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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 modifications to ;<br>; this file must be done via the web gui. There are alternative files to make ;<br>; custom modifications, details at: 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 sip general<br>;options that you might need set. For example: enable and force the sip jitterbuffer.<br>;If these settings are desired they should be set the 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 when going<br>;through a firewall. For nat'ing you'd need to add the following 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 upgrade<br>;from previous versions. If you have this file with lines in it please make<br>;sure they are not duplicated in sip_general_custom.conf, if so remove 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 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] 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 a<br>;extension to work to for example, those go here. So you have extension<br>;1000 defined in your system you start by creating a line [1000](+) in 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 of<br>;that extension.<br>;<br>#include sip_custom_post.conf<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>> > > 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>> > > > I put a post on here about my issues with outbound calls not<br>> > ringing<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. when<br>> > the<br>> > > > person picks up and starts to talk i can hear them fine. it sounds<br>> > > > great. How do I start to troubleshot this?<br>> > > <snip><br>> > > What type of phones are giving you the problem? If I recall<br>> > correctly,<br>> > > our SIP phones had this problem depending on how the destination<br>> > handled<br>> > > signaling. We resolved it by adding progressinband=no (as opposed to<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>> > helps<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 http://www.api-digital.com<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>> > HotmailŪ is up to 70% faster. Now good news travels really fast. Try<br>> > it now.<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>> 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>> 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 />Windows Live: Keep your friends up to date with what you do online. <a href='http://windowslive.com/Campaign/SocialNetworking?ocid=PID23285::T:WLMTAGL:ON:WL:en-US:SI_SB_online:082009' target='_new'>Find out more.</a></body>
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