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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.<br><br><br><br><br>> Date: Thu, 20 Aug 2009 10:51:25 +1200<br>> From: duncan@e-simple.co.nz<br>> To: asterisk-users@lists.digium.com<br>> Subject: Re: [asterisk-users] outbound calls not ringing<br>> <br>> Generally with FreePBX the ring options are set in the General Options - <br>> you can set the Dial options which are normally tr, but I guess that <br>> isn't working for you.<br>> <br>> The SIP files you could edit would have custom in their name, otherwise <br>> your changes will be overwritten when you reload freepbx<br>> <br>> You could put this in sip_general_custom.conf which will be included<br>> <br>> Cheers Duncan<br>> <br>> John A. Sullivan III wrote:<br>> > Oops! - You're using FreePBX - someone who knows more about FreePBX will<br>> > have to help you as I don't. May I also suggest that you bottom post in<br>> > future responses rather than top post; that makes it a little easier to<br>> > follow. Good luck - John<br>> ><br>> > On Wed, 2009-08-19 at 16:59 +0000, Ott Rose wrote:<br>> > <br>> >> here is my sip.conf. i don't see it.<br>> >> ;--------------------------------------------------------------------------------;<br>> >> ; Do NOT edit this file as it is auto-generated by FreePBX. All<br>> >> modifications to ;<br>> >> ; this file must be done via the web gui. There are alternative files<br>> >> to make ;<br>> >> ; custom modifications, details at:<br>> >> http://freepbx.org/configuration_files ;<br>> >> ;--------------------------------------------------------------------------------;<br>> >> ;<br>> >><br>> >> [general]<br>> >><br>> >> ; These files will all be included in the [general] context<br>> >> ;<br>> >> #include sip_general_additional.conf<br>> >><br>> >> ;sip_general_custom.conf is the proper file location for placing any<br>> >> sip general<br>> >> ;options that you might need set. For example: enable and force the<br>> >> sip jitterbuffer.<br>> >> ;If these settings are desired they should be set the<br>> >> sip_general_custom.conf file.<br>> >> ;<br>> >> ; jbenable=yes<br>> >> ; jbforce=yes<br>> >> ;<br>> >> ;It is also the proper place to add the lines needed for sip nat'ing<br>> >> when going<br>> >> ;through a firewall. For nat'ing you'd need to add the following<br>> >> lines:<br>> >> ; nat=yes , externip= , localhost= , and optionally fromdomain= .<br>> >> ;<br>> >> #include sip_general_custom.conf<br>> >><br>> >> ;sip_nat.conf is here for legacy support reasons and for those that<br>> >> upgrade<br>> >> ;from previous versions. If you have this file with lines in it<br>> >> please make<br>> >> ;sure they are not duplicated in sip_general_custom.conf, if so remove<br>> >> them<br>> >> ;from sip_nat.conf as sip_general_custom.conf will have precedence.<br>> >> #include sip_nat.conf<br>> >><br>> >> ;sip_registrations_custom.conf is for any customizations you might<br>> >> need to do to<br>> >> ;the automatically generated registrations that FreePBX makes.<br>> >> ;<br>> >> #include sip_registrations_custom.conf<br>> >> #include sip_registrations.conf<br>> >><br>> >> ; These files should all be expected to come after the [general]<br>> >> context<br>> >> ;<br>> >> #include sip_custom.conf<br>> >> #include sip_additional.conf<br>> >><br>> >> ;sip_custom_post.conf If you have extra parameters that are needed for<br>> >> a<br>> >> ;extension to work to for example, those go here. So you have<br>> >> extension<br>> >> ;1000 defined in your system you start by creating a line [1000](+) in<br>> >> this<br>> >> ;file. Then on the next line add the extra parameter that is needed.<br>> >> ;When the sip.conf is loaded it will append your additions to the end<br>> >> of<br>> >> ;that extension.<br>> >> ;<br>> >> #include sip_custom_post.conf<br>> >><br>> >><br>> >> <br>> >>> From: jsullivan@opensourcedevel.com<br>> >>> To: asterisk-users@lists.digium.com<br>> >>> Date: Wed, 19 Aug 2009 12:17:15 -0400<br>> >>> Subject: Re: [asterisk-users] outbound calls not ringing<br>> >>><br>> >>> sip.conf<br>> >>><br>> >>> On Wed, 2009-08-19 at 15:55 +0000, Ott Rose wrote:<br>> >>> <br>> >>>> we are using Aastra 57i<br>> >>>><br>> >>>> i don't see that setting. where is it at?