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спросил 2017-07-10 15:34:57 +0400

A66aT Gravatar A66aT

ACK в ответ на 100 Trying

Добрый день

Возникла проблема на одной АТС. Стоит сборка Elastix, сама АТС на виртуальной машине. Суть в чем, некоторые звонки оператору связи заканчиваются ошибкой 500 Internal Server Error, из-за того, что на пакет 100 Trying АТС отвечает пакетом АСК.

Reliably Transmitting (NAT) to operator_IP:5060:
INVITE sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.17.1)
Date: Tue, 04 Jul 2017 16:19:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 210920071 210920071 IN IP4 192.168.1.10
s=Asterisk PBX 11.17.1
c=IN IP4 192.168.1.10
t=0 0
m=audio 19224 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/to_Peru_SN/519961XXXXX
Retransmitting #1 (NAT) to operator_IP:5060:
INVITE sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.17.1)
Date: Tue, 04 Jul 2017 16:19:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 210920071 210920071 IN IP4 192.168.1.10
s=Asterisk PBX 11.17.1
c=IN IP4 192.168.1.10
t=0 0
m=audio 19224 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:operator_IP:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport=5060;received=192.168.1.10
From: <sip:username@192.168.1.10:5060>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 INVITE
Server: MediaCore/3.0.0
Content-Length: 0

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

<--- SIP read from UDP:operator_IP:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport=5060;received=192.168.1.10
From: <sip:username@192.168.1.10:5060>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 INVITE
WWW-Authenticate: Digest realm="MediaCore", domain="mediacore", nonce="ad2239f913f981f9e9c4179fce22868d.1499188754", stale=FALSE, algorithm=MD5
Server: MediaCore/3.0.0
Contact: <sip:519961XXXXX@operator_IP:5060>
Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO, REGISTER
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to operator_IP:5060:
ACK sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.17.1)
Content-Length: 0


---
Audio is at 19224
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to operator_IP:5060:
INVITE sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 103 INVITE
User-Agent: FPBX-2.11.0(11.17.1)
Authorization: Digest username="username_id", realm="MediaCore", algorithm=MD5, uri="sip:mediacore", nonce="ad2239f913f981f9e9c4179fce22868d.1499188754", response="secret"
Date: Tue, 04 Jul 2017 16:19:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 210920071 210920072 IN IP4 192.168.1.10
s=Asterisk PBX 11.17.1
c=IN IP4 192.168.1.10
t=0 0
m=audio 19224 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:operator_IP:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport=5060;received=192.168.1.10
From: <sip:username@192.168.1.10:5060>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 INVITE
WWW-Authenticate: Digest realm="MediaCore", domain="mediacore", nonce="ad2239f913f981f9e9c4179fce22868d.1499188754", stale=FALSE, algorithm=MD5
Server: MediaCore/3.0.0
Contact: <sip:519961XXXXX@operator_IP:5060>
Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO, REGISTER
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to operator_IP:5060:
ACK sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.17.1)
Content-Length: 0


---
Retransmitting #1 (NAT) to operator_IP:5060:
INVITE sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 103 INVITE
User-Agent: FPBX-2.11.0(11.17.1)
Authorization: Digest username="username_id", realm="MediaCore", algorithm=MD5, uri="sip:mediacore", nonce="ad2239f913f981f9e9c4179fce22868d.1499188754", response="secret"
Date: Tue, 04 Jul 2017 16:19:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 210920071 210920072 IN IP4 192.168.1.10
s=Asterisk PBX 11.17.1
c=IN IP4 192.168.1.10
t=0 0
m=audio 19224 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:operator_IP:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport=5060;received=192.168.1.10
From: <sip:username@192.168.1.10:5060>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 103 INVITE
Server: MediaCore/3.0.0
Content-Length: 0

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

<--- SIP read from UDP:operator_IP:5060 --->
SIP/2.0 500 Server Internal Error
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport=5060;received=192.168.1.10
From: <sip:username@192.168.1.10:5060>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 103 INVITE
Reason: Q.850;cause=34;text="No circuit/channel available"
Server: MediaCore/3.0.0
Contact: <sip:519961XXXXX@operator_IP:5060>
Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO, REGISTER
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
    -- Got SIP response 500 "Server Internal Error" back from operator_IP:5060
Transmitting (NAT) to operator_IP:5060:
ACK sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 103 ACK
User-Agent: FPBX-2.11.0(11.17.1)
Content-Length: 0

Насколько я понял, проблема в том, что изначально передается два инвайта, после чего начинаются две параллельные СИП-сессии, и в тот момент, когда сервер провайдера отвечает 100 Trying на INVITE, в параллельной СИП-сессии приходит АСК на пакет 401 Unauthorized

К сожлению, скрин из Sngrep`a с более наглядным логом звонков приложить не могу ибо карма меньше 10. Подскажите, пожалуйста, в чем может быть проблема ?

ACK в ответ на 100 Trying

Добрый день

Возникла проблема на одной АТС. Стоит сборка Elastix, сама АТС на виртуальной машине. Суть в чем, некоторые звонки оператору связи заканчиваются ошибкой 500 Internal Server Error, из-за того, что на пакет 100 Trying АТС отвечает пакетом АСК.

