Hi,
I’m configuration asterisk with voys service. I get busy/congestion errors for incoming calls. The outgoing call is working fine.
I get “SIP/2.0 503 Service Unavailable” errors. There are many instances of “SIP/2.0 100 Trying” and “SIP/2.0 407 Proxy Authentication Required” requests.
Following is the detailed debug lines.
<--- SIP read from UDP:195.35.114.119:6060 --->
INVITE sip:+31YYYYYYY498@95.211.XXX.XXX:5060 SIP/2.0
Record-Route: <sip:195.35.114.119:6060;r2=on;lr;ftag=as162de247;cc=39303438;eu=7369703a39352e3231312e3133322e3130373a35303630;did=b9a.b758105>
Record-Route: <sip:195.35.114.119;r2=on;lr;ftag=as162de247;cc=39303438;eu=7369703a39352e3231312e3133322e3130373a35303630;did=b9a.b758105>
Via: SIP/2.0/UDP 195.35.114.119:6060;branch=z9hG4bKb69b.ab49e274.0
Via: SIP/2.0/UDP 195.35.114.100:5060;received=195.35.114.100;branch=z9hG4bK274b392d;rport=5060
Max-Forwards: 70
From: <sip:2ZZZZZ0001@195.35.114.100>;tag=as162de247
To: <sip:+31YYYYYYY498@voipgrid.nl:5060>
Contact: <sip:2ZZZZZ0001@195.35.114.100:5060>
Call-ID: 27df162f346e49aa5646db6b7c3c9900@195.35.114.100:5060
CSeq: 102 INVITE
User-Agent: VGUA-
Date: Thu, 17 Apr 2014 11:29:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: <sip:0091994YYYYY77@voipgrid.nl>;privacy=off;screen=yes
Content-Type: application/sdp
Content-Length: 291
v=0
o=- 389444878 389444878 IN IP4 195.35.114.100
s=VGUA-
c=IN IP4 195.35.114.100
t=0 0
m=audio 15536 RTP/AVP 8 9 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
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
<------------->
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: --- (18 headers 14 lines) ---
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: Sending to 195.35.114.119:6060 (no NAT)
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: Using INVITE request as basis request - 27df162f346e49aa5646db6b7c3c9900@195.35.114.100:5060
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: Found peer '2ZZZZZ0001' for '2ZZZZZ0001' from 195.35.114.119:6060
[Apr 17 13:24:21] VERBOSE[16201] netsock2.c: == Using SIP RTP CoS mark 5
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: Found RTP audio format 8
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: Found RTP audio format 9
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: Found RTP audio format 18
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: Found RTP audio format 101
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: Found audio description format PCMA for ID 8
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: Found audio description format G722 for ID 9
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: Found audio description format G729 for ID 18
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: Found audio description format telephone-event for ID 101
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: Capabilities: us - 0x8 (alaw), peer - audio=0x1108 (alaw|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: Peer audio RTP is at port 195.35.114.100:15536
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: Looking for +31YYYYYYY498 in internal (domain 95.211.XXX.XXX)
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: list_route: hop: <sip:195.35.114.119:6060;r2=on;lr;ftag=as162de247;cc=39303438;eu=7369703a39352e3231312e3133322e3130373a35303630;did=b9a.b758105>
