[Freeswitch-users] SIP/2.0 480 Temporarily Unavailable

David Villasmil Govea david.villasmil at gmail.com
Mon Dec 29 01:46:44 MSK 2014 

Looks to that what you're dialing 404.... is not in your dialplan, you need
to add an extesion for that, like:

   <extension name="o <http://switch2voip.us/>utside_calls">

    <condition field="destination_number" expression="^(404.*)$">

      <action application="bridge" data="sofia/gateway/switch2voip.us/$1"/>

    </condition>

   </extension>


On Sun, Dec 28, 2014 at 11:42 PM, George F. Phelps <
GeorgePhelps at gfphelps.com> wrote:

> The full output from the “xml_locate dialplan” command is already in the
> previously pasted logfile.
>
>
>
> Below is the dialplan that I created, in
> /usr/local/freeswitch/conf/dialplan/switch2voip.us:
>
>
>
> <?xml version="1.0" encoding="utf-8"?>
>
>    <!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444)
> here -->
>
>    <extension name="switch2voip.us">
>
>     <condition field="destination_number" expression="^(1{0,1}\d{10})$">
>
>       <!-- If your provider does not provide ringback (180 or 183) you may
> simulate
>
>         ringback by uncommenting the following line. -->
>
>       <!-- action application="ringback" /-->
>
>       <action application="bridge" data="sofia/gateway/switch2voip.us/$1
> "/>
>
>     </condition>
>
>    </extension>
>
>
>
> My suspicion is that some other dialplan, other than my “switch2voip.us”
> dialplan, is being invoked.  My SIP Proxy is at 66.33.147.150.  IP address
> “172.31.33.109” is the local/internal IP address for my AWS virtual cloud
> server.  “4049392032” is a real phone number — not an extension.
>
>
>
> Thanks,
>
>
>
> George
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *David
> Villasmil Govea
> *Sent:* Sunday, December 28, 2014 5:05 PM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
>
>
>
> can you share your dialplan? It looks like you're dialing
>
> "To: sip:4049392032 at 172.31.33.109"
>
> but have no extension for that...
>
>
>
> On Sun, Dec 28, 2014 at 10:55 PM, George F. Phelps <
> GeorgePhelps at gfphelps.com> wrote:
>
> New “pastebin” created:
>
>
>
> http://pastebin.com/UwmgJGGg
>
>
>
> George
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *David
> Villasmil Govea
> *Sent:* Sunday, December 28, 2014 4:04 PM
>
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
>
>
>
> http://pastebin.com/E4sqTLa4 doesn't show anything. Comes back with "This
> is a private paste. If you created this paste, please login
> <http://pastebin.com/login.php?ref=L0U0c3FUTGE0> to view it."
>
>
>
> On Sun, Dec 28, 2014 at 3:22 PM, George F. Phelps <
> GeorgePhelps at gfphelps.com> wrote:
>
> Chris Tunbridge,
>
>
>
> 1)  I made the updates to my configuration, as suggested in the “
> https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2” link.
> I’m still not able to make a call to an outside number.  A call to an
> extension connects, but there is still no audio.
>
>
>
> 2)  Extension x9161 is one of the default dialplan applications.
>
>
>
> 3)  Call failure log posted at:  http://pastebin.com/E4sqTLa4
>
>
>
> Thanks,
>
>
>
> George
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chris
> Tunbridge
> *Sent:* Saturday, December 27, 2014 2:30 AM
>
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
>
>
>
> 1) This is an issue with the NAT, likely on the freeswitch side, see
> instructions here:
> https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 The
> important part is the external sip ip and external rtp ip.  Without this
> calls will connect, but audio will not pass.  I run dozens of servers on
> AWS without any issues as long as the external sip and rtp ip's are
> configured in the sip profile conf/sip_profiles/internal.xml
>
> 2) Your issue you said was with extension x9196, is this another sip
> endpoint or a dialplan application?  If this is a sip endpoint, please make
> some adjustments to the conf/dialplan/default.xml to address extra
> extensions outside of the 10XX range.
>
> 3) Can you post a log here http://pastebin.freeswitch.org of a call
> attempt?  My guess is that something's not matching the request, a complete
> log of a call attempt would help most here.
>
> 4) Glad to hear, its only used if you're using the JavaScript scripting
> engine for your scripts.
>
>
>
> On Fri, Dec 26, 2014 at 7:57 AM, George F. Phelps <
> GeorgePhelps at gfphelps.com> wrote:
>
> Chris Tunbridge, et al.,
>
>
>