Reliably Transmitting (NAT) to operator_IP:5060:
INVITE sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.17.1)
Date: Tue, 04 Jul 2017 16:19:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 210920071 210920071 IN IP4 192.168.1.10
s=Asterisk PBX 11.17.1
c=IN IP4 192.168.1.10
t=0 0
m=audio 19224 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/to_Peru_SN/519961XXXXX
Retransmitting #1 (NAT) to operator_IP:5060:
INVITE sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.17.1)
Date: Tue, 04 Jul 2017 16:19:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 210920071 210920071 IN IP4 192.168.1.10
s=Asterisk PBX 11.17.1
c=IN IP4 192.168.1.10
t=0 0
m=audio 19224 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:operator_IP:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport=5060;received=192.168.1.10
From: <sip:username@192.168.1.10:5060>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 INVITE
Server: MediaCore/3.0.0
Content-Length: 0

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

<--- SIP read from UDP:operator_IP:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport=5060;received=192.168.1.10
From: <sip:username@192.168.1.10:5060>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 INVITE
WWW-Authenticate: Digest realm="MediaCore", domain="mediacore", nonce="ad2239f913f981f9e9c4179fce22868d.1499188754", stale=FALSE, algorithm=MD5
Server: MediaCore/3.0.0
Contact: <sip:519961XXXXX@operator_IP:5060>
Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO, REGISTER
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to operator_IP:5060:
ACK sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.17.1)
Content-Length: 0


---
Audio is at 19224
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to operator_IP:5060:
INVITE sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 103 INVITE
User-Agent: FPBX-2.11.0(11.17.1)
Authorization: Digest username="username_id", realm="MediaCore", algorithm=MD5, uri="sip:mediacore", nonce="ad2239f913f981f9e9c4179fce22868d.1499188754", response="secret"
Date: Tue, 04 Jul 2017 16:19:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 210920071 210920072 IN IP4 192.168.1.10
s=Asterisk PBX 11.17.1
c=IN IP4 192.168.1.10
t=0 0
m=audio 19224 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:operator_IP:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport=5060;received=192.168.1.10
From: <sip:username@192.168.1.10:5060>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 INVITE
WWW-Authenticate: Digest realm="MediaCore", domain="mediacore", nonce="ad2239f913f981f9e9c4179fce22868d.1499188754", stale=FALSE, algorithm=MD5
Server: MediaCore/3.0.0
Contact: <sip:519961XXXXX@operator_IP:5060>
Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO, REGISTER
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to operator_IP:5060:
ACK sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.17.1)
Content-Length: 0


---
Retransmitting #1 (NAT) to operator_IP:5060:
INVITE sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 103 INVITE
User-Agent: FPBX-2.11.0(11.17.1)
Authorization: Digest username="username_id", realm="MediaCore", algorithm=MD5, uri="sip:mediacore", nonce="ad2239f913f981f9e9c4179fce22868d.1499188754", response="secret"
Date: Tue, 04 Jul 2017 16:19:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 210920071 210920072 IN IP4 192.168.1.10
s=Asterisk PBX 11.17.1
c=IN IP4 192.168.1.10
t=0 0
m=audio 19224 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:operator_IP:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport=5060;received=192.168.1.10
From: <sip:username@192.168.1.10:5060>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 103 INVITE
Server: MediaCore/3.0.0
Content-Length: 0

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

<--- SIP read from UDP:operator_IP:5060 --->
SIP/2.0 500 Server Internal Error
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport=5060;received=192.168.1.10
From: <sip:username@192.168.1.10:5060>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 103 INVITE
Reason: Q.850;cause=34;text="No circuit/channel available"
Server: MediaCore/3.0.0
Contact: <sip:519961XXXXX@operator_IP:5060>
Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO, REGISTER
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
    -- Got SIP response 500 "Server Internal Error" back from operator_IP:5060
Transmitting (NAT) to operator_IP:5060:
ACK sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 103 ACK
User-Agent: FPBX-2.11.0(11.17.1)
Content-Length: 0

Насколько я понял, проблема в том, что изначально передается два инвайта, после чего начинаются две параллельные СИП-сессии, и в тот момент, когда сервер провайдера отвечает 100 Trying на INVITE, в параллельной СИП-сессии приходит АСК на пакет 401 Unauthorized

Настройки транка на Elastix

[to_Peru_SN]
host=operator_IP
username=username_id
secret=secret
type=peer
nat=force_rport,comedia
canreinvite=no
qualify=yes

К сожлению, скрин из Sngrep`a с более наглядным логом звонков приложить не могу ибо карма меньше 10. Подскажите, пожалуйста, в чем может быть проблема ?

ACK в ответ на 100 Trying

Добрый день

Возникла проблема на одной АТС. Стоит сборка Elastix, сама АТС на виртуальной машине. Суть в чем, некоторые звонки оператору связи заканчиваются ошибкой 500 Internal Server Error, из-за того, что на пакет 100 Trying АТС отвечает пакетом АСК.

