[Asterisk-Users] Asterisk : No Sound No Dial

[niko singh]

Thanks for taking a look greg and hank. This seems to be getting bettre 
everyday..help please
My sjphone is running on the same box as asterisk...i believe then the red 
hat firewall should not be a problem.
Whenever i dial from CLI i get
#########

Executing Goto("OSS/dsp", "default|s|1") in new stack
    -- Goto (default,s,1)
    -- Executing Wait("OSS/dsp", "1") in new stack
  == Spawn extension (default, s, 1) exited non-zero on 'OSS/dsp'
<< Hangup on console >>
#################
but no sound
Another  problem i am having is that sjphone reports that another soft phone 
is running while asterisk is on and i need to start sjphone before asterisk. 
At this stage when i start asterisk i get the following error
###
WARNING[1116941120]: chan_sip.c:457 __sip_xmit: sip_xmit of 0x80ed894 (len 
363) to 192.246.69.223 returned -1: Bad file descriptor
###
Asterisk can register with fwd on its own but if the sjphone has been 
started it reports
###
NOTICE[1116941120]: chan_sip.c:3159 sip_reg_timeout: Registration for 
'mynumber at 192.246.69.223' timed out, trying again
###
My local ipbox address being 10.12.X.X the settings in my sjphone for 
proxydomain userdomain and registrar are all 10.12.X.X port being 5060...the 
sjphone shows in sip peers...
the relevant sections of sip.conf are:
#########
[general]
port = 5060			; Port to bind to
bindaddr = 10.12.X.X		; Address to bind SIP channel to
context = sip
    register => mynumber:mypasswd at fwd.pulver.com/1000
srvlookup = yes
maxexpirey=3600
deallow=ulaw
allow=ilbcfaultexpirey=1200
externip = 10.12.X.X
localnet = 10.12.X.X
localmask = 255.255.255.240
[fwd.pulver.com]

type=friend

secret=mypassword

username=mynumber

host=fwd.pulver.com
port = 5060

[zultys]

type=friend

host=dynamic
port = 5060
;defaultip=10.12.X.X

username=zultys

secret=blah

;dtmfmode=inband ; Choices are inband, rfc2833, or info

;mailbox=1000 ; Mailbox for message waiting indicator

context=sip
nat = yes
callerid="Me" <2124>





[mysjphone]
type=friend
host=dynamic
port = 5060
dtmfmode=inband
username=mysjphone
secret=mypassword
context = sip
careinvite = no
nat = yes
###################

the relevant section in extensions.conf are

#####
[sip]
exten => 1,1,Dial(SIP/zultys,20,tr)

exten => 2,1,Dial(SIP/mysjphone,20,tr)

exten => 1000,1,Dial(SIP/zultys&SIP/mysjphone,20,tr)
exten => _8.,1,Dial(SIP/${EXTEN-1}@fwd.pulver.com,tr)
exten => 100,1,dial(SIP/mysjphone)
exten => mysjphone,1,goto(100,1) ; To be able to dial with text, "mysjphone"
exten => 264,1,Answer
  exten => 264,2,Wait(1)
  exten => 264,3,Playtones(!950/330,!1400/330,!1800/330,0)
  exten => 264,4,Wait(5)
  exten => 264,5,StopPlaytones
  exten => 264,6,Wait(2)
  exten => 264,7,Playback(beep)
  exten => 264,8,Hangup

#############

help please..any suggestions r welcome ( i did do a ip tables -f -x )
thanks
niko






>--__--__--
>
>Message: 1
>Date: Sat, 7 Aug 2004 10:49:14 -0600 (MDT)
>From: Greg Hill <gregh-asterisk at hillnet.us>
>To: asterisk-users at lists.digium.com
>Subject: Re: [Asterisk-Users] Asterisk : No Sound Issues
>Reply-To: asterisk-users at lists.digium.com
>
>On Sat, 7 Aug 2004, niko singh wrote:
>
> > Thanks greg , for pointing out the valuable resources for reference. I
> > tried SJphone in a windows environment to connect to fwd and it worked
> > fine(including (audio). Now have to do the same thing for linux(red hat
> > 9 )  and hope the nat issue is resolved.
>
>your mention of firewalls below reminded me of a certain "feature"
>i

ncluded with RedHat 9. The installer likes to set up a firewall (using
>the ipchains tools) to help protect the machine against attacks. This
>could potentially cause problems if the firewall blocks connections when
>your softphones try to register with asterisk. A quick-and-easy temporary
>fix is to remove the firewall rules entirely by using "iptables -F;
>iptables -X" as root. The firewall rules are restored the next time you
>reboot. Long term, it would definitely be a good idea to read about
>firewalls with ipchains and get yours set up as you need.
>
> > Now i would like to connect asterisk to fwd and instead of the SJ phone
> > connecting to fwd directly i would wish to connect through asterisk, 
>writing
> > the extensions to transfer all dailled numbers from my SJphone to fwd. 
>At a
> > later stage make asterisk accept calls dialled to my fwd number and 
>operate
> > thm through the SJ phone
>
>register your box to fwd (for incoming calls to your fwd number): add to
>sip.conf in the [general] secion
>register => fwdnum:fwdpass at fwd.pulver.com
>calls to your fwd number will be routed to your context specified in the
>[general] section.
>
>To make calls to the fwd network, you'll need something like this in
>sip.conf:
>[fwd]
>type=friend
>secret=
>username=
>host=fwd.pulver.com
>context=incoming
>
>and then in your extensions.conf something like:
>exten => _8.,1,Dial(SIP/${EXTEN:1}@fwd)
>
>then any number that starts with an 8 will be tried at fwd. This exten
>statement would need to be in the same context as your softphones in order
>for them to use it.
>
> > How can nat issues be resolved with asterisk.
>
>typically you have to set up port forwarding on your nat device and use
>externip= in sip.conf. You may also need to use canreinvite=no in some
>contexts of sip.conf as well as nat=yes. Keep browsing and searching,
>especially on the wiki but also on google.
>
>Greg
>
>
>
>--__

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