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Failed incoming calls from asterisk bridge

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Soporte Sinet Colombia

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Oct 16, 2017
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Greetings,

I have a Linux 3CX 15.5 connected with an Asterisk 1.8 through bridge. Registration and calls from 3CX to Asterisk are OK, but calls from Asterisk to 3CX stay Trying. Servers are connected throuhg VPN and located in different cities. Some details:
  • Asterisk box -> 192.168.21.200/24
  • 3CX box -> 147.122.1.200/16
Calling to 3CX 607 extension:
192.168.21.200:5060 147.122.1.200:5060
=================== ==================
INVITE (SDP)
──────────────>
INVITE (SDP)
──────────────>>>
407 Proxy Authentication R
<────────────────
ACK
────────────────>
INVITE (SDP)
────────────────>
407 Proxy Authentication R
<<<──────────────
ACK
────────────────>
INVITE (SDP)
────────────────>
100 Trying
<───────────────
100 Trying
<<<─────────────
 
What does the 3CX Activity Log show for the incoming call/ Does it even show up?
 
Hi leejor, in 3CX log we only see this:

10/17/2017 8:22:10 AM - Source is identified as trunk Lc:10002(@BRIDGE_DIALVOX_CALI[<sip:[email protected]:5060>])
10/17/2017 8:22:10 AM - Source is identified as trunk Lc:10002(@BRIDGE_DIALVOX_CALI[<sip:[email protected]:5060>])
10/17/2017 8:22:10 AM - IPs do not match!
10/17/2017 8:22:10 AM - Compare IPs: incoming=192.168.21.200; external=0.0.0.0
10/17/2017 8:22:10 AM - IPs do not match!
10/17/2017 8:22:10 AM - Compare IPs: incoming=192.168.21.200; external=200.xxx.xxx.xxx

Using sngrep to capture traffic, we saw call stuck in Trying.
 
inside of the 3CX logs you will not be able to determine why the Asterisk remains in Trying.
You need to debug the Asterisk logs to get an idea where the call would be routed to and why there is no response...
 
Hi StefanW, this is what I see from asterisk side:

2017/10/17 10:12:13.744656 192.168.21.200:5060 -> 147.122.1.200:5060
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK59e507ac;rport
Max-Forwards: 70
From: "Adminfo" <sip:[email protected]>;tag=as78f7823c
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.17.0
Date: Tue, 17 Oct 2017 15:12:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: "Adminfo" <sip:[email protected]>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 999649886 999649886 IN IP4 192.168.21.200
s=Asterisk PBX 1.8.17.0
c=IN IP4 192.168.21.200
t=0 0
m=audio 18896 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/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

2017/10/17 10:12:13.855536 192.168.21.200:5060 -> 147.122.1.200:5060
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK59e507ac;rport
Max-Forwards: 70
From: "Adminfo" <sip:[email protected]>;tag=as78f7823c
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.17.0
Date: Tue, 17 Oct 2017 15:12:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: "Adminfo" <sip:[email protected]>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 999649886 999649886 IN IP4 192.168.21.200
s=Asterisk PBX 1.8.17.0
c=IN IP4 192.168.21.200
t=0 0
m=audio 18896 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/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

2017/10/17 10:12:13.870059 147.122.1.200:5060 -> 192.168.21.200:5060
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK59e507ac;rport=5060
Proxy-Authenticate: Digest nonce="414d535959e61dcf31:224747789b410d7e54d28f0856656764",algorithm=MD5,
alm="3CXPhoneSystem"
To: <sip:[email protected]:5060>;tag=59c6677f
From: "Adminfo" <sip:[email protected]>;tag=as78f7823c
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: 3CXPhoneSystem 15.5.3849.1 (3849)
Content-Length: 0

2017/10/17 10:12:13.870213 192.168.21.200:5060 -> 147.122.1.200:5060
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK59e507ac;rport
Max-Forwards: 70
From: "Adminfo" <sip:[email protected]>;tag=as78f7823c
To: <sip:[email protected]:5060>;tag=59c6677f
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.17.0
Content-Length: 0

