Call files issue

I’m trying to use call files for a charity autodialer. They work fine, one at a time, but if I enter more than one at a time, Asterisk tries to open all of the channels at once on the broadvoice SIP trunk. Moving them into the directory one at time or changing to spaced sequential timestamps is OK until I get some unanswered calls. If two calls go out at the same time, Broadvoice suspends my account, so a collision is not acceptable.

I don’t understand why Asterisk does not detect a busy condition and wait, but failing that, are there any settings to limit calls to the trunk? Using Asterisk 1.14

Something that can only handle one call at a time is not a trunk!!

A real trunk gets congested, rather than busy, and only when it is handling a considerable number of calls. I think what you have is an extension line from the service provider, not a trunk.

Have a look at the calllimit (spelling?) option, in sip.conf. Note details of this have changed between versions. Also note that version 1.14 won’t be released for several years yet, so I don’t know what facilities that provides in this respect!

My bad, version 1.4.40

I have not found any documentation that suggests that Call Limit can be applied to outbound calls on the trunk, just inbound SIP connections. Not sure how to apply it.

Certainly, a better SIP truck is the lowest cost option, do you have a recommendation for a single line account?


Noting that it is deprecated in

[quote];call-limit=1 ; permit only 1 [color=#FF0000]outgoing[/color] call and 1 incoming call at a time
; from the phone to asterisk (deprecated)
; 1 for the explicit peer, 1 for the explicit user,
; remember that a friend equals 1 peer and 1 user in
; memory
; There is no combined call counter for a “friend”
; so there’s currently no way in sip.conf to limit
; to one inbound or outbound call per phone. Use
; the group counters in the dial plan for that.

(This should be correct for 1.4, but may not be quite right for as there were extensive changes to the peer/user concept.)

call-limit had no effect in sip.conf, buf did work in users.conf. Thanks again for your help.