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Summary:ASTERISK-00533: Call park request gets reorder tone on sipura SPA-2000 SIP phone adapter
Reporter:hwstar (hwstar)Labels:
Date Opened:2003-11-16 14:23:13.000-0600Date Closed:2011-06-07 14:10:09
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:If I dial an outside destination from the SIP adapter (e.g. 6192891212), flash the hookswitch, and then attempt to park it by dialing the park extension (in my case 190). I get a reorder tone.

The context 'parkedcalls' is included in the
context 'house-toll' in extensions.conf so I don't understand why asterisk is complaining that the 190 extension is not found.


****** ADDITIONAL INFORMATION ******


Output from sip debug commend:

*CLI> sip debug
SIP Debugging Enabled
Sip read:
INVITE sip:92891212@192.168.17.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.17.6:5061;branch=z9hG4bk-6306a05c
From: SIP LINE 2 <sip:102@gw2.skrodgerslmca.org>;tag=ef3490f34f3b2031
To: <sip:92891212@gw2.skrodgerslmca.org>;tag=as442fcd3d
Call-ID: cd79426d-ca22e507@192.168.17.6
CSeq: 103 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest username="102",realm="asterisk",nonce="6b118500",uri="sip:92891212@gw2.skrodgerslmca.org",algorithm=MD5,response="1668b061dece85dafc0a68d831f856d2"
Contact: SIP LINE 2 <sip:102@192.168.17.6:5061>
Expires: 30
User-Agent: Sipura/SPA2000-1.0.9
Content-Length: 395
Content-Type: application/sdp
                                                                               
v=0
o=- 8806596 8806596 IN IP4 192.168.17.6
s=-
c=IN IP4 0.0.0.0
t=0 0
m=audio 16460 RTP/AVP 0 8 96 2 97 98 18 101 100
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:18 G729a/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 NSE/8000
a=ptime:20
a=sendonly
                                                                               
13 headers, 18 lines
Using latest request as basis request
Sending to 192.168.17.6 : 5061 (non-NAT)
Found audio format UNKN
Found audio format ALAW
Found audio format UNKN
Found audio format GSM
Found audio format UNKN
Found audio format UNKN
Found audio format UNKN
Found audio format UNKN
Found audio format UNKN
Found description format PCMU
Found description format PCMA
Found description format G726-40
Found description format G726-32
Found description format G726-24
Found description format G726-16
Found description format G729a
Found description format telephone-event
Found description format NSE
Capabilities: us - 4, them - 1308/0, combined - 4
Non-codec capabilities: us - 1, them - 1, combined - 1
   -- Started music on hold, class 'default', on Zap/1-1
We're at 192.168.17.2 port 10822
Answering with preferred capability 4
Answering with non-codec capability 1
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.17.6:5061;branch=z9hG4bk-6306a05c
From: SIP LINE 2 <sip:102@gw2.skrodgerslmca.org>;tag=ef3490f34f3b2031
To: <sip:92891212@gw2.skrodgerslmca.org>;tag=as442fcd3d
Call-ID: cd79426d-ca22e507@192.168.17.6
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:92891212@192.168.17.2>
Content-Type: application/sdp
Content-Length: 189
                                                                               
v=0
o=root 14972 14972 IN IP4 192.168.17.2
s=session
c=IN IP4 192.168.17.2
t=0 0
m=audio 10822 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
                                                                               
to 192.168.17.6:5061
Sip read:
ACK sip:92891212@192.168.17.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.17.6:5061;branch=z9hG4bk-6306a05c
From: SIP LINE 2 <sip:102@gw2.skrodgerslmca.org>;tag=ef3490f34f3b2031
To: <sip:92891212@gw2.skrodgerslmca.org>;tag=as442fcd3d
Call-ID: cd79426d-ca22e507@192.168.17.6
CSeq: 103 ACK
Max-Forwards: 70
Proxy-Authorization: Digest username="102",realm="asterisk",nonce="6b118500",uri="sip:92891212@gw2.skrodgerslmca.org",algorithm=MD5,response="903f472ad8de2b5a049719f8a3eecccb"
Contact: SIP LINE 2 <sip:102@192.168.17.6:5061>
User-Agent: Sipura/SPA2000-1.0.9
Content-Length: 0
                                                                               
