*->Шлюз->PBX->ШЛЮЗ->* звонки не ходят.

[fastbusy]

День Добрый!

Прошу помочь разобраться, укажите куда смотреть. Схема работы следующая:

Есть АТС традиционная, к ней через Mediant подключен FreePBX. Звонки ходят в обе стороны без проблем. На стороне PBX есть номера переадресованные в *. Позвонить на них нельзя. Т.е. звонок из VoIP в VoIP через PSTN не проходит. Всю цепочки проверена, с Медианта инициируется вызов на *. Первый лог звонка из PSTN в VoIP, второй вся цепочка. звоним с 5000 на 1802. С 1802 переадресовано на 5000, но можно и на любой другой, это не принципиально проверено..

Во втором случае получаю:
<--- SIP read from UDP:192.168.0.246:5060 --->
SIP/2.0 403 Forbidden

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

<--- SIP read from UDP:1.1.1.246:5060 --->
INVITE sip:5000@1.1.1.209;user=phone SIP/2.0
Via: SIP/2.0/UDP 1.1.1.246:5060;branch=z9hG4bKac739482695
Max-Forwards: 70
From: <sip:1802@1.1.1.246>;tag=1c738894729
To: <sip:5000@1.1.1.209;user=phone>
Call-ID: 73885488223102014143245@1.1.1.246
CSeq: 1 INVITE
Contact: <sip:1802@1.1.1.246:5060>
Supported: em,100rel,timer,replaces,path,resource-priority,sdp-anat
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Mediant 1000/v.6.60A.228.011
Content-Type: application/sdp
Content-Length: 548

v=0
o=AudiocodesGW 737617232 737617126 IN IP4 1.1.1.246
s=Phone-Call
c=IN IP4 1.1.1.246
t=0 0
m=audio 7840 RTP/AVP 0 8 18 3 3 101
a=ptime:20
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=image 7842 udptl t38
a=sendrecv
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:238
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
--- (13 headers 24 lines) ---
Sending to 1.1.1.246:5060 (no NAT)
Sending to 1.1.1.246:5060 (no NAT)
Using INVITE request as basis request - 73885488223102014143245@1.1.1.246
Found peer 'From_ACM_1000' for '1802' from 1.1.1.246:5060
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 3
Found RTP audio format 3
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 G729 for ID 18
Found audio description format GSM for ID 3
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
[2014-10-23 14:32:45] WARNING[2224][C-00000a4c]: chan_sip.c:10230 process_sdp: Failed to initialize UDPTL, declining image stream
Capabilities: us - (gsm|ulaw|alaw|g729), peer - audio=(gsm|ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 1.1.1.246:7840
Looking for 5000 in from-trunk-sip-To_ACM_1000 (domain 1.1.1.209)
list_route: hop: <sip:1802@1.1.1.246:5060>

<--- Transmitting (no NAT) to 1.1.1.246:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 1.1.1.246:5060;branch=z9hG4bKac739482695;received=1.1.1.246
From: <sip:1802@1.1.1.246>;tag=1c738894729
To: <sip:5000@1.1.1.209;user=phone>
Call-ID: 73885488223102014143245@1.1.1.246
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:5000@1.1.1.209:5060>
Content-Length: 0


<------------>
Audio is at 11044
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100008 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.246.0.13:5060:
INVITE sip:5000@10.246.0.13:5060 SIP/2.0
Via: SIP/2.0/UDP 1.1.1.209:5060;branch=z9hG4bK3de53e12;rport
Max-Forwards: 70
From: <sip:1802@1.1.1.209>;tag=as1c6c91a8
To: <sip:5000@10.246.0.13:5060>
Contact: <sip:1802@1.1.1.209:5060>
Call-ID: 6c375dab7b2cf6143f9cfbb65dcc67b7@1.1.1.209:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.13.0)
Date: Thu, 23 Oct 2014 10:32:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Asserted-Identity: "1802" <sip:1802@1.1.1.209>
Content-Type: application/sdp
Content-Length: 332

v=0
o=root 2060861493 2060861493 IN IP4 1.1.1.209
s=Asterisk PBX 11.13.0
c=IN IP4 1.1.1.209
t=0 0
m=audio 11044 RTP/AVP 0 8 3 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- Transmitting (no NAT) to 1.1.1.246:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 1.1.1.246:5060;branch=z9hG4bKac739482695;received=1.1.1.246
From: <sip:1802@1.1.1.246>;tag=1c738894729
To: <sip:5000@1.1.1.209;user=phone>;tag=as6a13f9e8
Call-ID: 73885488223102014143245@1.1.1.246
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:5000@1.1.1.209:5060>
Content-Length: 0


