[Freeswitch-users] SIP/2.0 480 Temporarily Unavailable

Fri Jan 2 19:05:14 MSK 2015

For the default config the public context listens on port 5080, the default (internal) one on 5060. Which context you 
hit depends on which port you contact. The ports are defined in the sip_profiles xmls.

On 02.01.2015 12:49, George F. Phelps wrote:
> Additional “freeswitch.log” debug output is included below, for a failed external call.
>
> According to the debug output, the call attempt is indeed being made in the “public” context.  My outbound dialplan is
> located in the “default” context, so apparently no “action” match is found.
>
> *Questions***
>
> 1) Should my dialplan be located in the “public” or “default” context?
>
> 2) Should my call attempt be taking place in the “public” or “default” context?
>
> 3) What (how?) do I need to change to have both the dialplan and the dial attempt in the same (correct) context?
>
> For reference, I pasted my current “xml_locate dialplan” output here:
>
> *http://pastebin.com/BRB5Lpmg*
>
> And this is my Freeswitch version information:
>
> FreeSWITCH Version 1.5.15b+git~20141230T150632Z~1965b3b18d~64bit (git 1965b3b 2014-12-30 15:06:32Z 64bit)
>
> Thanks,
>
> George
>
> ----
>
> 2015-01-02 09:44:19.406789 [INFO] mod_dialplan_xml.c:635 Processing George F Phelps <1001>->14049392032 in context public
>
> Dialplan: sofia/external/1001 at 172.31.33.109 parsing [public->unloop] continue=false
>
> Dialplan: sofia/external/1001 at 172.31.33.109 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
>
> Dialplan: sofia/external/1001 at 172.31.33.109 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
>
> Dialplan: sofia/external/1001 at 172.31.33.109 parsing [public->outside_call] continue=true
>
> Dialplan: sofia/external/1001 at 172.31.33.109 Absolute Condition [outside_call]
>
> Dialplan: sofia/external/1001 at 172.31.33.109 Action set(outside_call=true)
>
> Dialplan: sofia/external/1001 at 172.31.33.109 Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
>
> Dialplan: sofia/external/1001 at 172.31.33.109 parsing [public->call_debug] continue=true
>
> Dialplan: sofia/external/1001 at 172.31.33.109 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never
>
> Dialplan: sofia/external/1001 at 172.31.33.109 parsing [public->public_extensions] continue=false
>
> Dialplan: sofia/external/1001 at 172.31.33.109 Regex (FAIL) [public_extensions] destination_number(14049392032) =~
> /^(10[01][0-9])$/ break=\
>
> Dialplan: sofia/external/1001 at 172.31.33.109 parsing [public->public_did] continue=false
>
> Dialplan: sofia/external/1001 at 172.31.33.109 Regex (FAIL) [public_did] destination_number(14049392032) =~ /^(5551212)$/
> break=on-false
>
> 2015-01-02 09:44:19.406789 [DEBUG] switch_core_state_machine.c:216 (sofia/external/1001 at 172.31.33.109) State Change
> CS_ROUTING -> CS_EXE\
>
> 2015-01-02 09:44:19.406789 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/1001 at 172.31.33.109 [BREAK]
>
> 2015-01-02 09:44:19.406789 [DEBUG] switch_core_state_machine.c:528 (sofia/external/1001 at 172.31.33.109) State ROUTING
> going to sleep
>
> 2015-01-02 09:44:19.406789 [DEBUG] switch_core_state_machine.c:472 (sofia/external/1001 at 172.31.33.109) Running State
> Change CS_EXECUTE
>
> 2015-01-02 09:44:19.406789 [DEBUG] switch_core_state_machine.c:535 (sofia/external/1001 at 172.31.33.109) State EXECUTE
>
> 2015-01-02 09:44:19.406789 [DEBUG] mod_sofia.c:178 sofia/external/1001 at 172.31.33.109 SOFIA EXECUTE
>
> 2015-01-02 09:44:19.406789 [DEBUG] switch_core_state_machine.c:258 sofia/external/1001 at 172.31.33.109 Standard EXECUTE
>
> EXECUTE sofia/external/1001 at 172.31.33.109 set(outside_call=true)
>
> 2015-01-02 09:44:19.406789 [DEBUG] mod_dptools.c:1435 sofia/external/1001 at 172.31.33.109 SET [outside_call]=[true]
>
> EXECUTE sofia/external/1001 at 172.31.33.109 export(RFC2822_DATE=Fri, 02 Jan 2015 09:44:19 -0500)
>
> 2015-01-02 09:44:19.406789 [DEBUG] switch_channel.c:1247 EXPORT (export_vars) [RFC2822_DATE]=[Fri, 02 Jan 2015 09:44:19
> -0500]
>
> 2015-01-02 09:44:19.406789 [NOTICE] switch_core_state_machine.c:315 sofia/external/1001 at 172.31.33.109 has executed the
> last dialplan ins\
>
> 2015-01-02 09:44:19.406789 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/1001 at 172.31.33.109
> [CS_EXECUTE] [NORMAL_CLEARI\
>
> 2015-01-02 09:44:19.406789 [DEBUG] switch_channel.c:3222 Send signal sofia/external/1001 at 172.31.33.109 [KILL]
>
