OS Version:
Linux AsteriskNow 2.6.22.13-0.1.gcc3.4.x86.i686 #1 Mon Nov 19 19:39:41 EST 2007 i686 i686 i386 GNU/Linux
Asterisk Build:
Asterisk 1.4.18.1
Asterisk GUI-version Revision: SVN-branches_asterisknow+r2546
Hi, I need help setting up my incoming context;
My provider says to add;
exten => 16307367939,1,Answer()
into extensions.conf which I did but incoming calls fail. Outgoing calls work fine.
My sip.conf
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
allow=ulaw
allow=g729
register=“MyUser”:"MyPassword@sip.flowroute.com
[flowrowte]
type=friend
secret="MyPassword"
username="MyUser"
host=sip.flowroute.com
dtmfmode=rfc2833
context=ext-did "I also tryied from-trunk"
canreinvite=no
allow=ulaw
allow=g729
insecure=port,invite
The error I receive is;
[Apr 17 00:56:00] NOTICE[2792] chan_sip.c: Call from ‘MyUser’ to extension ‘16307367939’ rejected because extension not found.
My users.conf
callwaiting=yes
cid_number=16307367939
context=numberplan-custom-1
email="MyEmailAddress@example.com
fullname="My Name"
hasagent=yes
hasdirectory=yes
hasiax=no
hasmanager=no
hassip=yes
hasvoicemail=yes
deletevoicemail=no
host=dynamic
mailbox=1867
secret=1234
threewaycalling=yes
vmsecret=1234
registeriax=no
registersip=yes
autoprov=no
canreinvite=yes
nat=yes
dtmfmode=rfc2833
disallow=all
allow=alaw,ilbc,lpc10,adpcm,speex,ulaw,gsm,g726
signalling=fxo_ks
I think I’m missing something in the extensions.conf but I don’t know what.
Can somebody please help me with this?
I would try it with just exten => s,1,answer.
Assuming you are in the U.S. you could also try it without the 1 on the front of the number, since TELCO would eat the 1 anyway.
I just tried that and I got the same results. I also tried it without the “1” and got the same error.
Error;
[Apr 17 09:31:20] NOTICE[2792] chan_sip.c: Call from ‘15713561’ to extension ‘16307367939’ rejected because extension not found.
I checked my incoming call rulles;
Route incoming calls from provider ‘Custom - Flowroute’ that match pattern ‘_16307367939’ to ‘1867 – Israel Alba’.
This is all my voip provider told me to do for my 20 some DID’s
[inbound] - add the following to [inbound] context for your DIDs
exten => 16305805680,1,Answer()
exten => 16307367939,1,Answer()
Is there something that needs to go inside the parenthesis such as extension number perhaps?
Also for my sip.conf my voip provider told me to do this;
[flowroute] ;keep this lowercase, do not change format
type=friend
secret="MyPassword"
username="MyUserName"
host=sip.flowroute.com
dtmfmode=rfc2833
context=inbound ;change to ‘ext-did’ or ‘from-trunk’ for asterisk@home
canreinvite=no
allow=ulaw
allow=g729
insecure=port,invite
I have changed teh “context” to ext-did or from-trunk and the result is either a busy signal or pickup and hangup immediately.
I also tried to set the number to go into auto attendant and that doesn’t even work.
Where do I start troubleshooting from here? Thanks
what does the [default] context of extensions.conf look like?
The section [default] reads;
exten=6050,1,VoiceMailMain
exten=7000,1,Goto(voicemenu-custom-1|s|1)
exten=1800,1,Goto(ringroups-custom-1|s|1)
exten=o,1,Goto(default,1867,1)
and just in case you need it the same extensions.conf;
[generals]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
[globals]
trunk_1=SIP/trunk_1
flowroute_1_cid=16307367939
trunk_1_cid=6307367939
[DID_trunk_1]
include=default
exten=_X.,1,Goto(voicemenu-custom-1|s|1)
exten=s,1,ExecIf($[ “${CALLERID(num)}”="" ],SetCallerPres,unavailable)
exten=s,2,ExecIf($[ “${CALLERID(num)}”="" ],Set,CALLERID(all)=unknown <0000000>)
exten=s,3,Goto(voicemenu-custom-1|s|1)
exten=_16307367939,1,Goto(default|1867|1)
Thanks!
try this snippet in the default section - do dialplan reload after inserting
exten => s,1,noop(got a call)
exten => s,2,ExecIf($[ “${CALLERID(num)}”="" ],SetCallerPres,unavailable)
exten => s,3,ExecIf($[ “${CALLERID(num)}”="" ],Set,CALLERID(all)=unknown <0000000>)
exten => s,4,playback(tt-monkeys)
exten => s,5,hangup
exten => s,6,Goto(voicemenu-custom-1|s|1)
This should set the callerid, play the monkeys sound and hangup. If that works, change playback(tt-monkeys) to dial(sip/123) where 123 is the internal phone you want to ring.
I added the snippet but how do I do the dialplan reload? I restarted asterisk and also typed dialplan reload in the asterisk GUI CLI.
I still get a busy tone and same error;
put this in [default] and reload asterisk
exten => _16307367939,1,Playback(tt-monkeys)
exten => _XXXXXXXXXX,1,Playback(tt-monkeys)
exten => _X.,1,Playback(tt-monkeys)
I put it in default, restarted asterisk and still get the same error for inbound.
The message you are getting is consistent with an outgoing call. What does your CLI output look like with verbose 3 (core set verbose 3) when you try the incoming call?
I typed into CLI set core verbose 3 and it says;
Command> core set verbose 3
Verbosity is at least 3
Asterisk logs report same error for incoming;
Apr 17 11:36:47] NOTICE[2789] chan_sip.c: Call from ‘15713561’ to extension ‘16307367939’ rejected because extension not found.
You have been very helpful and I would like to help you help me so it it makes it better I’m willing to reinstall AstriskNow from scratch.
I noticed there is a problem with the GUI with the Users field. I can’t select or modify my user extension. I even tried IE7.
Thanks!
Somewhere in your dialplan (extensions.conf) is a line that is doing a dial(SIP/16307367939). That line should be doing Dial(Sip/1867). If you find that line and change it, this will go away. More than I can say for the GUI problem 
I looked at every line but other then incoming there is nothing else I see close to what you pointed out.
I’m going to reinstall asterisk and start fresh, I know there is probably an easy solution but I don’t know what it is.
I’ll post my results after a fresh install. Thanks!
I installed Asterisk (Ver. 1.4.24.1) FreePBX 2.5.1.2 and everything works wonderful now.
I registered my SIP trunks without a problem.
I have one item left to work on and that is music on hold.
I saw another post where you instructed someone to use sox to play mp3’s
My problem is that I’m able to upload .WAV but not .MP3’s. When I click on upload box it takes me back to the main screen of the GUI.
I’m able to upload WAV files I had laying around from Gizmo sofphone had isntalled once. The difference is the size of the WAV files.
Can you point me to the right direction on fixing this issue?
Thanks!