[Freeswitch-users] SIP/2.0 480 Temporarily Unavailable

Tue Dec 30 18:27:44 MSK 2014

Follow on…

It appears, to me, that my outbound call is being processed in the “public” context:

2014-12-30 10:08:58.019736 [INFO] mod_dialplan_xml.c:635 Processing George F Phelps <1001>->4049392032 in context public

Don’t I want it be processed in my “default” context?  My local extensions are in the “default” context.  My dialplan (for my gateway) is in the “default” context.

(Trace segment below.)

Thanks,

George

send 405 bytes to udp/[50.160.141.159]:48815 at 10:08:58.020862:

   ------------------------------------------------------------------------

   SIP/2.0 100 Trying

   Via: SIP/2.0/UDP 50.160.141.159:48815;rport=48815;branch=z9hG4bKPjZkkP4e5qeJ4naSA65qJHum-J56cEex7b

   From: "George F Phelps" <sip:1001 at 172.31.33.109>;tag=BH8vkcea3mvKx.nr5KHQUr5im7K4Kr5U

   To: sip:4049392032 at 172.31.33.109

   Call-ID: .qBtG1r2kiKaorRWP0SleO7zUHa0i5mQ

   CSeq: 10680 INVITE

   User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141225T071317Z~d88bae1a62~64bit

   Content-Length: 0

   ------------------------------------------------------------------------

nta.c:6791 incoming_reply() nta: sent 100 Trying for INVITE (10680)

nua_stack.c:271 nua_stack_event() nua(0xce4030): event i_invite 100 Trying

nua_session.c:4139 signal_call_state_change() nua(0xce4030): call state changed: init -> received, received offer

soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0xc261c0, [0x7fce4b5bf598], [0x7fce4b5bf5a0], [(nil)]) called

nua_stack.c:271 nua_stack_event() nua(0xce4030): event i_state 100 Trying

nua_stack.c:359 nua_application_event() nua: nua_application_event: entering

nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering

2014-12-30 10:08:58.019736 [NOTICE] switch_channel.c:1055 New Channel sofia/external/1001 at 172.31.33.109 [3a7e8146-7167-4276-a90c-70955ed5c250]

nua_stack.c:359 nua_application_event() nua: nua_application_event: entering

nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering

nua.c:610 nua_set_hparams() nua: nua_set_hparams: entering

nua.c:610 nua_set_hparams() nua: nua_r_set_params with invalid handle (nil)

nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering

nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering

nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering

2014-12-30 10:08:58.019736 [INFO] mod_dialplan_xml.c:635 Processing George F Phelps <1001>->4049392032 in context public

2014-12-30 10:08:58.019736 [NOTICE] switch_core_state_machine.c:315 sofia/external/1001 at 172.31.33.109 has executed the last dialplan instruction, hanging up.

2014-12-30 10:08:58.019736 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/1001 at 172.31.33.109 [CS_EXECUTE] [NORMAL_CLEARING]

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of George F. Phelps
Sent: Monday, December 29, 2014 7:21 AM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable

Chris Tunbridge,

I move my dialplan to the other folder.  I am still not able to place a call.  Is it still trying to dial a local extension?

recv 854 bytes from udp/[50.160.141.159]:13130 at 06:23:35.849167:

   ------------------------------------------------------------------------

   INVITE sip:17708410143 at 172.31.33.109 SIP/2.0

   Via: SIP/2.0/UDP 192.168.1.100:13130;branch=z9hG4bK-d8754z-30c3a66244749246-1---d8754z-;rport

   Max-Forwards: 70

   Contact: <sip:1000 at 50.160.141.159:13130>

   To: "George Phelps"<sip:17708410143 at 172.31.33.109>

   From: "George F Phelps"<sip:1000 at 172.31.33.109>;tag=13860149

   Call-ID: Y2Y3NTAzNWUxZDJhNDk1YjMzYzE4OWMxMTk5MzUwMTk

   CSeq: 1 INVITE

   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO

   Content-Type: application/sdp

   Supported: replaces

   User-Agent: Bria 3 release 3.5.5  stamp 71238

   Content-Length: 264

   v=0

   o=- 13064325826971649 1 IN IP4 192.168.1.100

   s=Bria 3 release 3.5.5  stamp 71238

   c=IN IP4 192.168.1.100

   t=0 0

   m=audio 50404 RTP/AVP 9 8 0 18 101

   a=rtpmap:18 G729/8000

   a=fmtp:18 annexb=yes

   a=rtpmap:101 telephone-event/8000

   a=fmtp:101 0-15

   a=sendrecv

   ------------------------------------------------------------------------

Thanks,

George

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Tunbridge
Sent: Sunday, December 28, 2014 11:07 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable

George, can you move your /usr/local/freeswitch/conf/dialplan/switch2voip.us into the 
/usr/local/freeswitch/conf/dialplan/default/ folder?

The main folder (not /default/) is used for context's, so it wouldn't get included in your default context.

On Sun, Dec 28, 2014 at 3:46 PM, David Villasmil Govea <david.villasmil at gmail.com> wrote:

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> http://pastebin.com/E4sqTLa4 doesn't show anything. Comes back with "This is a private paste. If you created this paste, please  <http://pastebin.com/login.php?ref=L0U0c3FUTGE0> login 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!

_________________________________________________________________________
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_________________________________________________________________________
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Official FreeSWITCH Sites
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_________________________________________________________________________
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Official FreeSWITCH Sites
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-- 
Imagination is more important than knowledge
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
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Official FreeSWITCH Sites
http://www.freeswitch.org
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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-- 

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
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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
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