> 2015-01-02 09:44:19.406789 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/1001 at 172.31.33.109 [BREAK]
>
> 2015-01-02 09:44:19.406789 [DEBUG] switch_core_state_machine.c:535 (sofia/external/1001 at 172.31.33.109) State EXECUTE
> going to sleep
>
> ----
>
> *From:*freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf
> Of *George F. Phelps
> *Sent:* Thursday, January 01, 2015 8:13 AM
> *To:* 'FreeSWITCH Users Help'
> *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
>
> I added a debug log message to my dialplan:
>
> <include>
>
>    <extension name="outbound_calls">
>
>      <condition field="destination_number" expression="^1?([2-9]\d{2}[2-9]\d{6})$">
>
> <action application="log" data="DIALING via switch2voip.us"/>
>
>      <action application="bridge" data="sofia/gateway/switch2voip.us/$1"/>
>
>      </condition>
>
>    </extension>
>
> </include>
>
> The log message is not showing up when I dial, as an example, “14049392032”.  So I assume that some other section of the
> dialplan is handling my outbound call?  How can I determine what section is handling the call, without manually adding
> multiple, additional log messages?
>
> I pasted my current “xml_locate dialplan” output here:
>
> http://pastebin.com/BRB5Lpmg
>
> Extracted “condition field="destination_number"”lines from the pastebin:
>
> <action application="deflect" data="${destination_number}"></action>
>
> <condition field="destination_number" expression="^886$">
>
> <condition field="destination_number" expression="^\*8$">
>
> <condition field="destination_number" expression="^\*\*(\d+)$">
>
> <condition field="destination_number" expression="^(redial|870)$">
>
> <action application="hash" data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"></action>
>
> <condition field="destination_number" expression="^9001$">
>
> <condition field="destination_number" expression="^9000$">
>
> <condition field="destination_number" expression="^88(\d{4})$|^\*0(.*)$">
>
> <condition field="destination_number" expression="^779$">
>
> <condition field="destination_number" expression="^\*69$|^869$|^lcr$">
>
> <condition field="destination_number" expression="^80(\d{2})$">
>
> <condition field="destination_number" expression="^81(\d{2})$">
>
> <condition field="destination_number" expression="^82(\d{2})$">
>
> <condition field="destination_number" expression="^83(\d{2})$">
>
> <condition field="destination_number" expression="^8(10[01][0-9])$">
>
> <condition field="destination_number" expression="^(10[01][0-9])$">
>
> <condition field="destination_number" expression="^(11[01][0-9])$">
>
> <action application="bridge" data="skinny/internal/${destination_number}"></action>
>
> <condition field="destination_number" expression="^2000$">
>
> <condition field="destination_number" expression="^2001$">
>
> <condition field="destination_number" expression="^2002$">
>
> <condition field="destination_number" expression="^(operator|0)$">
>
> <condition field="destination_number" expression="^vmain$|^4000$|^\*98$">
>
> <condition field="destination_number" expression="^sip:(.*)$">
>
> <condition field="destination_number" expression="^(30\d{2})$">
>
> <condition field="destination_number" expression="^(31\d{2})$">
>
> <condition field="destination_number" expression="^(32\d{2})$">
>
> <condition field="destination_number" expression="^(33\d{2})$">
>
> <condition field="destination_number" expression="^9(888|8888|1616|3232)$">
>
> <condition field="destination_number" expression="^0911$">
>
> <condition field="destination_number" expression="^0912$">
>
> <condition field="destination_number" expression="^0913$">
>
> <condition field="destination_number" expression="^5000$">
>
> <condition field="destination_number" expression="^5001$">
>
> <condition field="destination_number" expression="^pagegroup$|^7243$">
>
> <condition field="destination_number" expression="^5900$">
>
> <condition field="destination_number" expression="^5901$">
>
> <condition field="destination_number" expression="^(6000)$">
>
> <condition field="destination_number" expression="^((?!6000)60\d{2})$">
>
> <condition field="destination_number" expression="park\+(\d+)">
>
> <condition field="destination_number" expression="^parking$"></condition>
>
> <condition field="destination_number" expression="callpark"></condition>
>
> <condition field="destination_number" expression="pickup"></condition>
>
> <condition field="destination_number" expression="^wait$">
>
