Hello,
I’m trying to set up an Asterisk server (not for productive use, at the moment just for testing). I followed the instructions in Asterisk: The Definitive Guide (3rd version) until the point where two endpoints call each other. That’s where I ran into problems. I managed to install
Asterisk 11 (from source)
Dahdi 2.6.1+2.6.1
Libpri 1.4.14
allowed the ports 5060, 5061 in iptables. Both my server and my two softphones (Ekiga) are in the same local network and don’t need to be connected to the internet for this setup. From the console I can place a test call, but I neither can dial any of the softphones, nor does any call from a softphone work.
Calling a softphone results in
*CLI> console dial 100
Unable to re-open DSP device /dev/dsp: No such file or directory
voice only, console video support not present
*CLI> -- Executing [100@LocalSets:1] Dial("Console/dsp", "SIP/100") in new stack
Agent policy for Console/dsp is 'never'. CC not possible
No RTP engine was found. Do you have one loaded?
Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'Console/dsp' status is 'CHANUNAVAIL'
Unable to re-open DSP device /dev/dsp: No such file or directory
Unable to re-open DSP device /dev/dsp: No such file or directory
Unable to re-open DSP device /dev/dsp: No such file or directory
[...]
<< Hangup on console >> [/code]
My sip.conf
[code][general]
context=unauthenticated ; default context for incoming calls
allowguest=no ; disable unauthenticated calls
srvlookup=yes ; enabled DNS SRV record lookup on outbound calls
udpbindaddr=0.0.0.0 ; listen for UDP requests on all interfaces
tcpenable=no ; disable TCP support
[office-phone](!) ; create a template for our devices
type=friend ; the channel driver will match on username first, IP second
context=LocalSets ; this is where calls from the device will enter the dialplan
host=dynamic ; the device will register with asterisk
nat=force_rport,comedia ; assume device is behind NAT
; *** NAT stands for Network Address Translation, which allows
; multiple internal devices to share an external IP address.
; secret= ; a secure password for this device
dtmfmode=auto ; accept touch-tones from the devices, negotiated automatically
disallow=all ; reset which voice codecs this device will accept or offer
allow=alaw ; which audio codecs to accept from, and request to, the device
allow=ulaw ; in the order we prefer
; define a device name and use the office-phone template
[100](office-phone)
secret=
allow=gsm
; define another device name using the same template
[101](office-phone)
secret=
allow=gsmMy extensions.conf
[code][LocalSets]
exten => 100,1,Dial(SIP/100)
exten => 101,1,Dial(SIP/101)
exten => 200,1,Answer()
same => n,Playback(hello-world)
same => n,Hangup()[/code]
(I know device name = phone number is not the best idea, this will be changed at some point)
Another strange thing: just one softphone is online at the moment, but it shows 2 as online
*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status Description
100/100 ....... D N 5060 Unmonitored
101/101 ....... D N 5060 Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
I hope this information is sufficient. Any help and ideas is appreciated!
ykkn