2017/10/17 10:12:13.870400 192.168.21.200:5060 -> 147.122.1.200:5060
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK5f81d547;rport
Max-Forwards: 70
From: "Adminfo" <sip:[email protected]>;tag=as78f7823c
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.17.0
Proxy-Authorization: Digest username="10002", realm="3CXPhoneSystem", algorithm=MD5, uri="sip:000@147
22.1.200:5060", nonce="414d535959e61dcf31:224747789b410d7e54d28f0856656764", response="c2593e4e47f3a0
bc8669f33e198e8a"
Date: Tue, 17 Oct 2017 15:12:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: "Adminfo" <sip:[email protected]>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 999649886 999649887 IN IP4 192.168.21.200
s=Asterisk PBX 1.8.17.0
c=IN IP4 192.168.21.200
t=0 0
m=audio 18896 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/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

2017/10/17 10:12:13.881841 147.122.1.200:5060 -> 192.168.21.200:5060
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK59e507ac;rport=5060
Proxy-Authenticate: Digest nonce="414d535959e61dcf31:224747789b410d7e54d28f0856656764",algorithm=MD5,
alm="3CXPhoneSystem"
To: <sip:[email protected]:5060>;tag=59c6677f
From: "Adminfo" <sip:[email protected]>;tag=as78f7823c
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: 3CXPhoneSystem 15.5.3849.1 (3849)
Content-Length: 0

2017/10/17 10:12:13.881985 192.168.21.200:5060 -> 147.122.1.200:5060
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK5f81d547;rport
Max-Forwards: 70
From: "Adminfo" <sip:[email protected]>;tag=as78f7823c
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.17.0
Content-Length: 0

2017/10/17 10:12:13.980507 192.168.21.200:5060 -> 147.122.1.200:5060
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK5f81d547;rport
Max-Forwards: 70
From: "Adminfo" <sip:[email protected]>;tag=as78f7823c
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.17.0
Proxy-Authorization: Digest username="10002", realm="3CXPhoneSystem", algorithm=MD5, uri="sip:000@147
22.1.200:5060", nonce="414d535959e61dcf31:224747789b410d7e54d28f0856656764", response="c2593e4e47f3a0
bc8669f33e198e8a"
Date: Tue, 17 Oct 2017 15:12:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: "Adminfo" <sip:[email protected]>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 999649886 999649887 IN IP4 192.168.21.200
s=Asterisk PBX 1.8.17.0
c=IN IP4 192.168.21.200
t=0 0
m=audio 18896 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/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

2017/10/17 10:12:14.020568 147.122.1.200:5060 -> 192.168.21.200:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK5f81d547;rport=5060
To: <sip:[email protected]:5060>
From: "Adminfo" <sip:[email protected]>;tag=as78f7823c
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Content-Length: 0

2017/10/17 10:12:14.045784 147.122.1.200:5060 -> 192.168.21.200:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK5f81d547;rport=5060
To: <sip:[email protected]:5060>
From: "Adminfo" <sip:[email protected]>;tag=as78f7823c
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Content-Length: 0
 
Hi leejor,

Yes, it's a valid extension. I'm attaching a log for extension 604 anyway.

2017/10/17 11:13:47.885228 192.168.21.200:5060 -> 147.122.1.200:5060
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK0af942b7;rport
Max-Forwards: 70
From: "Adminfo" <sip:[email protected]>;tag=as1e851238
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>

Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.17.0
Date: Tue, 17 Oct 2017 16:13:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: "Adminfo" <sip:[email protected]>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 139235737 139235737 IN IP4 192.168.21.200
s=Asterisk PBX 1.8.17.0
c=IN IP4 192.168.21.200
t=0 0
m=audio 17500 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/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

2017/10/17 11:13:48.028263 192.168.21.200:5060 -> 147.122.1.200:5060
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK0af942b7;rport
Max-Forwards: 70
From: "Adminfo" <sip:[email protected]>;tag=as1e851238
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.17.0
Date: Tue, 17 Oct 2017 16:13:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: "Adminfo" <sip:[email protected]>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 139235737 139235737 IN IP4 192.168.21.200
s=Asterisk PBX 1.8.17.0
c=IN IP4 192.168.21.200
t=0 0
m=audio 17500 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/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