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: list_route: hop: <sip:195.35.114.119;r2=on;lr;ftag=as162de247;cc=39303438;eu=7369703a39352e3231312e3133322e3130373a35303630;did=b9a.b758105>
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c:
<--- Transmitting (no NAT) to 195.35.114.119:6060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 195.35.114.119:6060;branch=z9hG4bKb69b.ab49e274.0;received=195.35.114.119
Via: SIP/2.0/UDP 195.35.114.100:5060;received=195.35.114.100;branch=z9hG4bK274b392d;rport=5060
Record-Route: <sip:195.35.114.119:6060;r2=on;lr;ftag=as162de247;cc=39303438;eu=7369703a39352e3231312e3133322e3130373a35303630;did=b9a.b758105>
Record-Route: <sip:195.35.114.119;r2=on;lr;ftag=as162de247;cc=39303438;eu=7369703a39352e3231312e3133322e3130373a35303630;did=b9a.b758105>
From: <sip:2ZZZZZ0001@195.35.114.100>;tag=as162de247
To: <sip:+31YYYYYYY498@voipgrid.nl:5060>
Call-ID: 27df162f346e49aa5646db6b7c3c9900@195.35.114.100:5060
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:+31YYYYYYY498@95.211.XXX.XXX:5060>
Content-Length: 0
<------------>
[Apr 17 13:24:21] VERBOSE[16310] pbx.c: -- Executing [+31YYYYYYY498@internal:1] Dial("SIP/2ZZZZZ0001-00000000", "SIP/+31YYYYYYY498@2ZZZZZ0001,60,t") in new stack
[Apr 17 13:24:21] VERBOSE[16310] netsock2.c: == Using SIP RTP CoS mark 5
[Apr 17 13:24:21] VERBOSE[16310] chan_sip.c: Audio is at 11052
[Apr 17 13:24:21] VERBOSE[16310] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Apr 17 13:24:21] VERBOSE[16310] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Apr 17 13:24:21] VERBOSE[16310] chan_sip.c: Reliably Transmitting (no NAT) to 195.35.114.119:6060:
INVITE sip:+31YYYYYYY498@sipproxy.voipgrid.nl SIP/2.0
Via: SIP/2.0/UDP 95.211.XXX.XXX:5060;branch=z9hG4bK224ac8fd
Max-Forwards: 70
From: "0091994YYYYY77" <sip:2ZZZZZ0001@95.211.XXX.XXX>;tag=as350b5c59
To: <sip:+31YYYYYYY498@sipproxy.voipgrid.nl>
Contact: <sip:2ZZZZZ0001@95.211.XXX.XXX:5060>
Call-ID: 73b205fd474bfc8705160c2f276f71ae@95.211.XXX.XXX:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Date: Thu, 17 Apr 2014 11:24:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Remote-Party-ID: "0091994YYYYY77" <sip:0091994YYYYY77@95.211.XXX.XXX>;party=calling;privacy=off;screen=yes
Content-Type: application/sdp
Content-Length: 255
v=0
o=root 1305983722 1305983722 IN IP4 95.211.XXX.XXX
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 95.211.XXX.XXX
t=0 0
m=audio 11052 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
[Apr 17 13:24:21] VERBOSE[16310] app_dial.c: -- Called SIP/+31YYYYYYY498@2ZZZZZ0001
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c:
<--- SIP read from UDP:195.35.114.119:6060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 95.211.XXX.XXX:5060;branch=z9hG4bK224ac8fd
From: "0091994YYYYY77" <sip:2ZZZZZ0001@95.211.XXX.XXX>;tag=as350b5c59
To: <sip:+31YYYYYYY498@sipproxy.voipgrid.nl>
Call-ID: 73b205fd474bfc8705160c2f276f71ae@95.211.XXX.XXX:5060
CSeq: 102 INVITE
Server: VGPXY-e2d9677m-OS
Content-Length: 0
<------------->
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: --- (8 headers 0 lines) ---
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c:
<--- SIP read from UDP:195.35.114.119:6060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 95.211.XXX.XXX:5060;received=95.211.XXX.XXX;rport=5060;branch=z9hG4bK224ac8fd
From: "0091994YYYYY77" <sip:2ZZZZZ0001@95.211.XXX.XXX>;tag=as350b5c59
To: <sip:+31YYYYYYY498@sipproxy.voipgrid.nl>;tag=9e207c4db3df57e12fedd0328727cc06-b126