Reliably Transmitting (NAT) to operator_IP:5060:
INVITE sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.17.1)
Date: Tue, 04 Jul 2017 16:19:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 210920071 210920071 IN IP4 192.168.1.10
s=Asterisk PBX 11.17.1
c=IN IP4 192.168.1.10
t=0 0
m=audio 19224 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/to_Peru_SN/519961XXXXX
Retransmitting #1 (NAT) to operator_IP:5060:
INVITE sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.17.1)
Date: Tue, 04 Jul 2017 16:19:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 210920071 210920071 IN IP4 192.168.1.10
s=Asterisk PBX 11.17.1
c=IN IP4 192.168.1.10
t=0 0
m=audio 19224 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:operator_IP:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport=5060;received=192.168.1.10
From: <sip:username@192.168.1.10:5060>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 INVITE
Server: MediaCore/3.0.0
Content-Length: 0

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

<--- SIP read from UDP:operator_IP:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport=5060;received=192.168.1.10
From: <sip:username@192.168.1.10:5060>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 INVITE
WWW-Authenticate: Digest realm="MediaCore", domain="mediacore", nonce="ad2239f913f981f9e9c4179fce22868d.1499188754", stale=FALSE, algorithm=MD5
Server: MediaCore/3.0.0
Contact: <sip:519961XXXXX@operator_IP:5060>
Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO, REGISTER
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to operator_IP:5060:
ACK sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.17.1)
Content-Length: 0


---
Audio is at 19224
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to operator_IP:5060:
INVITE sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 103 INVITE
User-Agent: FPBX-2.11.0(11.17.1)
Authorization: Digest username="username_id", realm="MediaCore", algorithm=MD5, uri="sip:mediacore", nonce="ad2239f913f981f9e9c4179fce22868d.1499188754", response="secret"
Date: Tue, 04 Jul 2017 16:19:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 210920071 210920072 IN IP4 192.168.1.10
s=Asterisk PBX 11.17.1
c=IN IP4 192.168.1.10
t=0 0
m=audio 19224 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:operator_IP:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport=5060;received=192.168.1.10
From: <sip:username@192.168.1.10:5060>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 INVITE
WWW-Authenticate: Digest realm="MediaCore", domain="mediacore", nonce="ad2239f913f981f9e9c4179fce22868d.1499188754", stale=FALSE, algorithm=MD5
Server: MediaCore/3.0.0
Contact: <sip:519961XXXXX@operator_IP:5060>
Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO, REGISTER
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to operator_IP:5060:
ACK sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.17.1)
Content-Length: 0


---
Retransmitting #1 (NAT) to operator_IP:5060:
INVITE sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 103 INVITE
User-Agent: FPBX-2.11.0(11.17.1)
Authorization: Digest username="username_id", realm="MediaCore", algorithm=MD5, uri="sip:mediacore", nonce="ad2239f913f981f9e9c4179fce22868d.1499188754", response="secret"
Date: Tue, 04 Jul 2017 16:19:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 210920071 210920072 IN IP4 192.168.1.10
s=Asterisk PBX 11.17.1
c=IN IP4 192.168.1.10
t=0 0
m=audio 19224 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:operator_IP:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport=5060;received=192.168.1.10
From: <sip:username@192.168.1.10:5060>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 103 INVITE
Server: MediaCore/3.0.0
Content-Length: 0

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

<--- SIP read from UDP:operator_IP:5060 --->
SIP/2.0 500 Server Internal Error
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport=5060;received=192.168.1.10
From: <sip:username@192.168.1.10:5060>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 103 INVITE
Reason: Q.850;cause=34;text="No circuit/channel available"
Server: MediaCore/3.0.0
Contact: <sip:519961XXXXX@operator_IP:5060>
Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO, REGISTER
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
    -- Got SIP response 500 "Server Internal Error" back from operator_IP:5060
Transmitting (NAT) to operator_IP:5060:
ACK sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 103 ACK
User-Agent: FPBX-2.11.0(11.17.1)
Content-Length: 0

Насколько я понял, проблема в том, что изначально передается два инвайта, после чего начинаются две параллельные СИП-сессии, и в тот момент, когда сервер провайдера отвечает 100 Trying на INVITE, в параллельной СИП-сессии приходит АСК на пакет 401 Unauthorized

Настройки транка на Elastix

[to_Peru_SN]
host=operator_IP
username=username_id
secret=secret
type=peer
nat=force_rport,comedia
canreinvite=no
qualify=yes

К сожлению, скрин из Sngrep`a с более наглядным логом звонков приложить не могу ибо карма меньше 10. Подскажите, пожалуйста, в чем может быть проблема ?

нажмите, чтобы скрыть/показать версии 4
теги изменены
редактировать

обновил 2017-07-11 11:14:46 +0400

ACK в ответ на 100 Trying

Добрый день

Возникла проблема на одной АТС. Стоит сборка Elastix, сама АТС на виртуальной машине. Суть в чем, некоторые звонки оператору связи заканчиваются ошибкой 500 Internal Server Error, из-за того, что на пакет 100 Trying АТС отвечает пакетом АСК.

