Incoming call - Trixbox receiving, but busy signal
Submitted by lukey on Thu, 04/17/2008 - 4:35am.
Hi!
I've got the following SIP Trunk configuration in TrixboxCE v2.2:
In FreePBX module, Basic > Trunks:
OUTGOING SETTINGS:
fromdomain=sip.halonet.pl
fromuser=sanbetsolt
host=sip.halonet.pl
insecure=very
secret=mypassword
type=friend
username=sanbetsolt
INCOMING SETTINGS:
context=from-trunk
secret=mypassword
type=user
REGISTER STRING:
sanbetsolt:mypassword@sip.halonet.pl/sanbetsolt
Allow Anonymous Inbound SIP Calls: YES
The outgoing calls are working fine. When I call external number through this trunk, the remote party sees the PSTN number , which I have received from Halonet VoIP provider.
The problem is with incoming calls, placed to that same PSTN number mentioned above.
The Trixbox is seeing the calls, but it gives busy signal.
When I look to Reports section of FreePBX, just after trying to make an incoming call I can see:
Channel Source Clid Dst Disposition Duration
SIP/sanbetsolt-09601328 0223980381 0223980381 sanbetsolt NO ANSWER 00:00
At the same time, I have zaptel channel configured for receiving calls from analogue PSTN line. It works fine. A report from a successful connection, when I can hear my IVR looks like this:
Channel Source Clid Dst Disposition Duration
Zap/2-1 0223980381 0223980381 s ANSWERED 00:07
I noticed, that Dst is different. To me, "s" is my IVR menu, but "sanbetsolt" is just the SIP Trunk account.. I've got busy signal, nothing more.
In InboundSettings menu, I tried to send incoming calls (any DID/ any CID) directly to Extensions, or to IVR. But each time I got the busy signal.
A few lines from /var/log/asterisk/full, just after a try for incoming call to SIP trunk:
Apr 17 13:31:49 DEBUG[3153] chan_sip.c: Setting NAT on RTP to 524288
Apr 17 13:31:49 DEBUG[3153] chan_sip.c: Checking SIP call limits for device sanbetsolt
Apr 17 13:31:49 DEBUG[3153] chan_sip.c: build_route: Record-Route hop:
Apr 17 13:31:49 DEBUG[6259] pbx.c: Expression result is '0'
Apr 17 13:31:49 DEBUG[6259] pbx.c: Function result is 'sanbetsolt'
Apr 17 13:31:49 DEBUG[6259] pbx.c: Expression result is '1'
Apr 17 13:31:49 WARNING[6259] pbx.c: Channel 'SIP/sanbetsolt-0958bce0' sent into invalid extension 'sanbetsolt' in context 'from-trunk', but no invalid handler
Apr 17 13:31:49 DEBUG[6259] cdr_addon_mysql.c: cdr_mysql: inserting a CDR record.
Apr 17 13:31:49 DEBUG[6259] cdr_addon_mysql.c: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid) VALUES ('2008-04-17 13:31:49','0223980381 ','0223980381 ','sanbetsolt','from-trunk', 'SIP/sanbetsolt-0958bce0','','GotoIf','1?from-trunk|sanbetsolt|1',0,0,'NO ANSWER',3,'','1208431909.214')
Apr 17 13:31:49 DEBUG[6259] chan_sip.c: update_call_counter(sanbetsolt) - decrement call limit counter
Any help with this will be apreciated...
I had a similar problem and
Fri, 05/16/2008 - 5:38am
I had a similar problem and was racking my brain on it.
It turned out it was the context for the "OUTGOING" Peer which I needed to change. Once I changed this. I went to /etc/asterisk/extensions_custom.conf. Created a new stanza for this context (i.e. from-gizmo):
[from-gizmo]
exten => s,1,Goto(ext-group,6001,1)
or
[from-gizmo] ; Tried this first
exten => s,1,Dial(SIP/xxxx)
I then restarted asterisk and my phone's rang.
@james968:
Are you sure that
Fri, 05/16/2008 - 10:30am
@james968:
Are you sure that we are talking about the same issue?
To remind my problem: someone is calling FROM outside TO my Trixbox machine. That call is placed on a SIP trunk PSTN number. Instead of hearing the IVR menu (which is at the same time heard, when somebody is calling from outside to another PSTN number on zaptel channel), the person gets a busy tone.
If it's the same issue you have found a solution to, then please explain what should I put in extensions_custom.conf, to get my system working properly?
You showed an example with [from-gizmo]. What is gizmo in my system? Where can I find it?
And what should be put in the exten =>, to make that incoming call be redirected to default IVR?
Thanks in advance...
Member Since:
2008-04-02