Call-ID: 73b205fd474bfc8705160c2f276f71ae@95.211.XXX.XXX:5060
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="voipgrid.nl", nonce="534fbb510000bdb44ce42807939e86a102501ab2b800ffbc"
Server: VGPXY-e2d9677m-OS
Content-Length: 0
<------------->
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: --- (9 headers 0 lines) ---
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: set_destination: Parsing <sip:+31YYYYYYY498@sipproxy.voipgrid.nl> for address/port to send to
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: set_destination: set destination to 195.35.115.119:5060
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: Transmitting (no NAT) to 195.35.115.119:5060:
ACK sip:+31YYYYYYY498@sipproxy.voipgrid.nl SIP/2.0
Via: SIP/2.0/UDP 95.211.XXX.XXX:5060;branch=z9hG4bK224ac8fd
Max-Forwards: 70
From: "0091994YYYYY77" <sip:2ZZZZZ0001@95.211.XXX.XXX>;tag=as350b5c59
To: <sip:+31YYYYYYY498@sipproxy.voipgrid.nl>;tag=9e207c4db3df57e12fedd0328727cc06-b126
Contact: <sip:2ZZZZZ0001@95.211.XXX.XXX:5060>
Call-ID: 73b205fd474bfc8705160c2f276f71ae@95.211.XXX.XXX:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Content-Length: 0
---
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: Audio is at 11052
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: Reliably Transmitting (no NAT) to 195.35.115.119:5060:
INVITE sip:+31YYYYYYY498@sipproxy.voipgrid.nl SIP/2.0
Via: SIP/2.0/UDP 95.211.XXX.XXX:5060;branch=z9hG4bK20587496
Max-Forwards: 70
From: "0091994YYYYY77" <sip:2ZZZZZ0001@95.211.XXX.XXX>;tag=as350b5c59
To: <sip:+31YYYYYYY498@sipproxy.voipgrid.nl>
Contact: <sip:2ZZZZZ0001@95.211.XXX.XXX:5060>
Call-ID: 73b205fd474bfc8705160c2f276f71ae@95.211.XXX.XXX:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Proxy-Authorization: Digest username="2ZZZZZ0001", realm="voipgrid.nl", algorithm=MD5, uri="sip:+31YYYYYYY498@sipproxy.voipgrid.nl", nonce="534fbb510000bdb44ce42807939e86a102501ab2b800ffbc", response="3a7a504285d0719e9c2522b056c4351f"
Date: Thu, 17 Apr 2014 11:24:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Remote-Party-ID: "0091994YYYYY77" <sip:0091994YYYYY77@95.211.XXX.XXX>;party=calling;privacy=off;screen=yes
Content-Type: application/sdp
Content-Length: 255
v=0
o=root 1305983722 1305983723 IN IP4 95.211.XXX.XXX
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 95.211.XXX.XXX
t=0 0
m=audio 11052 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c:
<--- SIP read from UDP:195.35.115.119:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 95.211.XXX.XXX:5060;branch=z9hG4bK20587496
From: "0091994YYYYY77" <sip:2ZZZZZ0001@95.211.XXX.XXX>;tag=as350b5c59
To: <sip:+31YYYYYYY498@sipproxy.voipgrid.nl>
Call-ID: 73b205fd474bfc8705160c2f276f71ae@95.211.XXX.XXX:5060
CSeq: 103 INVITE
Server: VGPXY-8a541e8m-OS
Content-Length: 0
<------------->
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: --- (8 headers 0 lines) ---
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c:
<--- SIP read from UDP:195.35.115.119:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 95.211.XXX.XXX:5060;received=95.211.XXX.XXX;rport=5060;branch=z9hG4bK20587496
From: "0091994YYYYY77" <sip:2ZZZZZ0001@95.211.XXX.XXX>;tag=as350b5c59
To: <sip:+31YYYYYYY498@sipproxy.voipgrid.nl>;tag=32bbfaed647d04e1dda1989fc0442882-c765
Call-ID: 73b205fd474bfc8705160c2f276f71ae@95.211.XXX.XXX:5060
CSeq: 103 INVITE