<------------>

<--- SIP read from UDP:10.246.0.13:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 1.1.1.209:5060;branch=z9hG4bK3de53e12;rport=5060
From: <sip:1802@1.1.1.209>;tag=as1c6c91a8
To: <sip:5000@10.246.0.13:5060>
Call-ID: 6c375dab7b2cf6143f9cfbb65dcc67b7@1.1.1.209:5060
CSeq: 102 INVITE
Supported: replaces, path, timer
User-Agent: Grandstream GXP2130 1.0.4.15
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:10.246.0.13:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 1.1.1.209:5060;branch=z9hG4bK3de53e12;rport=5060
From: <sip:1802@1.1.1.209>;tag=as1c6c91a8
To: <sip:5000@10.246.0.13:5060>;tag=1801120698
Call-ID: 6c375dab7b2cf6143f9cfbb65dcc67b7@1.1.1.209:5060
CSeq: 102 INVITE
Contact: <sip:5000@10.246.0.13:5060>
Supported: replaces, path, timer
User-Agent: Grandstream GXP2130 1.0.4.15
Allow-Events: talk, hold
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
list_route: hop: <sip:5000@10.246.0.13:5060>

<--- Transmitting (no NAT) to 1.1.1.246:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 1.1.1.246:5060;branch=z9hG4bKac739482695;received=1.1.1.246
From: <sip:1802@1.1.1.246>;tag=1c738894729
To: <sip:5000@1.1.1.209;user=phone>;tag=as6a13f9e8
Call-ID: 73885488223102014143245@1.1.1.246
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:5000@1.1.1.209:5060>
Content-Length: 0


<------------>

<--- SIP read from UDP:10.246.0.13:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 1.1.1.209:5060;branch=z9hG4bK3de53e12;rport=5060
From: <sip:1802@1.1.1.209>;tag=as1c6c91a8
To: <sip:5000@10.246.0.13:5060>;tag=1801120698
Call-ID: 6c375dab7b2cf6143f9cfbb65dcc67b7@1.1.1.209:5060
CSeq: 102 INVITE
Contact: <sip:5000@10.246.0.13:5060>
Supported: replaces, path, timer
User-Agent: Grandstream GXP2130 1.0.4.15
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 280

v=0
o=5000 8000 8000 IN IP4 10.246.0.13
s=SIP Call
c=IN IP4 10.246.0.13
t=0 0
m=audio 5004 RTP/AVP 0 8 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (12 headers 14 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
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 G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|g729), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.246.0.13:5004
list_route: hop: <sip:5000@10.246.0.13:5060>
set_destination: Parsing <sip:5000@10.246.0.13:5060> for address/port to send to
set_destination: set destination to 10.246.0.13:5060
Transmitting (NAT) to 10.246.0.13:5060:
ACK sip:5000@10.246.0.13:5060 SIP/2.0
Via: SIP/2.0/UDP 1.1.1.209:5060;branch=z9hG4bK7426f60c;rport
Max-Forwards: 70
From: <sip:1802@1.1.1.209>;tag=as1c6c91a8
To: <sip:5000@10.246.0.13:5060>;tag=1801120698
Contact: <sip:1802@1.1.1.209:5060>
Call-ID: 6c375dab7b2cf6143f9cfbb65dcc67b7@1.1.1.209:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.13.0)
Content-Length: 0


