Can’t answer calls or call out!

[oskare100]

Hello,
I have two problems; I can't answer calls and I can't call anyone on the outside. But I can call the server and navigate through the menu and hear the recorded voice messages - but as soon as I try to answer from a softphone the call gets disconnected. Here is what I get from the asterisk CLI when I try to call out:

========================================================== ===============
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
Connected to Asterisk 1.4.11 currently running on localhost (pid = 15654)
Verbosity was 0 and is now 3
-- Executing [0410964106@digisip-out:1] Dial("SIP/client1-095f4a98", "SIP/0410964106@digisip") in new stack
-- Called 0410964106@digisip
[Sep 30 22:46:05] WARNING[15678]: chan_sip.c:12016 handle_response_invite: Received response: "Forbidden" from '"client1" <sip:client1@xxx.xxx.xxx.xxx>;tag=as1806b7 0d'
-- SIP/digisip-095fd960 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)

Here is what I get when I try to answer:
-- Executing [8@digisip-in:1] Dial("SIP/client1-08cefb98", "SIP/client1") in new stack
-- Called client1
-- SIP/client1-08cf1368 is ringing
-- SIP/client1-08cf1368 is ringing
-- SIP/client1-08cf1368 is ringing
== Spawn extension (digisip-in, 8, 1) exited non-zero on 'SIP/client1-08cefb98'

This is the settings in sip.conf:
register => xxxxxx:xxxxxxx:xxxx@proxy.digisip.net/5

[digisip]
type=friend
secret=xxxxx
username=xxxx
host=proxy.digisip.net
context=digisip-in
insecure=very

[client1]
type=friend
username=client1
secret=xxxxx
host=dynamic
context=digisip-out

And this is the extensions.conf:
[digisip-out]
exten => _0.,1,Dial(SIP/${EXTEN}@digisip)
exten => _0.,3,Playback(invalid)
exten => _0.,4,Hangup

[digisip-in]
exten => 5,1,Playback(welcome.gsm,answer)
exten => 5,2,Playback(if-u-know-ext-dial.gsm,skip)
exten => 5,3,Playback(or.gsm,skip)
exten => 5,4,Playback(press-2.gsm,skip)
exten => 5,5,Playback(for.gsm,skip)
exten => 5,6,Playback(service.gsm,skip)
exten => 2,7,Playback(transfer.gsm,skip)
exten => 2,2,Goto(7,1)
exten => 7,1,Dial(SIP/client1)
exten => 7-NOANSWER,n,Goto(10,1)
exten => 7-BUSY,n,Goto(11,1)
exten => 10,1,Playback(nbdy-avail-to-take-cal.gsm,skip)
exten => 10,2,Playback(please-try-again-later.gsm,skip)
exten => 10,3,Playback(or.gsm,skip)
exten => 10,4,Playback(press-1.gsm,skip)
exten => 10,5,Playback(T-to-leave-msg.gsm,skip)
exten => 10,6,Wait,2
exten => 10,7,Playback(thank-you-for-calling.gsm,skip)
exten => 11,1,Playback(busy-pls-hold.gsm,skip)
exten => 11,2,Playback(or.gsm,skip)
exten => 11,3,Playback(press-1.gsm,skip)
exten => 11,4,Playback(T-to-leave-msg.gsm,skip)
exten => 11,5,Goto(7,1)
exten => 1,1,Voicemail(default)
exten => 1,2,Playback(your-msg-has-been-saved.gsm,skip)
exten => 1,3,Playback(thank-you-for-calling.gsm,skip)

SIP show registry:
localhost*CLI> sip show registry
Host Username Refresh State Reg.Time
proxy.digisip.net:5060 0XXXXXXX 105 Registered Mon, 01 Oct 2007 10:41:36
localhost*CLI>


SIP show peers:
localhost*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
client1/client1 192.168.0.8 D N 5060 Unmonitored
digisip/XXXXX 82.209.165.194 N 5060 Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
localhost*CLI>


I really don't know how to solve this so I really need help, I can't find anywhere what I causing these errors.

Best Regards
Oskar R

 

[dthompson82]

Sounds like a codec issue. Check what codecs you have allowed for your provider and for you extensions. I usually test with ulaw or alaw if the provider supports it. Seems to be pretty common and stable.