> 1)  Freeswitch is running is running on an Amazon Web Services (AWS) Linux
> virtual cloud server.  I am testing with Bria softphones (both Windows PC
> and Android smartphone) from my home network (behind a Netgear wireless
> router).  The Freeswitch “show codecs” command indicates support for
> “codec, G.711 ulaw, CORE_PCM_MODULE” — which is the codec that I am using
> with Bria.  I am able to successfully connect with Bria to my other VoIP
> services, such as VoIP.ms.
>
>
>
> 2)  I am using mostly a default configuration, i.e., extensions 1000
> through 1019 are configured with updated passwords.
>
>
>
> 3)  This is my outbound dialplan.  How do I know if this is the dialplan
> that is actually being used for dialing?  It shows up in the “xml_locate
> dialplan” output — but as the very last entry.  My guess is that Freeswitch
> is attempting to us some other (default, example?) gateway instead of my
> desired (switch2voip.us) gateway.
>
>
>
>   <?xml version="1.0" encoding="utf-8"?>
>
>   <!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444)
> here -->
>
>   <extension name="switch2voip">
>
>     <condition field="destination_number" expression="^(1{0,1}\d{10})$">
>
>       <!-- If your provider does not provide ringback (180 or 183) you may
> simulate
>
>         ringback by uncommenting the following line. -->
>
>       <!-- action application="ringback" /-->
>
>       <action application="bridge" data="sofia/gateway/switch2voip.us/$1
> "/>
>
>     </condition>
>
>   </extension>
>
>
>
> 4)  The “mod_v8” issue is now resolved.  The module was not being built.
> I’m not sure why the downloaded default build/install files were not
> building it, but were attempting to load it.  Sounds like a bug to me…
>
>
>
> Thanks,
>
>
>
> George
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chris
> Tunbridge
> *Sent:* Thursday, December 25, 2014 9:25 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
>
>
>
> 1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate
> on your topology and configuration
>
> 2) If you're using default configs, its configured to look for extensions
> 10XX, you can see this in conf/dialplan/default.xml (and in
> conf/dialplan/public.xml for calls coming from the outside)
>
> 3) Do you have an outbound route configured that matches your dial string?
>
> 4) This just means the module wasn't configured, you can comment out the
> line in conf/autoload_configs/modules.conf.xml find the line that says
> mod_v8 and put a <!-- at the beginng and a -> at the end
>
>
>
> On Wed, Dec 24, 2014 at 9:39 AM, George F. Phelps <
> GeorgePhelps at gfphelps.com> wrote:
>
> I am debugging a new/initial Freeswitch configuration.
>
>
>
> I believe that I have successfully registered with my VoIP provider —
> “State=REGED”.
>
>
>
> I am able to dial from one extension (x1000) to a different extension
> (x1001), but after answering, there is NO AUDIO at either end of the call.
> Problem #1.
>
>
>
> When I test call to extension x9196, for example, I get an immediate
> hang-up and SIP response of “SIP/2.0 480 Temporarily Unavailable”.  Problem
> #2.  Do I have to do anything to enable calling to x9196?
>
>
>
> And when I attempt to call an external phone number via my VoIP provider,
> I get the same immediate hang-up and SIP response.  Problem #3.
>
>
>
> I am getting this critical error on startup.  Problem #4.
>
>
>
> 2014-12-24 11:29:09.869357 [CRIT] switch_loadable_module.c:1447 Error
> Loading module /usr/local/freeswitch/mod/mod_v8.so
>
>
>
> **/usr/local/freeswitch/mod/mod_v8.so: cannot open shared object file: No
> such file or directory**
>
>
>
> Any suggestions as to what configuration might be wrong?  Or how I can get
> additional debug information?
>
>
>
> Version info:
>
>
>
> FreeSWITCH Version 1.5.15b+git~20141222T221908Z~067cb0f0f2~64bit (git
> 067cb0f 2014-12-22 22:19:08Z 64bit)
>
>
>
> Thanks!
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>
> --
>
> DVG
>
> --
> Imagination is more important than knowledge
> Albert Einstein
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>
> --
>
> DVG
>
> --
> Imagination is more important than knowledge
> Albert Einstein
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
DVG

-- 
Imagination is more important than knowledge
Albert Einstein
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141228/a42721f9/attachment-0001.html

Join us at ClueCon 2016 Aug 8-12, 2016
More information about the FreeSWITCH-users mailing list