Why a DID from trunk should be in 'default' context?

Asterisk Asterisk Support
1

Hi
Perhaps a newbie question:

I have an Asterisk system connected trunk to trunk to a Cisco router to be used as an ISDN gateway
The incoming calls from ISDN passing through Cisco and presented to Asterisk SIP “trunk1” are rejected by asterisk itself -because extension not found in context ‘default’-
Effectively, the ‘default’ context is empty

I have another asterisk system that normally accepts calls on SIP trunk without this error, simply configuring incoming called number in the related trunk DID section on extension.conf

The same way , on the first Asterisk, the incoming called number is assigned in extension.conf:

[DID_trunk_1]
include=DID_trunk_1_default

[DID_trunk_1_default]
exten=_0123456789,1,Goto(default,5001,1)

but it doesn’t work until the same string is inserted in ‘default’ context

Why is this ??

Should i
include=DID_trunk_1_default
on ‘default’ context ?

Why the other Asterisk works with nothing present on ‘default’ context ?

Thank you very much

2

Please post the sip.conf entry for the SIP Trunk peer and the extensions.conf file.

3

That is not the right syntax for sub-classing. It isn’t even the right syntax for diaplan includes, but this ins’t the dialplan.

It should be:

DID_trunk_1

4

Thanks for answers
This is a fresh ubuntu installation, just configured enabling GUI , so all configuration comes from installation default…

There are two SIP trunks
One called IPO500 the other trunk_1 (the one toward Cisco)

The only uncommented part of sip.conf is this :

[general]
context=default
allowoverlap=no
realm=ubuntu
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
transport=udp
srvlookup=yes
subscribecontext=default

extensions.conf is a little bit long to be posted here , is there any particular section you need other than these ??

[globals]
CONSOLE=Console/dsp
IAXINFO=guest
TRUNK=DAHDI/G2
TRUNKMSD=1
FEATURES=
DIALOPTIONS=
RINGTIME=20
FOLLOWMEOPTIONS=
PAGING_HEADER=Intercom
ipo500=SIP/ipo500
GLOBAL_OUTBOUNDCID=
GLOBAL_OUTBOUNDCIDNAME=
CID_5000=5000
CID_ipo500=ubuntu
FOLLOWME_5000=1
CID_5001=5001
trunk_1=SIP/trunk_1
CID_5002=5002
CID_trunk_1=ubuntu

[default]
(empty)

[DID_ipo500]
include = DID_ipo500_default

[DID_ipo500_default]
exten = _5XXX,1,Goto(default,${EXTEN:0},1)

[CallingRule_IPO200]
exten = _2XX,1,Macro(trunkdial-failover-0.3,${ipo500}/${EXTEN:0},ipo500,ubuntu12)

[CallingRule_IPO400]
exten = _4XX,1,Macro(trunkdial-failover-0.3,${ipo500}/${EXTEN:0},ipo500,ubuntu12)

[DID_trunk_1]
include = DID_trunk_1_default

[DID_trunk_1_default]
exten = _0123456789,1,Goto(ringroups-custom-1,s,1)

[CallingRule_Urbane]
exten = _0.,1,Macro(trunkdial-failover-0.3,${trunk_1}/${EXTEN:0},trunk_1,Ubuntu12)

[CallingRule_Cellulari]
exten = _3.,1,Macro(trunkdial-failover-0.3,${trunk_1}/${EXTEN:0},trunk_1,Ubuntu12)

[DLPN_DialPlan1]
include = CallingRule_IPO200
include = CallingRule_IPO400
include = CallingRule_Urbane
include = CallingRule_Cellulari
include = default
include = parkedcalls
include = conferences
include = ringgroups
include = voicemenus
include = queues
include = voicemailgroups
include = directory
include = pagegroups
include = page_an_extension

[ringroups-custom-1]
exten = s,1,NoOp(Test)
exten = s,n,Dial(SIP/5001&SIP/5002,50,${DIALOPTIONS}i)
exten = s,n,Hangup

Thank you for patience

5

Sorry. I misread it that that was sip.conf. Although plain = is OK in extensions.conf, the official syntax is =>.

However, as you now say this was done with a GUI, and the current GUI is FreePBX, you need to go to freepbx.org/forums/

6

No, it isn’t Freepbx,
I think it’s a genuine Asterisk distro :

Asterisk Build:
Asterisk/1.8.13.1~dfsg-1ubuntu2
Asterisk GUI-version : SVN–r

The gui included is provided by Digium (and the Digium community link leads to this forum…)

7

That’s end of life.

8

[quote=“david55”]Sorry. I misread it that that was sip.conf. Although plain = is OK in extensions.conf, the official syntax is =>.

However, as you now say this was done with a GUI, and the current GUI is FreePBX, you need to go to freepbx.org/forums/[/quote]

I’m a little bit confused…

Asterisk is the PBX software

AsteriskNow is the pre-packed linux system including Asterisk

They can be managed via command-line but if you want a graphic environment you have to go to official GUI FreePBX (as indicated on AsteriskNOW post installation webpage)

FreePBX , the other way, seems to have an its own distro…

Anyway FreePBX goes to modify the same files that AsteriskNOW has and also Asterisk has

So why is there a forum for the only GUI , or better other forum (you suggest me to go) when the “core” is the original Asterisk discussed here ???

Let me help to understand…
Thank you

9

Because the GUIs maintain the configuration files in a way that is specific to the GUI, and directly modifying them is likely to break the GUI. They also tend to use terms like DID, trunk, and ring group, that are not terms used by Asterisk itself. In the case of the dialplans, most of the dialplan is part of the GUI, rather than Asterisk.

Also, as I understand it, AsteriskNOW is not just the third party OS, plus Asterisk, it is the OS, Asterisk, and, currently, the third party FreePBX. I believe it is an installation choice as to whether or not you use FreePBX.