Remote user rejected the call [SOLVED]

Hi All,

Sometimes when I am trying to make call from softphone ( ekiga )

I used to get the error

here is the debug

[code]my-machine*CLI> sip set debug on
SIP Debugging re-enabled

<— SIP read from UDP:192.168.1.2:5061 —>
INVITE sip:567@192.168.1.2 SIP/2.0
Date: Tue, 26 Jul 2011 08:47:09 GMT
CSeq: 1 INVITE
Via: SIP/2.0/UDP 192.168.1.2:5061;branch=z9hG4bK90468184-d1b5-e011-99ee-7071bc86c9c0;rport
User-Agent: Ekiga/3.2.6
From: “root” sip:ivan@192.168.1.2;tag=e2f98084-d1b5-e011-99ee-7071bc86c9c0
Call-ID: 76fc8084-d1b5-e011-99ee-7071bc86c9c0@my-machine
To: sip:567@192.168.1.2
Contact: sip:ivan@192.168.1.2:5061
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
Content-Type: application/sdp
Content-Length: 553
Max-Forwards: 70

v=0
o=- 1311670029 1 IN IP4 192.168.1.2
s=Opal SIP Session
c=IN IP4 192.168.1.2
t=0 0
m=audio 5066 RTP/AVP 112 0 8 9 101 120
a=sendrecv
a=rtpmap:112 Speex/16000/1
a=fmtp:112 sr=16000,mode=any
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:9 G722/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32,36
a=rtpmap:120 NSE/8000
a=fmtp:120 192-193
m=video 5068 RTP/AVP 99 31
b=AS:4096
b=TIAS:4096000
a=sendrecv
a=rtpmap:99 theora/90000
a=fmtp:99 height=576;width=704
a=rtpmap:31 h261/90000
a=fmtp:31 CIF=1;QCIF=1
<------------->
— (13 headers 24 lines) —
Sending to 192.168.1.2:5061 (no NAT)
Using INVITE request as basis request - 76fc8084-d1b5-e011-99ee-7071bc86c9c0@my-machine
Found peer ‘ivan’ for ‘ivan’ from 192.168.1.2:5061

<— Reliably Transmitting (no NAT) to 192.168.1.2:5061 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:5061;branch=z9hG4bK90468184-d1b5-e011-99ee-7071bc86c9c0;received=192.168.1.2;rport=5061
From: “root” sip:ivan@192.168.1.2;tag=e2f98084-d1b5-e011-99ee-7071bc86c9c0
To: sip:567@192.168.1.2;tag=as51c2a1ca
Call-ID: 76fc8084-d1b5-e011-99ee-7071bc86c9c0@my-machine
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="047670cf"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘76fc8084-d1b5-e011-99ee-7071bc86c9c0@my-machine’ in 32000 ms (Method: INVITE)

<— SIP read from UDP:192.168.1.2:5061 —>
ACK sip:567@192.168.1.2 SIP/2.0
CSeq: 1 ACK
Via: SIP/2.0/UDP 192.168.1.2:5061;branch=z9hG4bK90468184-d1b5-e011-99ee-7071bc86c9c0;rport
From: “root” sip:ivan@192.168.1.2;tag=e2f98084-d1b5-e011-99ee-7071bc86c9c0
Call-ID: 76fc8084-d1b5-e011-99ee-7071bc86c9c0@my-machine
To: sip:567@192.168.1.2;tag=as51c2a1ca
Content-Length: 0
Max-Forwards: 70

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

<— SIP read from UDP:192.168.1.2:5061 —>
INVITE sip:567@192.168.1.2 SIP/2.0
Date: Tue, 26 Jul 2011 08:47:09 GMT
CSeq: 2 INVITE
Via: SIP/2.0/UDP 192.168.1.2:5061;branch=z9hG4bK6a608384-d1b5-e011-99ee-7071bc86c9c0;rport
User-Agent: Ekiga/3.2.6
Authorization: Digest username=“ivan”, realm=“asterisk”, nonce=“047670cf”, uri="sip:567@192.168.1.2", algorithm=MD5, response="8b0942c497e86bfc5339eb556c045cf2"
From: “root” sip:ivan@192.168.1.2;tag=e2f98084-d1b5-e011-99ee-7071bc86c9c0
Call-ID: 76fc8084-d1b5-e011-99ee-7071bc86c9c0@my-machine
To: sip:567@192.168.1.2
Contact: sip:ivan@192.168.1.2:5061
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
Content-Type: application/sdp
Content-Length: 553
Max-Forwards: 70

