[Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
Brian West
brian at freeswitch.org
Tue Dec 30 21:23:36 MSK 2014
Wipe out your config and start over from our samples, clearly something was
changed that wasn't understood. I could only possibly help if I had access
to the system and understood the topology completely. I've only looked
over what you've posted and it would seem to me someone has modified the
configs and doesn't fully understand how things interact.
On Tue, Dec 30, 2014 at 10:44 AM, George F. Phelps <
GeorgePhelps at gfphelps.com> wrote:
> Brian West,
>
>
>
> Okay, I’m sure there is an answer/solution there, but it’s over my head…
>
>
>
> *Questions*
>
>
>
> How do I check to see if I have inadvertently disabled “auth”? I am 99%
> sure that I have do changed it.
>
>
>
> I am 100% sure that I have not touched anything to do with “allow acl”.
> But how do I check?
>
>
>
> So are you saying that dialing an outside in the in the “public” context
> is correct?
>
>
>
> Thanks,
>
>
>
> George
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West
> *Sent:* Tuesday, December 30, 2014 10:32 AM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
>
>
>
> Smells like someone has either disabled auth, or setup an allow acl,
> because the internal profile in the defaults have the context set to public
> unless you auth to prevent someone from opening up their default context by
> accident if they happen to turn off auth.
>
>
>
> On Tue, Dec 30, 2014 at 9:27 AM, George F. Phelps <
> GeorgePhelps at gfphelps.com> wrote:
>
> 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
> <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 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
>
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> 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
>
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>
>
>
>
> _________________________________________________________________________
> 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
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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
> 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
>
>
>
>
> _________________________________________________________________________
> 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
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>
>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> 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
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>
--
*Brian West*
brian at freeswitch.org
*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com
*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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