Outbound calls end in congestion/busy with that error.
I changed voip operator. With old one it works correctly
I have not changed the format of Register string
username:[email protected]/username
Where can I can look ? Thanks
I noticed you forgot to attach the logs where the call fails. If only we had those, we wouldn’t have to guess.
The easiest way to start would be call your new provider and ask them to help you with an Asterisk implementation. They should be able to get you started.
For example, your new provider may not want you to register, or your password could be wrong, or the username could be in the wrong format, or they may be expecting the user information to come in from a different setting in the User settings, or they may be expecting a single-direction user or peer and you might be using “type=friend”.
Without more information, that’s all I’ve got for you.
Here enclosed the log.
The operator has sent me 
Username
Password
Server
STUN server
I called them and they said that they haven’t other info and should work fine.
`
<— SIP read from UDP:37.176.214.47:1024 —>
SUBSCRIBE sip:[email protected]:7778 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.29:7778;rport;branch=z9hG4bKd7b0363b64
From: sip:[email protected]:7778;tag=127e8bf8
To: sip:[email protected]:7778
Call-ID: [email protected]
Contact: sip:[email protected]:7778
CSeq: 1 SUBSCRIBE
Max-Forwards: 70
Expires: 60
Accept: application/simple-message-summary
Event: message-summary
User-Agent: CM5K-PHONE (810170)
Content-Length: 0
<------------->
— (13 headers 0 lines) —
Ignoring this SUBSCRIBE request
Found peer ‘58’ for ‘58’ from 37.176.214.47:1024
<— Transmitting (NAT) to 37.176.214.47:1024 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.29:7778;branch=z9hG4bKd7b0363b64;received=37.176.214. 47;rport=1024
From: sip:[email protected]:7778;tag=127e8bf8
To: sip:[email protected]:7778;tag=as1669d022
Call-ID: [email protected]
CSeq: 1 SUBSCRIBE
Server: FPBX-2.11.0(11.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="2d1b679b"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘[email protected] .29’ in 9728 ms (Method: SUBSCRIBE)
<— SIP read from UDP:37.176.214.47:1024 —>
SUBSCRIBE sip:[email protected]:7778 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.29:7778;rport;branch=z9hG4bKd7b0363b64
From: sip:[email protected]:7778;tag=127e8bf8
To: sip:[email protected]:7778
Call-ID: [email protected]
Contact: sip:[email protected]:7778
CSeq: 1 SUBSCRIBE
Max-Forwards: 70
Expires: 60
Accept: application/simple-message-summary
Event: message-summary
User-Agent: CM5K-PHONE (810170)
Content-Length: 0
<------------->
— (13 headers 0 lines) —
Ignoring this SUBSCRIBE request
Found peer ‘58’ for ‘58’ from 37.176.214.47:1024
<— Transmitting (NAT) to 37.176.214.47:1024 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.29:7778;branch=z9hG4bKd7b0363b64;received=37.176.214. 47;rport=1024
From: sip:[email protected]:7778;tag=127e8bf8
To: sip:[email protected]:7778;tag=as1669d022
Call-ID: [email protected]
CSeq: 1 SUBSCRIBE
Server: FPBX-2.11.0(11.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="2d1b679b"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘[email protected] .29’ in 9728 ms (Method: SUBSCRIBE)
<— SIP read from UDP:192.168.1.7:5060 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK7aca3e8c2e345664
From: “Uff.Segr” sip:[email protected];tag=8f13dfb8cf4a7339
To: sip:[email protected]
Contact: sip:[email protected]:5060;transport=udp
Supported: replaces, timer, path
P-Early-Media: Supported
Call-ID: [email protected]
CSeq: 32122 INVITE
User-Agent: Grandstream GXP2000 1.2.5.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,ME SSAGE
Content-Type: application/sdp
Content-Length: 400
v=0
o=57 8000 8000 IN IP4 192.168.1.7
s=SIP Call
c=IN IP4 192.168.1.7
t=0 0
m=audio 5036 RTP/AVP 0 8 4 18 2 97 9 3 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
<------------->
— (14 headers 19 lines) —
Sending to 192.168.1.7:5060 (NAT)
Sending to 192.168.1.7:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘57’ for ‘57’ from 192.168.1.7:5060
<— Reliably Transmitting (NAT) to 192.168.1.7:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK7aca3e8c2e345664;received=192.16 8.1.7;rport=5060
From: “Uff.Segr” sip:[email protected];tag=8f13dfb8cf4a7339
