[Asterisk-Users] no SUBSCRIBE request sent

Wed May 17 03:27:55 MST 2006

Hi,

first of all, sorry for this long thread... I have
changed my extensions.conf like you suggested and
delete the line with subscribecontext=notify. But
unfortunately i still don't see subscribe request in
the sip debug trace.

SIP Debugging enabled
kingcoco*CLI>
<-- SIP read from 192.168.204.5:6108:

--- (0 headers 0 lines) Nat keepalive ---
kingcoco*CLI>
<-- SIP read from 192.168.204.100:5060:
INVITE sip:2002 at 192.168.204.223 SIP/2.0
Max-Forwards: 70
Content-Length: 307
Via: SIP/2.0/UDP
192.168.204.100:5060;branch=z9hG4bK4c6bfc983
Call-ID: 7e6c264483fd010
From: OptiPoint410std
<sip:2001 at 192.168.204.223>;tag=c2a05e95916bbfa;epid=SC22390b
To: sip:2002 at 192.168.204.223
CSeq: 1 INVITE
Supported: timer
Min-SE: 90
Supported: 100rel
Allow-Events: talk, hold, conference
Allow:
INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,MESSAGE,UPDATE
Content-Type: application/sdp
Contact: OptiPoint410std
<sip:2001 at 192.168.204.100:5060;transport=udp>
Supported: replaces
User-Agent: optiPoint 410_420/v4 4.1.66

v=0
o=MxSIP 0 1595508908 IN IP4 192.168.204.100
s=SIP Call
c=IN IP4 192.168.204.100
t=0 0
m=audio 5004 RTP/AVP 9 8 0 18 4 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

--- (17 headers 14 lines)---
Using INVITE request as basis request -
7e6c264483fd010
Sending to 192.168.204.100 : 5060 (non-NAT)
Found user '2001'
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 101
Peer audio RTP is at port 192.168.204.100:5004
Found description format G722
Found description format PCMA
Found description format PCMU
Found description format G729
Found description format G723
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10d
(g723|ulaw|alaw|g729)/video=0x0 (nothing), combined -
0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event),
peer - 0x1 (telephone-event), combined - 0x1
(telephone-event)
Looking for 2002 in local (domain 192.168.204.223)
list_route: hop:
<sip:2001 at 192.168.204.100:5060;transport=udp>
Transmitting (no NAT) to 192.168.204.100:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.204.100:5060;branch=z9hG4bK4c6bfc983;received=192.168.204.100
From: OptiPoint410std
<sip:2001 at 192.168.204.223>;tag=c2a05e95916bbfa;epid=SC22390b
To: sip:2002 at 192.168.204.223
Call-ID: 7e6c264483fd010
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY
Contact: <sip:2002 at 192.168.204.223>
Content-Length: 0

---
    -- Executing Dial("SIP/2001-65fe",
"SIP/2002|10|tr") in new stack
We're at 192.168.204.223 port 10830
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 11 lines
Reliably Transmitting (no NAT) to 192.168.204.5:6108:
INVITE sip:2002 at 192.168.204.5:6108 SIP/2.0
Via: SIP/2.0/UDP
192.168.204.223:5060;branch=z9hG4bK153d90a3;rport
From: "OptiPoint410std"
<sip:2001 at 192.168.204.223>;tag=as29a3f9ee
To: <sip:2002 at 192.168.204.5:6108>
Contact: <sip:2001 at 192.168.204.223>
Call-ID:
6b065f3a7c4047a52826d61c5066975d at 192.168.204.223
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 17 May 2006 08:58:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 246

v=0
o=root 24071 24071 IN IP4 192.168.204.223
s=session
c=IN IP4 192.168.204.223
t=0 0
m=audio 10830 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Called 2002
Transmitting (no NAT) to 192.168.204.100:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
192.168.204.100:5060;branch=z9hG4bK4c6bfc983;received=192.168.204.100
From: OptiPoint410std
<sip:2001 at 192.168.204.223>;tag=c2a05e95916bbfa;epid=SC22390b
To: sip:2002 at 192.168.204.223;tag=as5094780f
Call-ID: 7e6c264483fd010
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY
Contact: <sip:2002 at 192.168.204.223>
Content-Length: 0

