Call failed: declined


#1

Hi, we have asterisk 1.4 and we try to make a call between 2 extensions registered on the sever using x-lite (ext 783 and 784)

when we try to call we have an error: call failed: declined

If we active sip debug we have this … any help? Tks

(test.com is not the real domain, we’ve just changed here for privacy)

<— SIP read from 192.168.0.112:48740 —>
INVITE sip:784@voip.test.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.112:48740;branch=z9hG4bK-d87543-686c153e1808ab75-1–d87543-;rport
Max-Forwards: 70
Contact: sip:783@192.168.0.112:48740
To: "784"sip:784@voip.test.com
From: "783"sip:783@voip.test.com;tag=c61b4975
Call-ID: ZWZhNjI0ZmI0N2IxMWYxMzlhNWI2YzJkMWZlNjAyZGE.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 328

v=0
o=- 3 2 IN IP4 192.168.0.112
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.0.112
t=0 0
m=audio 36234 RTP/AVP 107 119 0 98 8 3 101
a=alt:1 1 : yXdJeykL Tv2dD6o/ 192.168.0.112 36234
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
— (12 headers 13 lines) —
Sending to 192.168.0.112 : 48740 (NAT)
Using INVITE request as basis request - ZWZhNjI0ZmI0N2IxMWYxMzlhNWI2YzJkMWZlNjAyZGE.

<— Reliably Transmitting (no NAT) to 192.168.0.112:48740 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.112:48740;branch=z9hG4bK-d87543-686c153e1808ab75-1–d87543-;received=192.168.0.112;rport=48740
From: "783"sip:783@voip.test.com;tag=c61b4975
To: "784"sip:784@voip.test.com;tag=as4795269d
Call-ID: ZWZhNjI0ZmI0N2IxMWYxMzlhNWI2YzJkMWZlNjAyZGE.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“mydomain.tld”, nonce="62f496e5"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘ZWZhNjI0ZmI0N2IxMWYxMzlhNWI2YzJkMWZlNjAyZGE.’ in 32000 ms (Method: INVITE)
Found user '783’
voip*CLI>
<— SIP read from 192.168.0.112:48740 —>
ACK sip:784@voip.test.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.112:48740;branch=z9hG4bK-d87543-686c153e1808ab75-1–d87543-;rport
To: "784"sip:784@voip.test.com;tag=as4795269d
From: "783"sip:783@voip.test.com;tag=c61b4975
Call-ID: ZWZhNjI0ZmI0N2IxMWYxMzlhNWI2YzJkMWZlNjAyZGE.
CSeq: 1 ACK
Content-Length: 0

<------------->
— (7 headers 0 lines) —
voip*CLI>
<— SIP read from 192.168.0.112:48740 —>
INVITE sip:784@voip.test.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.112:48740;branch=z9hG4bK-d87543-e646c010c96de422-1–d87543-;rport
Max-Forwards: 70
Contact: sip:783@192.168.0.112:48740
To: "784"sip:784@voip.test.com
From: “783"sip:783@voip.test.com;tag=c61b4975
Call-ID: ZWZhNjI0ZmI0N2IxMWYxMzlhNWI2YzJkMWZlNjAyZGE.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest username=“783”,realm=“mydomain.tld”,nonce=“62f496e5”,uri="sip:784@voip.test.com”,response=“a3567f0216411c1ce4335ee28f8874c6”,algorithm=MD5
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 328

v=0
o=- 3 2 IN IP4 192.168.0.112
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.0.112
t=0 0
m=audio 36234 RTP/AVP 107 119 0 98 8 3 101
a=alt:1 1 : yXdJeykL Tv2dD6o/ 192.168.0.112 36234
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
— (13 headers 13 lines) —
Sending to 192.168.0.112 : 48740 (NAT)
Using INVITE request as basis request - ZWZhNjI0ZmI0N2IxMWYxMzlhNWI2YzJkMWZlNjAyZGE.
Found user '783’
Found RTP audio format 107
Found RTP audio format 119
Found RTP audio format 0
Found RTP audio format 98
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.112:36234
Found description format BV32 for ID 107
Found description format BV32-FEC for ID 119
Found description format iLBC for ID 98
Found description format telephone-event for ID 101
Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0x40e (gsm|ulaw|alaw|ilbc)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.0.112:36234
Looking for 784 in DefaultOutgoingRule (domain voip.test.com)
list_route: hop: sip:783@192.168.0.112:48740

<— Transmitting (no NAT) to 192.168.0.112:48740 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.112:48740;branch=z9hG4bK-d87543-e646c010c96de422-1–d87543-;received=192.168.0.112;rport=48740
From: "783"sip:783@voip.test.com;tag=c61b4975
To: "784"sip:784@voip.test.com
Call-ID: ZWZhNjI0ZmI0N2IxMWYxMzlhNWI2YzJkMWZlNjAyZGE.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:784@192.168.1.5
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘ZWZhNjI0ZmI0N2IxMWYxMzlhNWI2YzJkMWZlNjAyZGE.’ in 32000 ms (Method: INVITE)
voip*CLI>
<— Reliably Transmitting (no NAT) to 192.168.0.112:48740 —>
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.0.112:48740;branch=z9hG4bK-d87543-e646c010c96de422-1–d87543-;received=192.168.0.112;rport=48740
From: "783"sip:783@voip.test.com;tag=c61b4975
To: "784"sip:784@voip.test.com;tag=as2dbb4a86
Call-ID: ZWZhNjI0ZmI0N2IxMWYxMzlhNWI2YzJkMWZlNjAyZGE.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:784@192.168.1.5
Content-Length: 0

<------------>
voip*CLI>
<— SIP read from 192.168.0.112:48740 —>
ACK sip:784@voip.test.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.112:48740;branch=z9hG4bK-d87543-e646c010c96de422-1–d87543-;rport
To: "784"sip:784@voip.test.com;tag=as2dbb4a86
From: "783"sip:783@voip.test.com;tag=c61b4975
Call-ID: ZWZhNjI0ZmI0N2IxMWYxMzlhNWI2YzJkMWZlNjAyZGE.
CSeq: 2 ACK
Content-Length: 0

<------------->
— (7 headers 0 lines) —


#2

The problem is with voiceone … (voiceone.it) … without it we can call, with it no …

:frowning:

Ste


#3

I am facing same issue. How did you resolve this issue?