I try to make the server autodail a number and play a message. But I get the following message:
Call failed to go through, reason (5) Remote end is Busy
But im 100% sure that its not busy. What can be wrong. This is the code that I add to a file in /var/spool/asterix/outbound
Channel: SIP/Phonzo2/0735453334
CallerID: “Linus” <0382788032>
Application: Playback
Data: hello-world
And this is the debug message that I get:
[2016-06-09 16:14:13] WARNING[1937]: pbx_spool.c:309 safe_append: Unable to set utime on /var/spool/asterisk/outgoing/test: Operation not permitted
Audio is at 13396
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 80.232.37.178:5060:
INVITE sip:0735453334@sip.phonzo.com SIP/2.0
Via: SIP/2.0/UDP 192.168.2.52:5060;branch=z9hG4bK19c331ae;rport
Max-Forwards: 70
From: “Linus” < sip:46382799032@80.232.37.178>;tag=as494f4a16
To: < sip:0735453334@sip.phonzo.com>
Contact: < sip:46382799032@192.168.2.52:5060>
Call-ID: 50c06e960a4db12d2fedc8450dc11f2a@80.232.37.178
CSeq: 102 INVITE
User-Agent: it-slav PBX
Date: Thu, 09 Jun 2016 14:14:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 249
v=0
o=root 201537809 201537809 IN IP4 192.168.2.52
s=it-slav PBX
c=IN IP4 192.168.2.52
t=0 0
m=audio 13396 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=ptime:20
a=sendrecv
<— SIP read from UDP:80.232.37.178:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.52:5060;branch=z9hG4bK19c331ae;rport=23536;received=88.206.243.45
To: < sip:0735453334@sip.phonzo.com>
From: “Linus”< sip:46382799032@80.232.37.178>;tag=as494f4a16
Call-ID: 50c06e960a4db12d2fedc8450dc11f2a@80.232.37.178
CSeq: 102 INVITE
Content-Length: 0
<------------->
— (7 headers 0 lines) —
<— SIP read from UDP:80.232.37.178:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.52:5060;received=88.206.243.45;branch=z9hG4bK19c331ae;rport=23536
Record-Route: < sip:80.232.37.178;lr>
To: < sip:0735453334@sip.phonzo.com>
From: “Linus”< sip:46382799032@80.232.37.178>;tag=as494f4a16
Call-ID: 50c06e960a4db12d2fedc8450dc11f2a@80.232.37.178
CSeq: 102 INVITE
Server: Sippy
WWW-Authenticate: Digest realm=“sip.phonzo.com”,nonce="a346c623f3dce8987c99f669701552931929"
Content-Length: 0
<------------->
— (10 headers 0 lines) —
Transmitting (NAT) to 80.232.37.178:5060:
ACK sip:0735453334@sip.phonzo.com SIP/2.0
Via: SIP/2.0/UDP 192.168.2.52:5060;branch=z9hG4bK19c331ae;rport
Max-Forwards: 70
From: “Linus” < sip:46382799032@80.232.37.178>;tag=as494f4a16
To: < sip:0735453334@sip.phonzo.com >
Contact: < sip:46382799032@192.168.2.52:5060>
Call-ID: 50c06e960a4db12d2fedc8450dc11f2a@80.232.37.178
CSeq: 102 ACK
User-Agent: it-slav PBX
Content-Length: 0
Audio is at 13396
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 80.232.37.178:5060:
INVITE sip:0735453334@sip.phonzo.com SIP/2.0
Via: SIP/2.0/UDP 192.168.2.52:5060;branch=z9hG4bK0fd84f18;rport
Max-Forwards: 70
