[asterisk-bugs] [Asterisk 0013396]: Not able to put call on hold
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon Sep 1 05:46:32 CDT 2008
A NOTE has been added to this issue.
======================================================================
http://bugs.digium.com/view.php?id=13396
======================================================================
Reported By: sujit
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 13396
Category: Applications/app_transfer
Reproducibility: always
Severity: minor
Priority: normal
Status: feedback
Asterisk Version: 1.2.X
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Disclaimer on File?: N/A
Request Review:
======================================================================
Date Submitted: 2008-08-28 21:47 CDT
Last Modified: 2008-09-01 05:46 CDT
======================================================================
Summary: Not able to put call on hold
Description:
Hi there,
I have registered my asterisk box as SIP client to 3rd party live SIP
server and registered a FXS number.
I can make incoming call to the number from my mobile and outgoing call
from the number to my mobile successfully.
But I can not put call oh hold from . Please help.
Thanks in advance.
~Sujit Das
===============
extensions.conf
===============
;
;DO NOT MODIFY, FILE AUTOMATICALLY GENERATED BY SCRIPT configure_asterisk
;
globals ;Sujit
general
static=yes
writeprotect=no
;autofallthrough=yes ;Sujit
default
include => phones
include => parkedcalls
;sujit - start
phones ; sujit
include => internal ; sujit
include => remote ; sujit
internal
;exten => _1XX,1,NoOp()
;exten => _1XX,n,Macro(stdexten, SIP/${EXTEN},30)
;exten => _1XX,n,Playback(the-number-is-unavail)
;exten => _1XX,n,Hangup()
remote
;exten => _7XX,1,NoOp()
;exten => _7XX,n,Macro(stdexten,SIP/${EXTEN}@192.168.2.200)
;exten => _7XX,n,Hangup()
;exten => _X.,1,NoOp()
;exten => _X.,n,Macro(stdexten,SIP/${EXTEN}@my-singtel-server,30,tr)
;exten => _X.,n,Hangup()
exten => _9XXXXXXXX,1,NoOp()
exten => _9XXXXXXXX,n,Dial(SIP/${EXTEN:1}@my-singtel-server,30,tr)
exten => _9XXXXXXXC,n,Hangup()
;exten => _9XXXXXXX,n,Macro(stdexten,SIP/${EXTEN}@my-singtel-server)
;exten => _9XXXXXXX,n,Hangup()
;AsteriskSER_incoming ; sujit
;exten => 101,1,NoOp()
;exten => 101,n,Macro(stdexten,MSPD/phone0,,101)
;exten => 102,1,NoOp()
;exten => 102,n,Macro(stdexten,MSPD/phone1,,102)
;exten => 103,1,NoOp()
;exten => 103,n,Macro(stdexten,MSPD/phone2,,103)
;exten => 104,1,NoOp()
;exten => 104,n,Macro(stdexten,MSPD/phone3,,104)
;sujit - end
;connecting to other network which has 1XX numbers thru SIP protocol
;A.B.C.D is IP-Address of other board
;replace A.B.C.D and reload configuration files
;exten => _4xx,1,Macro(stdexten,SIP/${EXTEN}@A.B.C.D,,401)
;connecting to other network which has 2XX numbers thru SIP protocol
exten => _5xx,1,Macro(stdexten,SIP/${EXTEN}@A.B.C.D,,401)
exten => 101,1,Macro(stdexten,MSPD/phone0,,101)
exten => AAAAAAAA,1,Macro(stdexten,MSPD/phone1,,AAAAAAAA)
exten => 104,1,Macro(stdexten,MSPD/phone3,,104)
exten => 202,1,Macro(stdexten,SIP/202,,202)
exten => 203,1,Macro(stdexten,SIP/203,,203)
exten => 204,1,Macro(stdexten,SIP/204,,204)
exten => s,1,GotoIf($${LEN(${ARG3})} > 0?4)
exten => s,2,SetVar(VMBOX=${MACRO_EXTEN})
exten => s,3,Goto(5)
exten => s,4,SetVar(VMBOX=${ARG3})
exten => s,5,Dial(${ARG1},20,t${ARG2})
exten => s,6,Goto(s-${DIALSTATUS},1)
exten => s-ANSWER,1,Hangup
exten => s-BUSY,1,Voicemail(b${VMBOX})
exten => s-BUSY,2,Hangup
exten => _s-.,1,Voicemail(u${VMBOX})
exten => _s-.,2,Hangup
exten => 1234,1,VoiceMailMain()
exten => 1234,1,NoOp(${EXTEN})
exten => 1234,2,NoOp(${MACRO_EXTEN})
exten => 1234,3,Hangup()
macro-stdexten
exten => s, 1, Dial(${ARG1}, 25, tT)
