[Freeswitch-users] SIP/2.0 480 Temporarily Unavailable

Sat Dec 27 10:29:41 MSK 2014

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!
>
>
>
>
> _________________________________________________________________________
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>
>
> _________________________________________________________________________
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> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
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> http://confluence.freeswitch.org
> http://www.cluecon.com
>
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