> <condition field="destination_number" expression="^9178$">
>
> <condition field="destination_number" expression="^9179$">
>
> <condition field="destination_number" expression="^9180$">
>
> <condition field="destination_number" expression="^9181$">
>
> <condition field="destination_number" expression="^9182$">
>
> <condition field="destination_number" expression="^9183$">
>
> <condition field="destination_number" expression="^9184$">
>
> <condition field="destination_number" expression="^9191$">
>
> <condition field="destination_number" expression="^9192$">
>
> <condition field="destination_number" expression="^9193$">
>
> <condition field="destination_number" expression="^9194$">
>
> <condition field="destination_number" expression="^9195$">
>
> <condition field="destination_number" expression="^9196$">
>
> <condition field="destination_number" expression="^9197$">
>
> <condition field="destination_number" expression="^9198$">
>
> <condition field="destination_number" expression="^9787$">
>
> <condition field="destination_number" expression="^9664$"></condition>
>
> <condition field="destination_number" expression="^9386$">
>
> <condition field="destination_number" expression="^1?([2-9]\d{2}[2-9]\d{6})$">
>
> <condition field="destination_number" expression="^9170$">
>
> <condition field="destination_number" expression="^9171$">
>
> <condition field="destination_number" expression="^9172$">
>
> <condition field="destination_number" expression="^(\d{7})$">
>
> <condition field="destination_number" expression="^(\d{11})$">
>
> <condition field="destination_number" expression="^(011\d+)$">
>
> <condition field="destination_number" expression="^101$">
>
> <condition field="destination_number" expression="^(pizza|74992)$"></condition>
>
> <condition field="destination_number" expression="^(.*)$">
>
> <condition field="destination_number" expression="^dx$">
>
> <condition field="destination_number" expression="^att_xfer$">
>
> <condition field="destination_number" expression="^is_transfer$"></condition>
>
> <condition field="destination_number" expression="^cf$">
>
> <condition field="destination_number" expression="^(10[01][0-9])$">
>
> <action application="deflect" data="${destination_number}"></action>
>
> <condition field="destination_number" expression="^(10[01][0-9])$">
>
> <condition field="destination_number" expression="^(5551212)$">
>
> <condition field="destination_number" expression="^(9[0-9]{3})$">
>
> <condition field="destination_number" expression="^(10[01][0-9])$">
>
> <condition field="destination_number" expression="^(11[01][0-9])$">
>
> <condition field="destination_number" expression="^(0[0-9]{10,})$">
>
> <condition field="destination_number" expression="^(.{14,})$">
>
> Thanks,
>
> George
>
> *From:*freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *George F. Phelps
> *Sent:* Wednesday, December 31, 2014 4:37 PM
> *To:* 'FreeSWITCH Users Help'
> *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
>
> Brian West,
>
> I wiped out my configuration and started over.  I am able to dial from one extension to another — with 2-way audio.
> However, I am still not able to dial an external phone number.  I also cannot dial the example extensions, x9196, x5000,
> etc.
>
> I am able (via IPv6) to register with my VoIP gateway:
>
>                       Name          Type                                       Data      State
>
> =================================================================================================
>
>              external-ipv6       profile sip:mod_sofia@[::1]:5080      RUNNING (0)
>
> external-ipv6::switch2voip.us   gateway sip:9221591504 at 66.33.147.150      REGED
>
>                   external       profile sip:mod_sofia at 54.174.255.168:5080      RUNNING (0)
>
> external::switch2voip.us       gateway sip:9221591504 at 66.33.147.150      FAIL_WAIT
>
>              172.31.33.109         alias                                   internal      ALIASED
>
>              internal-ipv6       profile sip:mod_sofia@[::1]:5060      RUNNING (0)
>
>                   internal       profile sip:mod_sofia at 54.174.255.168:5060      RUNNING (0)
>
> =================================================================================================
>
> 4 profiles 1 alias
>
> A failed dial attempt is shown below.  Freeswitch is apparently still attempting to route the call to a non-existent
> local extension (sip:14049392032 at 172.31.33.109) instead of my VoIP provider.