2017/10/17 11:13:48.028761 147.122.1.200:5060 -> 192.168.21.200:5060
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK0af942b7;rport=5060
Proxy-Authenticate: Digest nonce="414d535959e62c3e38:7092c032f85c525ca9f0a5d56147aedb",algorithm=MD5,
alm="3CXPhoneSystem"
To: <sip:[email protected]:5060>;tag=7c726410
From: "Adminfo" <sip:[email protected]>;tag=as1e851238
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: 3CXPhoneSystem 15.5.3849.1 (3849)
Content-Length: 0

2017/10/17 11:13:48.028895 192.168.21.200:5060 -> 147.122.1.200:5060
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK0af942b7;rport
Max-Forwards: 70
From: "Adminfo" <sip:[email protected]>;tag=as1e851238
To: <sip:[email protected]:5060>;tag=7c726410
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.17.0
Content-Length: 0

2017/10/17 11:13:48.029069 192.168.21.200:5060 -> 147.122.1.200:5060
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK78f4c9de;rport
Max-Forwards: 70
From: "Adminfo" <sip:[email protected]>;tag=as1e851238
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.17.0
Proxy-Authorization: Digest username="10002", realm="3CXPhoneSystem", algorithm=MD5, uri="sip:604@147
22.1.200:5060", nonce="414d535959e62c3e38:7092c032f85c525ca9f0a5d56147aedb", response="793ff613c03452
682c776c03942ab8"
Date: Tue, 17 Oct 2017 16:13:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: "Adminfo" <sip:[email protected]>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 139235737 139235738 IN IP4 192.168.21.200
s=Asterisk PBX 1.8.17.0
c=IN IP4 192.168.21.200
t=0 0
m=audio 17500 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/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

2017/10/17 11:13:48.046709 147.122.1.200:5060 -> 192.168.21.200:5060
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK0af942b7;rport=5060
Proxy-Authenticate: Digest nonce="414d535959e62c3e38:7092c032f85c525ca9f0a5d56147aedb",algorithm=MD5,
alm="3CXPhoneSystem"
To: <sip:[email protected]:5060>;tag=7c726410
From: "Adminfo" <sip:[email protected]>;tag=as1e851238
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: 3CXPhoneSystem 15.5.3849.1 (3849)
Content-Length: 0

2017/10/17 11:13:48.046830 192.168.21.200:5060 -> 147.122.1.200:5060
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK78f4c9de;rport
Max-Forwards: 70
From: "Adminfo" <sip:[email protected]>;tag=as1e851238
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.17.0
Content-Length: 0

2017/10/17 11:13:48.172066 192.168.21.200:5060 -> 147.122.1.200:5060
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK78f4c9de;rport
Max-Forwards: 70
From: "Adminfo" <sip:[email protected]>;tag=as1e851238
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.17.0
Proxy-Authorization: Digest username="10002", realm="3CXPhoneSystem", algorithm=MD5, uri="sip:604@147
22.1.200:5060", nonce="414d535959e62c3e38:7092c032f85c525ca9f0a5d56147aedb", response="793ff613c03452
682c776c03942ab8"
Date: Tue, 17 Oct 2017 16:13:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: "Adminfo" <sip:[email protected]>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 139235737 139235738 IN IP4 192.168.21.200
s=Asterisk PBX 1.8.17.0
c=IN IP4 192.168.21.200
t=0 0
m=audio 17500 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/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

2017/10/17 11:13:48.184206 147.122.1.200:5060 -> 192.168.21.200:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK78f4c9de;rport=5060
To: <sip:[email protected]:5060>
From: "Adminfo" <sip:[email protected]>;tag=as1e851238
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Content-Length: 0


2017/10/17 11:13:48.234734 147.122.1.200:5060 -> 192.168.21.200:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.21.200:5060;branch=z9hG4bK78f4c9de;rport=5060
To: <sip:[email protected]:5060>
From: "Adminfo" <sip:[email protected]>;tag=as1e851238
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Content-Length: 0
 
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