To: sip:[email protected];tag=as2a1fc1f9
Call-ID: [email protected]
CSeq: 32122 INVITE
Server: FPBX-2.11.0(11.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="3bf87e74"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 6400 ms ( Method: INVITE)
<— SIP read from UDP:192.168.1.7:5060 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK7aca3e8c2e345664
From: “Uff.Segr” sip:[email protected];tag=8f13dfb8cf4a7339
To: sip:[email protected];tag=as2a1fc1f9
Contact: sip:[email protected]:5060;transport=udp
Supported: path
Call-ID: [email protected]
CSeq: 32122 ACK
User-Agent: Grandstream GXP2000 1.2.5.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,ME SSAGE
Content-Length: 0
<------------->
— (12 headers 0 lines) —
<— SIP read from UDP:192.168.1.7:5060 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK38ef8efe3e279a00
From: “Uff.Segr” sip:[email protected];tag=8f13dfb8cf4a7339
To: sip:[email protected]
Contact: sip:[email protected]:5060;transport=udp
Supported: replaces, timer, path
P-Early-Media: Supported
Authorization: Digest username=“57”, realm=“asterisk”, algorithm=MD5, uri=“sip:7 [email protected]”, nonce=“3bf87e74”, response="f7f28cdf4be954044a00c0fd40624413 "
Call-ID: [email protected]
CSeq: 32123 INVITE
User-Agent: Grandstream GXP2000 1.2.5.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,ME SSAGE
Content-Type: application/sdp
Content-Length: 400
v=0
o=57 8000 8001 IN IP4 192.168.1.7
s=SIP Call
c=IN IP4 192.168.1.7
t=0 0
m=audio 5036 RTP/AVP 0 8 4 18 2 97 9 3 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
<------------->
— (15 headers 19 lines) —
Sending to 192.168.1.7:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘57’ for ‘57’ from 192.168.1.7:5060
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 97
Found RTP audio format 9
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 G723 for ID 4
Found audio description format G729 for ID 18
Found audio description format G726-32 for ID 2
Found audio description format iLBC for ID 97
Found audio description format G722 for ID 9
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw), peer - audio=(g723|gsm|ulaw|alaw|g726|g729|i lbc|g722)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephon e-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.7:5036
Looking for 7191 in from-internal (domain 192.168.1.135)
list_route: hop: sip:[email protected]:5060;transport=udp
<— Transmitting (NAT) to 192.168.1.7:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK38ef8efe3e279a00;received=192.16 8.1.7;rport=5060
From: “Uff.Segr” sip:[email protected];tag=8f13dfb8cf4a7339
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 32123 INVITE
Server: FPBX-2.11.0(11.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5060
Content-Length: 0
<------------>
Audio is at 10338
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 83.211.227.21:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP X.XXX.XX.XXX:5060;branch=z9hG4bK7d2b91d9;rport
Max-Forwards: 70
From: sip:[email protected];tag=as4f5c7564
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.5.0)
Date: Thu, 04 May 2017 13:53:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284
v=0
o=root 942349671 942349671 IN IP4 X.XXX.XX.XXX
s=Asterisk PBX 11.5.0
c=IN IP4 X.XXX.XX.XXX
t=0 0
m=audio 10338 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<— SIP read from UDP:83.211.227.21:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP X.XXX.XX.XXX:5060;received=X.XXX.XX.XXX;branch=z9hG4bK7d2b91d9; rport=5060
From: sip:[email protected];tag=as4f5c7564
To: sip:[email protected];tag=c040a69dfc7733bdec8c921a7a9f2d3a.371b
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm=“voip.eutelia.it”, nonce=“590b32b0b21c29056ede2 36a1f5c907e625af089”, qop="auth"
Server: SPS CI AR GW 04
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Transmitting (NAT) to 83.211.227.21:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP X.XXX.XX.XXX:5060;branch=z9hG4bK7d2b91d9;rport
Max-Forwards: 70
From: sip:[email protected];tag=as4f5c7564
To: sip:[email protected];tag=c040a69dfc7733bdec8c921a7a9f2d3a.371b
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.5.0)
Content-Length: 0
Audio is at 10338