From: “Linus” < sip:46382799032@80.232.37.178>;tag=as494f4a16
To: < sip:0735453334@sip.phonzo.com >
Contact: < sip:46382799032@192.168.2.52:5060 >
Call-ID: 50c06e960a4db12d2fedc8450dc11f2a@80.232.37.178
CSeq: 103 INVITE
User-Agent: it-slav PBX
Authorization: Digest username=“46382799032”, realm=“sip.phonzo.com”, algorithm=MD5, uri="sip:0735453334@sip.phonzo.com", nonce=“a346c623f3dce8987c99f669701552931929”, response="10d1b1d4a4455561870e264a3d46375b"
Date: Thu, 09 Jun 2016 14:14:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 249
v=0
o=root 201537809 201537810 IN IP4 192.168.2.52
s=it-slav PBX
c=IN IP4 192.168.2.52
t=0 0
m=audio 13396 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=ptime:20
a=sendrecv
<— SIP read from UDP:80.232.37.178:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.52:5060;branch=z9hG4bK0fd84f18;rport=23536;received=88.206.243.45
To: < sip:0735453334@sip.phonzo.com >
From: “Linus”< sip:46382799032@80.232.37.178 >;tag=as494f4a16
Call-ID: 50c06e960a4db12d2fedc8450dc11f2a@80.232.37.178
CSeq: 103 INVITE
Content-Length: 0
<------------->
— (7 headers 0 lines) —
<— SIP read from UDP:80.232.37.178:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.2.52:5060;received=88.206.243.45;branch=z9hG4bK0fd84f18;rport=23536
Record-Route: < sip:80.232.37.178;lr >
To: < sip:0735453334@sip.phonzo.com >;tag=4quafrq5hp47dm2g.i
From: “Linus”< sip:46382799032@80.232.37.178 >;tag=as494f4a16
Call-ID: 50c06e960a4db12d2fedc8450dc11f2a@80.232.37.178
CSeq: 103 INVITE
Content-Type: application/sdp
Server: Sippy
Content-Length: 255
v=0
o=Sippy 739145560 1 IN IP4 80.232.37.178
s=session
t=0 0
m=audio 44734 RTP/AVP 0 8 101
c=IN IP4 80.232.37.178
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=sendrecv
<------------->
— (10 headers 12 lines) —
list_route: hop: < sip:80.232.37.178;lr >
Found RTP audio format 0
Found RTP audio format 8
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 telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 80.232.37.178:44734
<— SIP read from UDP:80.232.37.178:5060 —>
OPTIONS sip:80.232.37.178 SIP/2.0
Via: SIP/2.0/UDP 80.232.37.178:5060;branch=z9hG4bK-d8754z-26dddf06bb36490c-1—d8754z-;rport
Max-Forwards: 1
To: < sip:80.232.37.178 >
From: < sip:80.232.37.178:5060 >;tag=50c0895e
Call-ID: natpingNWQwMmRmY2Q0OGVlYzdkMmI0YTA5OGIyZGUwMTcyMTQ.
CSeq: 1 OPTIONS
Accept: application/sdp
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Sending to 80.232.37.178:5060 (NAT)
Looking for s in from-sip-external (domain 80.232.37.178)
<— Transmitting (NAT) to 80.232.37.178:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 80.232.37.178:5060;branch=z9hG4bK-d8754z-26dddf06bb36490c-1—d8754z-;received=80.232.37.178;rport=5060
From: < sip:80.232.37.178:5060 >;tag=50c0895e
To: < sip:80.232.37.178 >;tag=as730bb54d
Call-ID: natpingNWQwMmRmY2Q0OGVlYzdkMmI0YTA5OGIyZGUwMTcyMTQ.