exten => s, 2, SetVar(VMBOX=${MACRO_EXTEN})
exten => s, 3, NoOp(${MACRO_EXTEN})
exten => s, 4, NoOp(${VMBOX})
exten => s, 5, Goto(s-${DIALSTATUS},1)
;exten => s-ANSWER,1,Hangup ; sujit
exten => s-ANSWER,1,Goto(1) ; sujit
exten => s-BUSY,1,Voicemail(b${VMBOX})
;exten => s-BUSY,2,Hangup ; sujit
exten => s-BUSY,2,Goto(1) ; sujit
;exten => _s-.,1,Voicemail(u${VMBOX})
exten => _s-.,1,Goto(1) ; sujit
;exten => _s-.,2,Hangup ; sujit
exten => _s-.,2,Goto(1) ; sujit
1. exten => s, 2, Goto(s, 102)
2. exten => s, 102, Playback(vm-nobodyavail)
3. exten => s, 103, Hangup()
=================
sip.conf
=================
;
;DO NOT MODIFY, FILE AUTOMATICALLY GENERATED BY SCRIPT configure_asterisk
;
general
CONTEXt=default ; Default context for incoming calls
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
disallow=all
allow=ulaw,alaw,g729 ;also defines preference
dtmfmode=rfc2833
tos=0x10
defaultexpiry=3600 ;added by Sujit default registration expiry timer
register => AAAAAAAA:XXXXXXXXX at 203.126.17.242:5060/AAAAAAAA ;register to
external sip server
my-singtel-server
username=AAAAAAAA
type=friend
secret=XXXXXXXXX
host=203.126.17.242
fromuser=AAAAAAAA
fromdomain=203.126.17.242
dtmfmode=rfc2833
auth=md5
canreinvite=yes ;no
insecure=very
qualify=yes
nat=no
disallow=all
allow=gsm
allow=ulaw
allow=alaw
2345
type=peer
context=default ; Where to start in the dialplan when this phone calls
username=2345; SIP username for registration
secret=2345; SIP password for registration
host=dynamic ; Sip phone has a dynamic IP address
canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
insecure=invite
202
type=friend
context=default
username=202
secret=202
callerid=202
host=dynamic
canreinvite=no
203
type=friend
context=default
username=203
secret=203
callerid=203
host=dynamic
canreinvite=no
204
type=friend
context=default
username=204
secret=204
callerid=204
host=dynamic
canreinvite=no
======================================================================
----------------------------------------------------------------------
(0091949) sujit (reporter) - 2008-09-01 05:46
http://bugs.digium.com/view.php?id=13396#c91949
----------------------------------------------------------------------
Hi Leif,
Thanks for your reply. I am using board from vendor MindSpeed, In the
board, 1 FXO, 4 FXS, 4 LAN, 1 WAN port are present. I have connected
PSTN phone to the FXS port, which has number AAAAAAAA, and register to the
SingTel live SIP server 203.126.17.242, and the number is registered
successfully. As a result I can make incoming call from my mobile to the
FXS number and outgoing call to my mobile from the FXS number
successfully.
Now I am not able to put the outgoing call (to my mobile) on hold.
Please find the attached log for call hold.
In log, when hook-flash is pressed for putting the call on hold no SIP
reINVITE
is going from the broad to server though the hook-flash event is
recognized which is clear from the log.
For the expected behavior for call hold please find the attached console
and
ethereal log of testing carried out on other vendor's board. As a expected
behavior, when a outgoing call (B-party) is put on hold by pressing
hook-flash,
server will play music on-hold to the B-party (see play RTP packets in
the
ethereal log) and pressing hook-flash again, FXS will be able to talk to
B-party.
Please advise on this issue and let us know if any other information is
required.
~Sujit
Issue History
Date Modified Username Field Change
======================================================================
2008-09-01 05:46 sujit Note Added: 0091949
======================================================================
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