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Matteo Brancaleoni
2003-Jun-23 03:39 UTC
[Asterisk-Users] Budgetone + remote call pickup

Hi.

I've found a problem when I pickup a remote sip phone with *8.
There're both budgetones 102 and are both in the same group.
When one sip phone is ringing, I can pickup the call from
another sip phone, but the first one keeps playing a loud
busy signal... that don't go away until I receive another call
or go off hook and then on hook on the first phone.
I think that could be a budgetone bug on BYE command, since
the snom and the crisco works ok...

But anyway I attached the log file (233 is the called, 225 is
the one who pickups via *8).

Anyone experienced that?

Matteo.
-------------- next part --------------
asterisk*CLI> sip debug
SIP Debugging Enabled
    -- Accepting unauthenticated call from 213.140.14.155, requested format = 4,
actual format = 2
    -- Executing AGI("IAX2[guest@213.140.14.155:4569]/2",
"channel_lookup.agi")
in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/channel_lookup.agi
    -- AGI Script channel_lookup.agi completed, returning 0
    -- Executing Dial("IAX2[guest@213.140.14.155:4569]/2",
"Sip/233|30|m") in new stack
We're at 192.168.1.203 port 13938
Answering with preferred capability 4
10 headers, 7 lines
Reliably Transmitting:
INVITE sip:233@192.168.1.243 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK41388b4a
From: ""Guest IAX User""
<sip:asterisk@192.168.1.203>;tag=as3786d582
To: <sip:233@192.168.1.243>
Contact: <sip:asterisk@192.168.1.203>
Call-ID: 67ff8ad97a8890102acbd5277a959ea2@192.168.1.203
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Content-Type: application/sdp
Content-Length: 135
 
v=0
o=root 19057 19057 IN IP4 192.168.1.203
s=session
c=IN IP4 192.168.1.203
t=0 0
m=audio 13938 RTP/AVP 0
a=rtpmap:0 PCMU/8000
 (no NAT) to 192.168.1.243:5060
    -- Called 233
    -- Started music on hold, class 'default', on
IAX2[guest@213.140.14.155:4569]/2
Sip read: LI>
SIP/2.0 100 trying
Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK41388b4a
From: ""Guest IAX User""
<sip:asterisk@192.168.1.203>;tag=as3786d582
To: <sip:233@192.168.1.243>
Call-ID: 67ff8ad97a8890102acbd5277a959ea2@192.168.1.203
CSeq: 102 INVITE
User-Agent: Grandstream SIP UA 1.0.3.60
Content-Length: 0
 
 
8 headers, 0 lines
Sip read: LI>
SIP/2.0 180 ringing
Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK41388b4a
From: ""Guest IAX User""
<sip:asterisk@192.168.1.203>;tag=as3786d582
To: <sip:233@192.168.1.243>
Call-ID: 67ff8ad97a8890102acbd5277a959ea2@192.168.1.203
CSeq: 102 INVITE
User-Agent: Grandstream SIP UA 1.0.3.60
Content-Length: 0
 
 
8 headers, 0 lines
    -- SIP/233-a2f4 is ringing
 
10 headers, 0 lines
Sip read: LI>
INVITE sip:*8@192.168.1.203 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.235
From: <sip:225@192.168.1.203>;tag=0da17af0-4d2a-1a7d-9744-548fad8a59bb
To: <sip:*8@192.168.1.203>
Contact: <sip:225@192.168.1.235>
Call-ID: 7f17a2d7-f9b5-603d-77a5-15347af0d086@192.168.1.235
CSeq: 1320 INVITE
User-Agent: Grandstream SIP UA 1.0.3.60
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS
Content-Type: application/sdp
Content-Length: 314
 
v=0
o=225 0 0 IN IP4 192.168.1.235
s=-
c=IN IP4 192.168.1.235
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 2 15 101
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
 
12 headers, 15 lines
Using latest request as basis request
Sending to 192.168.1.235 : 5060 (non-NAT)
Capabilities: us - 4, them - 269, combined - 4
Non-codec capabilities: us - 1, them - 1, combined - 1
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.235
From: <sip:225@192.168.1.203>;tag=0da17af0-4d2a-1a7d-9744-548fad8a59bb
To: <sip:*8@192.168.1.203>;tag=as38509822
Call-ID: 7f17a2d7-f9b5-603d-77a5-15347af0d086@192.168.1.235
CSeq: 1320 INVITE
User-Agent: Asterisk PBX
Contact:
Proxy-Authenticate: Digest realm="asterisk",
nonce="466877d9"
Content-Length: 0
 
 
 to 192.168.1.235:5060
Sip read: LI>
ACK sip:*8@192.168.1.203 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.235
From: <sip:225@192.168.1.203>;tag=0da17af0-4d2a-1a7d-9744-548fad8a59bb
To: <sip:*8@192.168.1.203>;tag=as38509822
Contact: <sip:225@192.168.1.235>
Call-ID: 7f17a2d7-f9b5-603d-77a5-15347af0d086@192.168.1.235
CSeq: 1320 ACK
User-Agent: Grandstream SIP UA 1.0.3.60
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS
Content-Length: 0
 
 
11 headers, 0 lines
Sip read: LI>
INVITE sip:*8@192.168.1.203 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.235
From: <sip:225@192.168.1.203>;tag=1a7d0da1-548f-4d2a-59bb-9744a2d7ad8a
To: <sip:*8@192.168.1.203>
Contact: <sip:225@192.168.1.235>
Proxy-Authorization: DIGEST username="225",
realm="asterisk", algorithm=MD5, uri="sip:*8@192.168.1.203",
nonce="466877d9",
response="0c894fd4b402750275650e18a138123e"
Call-ID: 7f17a2d7-f9b5-603d-77a5-15347af0d086@192.168.1.235
CSeq: 1321 INVITE
User-Agent: Grandstream SIP UA 1.0.3.60
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS
Content-Type: application/sdp
Content-Length: 314
 
v=0
o=225 0 0 IN IP4 192.168.1.235
s=-
c=IN IP4 192.168.1.235
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 2 15 101
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
 
