[asterisk-users] outbound calls not ringing
Ott Rose
sixfourimpala at hotmail.com
Thu Aug 20 08:50:02 CDT 2009
Thanks that is very helpful info. I am still trying to figure out how asterisk and freepbx works together. what do I add in those files to get the ringing to work. I checked teh Dail options under General Options and its set to tr.
> Date: Thu, 20 Aug 2009 10:51:25 +1200
> From: duncan at e-simple.co.nz
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] outbound calls not ringing
>
> Generally with FreePBX the ring options are set in the General Options -
> you can set the Dial options which are normally tr, but I guess that
> isn't working for you.
>
> The SIP files you could edit would have custom in their name, otherwise
> your changes will be overwritten when you reload freepbx
>
> You could put this in sip_general_custom.conf which will be included
>
> Cheers Duncan
>
> John A. Sullivan III wrote:
> > Oops! - You're using FreePBX - someone who knows more about FreePBX will
> > have to help you as I don't. May I also suggest that you bottom post in
> > future responses rather than top post; that makes it a little easier to
> > follow. Good luck - John
> >
> > On Wed, 2009-08-19 at 16:59 +0000, Ott Rose wrote:
> >
> >> 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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> >>>>>
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> >>>>>
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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
> >>>
> >> --
> >>
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> >>>
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> >>>
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> >>
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