                                                                               
11 headers, 0 lines
Sip read:
INVITE sip:190@gw2.skrodgerslmca.org SIP/2.0
Via: SIP/2.0/UDP 192.168.17.6:5061;branch=z9hG4bk-80b260cc
From: SIP LINE 2 <sip:102@gw2.skrodgerslmca.org>;tag=ef3490f34f3b2031
To: <sip:190@gw2.skrodgerslmca.org>
Call-ID: 2d600f80-a9decaac@192.168.17.6
CSeq: 101 INVITE
Max-Forwards: 70
Contact: SIP LINE 2 <sip:102@192.168.17.6:5061>
Expires: 240
User-Agent: Sipura/SPA2000-1.0.9
Content-Length: 373
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER
Supported: 100rel
Content-Type: application/sdp
                                                                               
v=0
o=- 8806829 8806829 IN IP4 192.168.17.6
s=-
c=IN IP4 192.168.17.6
t=0 0
m=audio 16462 RTP/AVP 0 8 96 2 97 98 101 100
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 NSE/8000
a=ptime:20
a=sendrecv
                                                                               
14 headers, 17 lines
Using latest request as basis request
Sending to 192.168.17.6 : 5061 (non-NAT)
Found audio format UNKN
Found audio format ALAW
Found audio format UNKN
Found audio format GSM
Found audio format UNKN
Found audio format UNKN
Found audio format UNKN
Found audio format UNKN
Found description format PCMU
Found description format PCMA
Found description format G726-40
Found description format G726-32
Found description format G726-24
Found description format G726-16
Found description format telephone-event
Found description format NSE
Capabilities: us - 4, them - 1052/0, 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 192.168.17.6:5061;branch=z9hG4bk-80b260cc
From: SIP LINE 2 <sip:102@gw2.skrodgerslmca.org>;tag=ef3490f34f3b2031
To: <sip:190@gw2.skrodgerslmca.org>;tag=as62463f04
Call-ID: 2d600f80-a9decaac@192.168.17.6
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Proxy-Authenticate: Digest realm="asterisk", nonce="16bd7ec8"
Content-Length: 0
                                                                               
                                                                               
to 192.168.17.6:5061
Sip read:
ACK sip:190@gw2.skrodgerslmca.org SIP/2.0
Via: SIP/2.0/UDP 192.168.17.6:5061;branch=z9hG4bk-80b260cc
From: SIP LINE 2 <sip:102@gw2.skrodgerslmca.org>;tag=ef3490f34f3b2031
To: <sip:190@gw2.skrodgerslmca.org>;tag=as62463f04
Call-ID: 2d600f80-a9decaac@192.168.17.6
CSeq: 101 ACK
Max-Forwards: 70
Contact: SIP LINE 2 <sip:102@192.168.17.6:5061>
User-Agent: Sipura/SPA2000-1.0.9
Content-Length: 0
                                                                               
                                                                               
10 headers, 0 lines
Sip read:
INVITE sip:190@gw2.skrodgerslmca.org SIP/2.0
Via: SIP/2.0/UDP 192.168.17.6:5061;branch=z9hG4bk-9c0b6c90
From: SIP LINE 2 <sip:102@gw2.skrodgerslmca.org>;tag=ef3490f34f3b2031
To: <sip:190@gw2.skrodgerslmca.org>
Call-ID: 2d600f80-a9decaac@192.168.17.6
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest username="102",realm="asterisk",nonce="16bd7ec8",uri="sip:190@gw2.skrodgerslmca.org",algorithm=MD5,response="033bdfee9b8420517eb54b102b1b6b97"
Contact: SIP LINE 2 <sip:102@192.168.17.6:5061>
Expires: 240
User-Agent: Sipura/SPA2000-1.0.9
Content-Length: 373
Content-Type: application/sdp
                                                                               
v=0
o=- 8806829 8806829 IN IP4 192.168.17.6
s=-
c=IN IP4 192.168.17.6
t=0 0
m=audio 16462 RTP/AVP 0 8 96 2 97 98 101 100
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 NSE/8000
a=ptime:20
a=sendrecv
                                                                               
13 headers, 17 lines
Using latest request as basis request
Sending to 192.168.17.6 : 5061 (non-NAT)
Found audio format UNKN
Found audio format ALAW
Found audio format UNKN
Found audio format GSM
Found audio format UNKN
Found audio format UNKN
Found audio format UNKN
Found audio format UNKN
Found description format PCMU
Found description format PCMA
Found description format G726-40
Found description format G726-32
Found description format G726-24
Found description format G726-16
Found description format telephone-event
Found description format NSE
Capabilities: us - 4, them - 1052/0, combined - 4
Non-codec capabilities: us - 1, them - 1, combined - 1
Looking for 190 in house-toll
Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.17.6:5061;branch=z9hG4bk-9c0b6c90
From: SIP LINE 2 <sip:102@gw2.skrodgerslmca.org>;tag=ef3490f34f3b2031
To: <sip:190@gw2.skrodgerslmca.org>;tag=as62463f04
Call-ID: 2d600f80-a9decaac@192.168.17.6
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:@192.168.17.2>
Content-Length: 0
                                                                               