---
Audio is at 12048
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100008 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 1.1.1.246:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 1.1.1.246:5060;branch=z9hG4bKac739482695;received=1.1.1.246
From: <sip:1802@1.1.1.246>;tag=1c738894729
To: <sip:5000@1.1.1.209;user=phone>;tag=as6a13f9e8
Call-ID: 73885488223102014143245@1.1.1.246
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:5000@1.1.1.209:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 351

v=0
o=root 995360436 995360436 IN IP4 1.1.1.209
s=Asterisk PBX 11.13.0
c=IN IP4 1.1.1.209
t=0 0
m=audio 12048 RTP/AVP 0 8 3 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=image 0 udptl t38

<------------>

<--- SIP read from UDP:1.1.1.246:5060 --->
ACK sip:5000@1.1.1.209:5060 SIP/2.0
Via: SIP/2.0/UDP 1.1.1.246:5060;branch=z9hG4bKac1602678149
Max-Forwards: 70
From: <sip:1802@1.1.1.246>;tag=1c738894729
To: <sip:5000@1.1.1.209;user=phone>;tag=as6a13f9e8
Call-ID: 73885488223102014143245@1.1.1.246
CSeq: 1 ACK
Contact: <sip:1802@1.1.1.246:5060>
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Mediant 1000/v.6.60A.228.011
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---

<--- SIP read from UDP:10.246.0.13:5060 --->
BYE sip:1802@1.1.1.209:5060 SIP/2.0
Via: SIP/2.0/UDP 10.246.0.13:5060;branch=z9hG4bK1957653961;rport
From: <sip:5000@10.246.0.13:5060>;tag=1801120698
To: <sip:1802@1.1.1.209>;tag=as1c6c91a8
Call-ID: 6c375dab7b2cf6143f9cfbb65dcc67b7@1.1.1.209:5060
CSeq: 103 BYE
Contact: <sip:5000@10.246.0.13:5060>
X-Grandstream-PBX: true
Max-Forwards: 70
Supported: replaces, path, timer
User-Agent: Grandstream GXP2130 1.0.4.15
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 10.246.0.13:5060 (NAT)
Scheduling destruction of SIP dialog '6c375dab7b2cf6143f9cfbb65dcc67b7@1.1.1.209:5060' in 6400 ms (Method: BYE)

<--- Transmitting (NAT) to 10.246.0.13:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.246.0.13:5060;branch=z9hG4bK1957653961;received=10.246.0.13;rport=5060
From: <sip:5000@10.246.0.13:5060>;tag=1801120698
To: <sip:1802@1.1.1.209>;tag=as1c6c91a8
Call-ID: 6c375dab7b2cf6143f9cfbb65dcc67b7@1.1.1.209:5060
CSeq: 103 BYE
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '73885488223102014143245@1.1.1.246' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:1802@1.1.1.246:5060> for address/port to send to
set_destination: set destination to 1.1.1.246:5060
Reliably Transmitting (no NAT) to 1.1.1.246:5060:
BYE sip:1802@1.1.1.246:5060 SIP/2.0
Via: SIP/2.0/UDP 1.1.1.209:5060;branch=z9hG4bK7a6f9241
Max-Forwards: 70
From: <sip:5000@1.1.1.209;user=phone>;tag=as6a13f9e8
To: <sip:1802@1.1.1.246>;tag=1c738894729
Call-ID: 73885488223102014143245@1.1.1.246
CSeq: 102 BYE
User-Agent: FPBX-2.11.0(11.13.0)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:1.1.1.246:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 1.1.1.209:5060;branch=z9hG4bK7a6f9241
From: <sip:5000@1.1.1.209;user=phone>;tag=as6a13f9e8
To: <sip:1802@1.1.1.246>;tag=1c738894729
Call-ID: 73885488223102014143245@1.1.1.246
CSeq: 102 BYE
Contact: <sip:1802@1.1.1.246:5060>
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Mediant 1000/v.6.60A.228.011
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '73885488223102014143245@1.1.1.246' Method: ACK