---
kingcoco*CLI>
<-- SIP read from 192.168.204.5:6108:
SIP/2.0 180 Ringing
To: <sip:2002 at 192.168.204.5:6108>;tag=0a630b27
From:
"OptiPoint410std"<sip:2001 at 192.168.204.223>;tag=as29a3f9ee
Via: SIP/2.0/UDP
192.168.204.223:5060;branch=z9hG4bK153d90a3;rport=5060;received=192.168.204.223
Call-ID:
6b065f3a7c4047a52826d61c5066975d at 192.168.204.223
CSeq: 102 INVITE
Contact: <sip:2002 at 192.168.204.5:6108>
Content-Length: 0

--- (8 headers 0 lines)---
    -- SIP/2002-7bc1 is ringing
kingcoco*CLI>
<-- SIP read from 192.168.204.5:6108:
SIP/2.0 200 OK
To: <sip:2002 at 192.168.204.5:6108>;tag=0a630b27
From:
"OptiPoint410std"<sip:2001 at 192.168.204.223>;tag=as29a3f9ee
Via: SIP/2.0/UDP
192.168.204.223:5060;branch=z9hG4bK153d90a3;rport=5060;received=192.168.204.223
Call-ID:
6b065f3a7c4047a52826d61c5066975d at 192.168.204.223
CSeq: 102 INVITE
Contact: <sip:2002 at 192.168.204.5:6108>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Content-Length: 185

v=0
o=- 10603328 10603369 IN IP4 192.168.204.5
s=eyeBeam
c=IN IP4 192.168.204.5
t=0 0
m=audio 8702 RTP/AVP 8 0 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

--- (10 headers 9 lines)---
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.204.5:8702
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc
(ulaw|alaw)/video=0x0 (nothing), combined - 0xc
(ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event),
peer - 0x1 (telephone-event), combined - 0x1
(telephone-event)
list_route: hop: <sip:2002 at 192.168.204.5:6108>
set_destination: Parsing <sip:2002 at 192.168.204.5:6108>
for address/port to send to
set_destination: set destination to 192.168.204.5,
port 6108
Transmitting (no NAT) to 192.168.204.5:6108:
ACK sip:2002 at 192.168.204.5:6108 SIP/2.0
Via: SIP/2.0/UDP
192.168.204.223:5060;branch=z9hG4bK30ed871d;rport
From: "OptiPoint410std"
<sip:2001 at 192.168.204.223>;tag=as29a3f9ee
To: <sip:2002 at 192.168.204.5:6108>;tag=0a630b27
Contact: <sip:2001 at 192.168.204.223>
Call-ID:
6b065f3a7c4047a52826d61c5066975d at 192.168.204.223
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

---
    -- SIP/2002-7bc1 answered SIP/2001-65fe
We're at 192.168.204.223 port 13582
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to
192.168.204.100:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.204.100:5060;branch=z9hG4bK4c6bfc983;received=192.168.204.100
From: OptiPoint410std
<sip:2001 at 192.168.204.223>;tag=c2a05e95916bbfa;epid=SC22390b
To: sip:2002 at 192.168.204.223;tag=as5094780f
Call-ID: 7e6c264483fd010
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY
Contact: <sip:2002 at 192.168.204.223>
Content-Type: application/sdp
Content-Length: 246

v=0
o=root 24071 24071 IN IP4 192.168.204.223
s=session
c=IN IP4 192.168.204.223
t=0 0
m=audio 13582 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Attempting native bridge of SIP/2001-65fe and
SIP/2002-7bc1
kingcoco*CLI>
<-- SIP read from 192.168.204.100:5060:
ACK sip:2002 at 192.168.204.223 SIP/2.0
Max-Forwards: 70
Content-Length: 0
Via: SIP/2.0/UDP
192.168.204.100:5060;branch=z9hG4bK43676e2ac
Call-ID: 7e6c264483fd010
From: OptiPoint410std
<sip:2001 at 192.168.204.223>;tag=c2a05e95916bbfa;epid=SC22390b
To: sip:2002 at 192.168.204.223;tag=as5094780f
CSeq: 1 ACK
User-Agent: optiPoint 410_420/v4 4.1.66

