[asterisk-users] Ring forever
FERNANDO VILLARROEL
fvillarroel at yahoo.com
Thu Jul 26 15:51:55 CDT 2007
Hello list, i need help.
My problem is that when I want to call (using ISDN
phone or internal SIP client) via the Sip provider a
real phone number (ISDN phone or internal SIP
Asterisk >> SIP ), I get a ring tone. When I
now decide to hang up (e.g. if
nobody answers), the called telephone continues to
ring almost forever.
the sip.conf:
[2563105]
accountcode = 2563105
username = 2563105
secret = 135
callerid = 412563105
context = test
canreinvite = no
dtmfmode = rfc2833
host = dynamic
insecure = very
language = es
nat = yes
qualify = yes
type = friend
disallow=all
allow=g729
[nyphone]
accountcode=nyphone
canreinvite=no
reinvite=yes
dtmfmode=rfc2833
host=72.55.143.XXX
insecure=very
language=es
nat=no
qualify=no
type=peer
disallow=all
allow=g729
My extensions.conf
exten => _00X.,1,dial(sip/${EXTEN:2}@nyphone,45)
exten => _00X.,2,hangup
Nyphone is my provider for everyone calls
international.
Fernando Villarroel Noriel.
Chillan
Chile
Sorry my English.
This is log:
SIP Debugging Enabled for IP: 72.55.143.XXX:5060
-- Executing [005642325405 at test:1]
Dial("SIP/2563105-0819cf80",
"sip/5642325405 at nyphone|45") in new stack
Audio is at 164.77.171.XXX port 16548
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 72.55.143.XXX:5060:
INVITE sip:5642325405 at 72.55.143.XXX SIP/2.0
Via: SIP/2.0/UDP
164.77.171.XXX:5060;branch=z9hG4bK4ae5ea5c;rport
From: "2563105"
<sip:412563105 at 164.77.171.XXX>;tag=as726ac50a
To: <sip:5642325405 at 72.55.143.XXX>
Contact: <sip:412563105 at 164.77.171.XXX>
Call-ID:
5b7470690d6cd476346bc1113f609a4b at 164.77.171.XXX
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 20 Jul 2007 03:38:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 265
v=0
o=root 2475 2475 IN IP4 164.77.171.XXX
s=session
c=IN IP4 164.77.171.XXX
t=0 0
m=audio 16548 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called 5642325405 at nyphone
vaca*CLI>
<--- SIP read from 72.55.143.XXX:5060 --->
SIP/2.0 407 Proxy Authentication Required
CSeq: 102 INVITE
Via: SIP/2.0/UDP
164.77.171.XXX:5060;branch=z9hG4bK4ae5ea5c;rport
From: "2563105"
<sip:412563105 at 164.77.171.XXX>;tag=as726ac50a
Call-ID:
5b7470690d6cd476346bc1113f609a4b at 164.77.171.XXX
To: <sip:5642325405 at 72.55.143.XXX>
Contact: <sip:72.55.143.XXX:5060;transport=udp>
Proxy-Authenticate: DIGEST realm="VoipSwitch",
nonce="118490324119231120007472128429"
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 72.55.143.XXX:5060:
ACK sip:5642325405 at 72.55.143.XXX SIP/2.0
Via: SIP/2.0/UDP
164.77.171.XXX:5060;branch=z9hG4bK4ae5ea5c;rport
From: "2563105"
<sip:412563105 at 164.77.171.XXX>;tag=as726ac50a
To: <sip:5642325405 at 72.55.143.XXX>
Contact: <sip:412563105 at 164.77.171.XXX>
Call-ID:
5b7470690d6cd476346bc1113f609a4b at 164.77.171.XXX
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Audio is at 164.77.171.XXX port 16548
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 72.55.143.XXX:5060:
INVITE sip:5642325405 at 72.55.143.XXX SIP/2.0
Via: SIP/2.0/UDP
164.77.171.XXX:5060;branch=z9hG4bK3cb0e5aa;rport
From: "2563105"
<sip:412563105 at 164.77.171.XXX>;tag=as726ac50a
To: <sip:5642325405 at 72.55.143.XXX>
Contact: <sip:412563105 at 164.77.171.XXX>
Call-ID:
5b7470690d6cd476346bc1113f609a4b at 164.77.171.XXX
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="test770",
realm="VoipSwitch",
algorithm=MD5, uri="sip:5642325405 at 72.55.143.XXX",
nonce="118490324119231120007472128429",
response="413be923621811a639c3b0e83d3a2e74", opaque=""
Date: Fri, 20 Jul 2007 03:38:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 265
v=0
o=root 2475 2476 IN IP4 164.77.171.XXX
s=session
c=IN IP4 164.77.171.XXX
t=0 0
m=audio 16548 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
vaca*CLI>
<--- SIP read from 72.55.143.XXX:5060 --->
SIP/2.0 200 OK
CSeq: 103 INVITE
Via: SIP/2.0/UDP
164.77.171.XXX:5060;branch=z9hG4bK3cb0e5aa;rport
From: "2563105"
<sip:412563105 at 164.77.171.XXX>;tag=as726ac50a
Call-ID:
5b7470690d6cd476346bc1113f609a4b at 164.77.171.XXX
To:
<sip:5642325405 at 72.55.143.XXX>;tag=1907470723212675853288937
Contact: <sip:72.55.143.XXX:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 215
