[Freeswitch-users] Trouble Playing Audio to external dialler after bridged call to external SIP URI is hung up

[Sina Owolabi]

Fixed it.. I had to define the full path to the wav file. For good
measure I moved it to /tmp/.
<extension name="playafile">
    <condition field="${caller_id_number}"
expression="^0(\d{10})$"require-nested="false">
       <action application="set" data="effective_caller_id_number=+000${1}"/>
       <action application="set" data="effective_caller_id_name=+000${1}"/>
    </condition>
    <condition field="destination_number" expression="^012345678(9)(0)$">
      <action application="ring_ready" />
      <action application="sleep" data="3000"/>
      <action application="transfer" data="1212 XML default"/>
   </condition>
  </extension>


<extension name="1212">
     <condition field="destination_number" expression="^1212$">
        <action application="answer"/>
        <action application="playback"
data="ivr/ivr-thank_you_for_calling.wav"/>
     </condition>
 </extension>

On Mon, Dec 8, 2014 at 9:55 AM, Notify Me <notify.sina at gmail.com> wrote:
> Hi!
>
> I am still a Freeswitch Newbie, and I am trying to learn by doing.
> I have installed and using  version
> 1.5.15b+git~20141120T035109Z~79de78a0fb~64bit on a 64-bit CentOS 6.6
> kvm node.
>
> I have been able to successfully transfer calls for an external
> dialler, through a SIP trunk, to call an external SIP URI. This works
> and I can see it in the logs as very successful. The external SIP URI
> is supposed to process the caller_id, and send the dialler an SMS.
>
> The problem is the SIP URI being dialled ends the call very quickly
> and the external dialler has a busy tone and the call is dropped, or
> other signals that do not indicate that the call was successful and
> may dial again several times, if the SMS does not get delivered in
> time. The SIP URI sends back a 100 Trying and a 480 Temporarily
> Unavailable response as you can see below from a dump of the sequence.
> I have tried to introduce ivr sounds before and after the user
> connects and the bridge is dropped but is also not working. I can see
> that the wav files are being called from the logs but I dont hear
> anything at all. Please can anyone guide me?
>
>
> my dialplan:
>
> public:
>
> <include>
>  <extension name="inbound">
>   <condition field="destination_number"  expression="^012345(\d{2})$"/>
>    <action application="set" data="ringback=${us-ring}"/>
>   <condition field="caller_id_number"  expression="^0(\d{10})$"
> require-nested="true">
>      <action application="set" data="effective_caller_id_number=+234$1"/>
>      <action application="transfer" data="1212 XML default"/>
>   </condition>
>  </extension>
> </include>
>
> default:
> <extension name="1212">
>        <action application="set" data="hangup_after_bridge=true"/>
>        <action application="set" data="transfer_after_bridge=1213:XML:default"/>
>        <action application="set" data="exec_after_bridge_app=transfer"/>
>        <action application="set" data="exec_after_bridge_arg="1213"/>
>        <action application="bridge"
> data="sofia/external/sip:user.name at sip.othersipgw.ip.address.com;${effective_caller_id_number}"/>
>  </condition>
> </extension>
>
> <!-- What to do after othersipgw drops the call with -->
> <extension name="1213">
>   <condition field="destination_number" expression="1213">
>     <action application="set" data="instant_ringback=true"/>
>     <action application="set" data="transfer_ringback=$${us-ring}"/>
>     <action application="playback" data="ivr/ivr-thank_you.wav"/>
>     <action application="playback" data="ivr/ivr-thank_you.wav"/>
>    </condition>
> </extension>
>
>
>
> TCP DUMP
> 09:31:58.808760 IP (tos 0x0, ttl 64, id 34090, offset 0, flags [none],
> proto UDP (17), length 1146)
>     my.natted.priv.ip.address.5080 > othersipgw.ip.address.5060: SIP,
> length: 1118
>         INVITE sip:user.name at sip.othersipgw.com;+234diallednumber SIP/2.0
>         Via: SIP/2.0/UDP