Reliably Transmitting (NAT) to operator_IP:5060:
INVITE sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.17.1)
Date: Tue, 04 Jul 2017 16:19:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 210920071 210920071 IN IP4 192.168.1.10
s=Asterisk PBX 11.17.1
c=IN IP4 192.168.1.10
t=0 0
m=audio 19224 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/to_Peru_SN/519961XXXXX
Retransmitting #1 (NAT) to operator_IP:5060:
INVITE sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.17.1)
Date: Tue, 04 Jul 2017 16:19:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 210920071 210920071 IN IP4 192.168.1.10
s=Asterisk PBX 11.17.1
c=IN IP4 192.168.1.10
t=0 0
m=audio 19224 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:operator_IP:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport=5060;received=192.168.1.10
From: <sip:username@192.168.1.10:5060>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 INVITE
Server: MediaCore/3.0.0
Content-Length: 0

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

<--- SIP read from UDP:operator_IP:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport=5060;received=192.168.1.10
From: <sip:username@192.168.1.10:5060>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 INVITE
WWW-Authenticate: Digest realm="MediaCore", domain="mediacore", nonce="ad2239f913f981f9e9c4179fce22868d.1499188754", stale=FALSE, algorithm=MD5
Server: MediaCore/3.0.0
Contact: <sip:519961XXXXX@operator_IP:5060>
Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO, REGISTER
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to operator_IP:5060:
ACK sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.17.1)
Content-Length: 0


---
Audio is at 19224
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to operator_IP:5060:
INVITE sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 103 INVITE
User-Agent: FPBX-2.11.0(11.17.1)
Authorization: Digest username="username_id", realm="MediaCore", algorithm=MD5, uri="sip:mediacore", nonce="ad2239f913f981f9e9c4179fce22868d.1499188754", response="secret"
Date: Tue, 04 Jul 2017 16:19:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 210920071 210920072 IN IP4 192.168.1.10
s=Asterisk PBX 11.17.1
c=IN IP4 192.168.1.10
t=0 0
m=audio 19224 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:operator_IP:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport=5060;received=192.168.1.10
From: <sip:username@192.168.1.10:5060>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 INVITE
WWW-Authenticate: Digest realm="MediaCore", domain="mediacore", nonce="ad2239f913f981f9e9c4179fce22868d.1499188754", stale=FALSE, algorithm=MD5
Server: MediaCore/3.0.0
Contact: <sip:519961XXXXX@operator_IP:5060>
Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO, REGISTER
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to operator_IP:5060:
ACK sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.17.1)
Content-Length: 0


---
Retransmitting #1 (NAT) to operator_IP:5060:
INVITE sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 103 INVITE
User-Agent: FPBX-2.11.0(11.17.1)
Authorization: Digest username="username_id", realm="MediaCore", algorithm=MD5, uri="sip:mediacore", nonce="ad2239f913f981f9e9c4179fce22868d.1499188754", response="secret"
Date: Tue, 04 Jul 2017 16:19:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 210920071 210920072 IN IP4 192.168.1.10
s=Asterisk PBX 11.17.1
c=IN IP4 192.168.1.10
t=0 0
m=audio 19224 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:operator_IP:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport=5060;received=192.168.1.10
From: <sip:username@192.168.1.10:5060>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 103 INVITE
Server: MediaCore/3.0.0
Content-Length: 0

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

<--- SIP read from UDP:operator_IP:5060 --->
SIP/2.0 500 Server Internal Error
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport=5060;received=192.168.1.10
From: <sip:username@192.168.1.10:5060>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 103 INVITE
Reason: Q.850;cause=34;text="No circuit/channel available"
Server: MediaCore/3.0.0
Contact: <sip:519961XXXXX@operator_IP:5060>
Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO, REGISTER
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
    -- Got SIP response 500 "Server Internal Error" back from operator_IP:5060
Transmitting (NAT) to operator_IP:5060:
ACK sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 103 ACK
User-Agent: FPBX-2.11.0(11.17.1)
Content-Length: 0

Насколько я понял, проблема в том, что изначально передается два инвайта, после чего начинаются две параллельные СИП-сессии, и в тот момент, когда сервер провайдера отвечает 100 Trying на INVITE, в параллельной СИП-сессии приходит АСК на пакет 401 Unauthorized

Настройки транка на Elastix

[to_Peru_SN]
host=operator_IP
username=username_id
secret=secret
type=peer
nat=force_rport,comedia
canreinvite=no
qualify=yes

К сожлению, скрин из Sngrep`a с более наглядным логом звонков приложить не могу ибо карма меньше 10. Подскажите, пожалуйста, в чем может быть проблема ?

нажмите, чтобы скрыть/показать версии 5
Добавил "скрин" лога звонка
редактировать

обновил 2017-07-11 11:14:46 +0400

ACK в ответ на 100 Trying

Добрый день

Возникла проблема на одной АТС. Стоит сборка Elastix, сама АТС на виртуальной машине. Суть в чем, некоторые звонки оператору связи заканчиваются ошибкой 500 Internal Server Error, из-за того, что на пакет 100 Trying АТС отвечает пакетом АСК.