CSeq: 1 OPTIONS
Server: it-slav PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: < sip:192.168.2.52:5060 >
Accept: application/sdp
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘natpingNWQwMmRmY2Q0OGVlYzdkMmI0YTA5OGIyZGUwMTcyMTQ.’ in 32000 ms (Method: OPTIONS)
[2016-06-09 16:14:24] NOTICE[1936]: chan_sip.c:15089 sip_reregister: – Re-registration for 46382799032@80.232.37.178
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 80.232.37.178:5060:
REGISTER sip:80.232.37.178 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.52:5060;branch=z9hG4bK7b5768ec;rport
Max-Forwards: 70
From: < sip:46382799032@80.232.37.178 >;tag=as79185ba2
To: < sip:46382799032@80.232.37.178 >
Call-ID: 5ff5d51c1d539cf1474618651768f624@[::1]
CSeq: 116 REGISTER
User-Agent: it-slav PBX
Authorization: Digest username=“46382799032”, realm=“80.232.37.178”, algorithm=MD5, uri=“sip:80.232.37.178”, nonce=“1465481559:0e6077e64a854b78fd283fe40309f04d”, response="e0335a3a1ada13b9c154e3c20af81ec6"
Expires: 120
Contact: < sip:382799032@192.168.2.52:5060>
Content-Length: 0
<— SIP read from UDP:80.232.37.178:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.52:5060;branch=z9hG4bK7b5768ec;rport=23536;received=88.206.243.45
Contact: < sip:382799032@88.206.243.45:23536>;expires=300
To: < sip:46382799032@80.232.37.178>;tag=6d83bc30
From: < sip:46382799032@80.232.37.178>;tag=as79185ba2
Call-ID: 5ff5d51c1d539cf1474618651768f624@[::1]
CSeq: 116 REGISTER
Date: Thu, 09 Jun 2016 14:14:24 GMT
PortaBilling: available-funds:2499.97 currency:SEK
Content-Length: 0
<------------->
— (10 headers 0 lines) —
[2016-06-09 16:14:24] NOTICE[1936]: chan_sip.c:23631 handle_response_register: Outbound Registration: Expiry for 80.232.37.178 is 120 sec (Scheduling reregistration in 105 s)
Really destroying SIP dialog ‘5ff5d51c1d539cf1474618651768f624@[::1]’ Method: REGISTER
<— SIP read from UDP:80.232.37.178:5060 —>
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.2.52:5060;received=88.206.243.45;branch=z9hG4bK0fd84f18;rport=23536
Record-Route: < sip:80.232.37.178;lr>
To: < sip:0735453334@sip.phonzo.com>;tag=4quafrq5hp47dm2g.i
From: “Linus”< sip:46382799032@80.232.37.178>;tag=as494f4a16
Call-ID: 50c06e960a4db12d2fedc8450dc11f2a@80.232.37.178
CSeq: 103 INVITE
Server: Sippy
Content-Length: 0
<------------->
— (9 headers 0 lines) —
set_destination: Parsing < sip:80.232.37.178;lr> for address/port to send to
set_destination: set destination to 80.232.37.178:5060
Transmitting (NAT) to 80.232.37.178:5060:
ACK sip:0735453334@sip.phonzo.com SIP/2.0
Via: SIP/2.0/UDP 192.168.2.52:5060;branch=z9hG4bK0fd84f18;rport
Route: < sip:80.232.37.178;lr>
Max-Forwards: 70
From: “Linus” < sip:46382799032@80.232.37.178>;tag=as494f4a16
To: < sip:0735453334@sip.phonzo.com>;tag=4quafrq5hp47dm2g.i
Contact: < sip:46382799032@192.168.2.52:5060>
Call-ID: 50c06e960a4db12d2fedc8450dc11f2a@80.232.37.178
CSeq: 103 ACK
User-Agent: it-slav PBX
Content-Length: 0
[2016-06-09 16:14:25] NOTICE[2809]: pbx_spool.c:389 attempt_thread: Call failed to go through, reason (5) Remote end is Busy
[2016-06-09 16:14:25] NOTICE[2809]: pbx_spool.c:392 attempt_thread: Queued call to SIP/Phonzo2/0735453334 expired without completion after 0 attempts
Really destroying SIP dialog ‘50c06e960a4db12d2fedc8450dc11f2a@80.232.37.178’ Method: INVITEindent preformatted text by 4 spaces