Trunk between asterisk and Cisco CME question


Newbie here and I search for my question with no success so here goes.

I’ve got a trunk connection setup between my asterisk and a Cisco cme. There is a cisco phone registered to the CME and I’ve got a softphone (x-lite) registered to my asterisk

ext:1000 ext:2000

I used the info in this page: as a guide for my Asterisk config files.

!!! sip.conf !!!
context = default

type =friend


!!! extensions.conf !!!

defaultexten => 2XXX,1,Dial(SIP/$) ; x.x.x.x Cisco Callmanager IP exten => 1XXX,1,Dial(SIP/$@x.x.x.x)

So from the Cisco phone I dial extension 2000 and I see from my xlite PC that extension 1000 is trying to reach me. I can answer the call and everything works fine.
The problem I have is when I dial ext 1000 from my xlite PC I get a error on my asterisk console basically telling me he doesn’t know who extension 1000 is. And I’m assuming Asterisk is saying no one with extension 1000 is registered with me

So does anyone know how I can get my softphone to call the cisco phone on the other side of the trunk?

Your extensions.conf section as posted appears to be incorrect. Please provide it in a code section, and also provide the console output when attempting the failing call.

Not sure what you mean by my extensions.conf seems to be wrong, I do have an extry for my 2000 extension below. Also when you say “Please provide it in code section” what do you mean?

exten => 2000,1,NoOp(First line)
same => n,NoOp(Second Line)
same => n,Dial(Sip/sp2untrust)
same => n,Hangup

exten => 2xxx,1,DIAL(SIP/$); Cisco Callmanager IP exten => 1XXX,1,Dial(SIP/$@

I get the below sip console error when doing so:

4 sip peers [Monitored: 1 online, 3 offline Unmonitored: 0 online, 0 offline]
[Jun 12 13:05:01] NOTICE[29343]: chan_sip.c:22753 handle_request_invite: Call from ‘sp2untrust’ ( to extension ‘1000’ rejected because extension not found in context ‘phones’.

This message tells me my asterisk isn’t aware of any extension 1000, which is true because 1000 is registered on the Cisco CME…I was under the impression that my Asterisk should only know how to route back to where the 1000 is registered.

There appear to multiple priorities without newlines between them. There is a $ where there should be a peer name. There is a $ where there should be a user name. There are no _'s before pattern matches. I have a feeling that lower case x matches as a literal in patterns, but could be wrong on that.

Basically it was so messed up that Joshua felt it unnecessary to detail all the problems.

I also didn’t know if perhaps the forum was stripping some stuff but yes, your dialplan is incomplete/incorrect.

The following is what it should look like, or at least closer:

exten => 2000,1,NoOp(First line)
same => n,NoOp(Second Line)
same => n,Dial(Sip/sp2untrust)
same => n,Hangup

exten => _2XXX,1,DIAL(SIP/${EXTEN}@; Cisco Callmanager IP
exten => _1XXX,1,Dial(SIP/${EXTEN}@

I’d also suggest reading up on dialplan basics[1].

[1] … asics.html