Reliably Transmitting (NAT) to operator_IP:5060:
INVITE sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.17.1)
Date: Tue, 04 Jul 2017 16:19:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 210920071 210920071 IN IP4 192.168.1.10
s=Asterisk PBX 11.17.1
c=IN IP4 192.168.1.10
t=0 0
m=audio 19224 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/to_Peru_SN/519961XXXXX
Retransmitting #1 (NAT) to operator_IP:5060:
INVITE sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.17.1)
Date: Tue, 04 Jul 2017 16:19:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 210920071 210920071 IN IP4 192.168.1.10
s=Asterisk PBX 11.17.1
c=IN IP4 192.168.1.10
t=0 0
m=audio 19224 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:operator_IP:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport=5060;received=192.168.1.10
From: <sip:username@192.168.1.10:5060>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 INVITE
Server: MediaCore/3.0.0
Content-Length: 0

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

<--- SIP read from UDP:operator_IP:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport=5060;received=192.168.1.10
From: <sip:username@192.168.1.10:5060>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 INVITE
WWW-Authenticate: Digest realm="MediaCore", domain="mediacore", nonce="ad2239f913f981f9e9c4179fce22868d.1499188754", stale=FALSE, algorithm=MD5
Server: MediaCore/3.0.0
Contact: <sip:519961XXXXX@operator_IP:5060>
Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO, REGISTER
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to operator_IP:5060:
ACK sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.17.1)
Content-Length: 0


---
Audio is at 19224
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to operator_IP:5060:
INVITE sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 103 INVITE
User-Agent: FPBX-2.11.0(11.17.1)
Authorization: Digest username="username_id", realm="MediaCore", algorithm=MD5, uri="sip:mediacore", nonce="ad2239f913f981f9e9c4179fce22868d.1499188754", response="secret"
Date: Tue, 04 Jul 2017 16:19:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 210920071 210920072 IN IP4 192.168.1.10
s=Asterisk PBX 11.17.1
c=IN IP4 192.168.1.10
t=0 0
m=audio 19224 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:operator_IP:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport=5060;received=192.168.1.10
From: <sip:username@192.168.1.10:5060>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 INVITE
WWW-Authenticate: Digest realm="MediaCore", domain="mediacore", nonce="ad2239f913f981f9e9c4179fce22868d.1499188754", stale=FALSE, algorithm=MD5
Server: MediaCore/3.0.0
Contact: <sip:519961XXXXX@operator_IP:5060>
Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO, REGISTER
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to operator_IP:5060:
ACK sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.17.1)
Content-Length: 0


---
Retransmitting #1 (NAT) to operator_IP:5060:
INVITE sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 103 INVITE
User-Agent: FPBX-2.11.0(11.17.1)
Authorization: Digest username="username_id", realm="MediaCore", algorithm=MD5, uri="sip:mediacore", nonce="ad2239f913f981f9e9c4179fce22868d.1499188754", response="secret"
Date: Tue, 04 Jul 2017 16:19:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 210920071 210920072 IN IP4 192.168.1.10
s=Asterisk PBX 11.17.1
c=IN IP4 192.168.1.10
t=0 0
m=audio 19224 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:operator_IP:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport=5060;received=192.168.1.10
From: <sip:username@192.168.1.10:5060>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 103 INVITE
Server: MediaCore/3.0.0
Content-Length: 0

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

<--- SIP read from UDP:operator_IP:5060 --->
SIP/2.0 500 Server Internal Error
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport=5060;received=192.168.1.10
From: <sip:username@192.168.1.10:5060>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 103 INVITE
Reason: Q.850;cause=34;text="No circuit/channel available"
Server: MediaCore/3.0.0
Contact: <sip:519961XXXXX@operator_IP:5060>
Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO, REGISTER
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
    -- Got SIP response 500 "Server Internal Error" back from operator_IP:5060
Transmitting (NAT) to operator_IP:5060:
ACK sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 103 ACK
User-Agent: FPBX-2.11.0(11.17.1)
Content-Length: 0

Насколько я понял, проблема в том, что изначально передается два инвайта, после чего начинаются две параллельные СИП-сессии, и в тот момент, когда сервер провайдера отвечает 100 Trying на INVITE, в параллельной СИП-сессии приходит АСК на пакет 401 Unauthorized

Настройки транка на Elastix

[to_Peru_SN]
host=operator_IP
username=username_id
secret=secret
type=peer
nat=force_rport,comedia
canreinvite=no
qualify=yes

К сожлению, скрин из Sngrep`a с более наглядным логом звонков приложить не могу ибо карма меньше 10. Подскажите, пожалуйста, в чем может быть проблема ?