<--- SIP read from UDP:10.246.0.6:5062 --->


<------------->

<--- SIP read from UDP:10.246.0.12:5062 --->


<------------->

<--- SIP read from UDP:10.246.0.11:5062 --->


<------------->
astnodea*CLI> sip set debug off
SIP Debugging Disabled

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

<--- SIP read from UDP:10.246.0.13:5060 --->
INVITE sip:1802@1.1.1.209 SIP/2.0
Via: SIP/2.0/UDP 10.246.0.13:5060;branch=z9hG4bK1992589500;rport
From: <sip:5000@1.1.1.209>;tag=1497061180
To: <sip:1802@1.1.1.209>
Call-ID: 865111062-5060-786@BA.CEG.A.BD
CSeq: 2360 INVITE
Contact: <sip:5000@10.246.0.13:5060>
X-Grandstream-PBX: true
Max-Forwards: 70
User-Agent: Grandstream GXP2130 1.0.4.15
Privacy: none
P-Preferred-Identity: <sip:5000@1.1.1.209>
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: 331

v=0
o=5000 8000 8000 IN IP4 10.246.0.13
s=SIP Call
c=IN IP4 10.246.0.13
t=0 0
m=audio 5004 RTP/AVP 0 8 18 9 2 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (17 headers 16 lines) ---
Sending to 10.246.0.13:5060 (no NAT)
Sending to 10.246.0.13:5060 (no NAT)
Using INVITE request as basis request - 865111062-5060-786@BA.CEG.A.BD
Found peer '5000' for '5000' from 10.246.0.13:5060

<--- Reliably Transmitting (no NAT) to 10.246.0.13:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.246.0.13:5060;branch=z9hG4bK1992589500;received=10.246.0.13;rport=5060
From: <sip:5000@1.1.1.209>;tag=1497061180
To: <sip:1802@1.1.1.209>;tag=as7c304f60
Call-ID: 865111062-5060-786@BA.CEG.A.BD
CSeq: 2360 INVITE
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3c1b9313"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '865111062-5060-786@BA.CEG.A.BD' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:10.246.0.13:5060 --->
ACK sip:1802@1.1.1.209 SIP/2.0
Via: SIP/2.0/UDP 10.246.0.13:5060;branch=z9hG4bK1992589500;rport
From: <sip:5000@1.1.1.209>;tag=1497061180
To: <sip:1802@1.1.1.209>;tag=as7c304f60
Call-ID: 865111062-5060-786@BA.CEG.A.BD
CSeq: 2360 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:10.246.0.13:5060 --->
INVITE sip:1802@1.1.1.209 SIP/2.0
Via: SIP/2.0/UDP 10.246.0.13:5060;branch=z9hG4bK1284790516;rport
From: <sip:5000@1.1.1.209>;tag=1497061180
To: <sip:1802@1.1.1.209>
Call-ID: 865111062-5060-786@BA.CEG.A.BD
CSeq: 2361 INVITE
Contact: <sip:5000@10.246.0.13:5060>
Authorization: Digest username="5000", realm="asterisk", nonce="3c1b9313", uri="sip:1802@1.1.1.209", response="df26e6055260f468e0143d1696dcd1b1", algorithm=MD5
X-Grandstream-PBX: true
Max-Forwards: 70
User-Agent: Grandstream GXP2130 1.0.4.15
Privacy: none
P-Preferred-Identity: <sip:5000@1.1.1.209>
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: 331

v=0
o=5000 8000 8000 IN IP4 10.246.0.13
s=SIP Call
c=IN IP4 10.246.0.13
t=0 0
m=audio 5004 RTP/AVP 0 8 18 9 2 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (18 headers 16 lines) ---
Sending to 10.246.0.13:5060 (no NAT)
Using INVITE request as basis request - 865111062-5060-786@BA.CEG.A.BD
Found peer '5000' for '5000' from 10.246.0.13:5060
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 9
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 G729 for ID 18
Found audio description format G722 for ID 9
Found audio description format G726-32 for ID 2
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|g729), peer - audio=(ulaw|alaw|g726|g729|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.246.0.13:5004
Looking for 1802 in from-internal (domain 1.1.1.209)
list_route: hop: <sip:5000@10.246.0.13:5060>

<--- Transmitting (no NAT) to 10.246.0.13:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.246.0.13:5060;branch=z9hG4bK1284790516;received=10.246.0.13;rport=5060
From: <sip:5000@1.1.1.209>;tag=1497061180
To: <sip:1802@1.1.1.209>
Call-ID: 865111062-5060-786@BA.CEG.A.BD
CSeq: 2361 INVITE
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:1802@1.1.1.209:5060>
Content-Length: 0