<br>> >>>><br>> >>>> <br>> >>>>> From: jsullivan@opensourcedevel.com<br>> >>>>> To: asterisk-users@lists.digium.com<br>> >>>>> Date: Wed, 19 Aug 2009 11:07:21 -0400<br>> >>>>> Subject: Re: [asterisk-users] outbound calls not ringing<br>> >>>>><br>> >>>>> On Wed, 2009-08-19 at 13:54 +0000, Ott Rose wrote:<br>> >>>>> <br>> >>>>>> I put a post on here about my issues with outbound calls not<br>> >>>>>> <br>> >>>> ringing<br>> >>>> <br>> >>>>>> but i haven't resolved it. so i am trying again.<br>> >>>>>><br>> >>>>>> When i dial any outside number i dont get a ring tone at all.<br>> >>>>>> <br>> >> when<br>> >> <br>> >>>> the<br>> >>>> <br>> >>>>>> person picks up and starts to talk i can hear them fine. it<br>> >>>>>> <br>> >> sounds<br>> >> <br>> >>>>>> great. How do I start to troubleshot this?<br>> >>>>>> <br>> >>>>> <snip><br>> >>>>> What type of phones are giving you the problem? If I recall<br>> >>>>> <br>> >>>> correctly,<br>> >>>> <br>> >>>>> our SIP phones had this problem depending on how the destination<br>> >>>>> <br>> >>>> handled<br>> >>>> <br>> >>>>> signaling. We resolved it by adding progressinband=no (as<br>> >>>>> <br>> >> opposed to<br>> >> <br>> >>>>> the default never - at least I think it is the default) but this<br>> >>>>> produces the problem of duplicate ring tones at times. Hope this<br>> >>>>> <br>> >>>> helps<br>> >>>> <br>> >>>>> - John<br>> >>>>> -- <br>> >>>>> John A. Sullivan III<br>> >>>>> Open Source Development Corporation<br>> >>>>> +1 207-985-7880<br>> >>>>> jsullivan@opensourcedevel.com<br>> >>>>><br>> >>>>> http://www.spiritualoutreach.com<br>> >>>>> Making Christianity intelligible to secular society<br>> >>>>><br>> >>>>><br>> >>>>> _______________________________________________<br>> >>>>> -- Bandwidth and Colocation Provided by<br>> >>>>> <br>> >> http://www.api-digital.com<br>> >> <br>> >>>> --<br>> >>>> <br>> >>>>> AstriCon 2009 - October 13 - 15 Phoenix, Arizona<br>> >>>>> Register Now: http://www.astricon.net<br>> >>>>><br>> >>>>> asterisk-users mailing list<br>> >>>>> To UNSUBSCRIBE or update options visit:<br>> >>>>> http://lists.digium.com/mailman/listinfo/asterisk-users<br>> >>>>> <br>> >>>><br>> >>>> <br>> >> ______________________________________________________________________<br>> >> <br>> >>>> HotmailŪ is up to 70% faster. Now good news travels really fast.<br>> >>>> <br>> >> Try<br>> >> <br>> >>>> it now.<br>> >>>> _______________________________________________<br>> >>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com<br>> >>>> <br>> >> --<br>> >> <br>> >>>> AstriCon 2009 - October 13 - 15 Phoenix, Arizona<br>> >>>> Register Now: http://www.astricon.net<br>> >>>><br>> >>>> asterisk-users mailing list<br>> >>>> To UNSUBSCRIBE or update options visit:<br>> >>>> http://lists.digium.com/mailman/listinfo/asterisk-users<br>> >>>> <br>> >>> -- <br>> >>> John A. Sullivan III<br>> >>> Open Source Development Corporation<br>> >>> +1 207-985-7880<br>> >>> jsullivan@opensourcedevel.com<br>> >>><br>> >>> http://www.spiritualoutreach.com<br>> >>> Making Christianity intelligible to secular society<br>> >>><br>> >>><br>> >>> _______________________________________________<br>> >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com<br>> >>> <br>> >> --<br>> >> <br>> >>> AstriCon 2009 - October 13 - 15 Phoenix, Arizona<br>> >>> Register Now: http://www.astricon.net<br>> >>><br>> >>> asterisk-users mailing list<br>> >>> To UNSUBSCRIBE or update options visit:<br>> >>> http://lists.digium.com/mailman/listinfo/asterisk-users<br>> >>> <br>> >> ______________________________________________________________________<br>> >> Windows Live: Keep your friends up to date with what you do online.<br>> >> Find out more.<br>> >> _______________________________________________<br>> >> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> >><br>> >> AstriCon 2009 - October 13 - 15 Phoenix, Arizona<br>> >> Register Now: http://www.astricon.net<br>> >><br>> >> asterisk-users mailing list<br>> >> To UNSUBSCRIBE or update options visit:<br>> >> http://lists.digium.com/mailman/listinfo/asterisk-users<br>> >> <br>> <br>> _______________________________________________<br>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> <br>> AstriCon 2009 - October 13 - 15 Phoenix, Arizona<br>> Register Now: http://www.astricon.net<br>> <br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-users<br><br /><hr />Get back to school stuff for them and cashback for you. <a href='http://www.bing.com/cashback?form=MSHYCB&publ=WLHMTAG&crea=TEXT_MSHYCB_BackToSchool_Cashback_BTSCashback_1x1' target='_new'>Try BingT now.</a></body>
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