[asterisk-users] outbound calls not ringing

Wed Aug 19 17:51:25 CDT 2009

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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>>> -- 
>>> 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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