13 headers, 15 lines
Using latest request as basis request
Sending to 192.168.1.235 : 5060 (non-NAT)
Capabilities: us - 4, them - 269, combined - 4
Non-codec capabilities: us - 1, them - 1, combined - 1
Looking for *8 in interni
list_route: hop: <sip:225@192.168.1.235>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.235
From: <sip:225@192.168.1.203>;tag=1a7d0da1-548f-4d2a-59bb-9744a2d7ad8a
To: <sip:*8@192.168.1.203>;tag=as758dd6c0
Call-ID: 7f17a2d7-f9b5-603d-77a5-15347af0d086@192.168.1.235
CSeq: 1321 INVITE
User-Agent: Asterisk PBX
Contact: <sip:*8@192.168.1.203>
Content-Length: 0
 
 
 to 192.168.1.235:5060
We're at 192.168.1.203 port 12292
Answering with preferred capability 4
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.235
From: <sip:225@192.168.1.203>;tag=1a7d0da1-548f-4d2a-59bb-9744a2d7ad8a
To: <sip:*8@192.168.1.203>;tag=as758dd6c0
Call-ID: 7f17a2d7-f9b5-603d-77a5-15347af0d086@192.168.1.235
CSeq: 1321 INVITE
User-Agent: Asterisk PBX
Contact: <sip:*8@192.168.1.203>
Content-Type: application/sdp
Content-Length: 135
 
v=0
o=root 27619 27619 IN IP4 192.168.1.203
s=session
c=IN IP4 192.168.1.203
t=0 0
m=audio 12292 RTP/AVP 0
a=rtpmap:0 PCMU/8000
 
 to 192.168.1.235:5060
Reliably Transmitting:
BYE sip:233@192.168.1.243 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK41388b4a
From: ""Guest IAX User""
<sip:asterisk@192.168.1.203>;tag=as3786d582
To: <sip:233@192.168.1.243>
Contact: <sip:asterisk@192.168.1.203>
Call-ID: 67ff8ad97a8890102acbd5277a959ea2@192.168.1.203
CSeq: 103 BYE
User-Agent: Asterisk PBX
Content-Length: 0
 
 (no NAT) to 192.168.1.243:5060
    -- SIP/225-22fb answered IAX2[guest@213.140.14.155:4569]/2
    -- Stopped music on hold on IAX2[guest@213.140.14.155:4569]/2
Sip read: LI>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK41388b4a
From: ""Guest IAX User""
<sip:asterisk@192.168.1.203>;tag=as3786d582
To: <sip:233@192.168.1.243>;tag=7092c2e4-62ac-37ed-bb76-ef912374d82e
Call-ID: 67ff8ad97a8890102acbd5277a959ea2@192.168.1.203
CSeq: 103 BYE
User-Agent: Grandstream SIP UA 1.0.3.60
Contact: <sip:233@192.168.1.243:5060>
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS
Content-Length: 0
 
 
10 headers, 0 lines
Sip read: LI>
ACK sip:*8@192.168.1.203 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.235
From: <sip:225@192.168.1.203>;tag=1a7d0da1-548f-4d2a-59bb-9744a2d7ad8a
To: <sip:*8@192.168.1.203>;tag=as758dd6c0
Contact: <sip:225@192.168.1.235>
Proxy-Authorization: DIGEST username="225",
realm="asterisk", algorithm=MD5, uri="sip:*8@192.168.1.203",
nonce="466877d9",
response="9250d0ae14d8902d80a9bd94ed2e7fa2"
Call-ID: 7f17a2d7-f9b5-603d-77a5-15347af0d086@192.168.1.235
CSeq: 1321 ACK
User-Agent: Grandstream SIP UA 1.0.3.60
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS
Content-Length: 0
 
 
12 headers, 0 lines
set_destination: Parsing <sip:225@192.168.1.235> for address/port to send
to
set_destination: set destination to 192.168.1.235, port 5060
Reliably Transmitting:
BYE sip:225@192.168.1.203 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK76ac80bc
From: <sip:*8@192.168.1.203>;tag=as758dd6c0
To: <sip:225@192.168.1.203>;tag=1a7d0da1-548f-4d2a-59bb-9744a2d7ad8a
Contact: <sip:*8@192.168.1.203>
Call-ID: 7f17a2d7-f9b5-603d-77a5-15347af0d086@192.168.1.235
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0
 
 (no NAT) to 192.168.1.235:5060
  == Spawn extension (bri, 233, 2) exited non-zero on
'IAX2[guest@213.140.14.155:4569]/2'
    -- Hungup 'IAX2[guest@213.140.14.155:4569]/2'
Sip read: LI>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK76ac80bc
From: <sip:*8@192.168.1.203>;tag=as758dd6c0
To: <sip:225@192.168.1.203>;tag=1a7d0da1-548f-4d2a-59bb-9744a2d7ad8a
Call-ID: 7f17a2d7-f9b5-603d-77a5-15347af0d086@192.168.1.235
CSeq: 102 BYE
User-Agent: Grandstream SIP UA 1.0.3.60
Contact: <sip:225@192.168.1.235:5060>
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS
Content-Length: 0
 
 
10 headers, 0 lines
Message is BYE
asterisk*CLI> sip no debug
SIP Debugging Disabled
asterisk*CLI> quit
[root@asterisk asterisk]#
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