               My IP                Operator`s IP         

                    |        INVITE (SDP)         |         
  22:24:44.187562   | --------------------------> |        
        +0.008365   |         100 Trying          |         
  22:24:44.195927   | <-------------------------- |         
        +0.005987   |      401 Unauthorized       |         
  22:24:44.201914   | <-------------------------- |         
        +0.206224   |        INVITE (SDP)         |         
  22:24:44.408138   | ------------------------>>> |         
        +0.003815   |      401 Unauthorized       |         
  22:24:44.411953   | <<<------------------------ |         
        +0.005991   |             ACK             |         
  22:24:44.417944   | --------------------------> |         
        +0.000467   |        INVITE (SDP)         |         
  22:24:44.418411   | --------------------------> |         
        +0.009147   |         100 Trying          |         
  22:24:44.427558   | <-------------------------- |         
        +0.199511   |             ACK             |         
  22:24:44.627069   | --------------------------> |         
        +0.004688   |  500 Server Internal Error  |         
  22:24:44.631757   | <-------------------------- |         
        +0.007662   |        INVITE (SDP)         |         
  22:24:44.639419   | --------------------------> |        
        +0.002707   |  500 Server Internal Error  |        
  22:24:44.642126   | <<<------------------------ |        
        +0.204764   |             ACK             |        
  22:24:44.846890   | --------------------------> |         
        +0.010352   |             ACK             |         
  22:24:44.857242   | ------------------------>>> |         
                    |                             |

ACK в ответ на 100 Trying

Добрый день

Возникла проблема на одной АТС. Стоит сборка Elastix, сама АТС на виртуальной машине. Суть в чем, некоторые звонки оператору связи заканчиваются ошибкой 500 Internal Server Error, из-за того, что на пакет 100 Trying АТС отвечает пакетом АСК.

Reliably Transmitting (NAT) to operator_IP:5060:
INVITE sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.17.1)
Date: Tue, 04 Jul 2017 16:19:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 210920071 210920071 IN IP4 192.168.1.10
s=Asterisk PBX 11.17.1
c=IN IP4 192.168.1.10
t=0 0
m=audio 19224 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/to_Peru_SN/519961XXXXX
Retransmitting #1 (NAT) to operator_IP:5060:
INVITE sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.17.1)
Date: Tue, 04 Jul 2017 16:19:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 210920071 210920071 IN IP4 192.168.1.10
s=Asterisk PBX 11.17.1
c=IN IP4 192.168.1.10
t=0 0
m=audio 19224 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:operator_IP:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport=5060;received=192.168.1.10
From: <sip:username@192.168.1.10:5060>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 INVITE
Server: MediaCore/3.0.0
Content-Length: 0

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

<--- SIP read from UDP:operator_IP:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport=5060;received=192.168.1.10
From: <sip:username@192.168.1.10:5060>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 INVITE
WWW-Authenticate: Digest realm="MediaCore", domain="mediacore", nonce="ad2239f913f981f9e9c4179fce22868d.1499188754", stale=FALSE, algorithm=MD5
Server: MediaCore/3.0.0
Contact: <sip:519961XXXXX@operator_IP:5060>
Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO, REGISTER
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to operator_IP:5060:
ACK sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.17.1)
Content-Length: 0


---
Audio is at 19224
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to operator_IP:5060:
INVITE sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 103 INVITE
User-Agent: FPBX-2.11.0(11.17.1)
Authorization: Digest username="username_id", realm="MediaCore", algorithm=MD5, uri="sip:mediacore", nonce="ad2239f913f981f9e9c4179fce22868d.1499188754", response="secret"
Date: Tue, 04 Jul 2017 16:19:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 210920071 210920072 IN IP4 192.168.1.10
s=Asterisk PBX 11.17.1
c=IN IP4 192.168.1.10
t=0 0
m=audio 19224 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:operator_IP:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport=5060;received=192.168.1.10
From: <sip:username@192.168.1.10:5060>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 INVITE
WWW-Authenticate: Digest realm="MediaCore", domain="mediacore", nonce="ad2239f913f981f9e9c4179fce22868d.1499188754", stale=FALSE, algorithm=MD5
Server: MediaCore/3.0.0
Contact: <sip:519961XXXXX@operator_IP:5060>
Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO, REGISTER
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to operator_IP:5060:
ACK sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.17.1)
Content-Length: 0


---
Retransmitting #1 (NAT) to operator_IP:5060:
INVITE sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 103 INVITE
User-Agent: FPBX-2.11.0(11.17.1)
Authorization: Digest username="username_id", realm="MediaCore", algorithm=MD5, uri="sip:mediacore", nonce="ad2239f913f981f9e9c4179fce22868d.1499188754", response="secret"
Date: Tue, 04 Jul 2017 16:19:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 210920071 210920072 IN IP4 192.168.1.10
s=Asterisk PBX 11.17.1
c=IN IP4 192.168.1.10
t=0 0
m=audio 19224 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:operator_IP:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport=5060;received=192.168.1.10
From: <sip:username@192.168.1.10:5060>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 103 INVITE
Server: MediaCore/3.0.0
Content-Length: 0

