Есть Еластикс, есть 2 транка от разных провайдеров, один транк работает нормально, входящие и исходящие есть. Второй транк не получается настроить входящие звонки.
Много разных комбинаций пробовал, читал много тем на форуме, так же изучал данную схему, не получается.
Вот что у меня вываливается в консоль при входящем звонке.
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:
<--- SIP read from UDP:10.5.7.1:5060 --->
INVITE sip:308951@10.5.7.16:5060 SIP/2.0
Via: SIP/2.0/UDP 10.5.7.1:5060;branch=z9hG4bK021007cb
Max-Forwards: 70
From: "0979542756 " <sip:0979542756@10.5.7.1>;tag=as66b43588
To: <sip:308951@10.5.7.16:5060>
Contact: <sip:0979542756@10.5.7.1:5060>
Call-ID: 55b55fbe7b9c10592e8318ff4b265c1b@10.5.7.1:5060
CSeq: 102 INVITE
User-Agent: Asterisk_voip__PBX
Date: Mon, 15 Dec 2014 09:46:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 251
v=0
o=root 1488185884 1488185884 IN IP4 10.5.7.1
s=Asterisk PBX 11.8.1
c=IN IP4 10.5.7.1
t=0 0
m=audio 14446 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
Sending to 10.5.7.1:5060 (no NAT)
Using INVITE request as basis request - 55b55fbe7b9c10592e8318ff4b265c1b@10.5.7.1:5060
Found peer '308951' for '0979542756' from 10.5.7.1:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/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 10.5.7.1:14446
Looking for 308951 in incoming (domain 10.5.7.16)
<--- Reliably Transmitting (no NAT) to 10.5.7.1:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.5.7.1:5060;branch=z9hG4bK021007cb;received=10.5.7.1
From: "0979542756 " <sip:0979542756@10.5.7.1>;tag=as66b43588
To: <sip:308951@10.5.7.16:5060>;tag=as6018678c
Call-ID: 55b55fbe7b9c10592e8318ff4b265c1b@10.5.7.1:5060
CSeq: 102 INVITE
erver: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '55b55fbe7b9c10592e8318ff4b265c1b@10.5.7.1:5060' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:10.5.7.1:5060 --->
ACK sip:308951@10.5.7.16:5060 SIP/2.0
Via: SIP/2.0/UDP 10.5.7.1:5060;branch=z9hG4bK021007cb
Max-Forwards: 70
From: "0979542756 " <sip:0979542756@10.5.7.1>;tag=as66b43588
To: <sip:308951@10.5.7.16:5060>;tag=as6018678c
Contact: <sip:0979542756@10.5.7.1:5060>
Call-ID: 55b55fbe7b9c10592e8318ff4b265c1b@10.5.7.1:5060
CSeq: 102 ACK
User-Agent: Asterisk_voip__PBX
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '55b55fbe7b9c10592e8318ff4b265c1b@10.5.7.1:5060' Method: ACK
INVITE sip:308951@10.5.7.16:5060 SIP/2.0
Via: SIP/2.0/UDP 10.5.7.1:5060;branch=z9hG4bK021007cb
Max-Forwards: 70
From: "0979542756 " <sip:0979542756@10.5.7.1>;tag=as66b43588
To: <sip:308951@10.5.7.16:5060>
Contact: <sip:0979542756@10.5.7.1:5060>
Call-ID: 55b55fbe7b9c10592e8318ff4b265c1b@10.5.7.1:5060
CSeq: 102 INVITE
User-Agent: Asterisk_voip__PBX
Date: Mon, 15 Dec 2014 09:46:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 251
v=0
o=root 1488185884 1488185884 IN IP4 10.5.7.1
s=Asterisk PBX 11.8.1
c=IN IP4 10.5.7.1
t=0 0
m=audio 14446 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
Sending to 10.5.7.1:5060 (no NAT)
Using INVITE request as basis request - 55b55fbe7b9c10592e8318ff4b265c1b@10.5.7.1:5060
Found peer '308951' for '0979542756' from 10.5.7.1:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/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 10.5.7.1:14446
Looking for 308951 in incoming (domain 10.5.7.16)
<--- Reliably Transmitting (no NAT) to 10.5.7.1:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.5.7.1:5060;branch=z9hG4bK021007cb;received=10.5.7.1
From: "0979542756 " <sip:0979542756@10.5.7.1>;tag=as66b43588
To: <sip:308951@10.5.7.16:5060>;tag=as6018678c
Call-ID: 55b55fbe7b9c10592e8318ff4b265c1b@10.5.7.1:5060
CSeq: 102 INVITE
erver: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '55b55fbe7b9c10592e8318ff4b265c1b@10.5.7.1:5060' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:10.5.7.1:5060 --->
ACK sip:308951@10.5.7.16:5060 SIP/2.0
Via: SIP/2.0/UDP 10.5.7.1:5060;branch=z9hG4bK021007cb
Max-Forwards: 70
From: "0979542756 " <sip:0979542756@10.5.7.1>;tag=as66b43588
To: <sip:308951@10.5.7.16:5060>;tag=as6018678c
Contact: <sip:0979542756@10.5.7.1:5060>
Call-ID: 55b55fbe7b9c10592e8318ff4b265c1b@10.5.7.1:5060
CSeq: 102 ACK
User-Agent: Asterisk_voip__PBX
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '55b55fbe7b9c10592e8318ff4b265c1b@10.5.7.1:5060' Method: ACK