>
> recv 880 bytes from udp/[50.160.141.159]:49334 at 15:57:15.095108:
>
>     ------------------------------------------------------------------------
>
>     INVITE sip:14049392032 at 172.31.33.109 SIP/2.0
>
>     Via: SIP/2.0/UDP 50.160.141.159:49334;rport;branch=z9hG4bKPjHAihEWmq6a3YH4jMMahZ.0Uxem0XpIXo
>
>     Max-Forwards: 70
>
>     From: "George F Phelps" <sip:1001 at 172.31.33.109>;tag=8uylnZnGRKkfjtF.I4UerNH9JcR6oWkr
>
>     To: sip:14049392032 at 172.31.33.109
>
>     Contact: "George F Phelps" <sip:1001 at 50.160.141.159:49334;ob>
>
>     Call-ID: npLJzYL0ezArXbyAUG2DguVqmKqdtqXT
>
>     CSeq: 30128 INVITE
>
>     Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
>
>     Supported: replaces, 100rel, norefersub
>
>     User-Agent: Bria Android 3.2.1
>
>     Content-Type: application/sdp
>
>     Content-Length:   247
>
>     v=0
>
>     o=- 3629048232 3629048232 IN IP4 50.160.141.159
>
>     s=cpc_med
>
>     c=IN IP4 50.160.141.159
>
>     t=0 0
>
>     m=audio 4000 RTP/AVP 0 8 101
>
>     a=sendrecv
>
>     a=rtpmap:0 PCMU/8000
>
>     a=rtpmap:8 PCMA/8000
>
>     a=rtpmap:101 telephone-event/8000
>
>     a=fmtp:101 0-16
>
>     a=ptime:20
>
> send 406 bytes to udp/[50.160.141.159]:49334 at 15:57:15.095386:
>
>     ------------------------------------------------------------------------
>
>     SIP/2.0 100 Trying
>
>     Via: SIP/2.0/UDP 50.160.141.159:49334;rport=49334;branch=z9hG4bKPjHAihEWmq6a3YH4jMMahZ.0Uxem0XpIXo
>
>     From: "George F Phelps" <sip:1001 at 172.31.33.109>;tag=8uylnZnGRKkfjtF.I4UerNH9JcR6oWkr
>
>     To: sip:14049392032 at 172.31.33.109
>
>     Call-ID: npLJzYL0ezArXbyAUG2DguVqmKqdtqXT
>
>     CSeq: 30128 INVITE
>
>     User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141230T150632Z~1965b3b18d~64bit
>
>     Content-Length: 0
>
>     ------------------------------------------------------------------------
>
> send 802 bytes to udp/[50.160.141.159]:49334 at 15:57:15.097601:
>
>     ------------------------------------------------------------------------
>
>     SIP/2.0 480 Temporarily Unavailable
>
>     Via: SIP/2.0/UDP 50.160.141.159:49334;rport=49334;branch=z9hG4bKPjHAihEWmq6a3YH4jMMahZ.0Uxem0XpIXo
>
>     Max-Forwards: 70
>
>     From: "George F Phelps" <sip:1001 at 172.31.33.109>;tag=8uylnZnGRKkfjtF.I4UerNH9JcR6oWkr
>
>     To: <sip:14049392032 at 172.31.33.109>;tag=ctKjQv8vKyZtB
>
>     Call-ID: npLJzYL0ezArXbyAUG2DguVqmKqdtqXT
>
>     CSeq: 30128 INVITE
>
>     User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141230T150632Z~1965b3b18d~64bit
>
>     Accept: application/sdp
>
>     Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
>
>     Supported: timer, path, replaces
>
>     Allow-Events: talk, hold, conference, refer
>
>     Reason: Q.850;cause=16;text="NORMAL_CLEARING"
>
>     Content-Length: 0
>
>     Remote-Party-ID: "14049392032" <sip:14049392032 at 172.31.33.109>;party=calling;privacy=off;screen=no
>
>     ------------------------------------------------------------------------
>
> My outbound dialplan “/usr/local/freeswitch/conf/dialplan/default/00_outbound_calls.xml”:
>
> <include>
>
>    <extension name="outbound_calls">
>
>      <condition field="destination_number" expression="^1?([2-9]\d{2}[2-9]\d{6})$">
>
>      <action application="bridge" data="sofia/gateway/switch2voip.us/$1"/>
>
>      </condition>
>
>    </extension>
>
> </include>
>
> My gateway “/usr/local/freeswitch/conf/directory/default/example.com.xml”:
>
> <include>
>
>    <user id="$${default_provider}">
>
>      <gateways>
>
>        <gateway name="$${default_provider}">
>
>          <param name="username" value="$${default_provider_username}"/>
>
>          <param name="password" value="$${default_provider_password}"/>
>
>          <param name="from-user" value="$${default_provider_username}"/>
>
>          <param name="from-domain" value="$${default_provider_from_domain}"/>
>
>          <param name="realm" value="$${default_provider_from_domain}"/>
>
>          <param name="proxy" value="66.33.147.150"/>
>
>          <param name="expire-seconds" value="600"/>
>
>          <param name="register" value="$${default_provider_register}"/>
>
>          <param name="retry-seconds" value="30"/>
>
>          <param name="extension" value="$${default_provider_contact}"/>
>
>          <!--<param name="contact-params" value="domain_name=$${domain}"/>-->
>
>          <param name="context" value="public"/>
>
>        </gateway>
>
>      </gateways>
>
>      <params>
>
>        <param name="password" value="$${default_provider_password}"/>
>
>      </params>
>
>    </user>
>
> </include>
>
> And in “/usr/local/freeswitch/conf/vars.xml”:
>
>    <X-PRE-PROCESS cmd="set" data="default_provider=switch2voip.us"/>
>
>    <X-PRE-PROCESS cmd="set" data="default_provider_username=9221591504"/>
>
>    <X-PRE-PROCESS cmd="set" data="default_provider_password=XXXXXXXX"/>
>
>    <X-PRE-PROCESS cmd="set" data="default_provider_from_domain=switch2voip.us"/>
>
>    <!-- true or false -->
>
>    <X-PRE-PROCESS cmd="set" data="default_provider_register=true"/>
>
>    <X-PRE-PROCESS cmd="set" data="default_provider_contact=5000"/>
>
> I captured IP packets on my server, and during the dial attempt, there are no IP packets being sent to my VoIP provider
> (66.33.147.150).
>
> I did make the additional configuration updates as documented here:
>