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

<--- SIP read from UDP:operator_IP:5060 --->
SIP/2.0 500 Server Internal Error
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport=5060;received=192.168.1.10
From: <sip:username@192.168.1.10:5060>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 103 INVITE
Reason: Q.850;cause=34;text="No circuit/channel available"
Server: MediaCore/3.0.0
Contact: <sip:519961XXXXX@operator_IP:5060>
Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO, REGISTER
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
    -- Got SIP response 500 "Server Internal Error" back from operator_IP:5060
Transmitting (NAT) to operator_IP:5060:
ACK sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 103 ACK
User-Agent: FPBX-2.11.0(11.17.1)
Content-Length: 0

Насколько я понял, проблема в том, что изначально передается два инвайта, после чего начинаются две параллельные СИП-сессии, и в тот момент, когда сервер провайдера отвечает 100 Trying на INVITE, в параллельной СИП-сессии приходит АСК на пакет 401 Unauthorized

Настройки транка на Elastix

[to_Peru_SN]
host=operator_IP
username=username_id
secret=secret
type=peer
nat=force_rport,comedia
canreinvite=no
qualify=yes

К сожлению, скрин из Sngrep`a с более наглядным логом звонков приложить не могу ибо карма меньше 10. Подскажите, пожалуйста, в чем может быть проблема ?

               My IP                Operator`s IP         

                    |        INVITE (SDP)         |         
  22:24:44.187562   | --------------------------> |        
        +0.008365   |         100 Trying          |         
  22:24:44.195927   | <-------------------------- |         
        +0.005987   |      401 Unauthorized       |         
  22:24:44.201914   | <-------------------------- |         
        +0.206224   |        INVITE (SDP)         |         
  22:24:44.408138   | ------------------------>>> |         
        +0.003815   |      401 Unauthorized       |         
  22:24:44.411953   | <<<------------------------ |         
        +0.005991   |             ACK             |         
  22:24:44.417944   | --------------------------> |         
        +0.000467   |        INVITE (SDP)         |     branch=z9hG4bK23bb3ab4       
  22:24:44.418411   | --------------------------> |         
        +0.009147   |         100 Trying          |     branch=z9hG4bK23bb3ab4       
  22:24:44.427558   | <-------------------------- |         
        +0.199511   |             ACK             |      branch=z9hG4bK23bb3ab4    
  22:24:44.627069   | --------------------------> |         
        +0.004688   |  500 Server Internal Error  |     branch=z9hG4bK23bb3ab4     
  22:24:44.631757   | <-------------------------- |         
        +0.007662   |        INVITE (SDP)         |         
  22:24:44.639419   | --------------------------> |        
        +0.002707   |  500 Server Internal Error  |        
  22:24:44.642126   | <<<------------------------ |        
        +0.204764   |             ACK             |        
  22:24:44.846890   | --------------------------> |         
        +0.010352   |             ACK             |         
  22:24:44.857242   | ------------------------>>> |         
                    |                             |

ACK в ответ на 100 Trying

Добрый день

Возникла проблема на одной АТС. Стоит сборка Elastix, сама АТС на виртуальной машине. Суть в чем, некоторые звонки оператору связи заканчиваются ошибкой 500 Internal Server Error, из-за того, что на пакет 100 Trying АТС отвечает пакетом АСК.

Reliably Transmitting (NAT) to operator_IP:5060:
INVITE sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.17.1)
Date: Tue, 04 Jul 2017 16:19:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 210920071 210920071 IN IP4 192.168.1.10
s=Asterisk PBX 11.17.1
c=IN IP4 192.168.1.10
t=0 0
m=audio 19224 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/to_Peru_SN/519961XXXXX
Retransmitting #1 (NAT) to operator_IP:5060:
INVITE sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.17.1)
Date: Tue, 04 Jul 2017 16:19:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 210920071 210920071 IN IP4 192.168.1.10
s=Asterisk PBX 11.17.1
c=IN IP4 192.168.1.10
t=0 0
m=audio 19224 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:operator_IP:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport=5060;received=192.168.1.10
From: <sip:username@192.168.1.10:5060>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 INVITE
Server: MediaCore/3.0.0
Content-Length: 0

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

<--- SIP read from UDP:operator_IP:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport=5060;received=192.168.1.10
From: <sip:username@192.168.1.10:5060>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 INVITE
WWW-Authenticate: Digest realm="MediaCore", domain="mediacore", nonce="ad2239f913f981f9e9c4179fce22868d.1499188754", stale=FALSE, algorithm=MD5
Server: MediaCore/3.0.0
Contact: <sip:519961XXXXX@operator_IP:5060>
Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO, REGISTER
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to operator_IP:5060:
ACK sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.17.1)
Content-Length: 0


---
Audio is at 19224
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to operator_IP:5060:
INVITE sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 103 INVITE
User-Agent: FPBX-2.11.0(11.17.1)
Authorization: Digest username="username_id", realm="MediaCore", algorithm=MD5, uri="sip:mediacore", nonce="ad2239f913f981f9e9c4179fce22868d.1499188754", response="secret"
Date: Tue, 04 Jul 2017 16:19:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 210920071 210920072 IN IP4 192.168.1.10
s=Asterisk PBX 11.17.1
c=IN IP4 192.168.1.10
t=0 0
m=audio 19224 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:operator_IP:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47eec727;rport=5060;received=192.168.1.10
From: <sip:username@192.168.1.10:5060>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 INVITE
WWW-Authenticate: Digest realm="MediaCore", domain="mediacore", nonce="ad2239f913f981f9e9c4179fce22868d.1499188754", stale=FALSE, algorithm=MD5
Server: MediaCore/3.0.0
Contact: <sip:519961XXXXX@operator_IP:5060>
Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO, REGISTER
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to operator_IP:5060:
ACK sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.17.1)
Content-Length: 0


