CIKTel account

itjumper

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#1
I have an account from CIK (www.CIKTel.com). This account is supposed to be an extension in CIK system, i.e. using SIP phone. I want to config it as a trunk in asterisk. This is my trunk setting

canreinvite=no
context=from-trunk
fromdomain=CMTor1.ciktel.com
fromuser=xxxxxxx
host=CMTor1.ciktel.com
insecure=very
nat=yes
qualify=yes
secret=yyyy
type=friend
username=zzzzzzzzz

Note that CIK has different username and auth id. With the above setting, I can make call from asterisk. But I kept getting SERVICE UNAVAILABE in the log and I could not receive any call.

I got the same symptom when using other provider (extension, not real trunk)

Anyone has any idea.
 

jgutierrez

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Feb 28, 2008
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#2
Try the following:
1. Set on /etc/asterisk/sip_general_custom.conf
context=from-pstn
2. Execute from the shell:
asterisk -rx "reload"
3. Set the appropiate inbound routes, under PBX->Inbound Routes
I would confiure a route, that would accept any number, for instance:
any DID/any CID
 

itjumper

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#3
I should have mentioned that I have also defined an inbound route to an IVR. I think from-pstn is the same as from-trunk.

thanks
 

dicko

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#4
Maybe you need a registration entry also?
 

itjumper

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#5
I did have a registration entry. I should have mentioned that too. My bad.

Anyone try treating an extension on another system as a sip trunk?
 

dicko

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#6
Then does:

sip show registry

from asterisk CLI confirm it is working?
 

itjumper

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#7
yes, sip show peers confirms that it is registered.

I can make outgoing calls, just incoming cannot reach asterisk
 

dicko

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#8
If you are successfully registered with the provider (VSP, extension on another SIP device, gateway, whatever they are all essentially the same thing) then any ensuing sip "invite" to that resolvable SIP url, will be so passed to your system (your device is the ONLY device attempting to register to that SIP endpoint, correct?), then you maybe should do

sip debug ip <ip of the registered provider>

and call that "endpoint" to see how it is being handled/rejected, possibly incompatible codec or unavailable resource but the packets should definitely arrive and give you a clue (your SIP register packets were obviously sent and successfully acknowledged so it can't be a routing problem) as to what isn't working.
 

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