[asterisk-users] outbound calls not ringing

[Ott Rose]

here is my sip.conf. i don't see it.
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ;
; this file must be done via the web gui. There are alternative files to make    ;
; custom modifications, details at: http://freepbx.org/configuration_files       ;
;--------------------------------------------------------------------------------;
;

[general]

; These files will all be included in the [general] context
;
#include sip_general_additional.conf

;sip_general_custom.conf is the proper file location for placing any sip general
;options that you might need set. For example: enable and force the sip jitterbuffer.
;If these settings are desired they should be set the sip_general_custom.conf file.
;
; jbenable=yes
; jbforce=yes
;
;It is also the proper place to add the lines needed for sip nat'ing when going
;through a firewall.  For nat'ing you'd need to add the following lines:
; nat=yes , externip= , localhost= , and optionally fromdomain= .
;
#include sip_general_custom.conf

;sip_nat.conf is here for legacy support reasons and for those that upgrade
;from previous versions.  If you have this file with lines in it please make
;sure they are not duplicated in sip_general_custom.conf, if so remove them
;from sip_nat.conf as sip_general_custom.conf will have precedence.
#include sip_nat.conf

;sip_registrations_custom.conf is for any customizations you might need to do to
;the automatically generated registrations that FreePBX makes.
;
#include sip_registrations_custom.conf
#include sip_registrations.conf

; These files should all be expected to come after the [general] context
;
#include sip_custom.conf
#include sip_additional.conf

;sip_custom_post.conf If you have extra parameters that are needed for a
;extension to work to for example, those go here.  So you have extension
;1000 defined in your system you start by creating a line [1000](+) in this
;file.  Then on the next line add the extra parameter that is needed.
;When the sip.conf is loaded it will append your additions to the end of
;that extension.
;
#include sip_custom_post.conf


> From: jsullivan at opensourcedevel.com
> To: asterisk-users at lists.digium.com
> Date: Wed, 19 Aug 2009 12:17:15 -0400
> Subject: Re: [asterisk-users] outbound calls not ringing
> 
> sip.conf
> 
> On Wed, 2009-08-19 at 15:55 +0000, Ott Rose wrote:
> > 
> > we are using Aastra 57i
> > 
> > i don't see that setting. where is it at?
> > 
> > > From: jsullivan at opensourcedevel.com
> > > To: asterisk-users at lists.digium.com
> > > Date: Wed, 19 Aug 2009 11:07:21 -0400
> > > Subject: Re: [asterisk-users] outbound calls not ringing
> > > 
> > > On Wed, 2009-08-19 at 13:54 +0000, Ott Rose wrote:
> > > > I put a post on here about my issues with outbound calls not
> > ringing
> > > > but i haven't resolved it. so i am trying again.
> > > > 
> > > > When i dial any outside number i dont get a ring tone at all. when
> > the
> > > > person picks up and starts to talk i can hear them fine. it sounds
> > > > great. How do I start to troubleshot this?
> > > <snip>
> > > What type of phones are giving you the problem? If I recall
> > correctly,
> > > our SIP phones had this problem depending on how the destination
> > handled
> > > signaling. We resolved it by adding progressinband=no (as opposed to
> > > the default never - at least I think it is the default) but this
> > > produces the problem of duplicate ring tones at times. Hope this
> > helps
> > > - John
> > > -- 
> > > John A. Sullivan III
> > > Open Source Development Corporation
> > > +1 207-985-7880
> > > jsullivan at opensourcedevel.com
> > > 
> > > http://www.spiritualoutreach.com
> > > Making Christianity intelligible to secular society
> > > 
> > > 
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> -- 
> John A. Sullivan III
> Open Source Development Corporation
> +1 207-985-7880
> jsullivan at opensourcedevel.com
> 
> http://www.spiritualoutreach.com
> Making Christianity intelligible to secular society
> 
> 
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> Register Now: http://www.astricon.net
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users

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