<------------>
Audio is at 11704
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100008 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 1.1.1.246:5060:
INVITE sip:1802@1.1.1.246 SIP/2.0
Via: SIP/2.0/UDP 1.1.1.209:5060;branch=z9hG4bK47725e11
Max-Forwards: 70
From: <sip:5000@1.1.1.209>;tag=as3a2012e6
To: <sip:1802@1.1.1.246>
Contact: <sip:5000@1.1.1.209:5060>
Call-ID: 55d79c0b0436b6ff7b93b6471bc712e7@1.1.1.209:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.13.0)
Date: Thu, 23 Oct 2014 10:25:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 330

v=0
o=root 925160736 925160736 IN IP4 1.1.1.209
s=Asterisk PBX 11.13.0
c=IN IP4 1.1.1.209
t=0 0
m=audio 11704 RTP/AVP 0 8 3 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:1.1.1.246:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 1.1.1.209:5060;branch=z9hG4bK47725e11
From: <sip:5000@1.1.1.209>;tag=as3a2012e6
To: <sip:1802@1.1.1.246>;tag=1c227908595
Call-ID: 55d79c0b0436b6ff7b93b6471bc712e7@1.1.1.209:5060
CSeq: 102 INVITE
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Mediant 1000/v.6.60A.228.011
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:1.1.1.246:5060 --->
INVITE sip:5000@1.1.1.209;user=phone SIP/2.0
Via: SIP/2.0/UDP 1.1.1.246:5060;branch=z9hG4bKac273647519
Max-Forwards: 70
From: <sip:5000@1.1.1.246>;tag=1c272843815
To: <sip:5000@1.1.1.209;user=phone>
Call-ID: 27280473423102014142526@1.1.1.246
CSeq: 1 INVITE
Contact: <sip:5000@1.1.1.246:5060>
Supported: em,100rel,timer,replaces,path,resource-priority,sdp-anat
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Mediant 1000/v.6.60A.228.011
Content-Type: application/sdp
Content-Length: 548

v=0
o=AudiocodesGW 271264696 271264667 IN IP4 1.1.1.246
s=Phone-Call
c=IN IP4 1.1.1.246
t=0 0
m=audio 7820 RTP/AVP 0 8 18 3 3 101
a=ptime:20
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=image 7822 udptl t38
a=sendrecv
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:238
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
--- (13 headers 24 lines) ---
Sending to 1.1.1.246:5060 (no NAT)
Sending to 1.1.1.246:5060 (no NAT)
Using INVITE request as basis request - 27280473423102014142526@1.1.1.246
Found peer '5000' for '5000' from 1.1.1.246:5060

<--- Reliably Transmitting (no NAT) to 1.1.1.246:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 1.1.1.246:5060;branch=z9hG4bKac273647519;received=1.1.1.246
From: <sip:5000@1.1.1.246>;tag=1c272843815
To: <sip:5000@1.1.1.209;user=phone>;tag=as7e83481f
Call-ID: 27280473423102014142526@1.1.1.246
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4e6ee8b6"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '27280473423102014142526@1.1.1.246' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:1.1.1.246:5060 --->
ACK sip:5000@1.1.1.209;user=phone SIP/2.0
Via: SIP/2.0/UDP 1.1.1.246:5060;branch=z9hG4bKac273647519
Max-Forwards: 70
From: <sip:5000@1.1.1.246>;tag=1c272843815
To: <sip:5000@1.1.1.209;user=phone>;tag=as7e83481f
Call-ID: 27280473423102014142526@1.1.1.246
CSeq: 1 ACK
Contact: <sip:5000@1.1.1.246:5060>
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Mediant 1000/v.6.60A.228.011
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---