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 83.211.227.21:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP X.XXX.XX.XXX:5060;branch=z9hG4bK0ee22ec9;rport
Max-Forwards: 70
From: sip:[email protected];tag=as4f5c7564
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 103 INVITE
User-Agent: FPBX-2.11.0(11.5.0)
Proxy-Authorization: Digest username=“0123456789”, realm=“voip.eutelia.it”, algo rithm=MD5, uri="sip:[email protected]", nonce=“590b32b0b21c29056ede236a1f5c907 e625af089”, response=“89f29fa60e1e04a4194fc7655ead482b”, qop=auth, cnonce=“61f19 d02”, nc=00000001
Date: Thu, 04 May 2017 13:53:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284
v=0
o=root 942349671 942349672 IN IP4 X.XXX.XX.XXX
s=Asterisk PBX 11.5.0
c=IN IP4 X.XXX.XX.XXX
t=0 0
m=audio 10338 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<— SIP read from UDP:83.211.227.21:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP X.XXX.XX.XXX:5060;received=X.XXX.XX.XXX;branch=z9hG4bK0ee22ec9; rport=5060
From: sip:[email protected];tag=as4f5c7564
To: sip:[email protected];tag=c040a69dfc7733bdec8c921a7a9f2d3a.5e6f
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Proxy-Authenticate: Digest realm=“voip.eutelia.it”, nonce=“590b32b0b21c29056ede2 36a1f5c907e625af089”, qop="auth"
Server: SPS CI AR GW 04
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Transmitting (NAT) to 83.211.227.21:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP X.XXX.XX.XXX:5060;branch=z9hG4bK0ee22ec9;rport
Max-Forwards: 70
From: sip:[email protected];tag=as4f5c7564
To: sip:[email protected];tag=c040a69dfc7733bdec8c921a7a9f2d3a.5e6f
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 103 ACK
User-Agent: FPBX-2.11.0(11.5.0)
Content-Length: 0
Audio is at 10338
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 83.211.227.21:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP X.XXX.XX.XXX:5060;branch=z9hG4bK48cc2637;rport
Max-Forwards: 70
From: sip:[email protected];tag=as4f5c7564
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 104 INVITE
User-Agent: FPBX-2.11.0(11.5.0)
Proxy-Authorization: Digest username=“0123456789”, realm=“voip.eutelia.it”, algo rithm=MD5, uri="sip:[email protected]", nonce=“590b32b0b21c29056ede236a1f5c907 e625af089”, response=“7e162cdb90641db2de571410240841d5”, qop=auth, cnonce=“73d2d 18d”, nc=00000002
Date: Thu, 04 May 2017 13:53:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284
v=0
o=root 942349671 942349673 IN IP4 X.XXX.XX.XXX
s=Asterisk PBX 11.5.0
c=IN IP4 X.XXX.XX.XXX
t=0 0
m=audio 10338 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<— SIP read from UDP:83.211.227.21:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP X.XXX.XX.XXX:5060;received=X.XXX.XX.XXX;branch=z9hG4bK48cc2637; rport=5060
From: sip:[email protected];tag=as4f5c7564
To: sip:[email protected];tag=c040a69dfc7733bdec8c921a7a9f2d3a.96c8
Call-ID: [email protected]:5060
CSeq: 104 INVITE
Proxy-Authenticate: Digest realm=“voip.eutelia.it”, nonce=“590b32b0b21c29056ede2 36a1f5c907e625af089”, qop="auth"
Server: SPS CI AR GW 04
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Transmitting (NAT) to 83.211.227.21:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP X.XXX.XX.XXX:5060;branch=z9hG4bK48cc2637;rport
Max-Forwards: 70
From: sip:[email protected];tag=as4f5c7564
To: sip:[email protected];tag=c040a69dfc7733bdec8c921a7a9f2d3a.96c8
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 104 ACK
User-Agent: FPBX-2.11.0(11.5.0)
Content-Length: 0
Audio is at 10338
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 83.211.227.21:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP X.XXX.XX.XXX:5060;branch=z9hG4bK488a666a;rport
Max-Forwards: 70
From: sip:[email protected];tag=as4f5c7564
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 105 INVITE
User-Agent: FPBX-2.11.0(11.5.0)
Proxy-Authorization: Digest username=“0123456789”, realm=“voip.eutelia.it”, algo rithm=MD5, uri="sip:[email protected]", nonce=“590b32b0b21c29056ede236a1f5c907 e625af089”, response=“4c7f46864f9af7eb729eaf85f1970755”, qop=auth, cnonce=“17d5d 43e”, nc=00000003
Date: Thu, 04 May 2017 13:53:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284
v=0
o=root 942349671 942349674 IN IP4 X.XXX.XX.XXX
s=Asterisk PBX 11.5.0
c=IN IP4 X.XXX.XX.XXX
t=0 0