v=0
o=VoipSwitch 9936 9936 IN IP4 72.55.143.XXX
s=VoipSIP
i=Audio Session
c=IN IP4 72.55.143.XXX
t=0 0
m=audio 8936 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (9 headers 10 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 72.55.143.XXX:8936
Found description format G729 for ID 18
Found description format telephone-event for ID 101
Capabilities: us - 0x100 (g729), peer - audio=0x100
(g729)/video=0x0
(nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1
(telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 72.55.143.XXX:8936
list_route: hop:
<sip:72.55.143.XXX:5060;transport=udp>
set_destination: Parsing
<sip:72.55.143.XXX:5060;transport=udp> for
address/port to send to
set_destination: set destination to 72.55.143.XXX,
port 5060
Transmitting (no NAT) to 72.55.143.XXX:5060:
ACK sip:72.55.143.XXX:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP
164.77.171.XXX:5060;branch=z9hG4bK22458f4a;rport
From: "2563105"
<sip:412563105 at 164.77.171.XXX>;tag=as726ac50a
To:
<sip:5642325405 at 72.55.143.XXX>;tag=1907470723212675853288937
Contact: <sip:412563105 at 164.77.171.XXX>
Call-ID:
5b7470690d6cd476346bc1113f609a4b at 164.77.171.XXX
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- Call on SIP/nyphone-081a7768 left from hold
-- SIP/nyphone-081a7768 answered
SIP/2563105-0819cf80
-- Packet2Packet bridging SIP/2563105-0819cf80 and
SIP/nyphone-081a7768
vaca*CLI>
<--- SIP read from 72.55.143.XXX:5060 --->
SIP/2.0 183 Session Progress
CSeq: 103 INVITE
Via: SIP/2.0/UDP
164.77.171.XXX:5060;branch=z9hG4bK3cb0e5aa;rport
From: "2563105"
<sip:412563105 at 164.77.171.XXX>;tag=as726ac50a
Call-ID:
5b7470690d6cd476346bc1113f609a4b at 164.77.171.XXX
To:
<sip:5642325405 at 72.55.143.XXX>;tag=1907470723212675853288937
Contact: <sip:72.55.143.XXX:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 215
v=0
o=VoipSwitch 9936 9936 IN IP4 72.55.143.XXX
s=VoipSIP
=Audio Session
c=IN IP4 72.55.143.XXX
t=0 0
m=audio 8936 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (9 headers 10 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 72.55.143.XXX:8936
Found description format G729 for ID 18
Found description format telephone-event for ID 101
Capabilities: us - 0x100 (g729), peer - audio=0x100
(g729)/video=0x0
(nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1
(telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 72.55.143.XXX:8936
-- Packet2Packet bridging SIP/2563105-0819cf80 and
SIP/nyphone-081a7768
Scheduling destruction of SIP dialog
'5b7470690d6cd476346bc1113f609a4b at 164.77.171.XXX' in
32000 ms (Method:
INVITE)
set_destination: Parsing
<sip:72.55.143.XXX:5060;transport=udp> for
address/port to send to
set_destination: set destination to 72.55.143.XXX,
port 5060
Reliably Transmitting (no NAT) to 72.55.143.XXX:5060:
BYE sip:72.55.143.XXX:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP
164.77.171.XXX:5060;branch=z9hG4bK5fedb647;rport
From: "2563105"
<sip:412563105 at 164.77.171.XXX>;tag=as726ac50a
To:
<sip:5642325405 at 72.55.143.XXX>;tag=1907470723212675853288937
Call-ID:
5b7470690d6cd476346bc1113f609a4b at 164.77.171.XXX
CSeq: 104 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="test770",
realm="VoipSwitch",
algorithm=MD5, uri="sip:72.55.143.XXX:5060",
nonce="118490324119231120007472128429",
response="84f5246532adf64b1b47849b6fa03648", opaque=""
Content-Length: 0
---
== Spawn extension (test, 005642325405, 1) exited
non-zero on
'SIP/2563105-0819cf80'
vaca*CLI>
<--- SIP read from 72.55.143.XXX:5060 --->
SIP/2.0 200 OK
CSeq: 104 BYE
Via: SIP/2.0/UDP
164.77.171.XXX:5060;branch=z9hG4bK5fedb647;rport
From: "2563105"
<sip:412563105 at 164.77.171.XXX>;tag=as726ac50a
Call-ID:
5b7470690d6cd476346bc1113f609a4b at 164.77.171.XXX
To:
<sip:5642325405 at 72.55.143.XXX>;tag=1907470723212675853288937
Contact: <sip:72.55.143.XXX:5060;transport=udp>
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog
'5b7470690d6cd476346bc1113f609a4b at 164.77.171.XXX'
Method: INVITE
vaca*CLI>
I use Asterisk 1.2.13
I hope you understand me and help.
Best regards
Fernando Villarroel Noriel.
Chillan
Chile
Sorry my English.
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