v=0
o=- 1311670029 1 IN IP4 192.168.1.2
s=Opal SIP Session
c=IN IP4 192.168.1.2
t=0 0
m=audio 5066 RTP/AVP 112 0 8 9 101 120
a=sendrecv
a=rtpmap:112 Speex/16000/1
a=fmtp:112 sr=16000,mode=any
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:9 G722/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32,36
a=rtpmap:120 NSE/8000
a=fmtp:120 192-193
m=video 5068 RTP/AVP 99 31
b=AS:4096
b=TIAS:4096000
a=sendrecv
a=rtpmap:99 theora/90000
a=fmtp:99 height=576;width=704
a=rtpmap:31 h261/90000
a=fmtp:31 CIF=1;QCIF=1
<------------->
— (14 headers 24 lines) —
Sending to 192.168.1.2:5061 (no NAT)
Using INVITE request as basis request - 76fc8084-d1b5-e011-99ee-7071bc86c9c0@my-machine
Found peer ‘ivan’ for ‘ivan’ from 192.168.1.2:5061
== Using SIP RTP CoS mark 5
Found RTP audio format 112
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 101
Found RTP audio format 120
Found audio description format Speex for ID 112
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G722 for ID 9
Found audio description format telephone-event for ID 101
Found audio description format NSE for ID 120
Found RTP video format 99
Found RTP video format 31
Found video description format h261 for ID 31
Capabilities: us - 0x80030c7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719), peer - audio=0x20000100c (ulaw|alaw|speex16|g722)/video=0x240000 (h261|h264)/text=0x0 (nothing), combined - 0x20024100c (ulaw|alaw|speex16|g722|h261|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.2:5066
Looking for 567 in test (domain 192.168.1.2)
list_route: hop: sip:ivan@192.168.1.2:5061

<— Transmitting (no NAT) to 192.168.1.2:5061 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:5061;branch=z9hG4bK6a608384-d1b5-e011-99ee-7071bc86c9c0;received=192.168.1.2;rport=5061
From: “root” sip:ivan@192.168.1.2;tag=e2f98084-d1b5-e011-99ee-7071bc86c9c0
To: sip:567@192.168.1.2
Call-ID: 76fc8084-d1b5-e011-99ee-7071bc86c9c0@my-machine
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:567@192.168.1.2:5060
Content-Length: 0

<------------>
– Executing [567@test:1] AGI(“SIP/ivan-0000000c”, “agi://localhost/hello.agi”) in new stack
– <SIP/ivan-0000000c>AGI Script agi://localhost/hello.agi completed, returning 0
– Auto fallthrough, channel ‘SIP/ivan-0000000c’ status is 'UNKNOWN’
Scheduling destruction of SIP dialog ‘76fc8084-d1b5-e011-99ee-7071bc86c9c0@my-machine’ in 32000 ms (Method: INVITE)

<— Reliably Transmitting (no NAT) to 192.168.1.2:5061 —>
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.1.2:5061;branch=z9hG4bK6a608384-d1b5-e011-99ee-7071bc86c9c0;received=192.168.1.2;rport=5061
From: “root” sip:ivan@192.168.1.2;tag=e2f98084-d1b5-e011-99ee-7071bc86c9c0
To: sip:567@192.168.1.2;tag=as7c0ce660
Call-ID: 76fc8084-d1b5-e011-99ee-7071bc86c9c0@my-machine
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>

<— SIP read from UDP:192.168.1.2:5061 —>
ACK sip:567@192.168.1.2 SIP/2.0
CSeq: 2 ACK
Via: SIP/2.0/UDP 192.168.1.2:5061;branch=z9hG4bK6a608384-d1b5-e011-99ee-7071bc86c9c0;rport
Authorization: Digest username=“ivan”, realm=“asterisk”, nonce=“047670cf”, uri="sip:567@192.168.1.2", algorithm=MD5, response="4ee7b5f36a74c7d91f692f7156997ff4"
From: “root” sip:ivan@192.168.1.2;tag=e2f98084-d1b5-e011-99ee-7071bc86c9c0
Call-ID: 76fc8084-d1b5-e011-99ee-7071bc86c9c0@my-machine
To: sip:567@192.168.1.2;tag=as7c0ce660
Content-Length: 0
Max-Forwards: 70

<------------->
— (9 headers 0 lines) —
my-machine*CLI> [/code]

The AGI script didn’t answer the call.

thanks david55 for your reply …

BTW how to mark this post as solved