> https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2”
>
> Thanks,
>
> George
>
> *From:*freeswitch-users-bounces at lists.freeswitch.org <mailto: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 1:24 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable
>
> 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 <mailto: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>
> [mailto: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 <mailto: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 <tel: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 <mailto:sip%3A1001 at 172.31.33.109>>;tag=BH8vkcea3mvKx.nr5KHQUr5im7K4Kr5U
>
>     To: sip:4049392032 <tel:4049392032>@172.31.33.109 <http://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
> <mailto: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 <tel: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
> <mailto: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
> <mailto: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>
> [mailto: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>
> [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 <http://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 <mailto: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 <http://switch2voip.us/$1>"/>
>
>      </condition>
>
>     </extension>
>
> On Sun, Dec 28, 2014 at 11:42 PM, George F. Phelps <GeorgePhelps at gfphelps.com <mailto:GeorgePhelps at gfphelps.com>> wrote:
>
> The full output from the “xml_locate dialplan” command is already in the previously pasted logfile.
>
> Below is the dialplan that I created, in /usr/local/freeswitch/conf/dialplan/switch2voip.us <http://switch2voip.us>:
>
> <?xml version="1.0" encoding="utf-8"?>
>
>     <!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here -->
>
>     <extension name="switch2voip.us <http://switch2voip.us>">
>
>      <condition field="destination_number" expression="^(1{0,1}\d{10})$">
>
>        <!-- 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 <http://switch2voip.us/$1>"/>
>
>      </condition>
>
>     </extension>
>
> My suspicion is that some other dialplan, other than my “switch2voip.us <http://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>
> [mailto: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 <tel:4049392032>@172.31.33.109 <http://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 <mailto: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>
> [mailto: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 <mailto: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>
> [mailto: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 <mailto: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 <http://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 <http://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>
> [mailto: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 <mailto: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 <mailto: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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>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
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>
> FreeSWITCH-users mailing list
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>
>
> _________________________________________________________________________
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> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
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>
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> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org
>
>
>
> --
>
> DVG
>
> --
> Imagination is more important than knowledge
> Albert Einstein
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto: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 <mailto: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 <mailto: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 <mailto:FreeSWITCH-users at lists.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 <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
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>
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>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
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> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
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>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
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>
>
>
> --
>
> */Brian West/*
> brian at freeswitch.org <mailto:brian at freeswitch.org>
>
> */Twitter: @FreeSWITCH , @briankwest/*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> *T:*+19184209001 <tel:%2B19184209001> | *F:*+19184209002 <tel:%2B19184209002> | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
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>
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>
>
>
> --
>
> */Brian West/*
> brian at freeswitch.org <mailto:brian at freeswitch.org>
>
> */Twitter: @FreeSWITCH , @briankwest/*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
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>
>
>
> _________________________________________________________________________
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>
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