Proxy-Authenticate: Digest realm="voipgrid.nl", nonce="534fbb51000166f6f1bf77571eb98caccd7b5b5c7314fe82", stale=true
Server: VGPXY-8a541e8m-OS
Content-Length: 0
<------------->
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: --- (9 headers 0 lines) ---
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: set_destination: Parsing <sip:+31YYYYYYY498@sipproxy.voipgrid.nl> for address/port to send to
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: set_destination: set destination to 195.35.114.119:5060
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: Transmitting (no NAT) to 195.35.114.119:5060:
ACK sip:+31YYYYYYY498@sipproxy.voipgrid.nl SIP/2.0
Via: SIP/2.0/UDP 95.211.XXX.XXX:5060;branch=z9hG4bK20587496
Max-Forwards: 70
From: "0091994YYYYY77" <sip:2ZZZZZ0001@95.211.XXX.XXX>;tag=as350b5c59
To: <sip:+31YYYYYYY498@sipproxy.voipgrid.nl>;tag=32bbfaed647d04e1dda1989fc0442882-c765
Contact: <sip:2ZZZZZ0001@95.211.XXX.XXX:5060>
Call-ID: 73b205fd474bfc8705160c2f276f71ae@95.211.XXX.XXX:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Content-Length: 0
---
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: Audio is at 11052
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: Reliably Transmitting (no NAT) to 195.35.114.119:5060:
INVITE sip:+31YYYYYYY498@sipproxy.voipgrid.nl SIP/2.0
Via: SIP/2.0/UDP 95.211.XXX.XXX:5060;branch=z9hG4bK2b6f4f8d
Max-Forwards: 70
From: "0091994YYYYY77" <sip:2ZZZZZ0001@95.211.XXX.XXX>;tag=as350b5c59
To: <sip:+31YYYYYYY498@sipproxy.voipgrid.nl>
Contact: <sip:2ZZZZZ0001@95.211.XXX.XXX:5060>
Call-ID: 73b205fd474bfc8705160c2f276f71ae@95.211.XXX.XXX:5060
CSeq: 104 INVITE
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Proxy-Authorization: Digest username="2ZZZZZ0001", realm="voipgrid.nl", algorithm=MD5, uri="sip:+31YYYYYYY498@sipproxy.voipgrid.nl", nonce="534fbb51000166f6f1bf77571eb98caccd7b5b5c7314fe82", response="162df9bc701d59e6b08ebc04daf1fdf1"
Date: Thu, 17 Apr 2014 11:24:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Remote-Party-ID: "0091994YYYYY77" <sip:0091994YYYYY77@95.211.XXX.XXX>;party=calling;privacy=off;screen=yes
Content-Type: application/sdp
Content-Length: 255
v=0
o=root 1305983722 1305983724 IN IP4 95.211.XXX.XXX
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 95.211.XXX.XXX
t=0 0
m=audio 11052 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c:
<--- SIP read from UDP:195.35.114.119:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 95.211.XXX.XXX:5060;branch=z9hG4bK2b6f4f8d
From: "0091994YYYYY77" <sip:2ZZZZZ0001@95.211.XXX.XXX>;tag=as350b5c59
To: <sip:+31YYYYYYY498@sipproxy.voipgrid.nl>
Call-ID: 73b205fd474bfc8705160c2f276f71ae@95.211.XXX.XXX:5060
CSeq: 104 INVITE
Server: VGPXY-e2d9677m-OS
Content-Length: 0
<------------->
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: --- (8 headers 0 lines) ---
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c:
<--- SIP read from UDP:195.35.114.119:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 95.211.XXX.XXX:5060;received=95.211.XXX.XXX;rport=5060;branch=z9hG4bK2b6f4f8d
From: "0091994YYYYY77" <sip:2ZZZZZ0001@95.211.XXX.XXX>;tag=as350b5c59
To: <sip:+31YYYYYYY498@sipproxy.voipgrid.nl>;tag=9e207c4db3df57e12fedd0328727cc06-6707
Call-ID: 73b205fd474bfc8705160c2f276f71ae@95.211.XXX.XXX:5060
CSeq: 104 INVITE
Proxy-Authenticate: Digest realm="voipgrid.nl", nonce="534fbb510000bdb500f6d77db195321c4905df3188dfaed3", stale=true
Server: VGPXY-e2d9677m-OS