m=audio 10338 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<— SIP read from UDP:83.211.227.21:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP X.XXX.XX.XXX:5060;received=X.XXX.XX.XXX;branch=z9hG4bK488a666a; rport=5060
From: sip:[email protected];tag=as4f5c7564
To: sip:[email protected];tag=c040a69dfc7733bdec8c921a7a9f2d3a.24cc
Call-ID: [email protected]:5060
CSeq: 105 INVITE
Proxy-Authenticate: Digest realm=“voip.eutelia.it”, nonce=“590b32b0b21c29056ede2 36a1f5c907e625af089”, qop="auth"
Server: SPS CI AR GW 04
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Transmitting (NAT) to 83.211.227.21:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP X.XXX.XX.XXX:5060;branch=z9hG4bK488a666a;rport
Max-Forwards: 70
From: sip:[email protected];tag=as4f5c7564
To: sip:[email protected];tag=c040a69dfc7733bdec8c921a7a9f2d3a.24cc
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 105 ACK
User-Agent: FPBX-2.11.0(11.5.0)
Content-Length: 0
[2017-05-04 15:53:57] NOTICE[2995][C-000001c7]: chan_sip.c:22914 handle_response _invite: Failed to authenticate on INVITE to 'sip:[email protected];tag= as4f5c7564’
Audio is at 12052
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<— Transmitting (NAT) to 192.168.1.7:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK38ef8efe3e279a00;received=192.16 8.1.7;rport=5060
From: “Uff.Segr” sip:[email protected];tag=8f13dfb8cf4a7339
To: sip:[email protected];tag=as45897561
Call-ID: [email protected]
CSeq: 32123 INVITE
Server: FPBX-2.11.0(11.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Require: timer
Content-Length: 309
v=0
o=root 246841368 246841368 IN IP4 192.168.1.135
s=Asterisk PBX 11.5.0
c=IN IP4 192.168.1.135
t=0 0
m=audio 12052 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
<— SIP read from UDP:37.176.214.47:7778 —>
<------------->
Really destroying SIP dialog '[email protected]:5060 ’ Method: INVITE
<— SIP read from UDP:2.238.70.138:64378 —>
<------------->
<— SIP read from UDP:37.176.214.47:1024 —>
SUBSCRIBE sip:[email protected]:7778 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.29:7778;rport;branch=z9hG4bKd7b0363b64
From: sip:[email protected]:7778;tag=127e8bf8
To: sip:[email protected]:7778
Call-ID: [email protected]
Contact: sip:[email protected]:7778
CSeq: 1 SUBSCRIBE
Max-Forwards: 70
Expires: 60
Accept: application/simple-message-summary
Event: message-summary
User-Agent: CM5K-PHONE (810170)
Content-Length: 0
<------------->
— (13 headers 0 lines) —
Ignoring this SUBSCRIBE request
Found peer ‘58’ for ‘58’ from 37.176.214.47:1024
<— Transmitting (NAT) to 37.176.214.47:1024 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.29:7778;branch=z9hG4bKd7b0363b64;received=37.176.214. 47;rport=1024
From: sip:[email protected]:7778;tag=127e8bf8
To: sip:[email protected]:7778;tag=as1669d022
Call-ID: [email protected]
CSeq: 1 SUBSCRIBE
Server: FPBX-2.11.0(11.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="2d1b679b"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘[email protected] .29’ in 9728 ms (Method: SUBSCRIBE)
<— SIP read from UDP:192.168.1.7:5060 —>
CANCEL sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK38ef8efe3e279a00
From: “Uff.Segr” sip:[email protected];tag=8f13dfb8cf4a7339
To: sip:[email protected]
Supported: path
Call-ID: [email protected]
CSeq: 32123 CANCEL
User-Agent: Grandstream GXP2000 1.2.5.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,ME SSAGE
Content-Length: 0
<------------->
— (11 headers 0 lines) —
Sending to 192.168.1.7:5060 (NAT)
<— Reliably Transmitting (NAT) to 192.168.1.7:5060 —>
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK38ef8efe3e279a00;received=192.16 8.1.7;rport=5060
From: “Uff.Segr” sip:[email protected];tag=8f13dfb8cf4a7339
To: sip:[email protected];tag=as45897561
Call-ID: [email protected]
CSeq: 32123 INVITE
Server: FPBX-2.11.0(11.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H
Supported: replaces, timer
Content-Length: 0
<------------>
<— Transmitting (NAT) to 192.168.1.7:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK38ef8efe3e279a00;received=192.16 8.1.7;rport=5060
From: “Uff.Segr” sip:[email protected];tag=8f13dfb8cf4a7339
To: sip:[email protected];tag=as45897561
Call-ID: [email protected]
CSeq: 32123 CANCEL
Server: FPBX-2.11.0(11.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H
Supported: replaces, timer
