No Of SIP Channels In Trunk

I have an ACH call conferencing application which I have trunked to an Asterisk Box.

I created an extension on the asterisk like 2000 and on the ACH conf application created an account with the asterisk credentials.

Now on the inbound of my asterisk I have pointed a DID to go to 2000.

This 2000 then rings the call conf application .Now my problem is when I try to cal from another line i get a user is busy message.

My message is there a way I can define concurrent channels on my asterisk box to take more than one call ?


you can define the call limit, but this is normally more than 1 in default

do a sip show peer 2000 to whats set.

I set the connection as an extension or is it supposed to be trunk ? if so on the asterisk side that wil be sending calls to call conf app what are mu settings ?? peer ??

Name : 200400
Secret :
MD5Secret :
Context : from-internal
Subscr.Cont. :
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup : >
Pickupgroup :
Mailbox : 200400@default
VM Extension : *97
LastMsgsSent : 32767/65535
Call limit : 50
Dynamic : Yes
Callerid : “device” <200400>
MaxCallBR : 384 kbps
Expire : -1
Insecure : no
Nat : Always
ACL : No
T38 pt UDPTL : No
CanReinvite : No
PromiscRedir : No
User=Phone : No
Video Support: No
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
LastMsg : 0
ToHost :