<--- SIP read from UDP:1.1.1.246:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 1.1.1.209:5060;branch=z9hG4bK47725e11
From: <sip:5000@1.1.1.209>;tag=as3a2012e6
To: <sip:1802@1.1.1.246>;tag=1c227908595
Call-ID: 55d79c0b0436b6ff7b93b6471bc712e7@1.1.1.209:5060
CSeq: 102 INVITE
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Mediant 1000/v.6.60A.228.011
Reason: Q.850 ;cause=21
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (no NAT) to 1.1.1.246:5060:
ACK sip:1802@1.1.1.246 SIP/2.0
Via: SIP/2.0/UDP 1.1.1.209:5060;branch=z9hG4bK47725e11
Max-Forwards: 70
From: <sip:5000@1.1.1.209>;tag=as3a2012e6
To: <sip:1802@1.1.1.246>;tag=1c227908595
Contact: <sip:5000@1.1.1.209:5060>
Call-ID: 55d79c0b0436b6ff7b93b6471bc712e7@1.1.1.209:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.13.0)
Content-Length: 0


---
[2014-10-23 14:25:26] WARNING[2224][C-00000a40]: chan_sip.c:23019 handle_response_invite: Received response: "Forbidden" from '<sip:5000@1.1.1.209>;tag=as3a2012e6'
Scheduling destruction of SIP dialog '55d79c0b0436b6ff7b93b6471bc712e7@1.1.1.209:5060' in 32000 ms (Method: INVITE)
Audio is at 13874
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100008 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 10.246.0.13:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.246.0.13:5060;branch=z9hG4bK1284790516;received=10.246.0.13;rport=5060
From: <sip:5000@1.1.1.209>;tag=1497061180
To: <sip:1802@1.1.1.209>;tag=as3057a5bc
Call-ID: 865111062-5060-786@BA.CEG.A.BD
CSeq: 2361 INVITE
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:1802@1.1.1.209:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 309

v=0
o=root 1297429220 1297429220 IN IP4 1.1.1.209
s=Asterisk PBX 11.13.0
c=IN IP4 1.1.1.209
t=0 0
m=audio 13874 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:10.246.0.13:5060 --->
CANCEL sip:1802@1.1.1.209 SIP/2.0
Via: SIP/2.0/UDP 10.246.0.13:5060;branch=z9hG4bK1284790516;rport
From: <sip:5000@1.1.1.209>;tag=1497061180
To: <sip:1802@1.1.1.209>
Call-ID: 865111062-5060-786@BA.CEG.A.BD
CSeq: 2361 CANCEL
Max-Forwards: 70
User-Agent: Grandstream GXP2130 1.0.4.15
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 10.246.0.13:5060 (no NAT)

<--- Reliably Transmitting (no NAT) to 10.246.0.13:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.246.0.13:5060;branch=z9hG4bK1284790516;received=10.246.0.13;rport=5060
From: <sip:5000@1.1.1.209>;tag=1497061180
To: <sip:1802@1.1.1.209>;tag=as3057a5bc
Call-ID: 865111062-5060-786@BA.CEG.A.BD
CSeq: 2361 INVITE
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 10.246.0.13:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.246.0.13:5060;branch=z9hG4bK1284790516;received=10.246.0.13;rport=5060
From: <sip:5000@1.1.1.209>;tag=1497061180
To: <sip:1802@1.1.1.209>;tag=as3057a5bc
Call-ID: 865111062-5060-786@BA.CEG.A.BD
CSeq: 2361 CANCEL
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:10.246.0.13:5060 --->
ACK sip:1802@1.1.1.209 SIP/2.0
Via: SIP/2.0/UDP 10.246.0.13:5060;branch=z9hG4bK1284790516;rport
From: <sip:5000@1.1.1.209>;tag=1497061180
To: <sip:1802@1.1.1.209>;tag=as3057a5bc
Call-ID: 865111062-5060-786@BA.CEG.A.BD
CSeq: 2361 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '865111062-5060-786@BA.CEG.A.BD' Method: ACK

<--- SIP read from UDP:10.246.0.3:5062 --->


<------------->

[fastbusy]

Да, если позвонить в PSTN на живого человека, то он без проблем переведет звонок в VoIP.

[ded]

А как подключено к АТС, через экстеншны? Или через транки? По схеме -

[fastbusy]

Транк peer без авторизации.

