Trouble Playing Audio to external dialler after bridged call to external SIP URI is hung up

8 Dec 2014

      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@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@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@my.public.ip.address>;tag=6pKBvpXc32yeD
        To: <sip:user.name@sip.othersipgw.com;+234diallednumber>
        Call-ID: 840abf24-f957-1232-2ca9-525400ecad09
        CSeq: 68677695 INVITE
        Contact: <sip:mod_sofia@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@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@my.public.ip.address>;tag=6pKBvpXc32yeD
        To: <sip:user.name@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@my.public.ip.address>;tag=6pKBvpXc32yeD
        To: <sip:user.name@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@sip.othersipgw.com
        Remote-Party-ID: "user.name"
<sip:user.name@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@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@my.public.ip.address>;tag=6pKBvpXc32yeD
        To: <sip:user.name@sip.othersipgw.com;+234diallednumber>;tag=Ur840N1DvjvpH
        Call-ID: 840abf24-f957-1232-2ca9-525400ecad09
        CSeq: 68677695 ACK
        Content-Length: 0

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Sina Owolabi

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