-00000146", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 1") in new stack -- Executing [s@macro-dialout-trunk:24] Goto("SIP/123-00000146", "s-CHANUNAVAIL,1") in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/123-00000146", "RC=1") in new stack -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/123-00000146", "1,1") in new stack -- Goto (macro-dialout-trunk,1,1) -- Executing [1@macro-dialout-trunk:1] Goto("SIP/123-00000146", "continue,1") in new stack -- Goto (macro-dialout-trunk,continue,1) -- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/123-00000146", "1?noreport") in new stack -- Goto (macro-dialout-trunk,continue,3) -- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/123-00000146", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 1 - failing through to other trunks") in new stack -- Executing [continue@macro-dialout-trunk:4] Set("SIP/123-00000146", "CALLERID(number)=123") in new stack -- Executing [123456@from-internal:6] Macro("SIP/123-00000146", "outisbusy,") in new stack -- Executing [s@macro-outisbusy:1] Progress("SIP/123-00000146", "") in new stack -- Executing [s@macro-outisbusy:2] GotoIf("SIP/123-00000146", "0?emergency,1") in new stack -- Executing [s@macro-outisbusy:3] GotoIf("SIP/123-00000146", "0?intracompany,1") in new stack -- Executing [s@macro-outisbusy:4] Playback("SIP/123-00000146", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack -- <SIP/123-00000146> Playing 'all-circuits-busy-now.ulaw' (language 'en') -- <SIP/123-00000146> Playing 'pls-try-call-later.ulaw' (language 'en') == Spawn extension (macro-outisbusy, s, 4) exited non-zero on 'SIP/123-00000146' in macro 'outisbusy' == Spawn extension (from-internal, 123456, 6) exited non-zero on 'SIP/123-00000146' -- Executing [h@from-internal:1] Hangup("SIP/123-00000146", "") in new stack == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/123-00000146' 

дебаг транка pstn:

<--- SIP read from UDP:10.0.0.5:5060 ---> ACK sip:123456@10.0.0.12 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bK-aa0c42b2 From: <sip:123@10.0.0.12>;tag=b03b2195ea0779bao0 To: <sip:123456@10.0.0.12>;tag=as318bcb3b Call-ID: 939006f1-f5a740e@10.0.0.5 CSeq: 101 ACK Max-Forwards: 70 Contact: <sip:123@10.0.0.5:5060> User-Agent: Linksys/SPA3102-5.2.13(GW002) Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from UDP:10.0.0.5:5060 ---> INVITE sip:123456@10.0.0.12 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bK-7bd8bbf2 From: <sip:123@10.0.0.12>;tag=b03b2195ea0779bao0 To: <sip:123456@10.0.0.12> Remote-Party-ID: <sip:123@10.0.0.12>;screen=yes;party=calling Call-ID: 939006f1-f5a740e@10.0.0.5 CSeq: 102 INVITE Max-Forwards: 70 Authorization: Digest username="123",realm="asterisk",nonce="72caa058",uri="sip:123456@10.0.0.12",algorithm=MD5,response="42912e7f7bd20a39a0e18ce9113b7493" Contact: <sip:123@10.0.0.5:5060> Expires: 240 User-Agent: Linksys/SPA3102-5.2.13(GW002) Content-Length: 434 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 754636 754636 IN IP4 10.0.0.5 s=- c=IN IP4 10.0.0.5 t=0 0 m=audio 16476 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> --- (16 headers 20 lines) --- Sending to 10.0.0.5:5060 (no NAT) Using INVITE request as basis request - 939006f1-f5a740e@10.0.0.5 Found peer '123' for '123' from 10.0.0.5:5060 Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 100 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format G726-32 for ID 2 Found audio description format G723 for ID 4 Found audio description format PCMA for ID 8 Found audio description format G729a for ID 18 Found unknown media description format G726-40 for ID 96 Found unknown media description format G726-24 for ID 97 Found unknown media description format G726-16 for ID 98 Found unknown media description format NSE for ID 100 Found audio description format telephone-event for ID 101 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x90d (g723|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.0.0.5:16476 Looking for 123456 in from-internal (domain 10.0.0.12) list_route: hop: <sip:123@10.0.0.5:5060> <--- Transmitting (no NAT) to 10.0.0.5:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bK-7bd8bbf2;received=10.0.0.5 From: <sip:123@10.0.0.12>;tag=b03b2195ea0779bao0 To: <sip:123456@10.0.0.12> Call-ID: 939006f1-f5a740e@10.0.0.5 CSeq: 102 INVITE Server: FPBX-2.10.1(1.8.15.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:123456@10.0.0.12:5060> Content-Length: 0 <------------> Audio is at 14814 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.0.0.5:5060: INVITE sip:123456@10.0.0.5 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.12:5060;branch=z9hG4bK347fddc1;rport Max-Forwards: 70 From: "123" <sip:123@10.0.0.12>;tag=as0ff47d96 To: <sip:123456@10.0.0.5> Contact: <sip:123@10.0.0.12:5060> Call-ID: 4520235a65d167f846f293560f6fd293@10.0.0.12:5060 CSeq: 102 INVITE User-Agent: FPBX-2.10.1(1.8.15.0) Date: Mon, 03 Sep 2012 15:06:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 278 v=0 o=root 1524906032 1524906032 IN IP4 10.0.0.12 s=Asterisk PBX 1.8.15.0 c=IN IP4 10.0.0.12 t=0 0 m=audio 14814 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=ptime:20 a=sendrecv --- <--- SIP read from UDP:10.0.0.5:5060 ---> SIP/2.0 404 Not Found To: <sip:123456@10.0.0.5>;tag=d7b869f9d266a9d9i0 From: "123" <sip:123@10.0.0.12>;tag=as0ff47d96 Call-ID: 4520235a65d167f846f293560f6fd293@10.0.0.12:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.0.0.12:5060;branch=z9hG4bK347fddc1 Server: Linksys/SPA3102-5.2.13(GW002) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.0.0.5:5060: ACK sip:123456@10.0.0.5 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.12:5060;branch=z9hG4bK347fddc1;rport Max-Forwards: 70 From: "123" <sip:123@10.0.0.12>;tag=as0ff47d96 To: <sip:123456@10.0.0.5>;tag=d7b869f9d266a9d9i0 Contact: <sip:123@10.0.0.12:5060> Call-ID: 4520235a65d167f846f293560f6fd293@10.0.0.12:5060 CSeq: 102 ACK User-Agent: FPBX-2.10.1(1.8.15.0) Content-Length: 0 --- Scheduling destruction of SIP dialog '4520235a65d167f846f293560f6fd293@10.0.0.12:5060' in 32000 ms (Method: INVITE) Audio is at 14628 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (no NAT) to 10.0.0.5:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bK-7bd8bbf2;received=10.0.0.5 From: <sip:123@10.0.0.12>;tag=b03b2195ea0779bao0 To: <sip:123456@10.0.0.12>;tag=as0225c934 Call-ID: 939006f1-f5a740e@10.0.0.5 CSeq: 102 INVITE Server: FPBX-2.10.1(1.8.15.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:123456@10.0.0.12:5060> Content-Type: application/sdp Content-Length: 251 v=0 o=root 96821806 96821806 IN IP4 10.0.0.12 s=Asterisk PBX 1.8.15.0 c=IN IP4 10.0.0.12 t=0 0 m=audio 14628 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> <--- SIP read from UDP:10.0.0.5:5060 ---> CANCEL sip:123456@10.0.0.12 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bK-7bd8bbf2 From: <sip:123@10.0.0.12>;tag=b03b2195ea0779bao0 To: <sip:123456@10.0.0.12> Call-ID: 939006f1-f5a740e@10.0.0.5 CSeq: 102 CANCEL Max-Forwards: 70 Authorization: Digest username="123",realm="asterisk",nonce="72caa058",uri="sip:123456@10.0.0.12",algorithm=MD5,response="20d3b13e27589ab6b68bde6be33f57c8" User-Agent: Linksys/SPA3102-5.2.13(GW002) Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.0.0.5:5060 (no NAT) <--- Reliably Transmitting (no NAT) to 10.0.0.5:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bK-7bd8bbf2;received=10.0.0.5 From: <sip:123@10.0.0.12>;tag=b03b2195ea0779bao0 To: <sip:123456@10.0.0.12>;tag=as0225c934 Call-ID: 939006f1-f5a740e@10.0.0.5 CSeq: 102 INVITE Server: FPBX-2.10.1(1.8.15.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> <--- Transmitting (no NAT) to 10.0.0.5:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bK-7bd8bbf2;received=10.0.0.5 From: <sip:123@10.0.0.12>;tag=b03b2195ea0779bao0 To: <sip:123456@10.0.0.12>;tag=as0225c934 Call-ID: 939006f1-f5a740e@10.0.0.5 CSeq: 102 CANCEL Server: FPBX-2.10.1(1.8.15.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> <--- SIP read from UDP:10.0.0.5:5060 ---> ACK sip:123456@10.0.0.12 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bK-7bd8bbf2 From: <sip:123@10.0.0.12>;tag=b03b2195ea0779bao0 To: <sip:123456@10.0.0.12>;tag=as0225c934 Call-ID: 939006f1-f5a740e@10.0.0.5 CSeq: 102 ACK Max-Forwards: 70 Authorization: Digest username="123",realm="asterisk",nonce="72caa058",uri="sip:123456@10.0.0.12",algorithm=MD5,response="42912e7f7bd20a39a0e18ce9113b7493" Contact: <sip:123@10.0.0.5:5060> User-Agent: Linksys/SPA3102-5.2.13(GW002) Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '939006f1-f5a740e@10.0.0.5' Method: ACK 

входящие с spa3102 проходят... между экстеншенами звонки в обе стороны также проходят.