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проблемма с исходящим маршрутом

0

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.

удалить закрыть спам изменить тег редактировать

спросил 2012-09-03 19:00:43 +0400

jone31 Gravatar jone31
16 ● 13 ● 3 ● 8

обновил 2012-09-03 19:14:31 +0400

2 Ответа

Сорт. по » старые новые больше голосов
0

ну так он и настроен как ТРАНК (конфиг выше я выложил- pstn)!

сам spa3102 если на него позвонить(по номеру екстеншена 100) и донабрать номер - звонок проходит, и из этого я сделал вывод что на spa3102 все настроено верно, или я ошибаюсь!?

ссылка удалить спам редактировать

ответил 2012-09-04 09:07:32 +0400

jone31 Gravatar jone31
16 ● 13 ● 3 ● 8
0

читайте мануал по настройки спа. астериск тут вообещ нипричем.

1) если хотите звонить через гейт его надо настроить как ТРАНК а не екстеншен.

2) надо прочитайть мануал по правилам набора и прописать их на гейте. все что вы выложили - бесполезно. полезен конфиг устройства, но чиатть его врядли ктото будет, ибо ситуация стандартна и описана в официальной документации. проще настроить заново чем найти где вы чтото пропустили.

ссылка удалить спам редактировать

ответил 2012-09-03 21:58:25 +0400

meral Gravatar meral flag of Ukraine
23347 ● 24 ● 20 ● 177
http://pro-sip.net/

обновил 2012-09-03 23:23:08 +0400

Comments

разобрался... как раз таки и астериск все запарывал... создал custom trunk и прописал туда прямую строку вызова к spa3102 и все заработало...

jone31 ( 2012-09-04 21:27:44 +0400 )редактировать

запарывали вы. не надо кустом транки. надо правильно настроить ОДИН транк. блин ну мануалов по этой конкретной модели в сети десяток,в чем проблема?

meral ( 2012-09-05 11:03:55 +0400 )редактировать

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Задан: 2012-09-03 19:00:43 +0400

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Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией GNU GPL.