Content-Length: 0
<------------>
Retransmitting #1 (NAT) to 192.168.1.7:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK38ef8efe3e279a00;received=192.16 8.1.7;rport=5060
From: “Uff.Segr” sip:[email protected];tag=8f13dfb8cf4a7339
To: sip:[email protected];tag=as45897561
Call-ID: [email protected]
CSeq: 32123 INVITE
Server: FPBX-2.11.0(11.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H
Supported: replaces, timer
Content-Length: 0
<— SIP read from UDP:192.168.1.7:5060 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK38ef8efe3e279a00
From: “Uff.Segr” sip:[email protected];tag=8f13dfb8cf4a7339
To: sip:[email protected];tag=as45897561
Contact: sip:[email protected]:5060;transport=udp
Supported: path
Authorization: Digest username=“57”, realm=“asterisk”, algorithm=MD5, uri=“sip:7 [email protected]”, nonce=“3bf87e74”, response="f7f28cdf4be954044a00c0fd40624413 "
Call-ID: [email protected]
CSeq: 32123 ACK
User-Agent: Grandstream GXP2000 1.2.5.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,ME SSAGE
Content-Length: 0
<------------->
— (13 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]’ Method: ACK
Really destroying SIP dialog ‘2634574349@192_168_1_133’ Method: REGISTER
<— SIP read from UDP:37.176.214.47:1024 —>
SUBSCRIBE sip:[email protected]:7778 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.29:7778;rport;branch=z9hG4bKd7b0363b64
From: sip:[email protected]:7778;tag=127e8bf8
To: sip:[email protected]:7778
Call-ID: [email protected]
Contact: sip:[email protected]:7778
CSeq: 1 SUBSCRIBE
Max-Forwards: 70
Expires: 60
Accept: application/simple-message-summary
Event: message-summary
User-Agent: CM5K-PHONE (810170)
Content-Length: 0
<------------->
— (13 headers 0 lines) —
Ignoring this SUBSCRIBE request
Found peer ‘58’ for ‘58’ from 37.176.214.47:1024
<— Transmitting (NAT) to 37.176.214.47:1024 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.29:7778;branch=z9hG4bKd7b0363b64;received=37.176.214. 47;rport=1024
From: sip:[email protected]:7778;tag=127e8bf8
To: sip:[email protected]:7778;tag=as1669d022
Call-ID: [email protected]
CSeq: 1 SUBSCRIBE
Server: FPBX-2.11.0(11.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="2d1b679b"
Content-Length: 0
`
There’s something interesting here.
Are you sure you have your NAT set up correctly? Asterisk doesn’t use STUN, so you have to make sure your internal and external addresses are correct in the Advanced Settings → SIP configuration tab.
yes of course!
With Eutelia Clouditalia and FreeVoipDeal no problem.
What is wrong here from the log?
In the SIP Debug, you’ve got everything communicating with 192.168.0.x, but in your screenshot, you say the local network is 192.168.1.x.
Which is it?
Ah ok. Dont worry. That address is related to others phones joined to the same pbx but in another LAN . This is correct
so, any other suggestions?
The problem is that my asterisk requires a
Proxy Authentication because it receives an error 407.
I found on a website this meaning:
Failure to authenticate - 401 or 407?
If the origin server does not wish to accept the credentials sent with a request, it SHOULD return a 401 (Unauthorized) response. The response MUST include a WWW-Authenticate header field containing at least one (possibly new) challenge applicable to the requested resource. If a proxy does not accept the credentials sent with a request, it SHOULD return a 407 (Proxy Authentication Required). The response MUST include a Proxy-Authenticate header field containing a (possibly new) challenge applicable to the proxy for the requested resource.
Moreover appears that a
Proxy-Authorization: Digest username=“0123456789”, realm=“voip.eutelia.it”, algorithm=MD5, uri="sip:[email protected]"
Why in realm there is my other trunk ? eutelia? This one is vivavox.it…
Most likely you have your trunk settings wrong, perhaps mixing settings from your two providers.
Also you have obfuscated your logs too much with XXXXX, making it difficult to help.
Yes… I can post details… Anyway, I have not confused providers, only probably also ehiweb relies on eutelia behind the scenes
Thank you! You solved my issue years later…
I was coming crazy not understanding the reason of this issue affecting only the outgoing calls!
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