Content-Length: 0
<------------->
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: --- (9 headers 0 lines) ---
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: set_destination: Parsing <sip:+31YYYYYYY498@sipproxy.voipgrid.nl> for address/port to send to
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: set_destination: set destination to 195.35.115.119:5060
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: Transmitting (no NAT) to 195.35.115.119:5060:
ACK sip:+31YYYYYYY498@sipproxy.voipgrid.nl SIP/2.0
Via: SIP/2.0/UDP 95.211.XXX.XXX:5060;branch=z9hG4bK2b6f4f8d
Max-Forwards: 70
From: "0091994YYYYY77" <sip:2ZZZZZ0001@95.211.XXX.XXX>;tag=as350b5c59
To: <sip:+31YYYYYYY498@sipproxy.voipgrid.nl>;tag=9e207c4db3df57e12fedd0328727cc06-6707
Contact: <sip:2ZZZZZ0001@95.211.XXX.XXX:5060>
Call-ID: 73b205fd474bfc8705160c2f276f71ae@95.211.XXX.XXX:5060
CSeq: 104 ACK
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Content-Length: 0
---
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: Audio is at 11052
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: Reliably Transmitting (no NAT) to 195.35.115.119:5060:
INVITE sip:+31YYYYYYY498@sipproxy.voipgrid.nl SIP/2.0
Via: SIP/2.0/UDP 95.211.XXX.XXX:5060;branch=z9hG4bK6f3e4288
Max-Forwards: 70
From: "0091994YYYYY77" <sip:2ZZZZZ0001@95.211.XXX.XXX>;tag=as350b5c59
To: <sip:+31YYYYYYY498@sipproxy.voipgrid.nl>
Contact: <sip:2ZZZZZ0001@95.211.XXX.XXX:5060>
Call-ID: 73b205fd474bfc8705160c2f276f71ae@95.211.XXX.XXX:5060
CSeq: 105 INVITE
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Proxy-Authorization: Digest username="2ZZZZZ0001", realm="voipgrid.nl", algorithm=MD5, uri="sip:+31YYYYYYY498@sipproxy.voipgrid.nl", nonce="534fbb510000bdb500f6d77db195321c4905df3188dfaed3", response="fe40313a3da8005565cdca1584961526"
Date: Thu, 17 Apr 2014 11:24:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Remote-Party-ID: "0091994YYYYY77" <sip:0091994YYYYY77@95.211.XXX.XXX>;party=calling;privacy=off;screen=yes
Content-Type: application/sdp
Content-Length: 255
v=0
o=root 1305983722 1305983725 IN IP4 95.211.XXX.XXX
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 95.211.XXX.XXX
t=0 0
m=audio 11052 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c:
<--- SIP read from UDP:195.35.115.119:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 95.211.XXX.XXX:5060;branch=z9hG4bK6f3e4288
From: "0091994YYYYY77" <sip:2ZZZZZ0001@95.211.XXX.XXX>;tag=as350b5c59
To: <sip:+31YYYYYYY498@sipproxy.voipgrid.nl>
Call-ID: 73b205fd474bfc8705160c2f276f71ae@95.211.XXX.XXX:5060
CSeq: 105 INVITE
Server: VGPXY-8a541e8m-OS
Content-Length: 0
<------------->
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: --- (8 headers 0 lines) ---
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c:
<--- SIP read from UDP:195.35.115.119:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 95.211.XXX.XXX:5060;received=95.211.XXX.XXX;rport=5060;branch=z9hG4bK6f3e4288
From: "0091994YYYYY77" <sip:2ZZZZZ0001@95.211.XXX.XXX>;tag=as350b5c59
To: <sip:+31YYYYYYY498@sipproxy.voipgrid.nl>;tag=32bbfaed647d04e1dda1989fc0442882-ef87
Call-ID: 73b205fd474bfc8705160c2f276f71ae@95.211.XXX.XXX:5060
CSeq: 105 INVITE
Proxy-Authenticate: Digest realm="voipgrid.nl", nonce="534fbb51000166f708c5d77cb67f38cb0008a159fb09d2cf", stale=true
Server: VGPXY-8a541e8m-OS
Content-Length: 0
<------------->
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: --- (9 headers 0 lines) ---