                                                                               
to 192.168.17.6:5061
Sip read:
ACK sip:190@gw2.skrodgerslmca.org SIP/2.0
Via: SIP/2.0/UDP 192.168.17.6:5061;branch=z9hG4bk-9c0b6c90
From: SIP LINE 2 <sip:102@gw2.skrodgerslmca.org>;tag=ef3490f34f3b2031
To: <sip:190@gw2.skrodgerslmca.org>;tag=as62463f04
Call-ID: 2d600f80-a9decaac@192.168.17.6
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest username="102",realm="asterisk",nonce="16bd7ec8",uri="sip:190@gw2.skrodgerslmca.org",algorithm=MD5,response="6561295f713465d46e522fb84c569734"
Contact: SIP LINE 2 <sip:102@192.168.17.6:5061>
User-Agent: Sipura/SPA2000-1.0.9
Content-Length: 0
                                                                               
                                                                               
11 headers, 0 lines
                                                                               
*CLI> sip no debug
SIP Debugging Disabled
   -- Stopped music on hold on Zap/1-1
   -- Hungup 'Zap/1-1'
 == Spawn extension (macro-dialout-pstn, s, 1) exited non-zero on 'SIP/102-31f0' in macro 'dialout-pstn'
 == Spawn extension (house-toll, s, 102) exited non-zero on 'SIP/102-31f0'
                                                                               
Parkedcalls context inclusion in extensions.conf:

;
; Context for phones which can place toll calls on the pstn
;
                                                                                                                                         
[house-toll]
                                                                                                                                         
include => parkedcalls
include => internal-extensions
include => system-speeddials
include => testing-extensions
                                                                                                                                         
ignorepat => 9
include => emerg
include => pstn-local
include => iaxtel
ignorepat => 8
include => pstn-domestic-toll


My sip.conf file:

[general]
port=5060
bindaddr=192.168.17.2
tos=lowdelay
disallow=all
allow=ulaw
context=aliens
;
; SIP Entry for sipura line 1
; This phone is allowed to dial extensions and local phone numbers
;
[101]
type=friend
host=dynamic
context=house-toll
canreinvite=yes
qualify=300
secret=xxxxxx
callerid="Sipura Line 1" <101>
username=101
callgroup=1
pickupgroup=1
mailbox=101@default
nat=0
                                                                                                                                         
; Sample for sipura line 2
; This phone is allowed to dial extensions and local phone numbers
;
[102]
type=friend
host=dynamic
context=house-toll
canreinvite=yes
qualify=300
secret=yyyyyy
callerid="Sipura Line 2" <102>
username=102
callgroup=1
pickupgroup=1
mailbox=102@default


My parking.conf file:

                                                                                                                                         
[general]
parkext => 190                          ; What ext. to dial to park
parkpos => 191-195                      ; What extensions to park calls on
context => parkedcalls                  ; Which context parked calls are in
parkingtime => 120                      ; Number of seconds a call can be parked for (default is 45 seconds)





Comments:By: Brian West (bkw918) 2003-11-16 15:39:55.000-0600

You can't do native sip transfers to parking.  CANT BE DONE.  You will need to enable # transfers or:

exten => _2XX,1,Answer
exten => _2XX,2,Wait(3)
exten => _2XX,3,ParkAndAnnounce(pbx-transfer:PARKED|7200|SIP/${EXTEN:1}|default,${EXTEN:1},1)

I use the above with my 7960's native sip transfers.  So if i'm at exten 11, I transfer to 211 and it calls me back in 3 seconds with the parking number.

bkw
By: denon (denon) 2003-11-16 16:02:22.000-0600

I like the idea of using the 7960's new intercom features to ParkAndAnnounce .. though, you know my long-term ideas on a 7960 on-screen service. PHP interface tied to * management interface..
By: hwstar (hwstar) 2003-11-16 16:57:18.000-0600

I'm curious as to what the technical reasons are for not allowing native transfers. Is this a limitation of the SIP protocol, or is it something
in the way the parking code is implemented in Asterisk?
By: Mark Spencer (markster) 2003-11-17 00:01:04.000-0600

It is not clear that the SIP protocol permits us to send audio in between the "REFER" request and the "BYE" request we are obligated to send following.  It's *especially* not clearly whether we can send a new "INVITE" to get the audio stream back to coming from us instead of the other phone if it was native bridged.

If you want call features you should be thinking MGCP, not SIP.
By: hwstar (hwstar) 2003-11-17 09:28:30.000-0600

Thanks for the info Mark.

I'm in a learning mode trying to figure out what can and can't be done. There
are no customers waiting for a solution to this.

Can I close this, or does the buglist administrator do that?

Steve.

edited on: 11-17-03 09:22
By: Brian West (bkw918) 2003-11-20 12:58:04.000-0600

This feature is already on the list of things todo.  Its not that high priority.  I personally would like to see native siptransfers to parking.  I think it can be done.  I'm going to close this ticket for now.