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@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$1"/> <action application="transfer" data="1212 XML default"/> </condition> </extension> </include>
default: <extension name="1212"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="transfer_after_bridge=1213:XML:default"/> <action application="set" data="exec_after_bridge_app=transfer"/> <action application="set" data="exec_after_bridge_arg="1213"/> <action application="bridge" data="sofia/external/sip:user.name@sip.othersipgw.ip.address.com;${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