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: set_destination: Parsing <sip:+31YYYYYYY498@sipproxy.voipgrid.nl> for address/port to send to
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: set_destination: set destination to 195.35.115.119:5060
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: Transmitting (no NAT) to 195.35.115.119:5060:
ACK sip:+31YYYYYYY498@sipproxy.voipgrid.nl SIP/2.0
Via: SIP/2.0/UDP 95.211.XXX.XXX:5060;branch=z9hG4bK6f3e4288
Max-Forwards: 70
From: "0091994YYYYY77" <sip:2ZZZZZ0001@95.211.XXX.XXX>;tag=as350b5c59
To: <sip:+31YYYYYYY498@sipproxy.voipgrid.nl>;tag=32bbfaed647d04e1dda1989fc0442882-ef87
Contact: <sip:2ZZZZZ0001@95.211.XXX.XXX:5060>
Call-ID: 73b205fd474bfc8705160c2f276f71ae@95.211.XXX.XXX:5060
CSeq: 105 ACK
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Content-Length: 0
---
[Apr 17 13:24:21] NOTICE[16201] chan_sip.c: Failed to authenticate on INVITE to '"0091994YYYYY77" <sip:2ZZZZZ0001@95.211.XXX.XXX>;tag=as350b5c59'
[Apr 17 13:24:21] VERBOSE[16310] app_dial.c: -- SIP/2ZZZZZ0001-00000001 is circuit-busy
[Apr 17 13:24:21] VERBOSE[16310] app_dial.c: == Everyone is busy/congested at this time (1:0/1/0)
[Apr 17 13:24:21] VERBOSE[16310] pbx.c: -- Auto fallthrough, channel 'SIP/2ZZZZZ0001-00000000' status is 'CONGESTION'
[Apr 17 13:24:21] VERBOSE[16310] chan_sip.c:
<--- Reliably Transmitting (no NAT) to 195.35.114.119:6060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 195.35.114.119:6060;branch=z9hG4bKb69b.ab49e274.0;received=195.35.114.119
Via: SIP/2.0/UDP 195.35.114.100:5060;received=195.35.114.100;branch=z9hG4bK274b392d;rport=5060
From: <sip:2ZZZZZ0001@195.35.114.100>;tag=as162de247
To: <sip:+31YYYYYYY498@voipgrid.nl:5060>;tag=as70ea88d5
Call-ID: 27df162f346e49aa5646db6b7c3c9900@195.35.114.100:5060
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21
Content-Length: 0
<------------>
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: Really destroying SIP dialog '73b205fd474bfc8705160c2f276f71ae@95.211.XXX.XXX:5060' Method: INVITE
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c:
<--- SIP read from UDP:195.35.114.119:6060 --->
ACK sip:+31YYYYYYY498@95.211.XXX.XXX:5060 SIP/2.0
Via: SIP/2.0/UDP 195.35.114.119:6060;branch=z9hG4bKb69b.ab49e274.0
From: <sip:2ZZZZZ0001@195.35.114.100>;tag=as162de247
Call-ID: 27df162f346e49aa5646db6b7c3c9900@195.35.114.100:5060
To: <sip:+31YYYYYYY498@voipgrid.nl:5060>;tag=as70ea88d5
CSeq: 102 ACK
Max-Forwards: 70
User-Agent: VGPXY-e2d9677m-OS
Content-Length: 0
<------------->
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: --- (9 headers 0 lines) ---
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: Really destroying SIP dialog '27df162f346e49aa5646db6b7c3c9900@195.35.114.100:5060' Method: ACK
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c:
<--- SIP read from UDP:195.35.115.119:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 95.211.XXX.XXX:5060;received=95.211.XXX.XXX;rport=5060;branch=z9hG4bK20587496
From: "0091994YYYYY77" <sip:2ZZZZZ0001@95.211.XXX.XXX>;tag=as350b5c59
To: <sip:+31YYYYYYY498@sipproxy.voipgrid.nl>;tag=32bbfaed647d04e1dda1989fc0442882-c765
Call-ID: 73b205fd474bfc8705160c2f276f71ae@95.211.XXX.XXX:5060
CSeq: 103 INVITE
Proxy-Authenticate: Digest realm="voipgrid.nl", nonce="534fbb51000166f6f1bf77571eb98caccd7b5b5c7314fe82", stale=true
Server: VGPXY-8a541e8m-OS