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

sip show peer From_ACM_1000


* Name : From_ACM_1000
Description :
Secret : <Not set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : from-trunk-sip-To_ACM_1000
Record On feature : automon
Record Off feature : automon
Subscr.Cont. : <Not set>
Language : ru
Tonezone : <Not set>
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Named Callgr :
Nam. Pickupgr:
MOH Suggest :
Mailbox :
VM Extension : *97
LastMsgsSent : 0/0
Call limit : 0
Max forwards : 0
Dynamic : No
Callerid : "" <>
MaxCallBR : 384 kbps
Expire : -1
Insecure : no
Force rport : No
Symmetric RTP: No
ACL : No
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: 4294967295
DirectMedia : No
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID : No
TrustIDOutbnd: Legacy
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost : 1.1.1.246
Addr->IP : 1.1.1.246:5060
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username:
SIP Options : 100rel path replaces replace resource-priority sdp-anat timer
Codecs : (gsm|ulaw|alaw|g729)
Codec Order : (ulaw:20,alaw:20,gsm:20,g729:20)
Auto-Framing : No
Status : Unmonitored
Useragent :
Reg. Contact :
Qualify Freq : 60000 ms
Keepalive : 0 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No

[ded]

Я же спросил - с какой стороны подключен Астериск - со стороны СО транков или со стороны экстеншн?

[fastbusy]

* подключен двумя потоками PRI к АТС. Исходя из вашего вопроса со стороны транзита, но по факту увеличина емкость АТС. Внешние линки на операторов будут и напрямую с * на и по старым присоединениям через потоки Е 1 на АТС.. СО- аналоговых внешних линий нет.. там емкость большая.

[virus_net]

В очередной раз снимаю шляпу перед ded. Завидую его умению по таким описаниям ещё и вопросы какие то задавать.

Из описания я вот понял только то, что что-то не работает. Несколько раз перечитывал, но осознания так и не пришло, одни вопросы, куча вопросов. М.б. все все понимают, а это это я такой тупой ?

На стороне PBX есть номера переадресованные в *.

Читаем сабж:

*->Шлюз->PBX->ШЛЮЗ->*

Имеем * два раза. На какую * ?

Т.е. звонок из VoIP в VoIP через PSTN не проходит.

Откуда ?

Во втором случае получаю:
<--- SIP read from UDP:192.168.0.246:5060 --->
SIP/2.0 403 Forbidden

Разворачиваем спойлеры, жмем CTRL+F ищем 192.168.0.246, получаем "1 of 1".
Смотрим на встречающиеся адреса: 1.1.1.246, 1.1.1.209, 10.246.0.13, 10.246.0.12, 10.246.0.11, 10.246.0.6

Итог: прежде чем что-то можно было бы сказать нужно потратить уйму времени чтобы вкурить откуда и куда и кто есть кто.

[fastbusy]

* один. Два сегмента : VoIP&PSTN. Это понятно? Соединены через 2 потока. Звонки из одного сегмента в другой ходят без проблем. Понятно? Часть пользователей переехало в VoIP. Со старых добавочных стоит переадресация на новые. Так вот, когда они звонят своим коллегам также переехавшим в VoIP, звонок(сигнализаия) проходит по цепочке *->GW->PBX и возвращается после подмены б номера обратно ->GW->*. Так вот на последнем этапе, при вызове с GW на * на инвайт приходит отбой. Вопрос: зачем звонить на старый номер? - необсужается.

192.168.0. в логах подменил на 1.1.1.- очевидно не везде, мой косяк.

[ded]

Не понятно.
1) Где город в ваших сегментах?
2) Часть пользователей переехало в VoIP. ......

Так вот, когда они звонят своим коллегам также переехавшим в VoIP, звонок(сигнализаия) проходит по цепочке *->GW->PBX и возвращается после подмены б номера обратно ->GW->*

С чего бы? Надеюсь, что у этих всех VoIP пользователей номера +/- близкие, то есть единый диал-план? Тогда когда они звонят своим коллегам также переехавшим в VoIP звонок(сигнализация) должен оставаться в Астериске.

[fastbusy]

Не могу схемку прикрутить. Куда отправить?

"Достигнут максимальный общий размер ваших вложений."