--- (9 headers 0 lines)---
kingcoco*CLI>
<-- SIP read from 192.168.204.5:6108:

--- (0 headers 0 lines) Nat keepalive ---
kingcoco*CLI>
<-- SIP read from 192.168.204.5:6108:
BYE sip:2001 at 192.168.204.223 SIP/2.0
To:
"OptiPoint410std"<sip:2001 at 192.168.204.223>;tag=as29a3f9ee
From: <sip:2002 at 192.168.204.5:6108>;tag=0a630b27
Via: SIP/2.0/UDP
192.168.204.5:6108;branch=z9hG4bK-d87543-421140965-1--d87543-;rport
Call-ID:
6b065f3a7c4047a52826d61c5066975d at 192.168.204.223
CSeq: 2 BYE
Contact: <sip:2002 at 192.168.204.5:6108>
Max-Forwards: 70
User-Agent: eyeBeam release 3004t stamp 16741
Content-Length: 0

--- (10 headers 0 lines)---
Sending to 192.168.204.5 : 6108 (NAT)
Transmitting (NAT) to 192.168.204.5:6108:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.204.5:6108;branch=z9hG4bK-d87543-421140965-1--d87543-;received=192.168.204.5;rport=6108
From: <sip:2002 at 192.168.204.5:6108>;tag=0a630b27
To:
"OptiPoint410std"<sip:2001 at 192.168.204.223>;tag=as29a3f9ee
Call-ID:
6b065f3a7c4047a52826d61c5066975d at 192.168.204.223
CSeq: 2 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY
Contact: <sip:2001 at 192.168.204.223>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing

---
  == Spawn extension (local, 2002, 1) exited non-zero
on 'SIP/2001-65fe'
set_destination: Parsing
<sip:2001 at 192.168.204.100:5060;transport=udp> for
address/port to send to
set_destination: set destination to 192.168.204.100,
port 5060
Reliably Transmitting (no NAT) to
192.168.204.100:5060:
BYE sip:2001 at 192.168.204.100:5060 SIP/2.0
Via: SIP/2.0/UDP
192.168.204.223:5060;branch=z9hG4bK2010e979;rport
From: sip:2002 at 192.168.204.223;tag=as5094780f
To: OptiPoint410std
<sip:2001 at 192.168.204.223>;tag=c2a05e95916bbfa;epid=SC22390b
Contact: <sip:2002 at 192.168.204.223>
Call-ID: 7e6c264483fd010
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

---
kingcoco*CLI>
<-- SIP read from 192.168.204.100:5060:
SIP/2.0 200 OK
Call-ID: 7e6c264483fd010
CSeq: 102 BYE
From: sip:2002 at 192.168.204.223;tag=as5094780f
To: OptiPoint410std
<sip:2001 at 192.168.204.223>;tag=c2a05e95916bbfa;epid=SC22390b
Via: SIP/2.0/UDP
192.168.204.223:5060;branch=z9hG4bK2010e979;rport
Content-Length: 0
Supported: replaces
User-Agent: optiPoint 410_420/v4 4.1.66

--- (9 headers 0 lines)---
Destroying call
'6b065f3a7c4047a52826d61c5066975d at 192.168.204.223'
Destroying call '7e6c264483fd010'
kingcoco*CLI> sip no deb
<-- SIP read from 192.168.204.5:6108:

--- (0 headers 0 lines) Nat keepalive ---

> Try this:
> 
> [local]
> exten => 2001,1,Dial(SIP/2001,10,tr)
> exten => 2001,hint,SIP/2001
> exten => 2002,1,Dial(SIP/2002,10,tr)
> exten => 2002,hint,SIP/2002
> 
> cYa,
> Avi

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