Can't make external calls from my softphone

Asterisk Asterisk Support
1

Guys I have installed asterisk 4.1 on my qnap device and am having issues calling out (both uk landline and mobile numbers)

My sipgate trunk is showing registered and my 3cx on my windows desktop showing registered to the PBX.

When I call an external number I get the an error on the softphone of “SERVER UNREACHABLE”

On the Asterisk logs I am seeing:

[Dec 13 22:22:18] NOTICE[7464] chan_sip.c: Call from ‘’ to extension ‘99900972595301124’ rejected because extension not found.
[Dec 13 22:22:29] ERROR[6037] config.c: *********************************************************
[Dec 13 22:22:29] ERROR[6037] config.c: *********** YOU SHOULD REALLY READ THIS ERROR ***********
[Dec 13 22:22:29] ERROR[6037] config.c: Future versions of Asterisk will treat a #include of a file that does not exist as an error, and will fail to load that configuration file. Please ensure that the file ‘…/zaptel.conf’ exists, even if it is empty.
[Dec 13 22:22:29] ERROR[6037] config.c: *********** YOU SHOULD REALLY READ THIS ERROR ***********
[Dec 13 22:22:29] ERROR[6037] config.c: *********************************************************
[Dec 13 22:22:29] WARNING[6053] app_system.c: Unable to execute ‘ztscan > /etc/asterisk/ztscan.conf’
[Dec 13 22:39:11] NOTICE[7464] chan_sip.c: Call from ‘’ to extension ‘9900972595301124’ rejected because extension not found.
[Dec 13 22:55:46] NOTICE[7464] chan_sip.c: Call from ‘’ to extension ‘972595301124’ rejected because extension not found.
[Dec 13 22:56:02] WARNING[26586] channel.c: No channel type registered for ‘’
[Dec 13 22:56:02] WARNING[26586] app_dial.c: Unable to create channel of type ‘’ (cause 66 - Channel not implemented)

my sip.conf

context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes

my extensions.conf

[general]
static = yes
writeprotect = no
clearglobalvars = no

[globals]
CONSOLE = Console/dsp
IAXINFO = guest
TRUNK = Zap/G2
TRUNKMSD = 1
FEATURES =
DIALOPTIONS =
RINGTIME = 20
FOLLOWMEOPTIONS =
PAGING_HEADER = Intercom
PAGING_TIMEOUT = 60
GLOBAL_OUTBOUNDCID =
GLOBAL_OUTBOUNDCIDNAME =
CID_6005 = 08451547038
2823768 = SIP/2823768
exten => 2823768,1,Playback(vm-goodbye)
exten => 2823768,n,Hangup()
08444879795 = SIP/08444879795
;TRUNK=IAX2/user:pass@provider

I am new to asterisk, please help me understand whats going on here its been 3 days and I am not getting any closer. Assistance please and thank you

2

Please start here: www.asteriskdocs.org

3

Based on you configuration you are just copying and pasting, and unfortunately you don’t have no idea of what you are doing. You have the answer to your problem in your log messages.

[quote][Dec 13 22:56:02] WARNING[26586] channel.c: No channel type registered for ‘’
[Dec 13 22:56:02] WARNING[26586] app_dial.c: Unable to create channel of type ‘’ (cause 66 - Channel not implemented)[/quote]

Based on your extensions.conf you don’t have any context for handle this extension ‘99900972595301124’

And where is the SIP trunk configuration for your sipgate trunk

4

Thank you for pointing out the obvious, I am new and require a little help to get me started.

Issue1: chan_sip.c: Call from ‘’ to extension ‘99900972595301124’ rejected because extension not found.

I am unsure why the above number is showing up, not a number I am dialing from my softphone. however I need to understand how to insert a context to allow me to dial out all national uk numbers and mobiles - any guidance would be appreciated

The sipgate trunk is registered, all I have done with this through the GUI is to pop the host, user and password in at this point with the sipgate website showing my trunk as online. Where would you like me to pull the config from?

Thank you

5

Probably because you have allowguest=yes and are being attacked by a toll fraudster.

6

thanks David, allowguest=yes has now been changed. Could you please help me understand why I cannot call out? could I post anything else to help facilitate in getting this resolved?

Thank you for taking the time.

7

I think the reasons for not being able to call out have been explained. Your configuration is so far from a valid one that the only sensible advice has already been given, namely read asteriskdocs.org/ You should also look at the sample configuration files that come with Asterisk; you appear not to have used those as a starting point.