[Asterisk-Users] Asterisk : No Sound No Dial

Sun Aug 8 08:00:27 MST 2004

Niko,

Just jumping into this thread so excuse me if you have gone down this 
path... sounds like to me the sjphone and * are fighting to use 5060. 
Obviously the one that runs second is going to loose. I would try 
setting up the sjphone (or *) to use a different port for SIP and then 
point the other application to expect communications from this new 
port. (Most people would simply use 5061.) See if that helps.

Mike

On Aug 7, 2004, at 9:07 PM, niko singh wrote:

> 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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