---
Retransmitting #1 (NAT) to operator_IP:5060:
INVITE sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 103 INVITE
User-Agent: FPBX-2.11.0(11.17.1)
Authorization: Digest username="username_id", realm="MediaCore", algorithm=MD5, uri="sip:mediacore", nonce="ad2239f913f981f9e9c4179fce22868d.1499188754", response="secret"
Date: Tue, 04 Jul 2017 16:19:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 210920071 210920072 IN IP4 192.168.1.10
s=Asterisk PBX 11.17.1
c=IN IP4 192.168.1.10
t=0 0
m=audio 19224 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:operator_IP:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport=5060;received=192.168.1.10
From: <sip:username@192.168.1.10:5060>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 103 INVITE
Server: MediaCore/3.0.0
Content-Length: 0

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

<--- SIP read from UDP:operator_IP:5060 --->
SIP/2.0 500 Server Internal Error
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport=5060;received=192.168.1.10
From: <sip:username@192.168.1.10:5060>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 103 INVITE
Reason: Q.850;cause=34;text="No circuit/channel available"
Server: MediaCore/3.0.0
Contact: <sip:519961XXXXX@operator_IP:5060>
Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO, REGISTER
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
    -- Got SIP response 500 "Server Internal Error" back from operator_IP:5060
Transmitting (NAT) to operator_IP:5060:
ACK sip:519961XXXXX@operator_IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK046728ff;rport
Max-Forwards: 70
From: <sip:username@192.168.1.10>;tag=as44b7744e
To: <sip:519961XXXXX@operator_IP>
Contact: <sip:username@192.168.1.10:5060>
Call-ID: 042127176db723121508a2f20eb22b43@192.168.1.10:5060
CSeq: 103 ACK
User-Agent: FPBX-2.11.0(11.17.1)
Content-Length: 0

Насколько я понял, проблема в том, что изначально передается два инвайта, после чего начинаются две параллельные СИП-сессии, и в тот момент, когда сервер провайдера отвечает 100 Trying на INVITE, в параллельной СИП-сессии приходит АСК на пакет 401 Unauthorized

Настройки транка на Elastix

[to_Peru_SN]
host=operator_IP
username=username_id
secret=secret
type=peer
nat=force_rport,comedia
canreinvite=no
qualify=yes

К сожлению, скрин из Sngrep`a с более наглядным логом звонков приложить не могу ибо карма меньше 10. Подскажите, пожалуйста, в чем может быть проблема ?

               My IP                Operator`s IP         

                    |        INVITE (SDP)         |        branch=z9hG4bK1eb91a47    
  22:24:44.187562   | --------------------------> |        
        +0.008365   |         100 Trying          |        branch=z9hG4bK1eb91a47    
  22:24:44.195927   | <-------------------------- |         
        +0.005987   |      401 Unauthorized       |      |<--|    branch=z9hG4bK1eb91a47    
  22:24:44.201914   | <-------------------------- |    |      
        +0.206224   |        INVITE (SDP)         |    |    branch=z9hG4bK1eb91a47    
  22:24:44.408138   | ------------------------>>> |    |      
        +0.003815   |      401 Unauthorized       |      |<--|-|  branch=z9hG4bK1eb91a47    
  22:24:44.411953   | <<<------------------------ |   | |       
        +0.005991   |             ACK             |    | |  branch=z9hG4bK1eb91a47    
  22:24:44.417944   | --------------------------> |   |---| |       
        +0.000467   |        INVITE (SDP)         |     |  branch=z9hG4bK23bb3ab4       
  22:24:44.418411   | --------------------------> |     |        
        +0.009147   |         100 Trying          |     |  branch=z9hG4bK23bb3ab4       
  22:24:44.427558   | <-------------------------- |     |        
        +0.199511   |             ACK             |     |  branch=z9hG4bK23bb3ab4    
  22:24:44.627069   | --------------------------> | |-----|         
        +0.004688   |  500 Server Internal Error  |      branch=z9hG4bK23bb3ab4     
  22:24:44.631757   | <-------------------------- |         
        +0.007662   |        INVITE (SDP)         |        branch=z9hG4bK23bb3ab4    
  22:24:44.639419   | --------------------------> |        
        +0.002707   |  500 Server Internal Error  |        branch=z9hG4bK23bb3ab4   
  22:24:44.642126   | <<<------------------------ |        
        +0.204764   |             ACK             |        branch=z9hG4bK23bb3ab4   
  22:24:44.846890   | --------------------------> |         
        +0.010352   |             ACK             |        branch=z9hG4bK23bb3ab4    
  22:24:44.857242   | ------------------------>>> |         
                    |                             |
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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.