> my.public.ip.address:5080;rport;branch=z9hG4bK8eDcDvZpBm4yr
>         Max-Forwards: 5
>         From: "0diallednumber"
> <sip:+234diallednumber at my.public.ip.address>;tag=6pKBvpXc32yeD
>         To: <sip:user.name at sip.othersipgw.com;+234diallednumber>
>         Call-ID: 840abf24-f957-1232-2ca9-525400ecad09
>         CSeq: 68677695 INVITE
>         Contact: <sip:mod_sofia at my.public.ip.address:5080>
>         User-Agent:
> FreeSWITCH-mod_sofia/1.5.15b+git~20141120T035109Z~79de78a0fb~64bit
>         Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO,
> UPDATE, REGISTER, REFER, NOTIFY
>         Supported: timer, path, replaces
>         Allow-Events: talk, hold, conference, refer
>         Content-Type: application/sdp
>         Content-Disposition: session
>         Content-Length: 247
>         X-FS-Support: update_display,send_info
>         Remote-Party-ID: "0diallednumber"
> <sip:+234diallednumber at my.public.ip.address>;party=calling;screen=yes;privacy=off
>
>         v=0
>         o=FreeSWITCH 1418006836 1418006837 IN IP4 my.public.ip.address
>         s=FreeSWITCH
>         c=IN IP4 my.public.ip.address
>         t=0 0
>         m=audio 20682 RTP/AVP 8 0 101 13
>         a=rtpmap:8 PCMA/8000
>         a=rtpmap:0 PCMU/8000
>         a=rtpmap:101 telephone-event/8000
>         a=fmtp:101 0-16
>         a=ptime:20
>
> 09:31:59.080097 IP (tos 0x0, ttl 53, id 5734, offset 0, flags [none],
> proto UDP (17), length 384)
>     othersipgw.ip.address.5060 > my.natted.priv.ip.address.5080: SIP,
> length: 356
>         SIP/2.0 100 Trying
>         Via: SIP/2.0/UDP
> my.public.ip.address:5080;rport=5080;branch=z9hG4bK8eDcDvZpBm4yr
>         From: "0diallednumber"
> <sip:+234diallednumber at my.public.ip.address>;tag=6pKBvpXc32yeD
>         To: <sip:user.name at sip.othersipgw.com;+234diallednumber>
>         Call-ID: 840abf24-f957-1232-2ca9-525400ecad09
>         CSeq: 68677695 INVITE
>         User-Agent: FreeSWITCH-mod_sofia/1.2.14
>         Content-Length: 0
>
>
> 09:31:59.145629 IP (tos 0x0, ttl 53, id 5735, offset 0, flags [none],
> proto UDP (17), length 1015)
>     othersipgw.ip.address.5060 > my.natted.priv.ip.address.5080: SIP,
> length: 987
>         SIP/2.0 480 Temporarily Unavailable
>         Via: SIP/2.0/UDP
> my.public.ip.address:5080;rport=5080;branch=z9hG4bK8eDcDvZpBm4yr
>         Max-Forwards: 4
>         From: "0diallednumber"
> <sip:+234diallednumber at my.public.ip.address>;tag=6pKBvpXc32yeD
>         To: <sip:user.name at sip.othersipgw.com;+234diallednumber>;tag=Ur840N1DvjvpH
>         Call-ID: 840abf24-f957-1232-2ca9-525400ecad09
>         CSeq: 68677695 INVITE
>         User-Agent: FreeSWITCH-mod_sofia/1.2.14
>         Accept: application/sdp
>         Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO,
> UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>         Supported: timer, precondition, path, replaces
>         Allow-Events: talk, hold, conference, presence, dialog,
> line-seize, call-info, sla, include-session-description,
> presence.winfo, message-summary, refer
>         Reason: Q.850;cause=16;text="NORMAL_CLEARING"
>         Content-Length: 0
>         X-FS-Display-Name: user.name
>         X-FS-Display-Number: sip:user.name at sip.othersipgw.com
>         Remote-Party-ID: "user.name"
> <sip:user.name at sip.othersipgw.com>;party=calling;privacy=off;screen=no
>
>
> 09:31:59.146980 IP (tos 0x0, ttl 64, id 34091, offset 0, flags [none],
> proto UDP (17), length 412)
>     my.natted.priv.ip.address.5080 > othersipgw.ip.address.5060: SIP,
> length: 384
>         ACK sip:user.name at sip.othersipgw.com;+234diallednumber SIP/2.0
>         Via: SIP/2.0/UDP
> my.public.ip.address:5080;rport;branch=z9hG4bK8eDcDvZpBm4yr
>         Max-Forwards: 5
>         From: "0diallednumber"
> <sip:+234diallednumber at my.public.ip.address>;tag=6pKBvpXc32yeD
>         To: <sip:user.name at sip.othersipgw.com;+234diallednumber>;tag=Ur840N1DvjvpH
>         Call-ID: 840abf24-f957-1232-2ca9-525400ecad09
>         CSeq: 68677695 ACK
>         Content-Length: 0