Content-Length: 0
<------------->
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: --- (9 headers 0 lines) ---
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c:
<--- SIP read from UDP:195.35.114.119:6060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 95.211.XXX.XXX:5060;received=95.211.XXX.XXX;rport=5060;branch=z9hG4bK224ac8fd
From: "0091994YYYYY77" <sip:2ZZZZZ0001@95.211.XXX.XXX>;tag=as350b5c59
To: <sip:+31YYYYYYY498@sipproxy.voipgrid.nl>;tag=9e207c4db3df57e12fedd0328727cc06-b126
Call-ID: 73b205fd474bfc8705160c2f276f71ae@95.211.XXX.XXX:5060
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="voipgrid.nl", nonce="534fbb510000bdb44ce42807939e86a102501ab2b800ffbc"
Server: VGPXY-e2d9677m-OS
Content-Length: 0
<------------->
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: --- (9 headers 0 lines) ---
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c:
<--- SIP read from UDP:195.35.114.119:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 95.211.XXX.XXX:5060;received=95.211.XXX.XXX;rport=5060;branch=z9hG4bK2b6f4f8d
From: "0091994YYYYY77" <sip:2ZZZZZ0001@95.211.XXX.XXX>;tag=as350b5c59
To: <sip:+31YYYYYYY498@sipproxy.voipgrid.nl>;tag=9e207c4db3df57e12fedd0328727cc06-6707
Call-ID: 73b205fd474bfc8705160c2f276f71ae@95.211.XXX.XXX:5060
CSeq: 104 INVITE
Proxy-Authenticate: Digest realm="voipgrid.nl", nonce="534fbb510000bdb500f6d77db195321c4905df3188dfaed3", stale=true
Server: VGPXY-e2d9677m-OS
Content-Length: 0
<------------->
[Apr 17 13:24:21] VERBOSE[16201] chan_sip.c: --- (9 headers 0 lines) ---
[Apr 17 13:24:22] VERBOSE[16201] chan_sip.c:
<--- SIP read from UDP:195.35.115.119:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 95.211.XXX.XXX:5060;received=95.211.XXX.XXX;rport=5060;branch=z9hG4bK20587496
From: "0091994YYYYY77" <sip:2ZZZZZ0001@95.211.XXX.XXX>;tag=as350b5c59
To: <sip:+31YYYYYYY498@sipproxy.voipgrid.nl>;tag=32bbfaed647d04e1dda1989fc0442882-c765
Call-ID: 73b205fd474bfc8705160c2f276f71ae@95.211.XXX.XXX:5060
CSeq: 103 INVITE
Proxy-Authenticate: Digest realm="voipgrid.nl", nonce="534fbb51000166f6f1bf77571eb98caccd7b5b5c7314fe82", stale=true
Server: VGPXY-8a541e8m-OS
Content-Length: 0
<------------->
[Apr 17 13:24:22] VERBOSE[16201] chan_sip.c: --- (9 headers 0 lines) ---
[Apr 17 13:24:25] VERBOSE[16201] chan_sip.c:
[Apr 17 13:24:25] VERBOSE[16201] chan_sip.c: --- (13 headers 0 lines) ---
[Apr 17 13:24:25] VERBOSE[16201] chan_sip.c: Sending to 190.2.8.121:6062 (no NAT)
[Apr 17 13:24:25] VERBOSE[16201] chan_sip.c: sip.conf
[general]
dtmfmode=rfc2833
context=internal
register => UUUUUU:PPPPPPPP@sipproxy.vvvvvvv.nl/UUUUUU
allowguest=no
allowoverlap=yes
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
alwaysauthreject=yes
canreinvite=yes
nat=no
session-timers=refuse
externrefresh=15
localnet=192.168.1.0/255.255.255.0
disallow=all
allow=alaw
allowtransfer=yes
[UUUUUU]
type=friend
secret=PPPPPP
context=internal
host=sipproxy.vvvvvvvv.nl
username=UUUUUU
fromuser=UUUUUU
insecure=invite
nat=never
disallowed_methods=UPDATE
directmedia=no
videosupport=no
disallow=all
allow=alaw
sendrpid=yes
trustrpid=yes
permit=195.35.114.0/24
permit=195.35.115.0/24
permit=217.21.203.0/27
permit=217.21.192.64/27extensions.conf
[internal]
exten => UUUUUU,1,Answer()
exten => UUUUUU,2,Playback(vm-nobodyavail)
exten => UUUUUU,3,Hangup()
exten => _[0-9+]!,1,Dial(SIP/${EXTEN}@UUUUUU,60,t)What might be wrong with the configuration? Please help!