[Freeswitch-users] Newbie -- Help Needed Transferring Inbound Caller ID to external SIP Gateway URI

Sina Owolabi notify.sina at gmail.com
Mon Dec 8 17:07:57 MSK 2014 

I've tried to play a message after but I cannot hear it. Please can
you tell me if this is correct?
Thanks again for all your help:


<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 connection-->
<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>

On Mon, Dec 8, 2014 at 2:21 PM, Avi Marcus <avi at avimarcus.net> wrote:
> Since there's only a 100 trying and no 18X, you won't get a ringing tone.
> If it's short, just play a message after. If not, you can use lua/js and run
> this second call as a bgapi call and try to capture the response.
> -Avi
>
>
> On Sun, Dec 7, 2014 at 1:47 PM, Notify Me <notify.sina at gmail.com> wrote:
>>
>> The delay is very short, they send a100 Trying  and then a 480 temporarily
>> Unavailable back, as I can see in tcpdump. After they send the fail
>> and its acknowledged, they end it and send an SMS back.
>> This is a full dump
>>
>> 12:41:27.169928 IP (tos 0x0, ttl 64, id 34092, offset 0, flags [none],
>> proto UDP (17), length 1146)
>>     10.22.0.252.5080 > othersipgw.ip.address.5060: SIP, length: 1118
>>         INVITE sip:sina.nowahala at sip.othersipgw.com;+234diallednumber
>> SIP/2.0
>>         Via: SIP/2.0/UDP
>> my.public.ip.address:5080;rport;branch=z9hG4bK2S6KjD1jXFj1Q
>>         Max-Forwards: 5
>>         From: "234diallednumber"
>> <sip:+234diallednumber at my.public.ip.address>;tag=04tDF6pXQU4Ng
>>         To: <sip:sina.nowahala at sip.othersipgw.com;+234diallednumber>
>>         Call-ID: d1ef2fdb-f8a8-1232-80a6-525400ecad09
>>         CSeq: 68640179 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: "234diallednumber"
>>
>> <sip:+234diallednumber at my.public.ip.address>;party=calling;screen=yes;privacy=off
>>
>>         v=0
>>         o=FreeSWITCH 1417930197 1417930198 IN IP4 my.public.ip.address
>>         s=FreeSWITCH
>>         c=IN IP4 my.public.ip.address
>>         t=0 0
>>         m=audio 22290 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
>>
>> 12:41:27.385419 IP (tos 0x0, ttl 53, id 5736, offset 0, flags [none],
>> proto UDP (17), length 384)
>>     othersipgw.ip.address.5060 > 10.22.0.252.5080: SIP, length: 356
>>         SIP/2.0 100 Trying
>>         Via: SIP/2.0/UDP
>> my.public.ip.address:5080;rport=5080;branch=z9hG4bK2S6KjD1jXFj1Q
>>         From: "234diallednumber"
>> <sip:+234diallednumber at my.public.ip.address>;tag=04tDF6pXQU4Ng
>>         To: <sip:sina.nowahala at sip.othersipgw.com;+234diallednumber>
>>         Call-ID: d1ef2fdb-f8a8-1232-80a6-525400ecad09
>>         CSeq: 68640179 INVITE
>>         User-Agent: FreeSWITCH-mod_sofia/1.2.14
>>         Content-Length: 0
>>
>>
>> 12:41:27.424422 IP (tos 0x0, ttl 53, id 5737, offset 0, flags [none],
>> proto UDP (17), length 1015)
>>     othersipgw.ip.address.5060 > 10.22.0.252.5080: SIP, length: 987
>>         SIP/2.0 480 Temporarily Unavailable
>>         Via: SIP/2.0/UDP
>> my.public.ip.address:5080;rport=5080;branch=z9hG4bK2S6KjD1jXFj1Q
>>         Max-Forwards: 4
>>         From: "234diallednumber"
>> <sip:+234diallednumber at my.public.ip.address>;tag=04tDF6pXQU4Ng
>>         To:
>> <sip:sina.nowahala at sip.othersipgw.com;+234diallednumber>;tag=8c8DZK1aK9mKN
>>         Call-ID: d1ef2fdb-f8a8-1232-80a6-525400ecad09
>>         CSeq: 68640179 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: sina.nowahala
>>         X-FS-Display-Number: sip:sina.nowahala at sip.othersipgw.com
>>         Remote-Party-ID: "sina.nowahala"
>> <sip:sina.nowahala at sip.othersipgw.com>;party=calling;privacy=off;screen=no
>>
>>
>> 12:41:27.424789 IP (tos 0x0, ttl 64, id 34093, offset 0, flags [none],
>> proto UDP (17), length 412)
>>     10.22.0.252.5080 > othersipgw.ip.address.5060: SIP, length: 384
>>         ACK sip:sina.nowahala at sip.othersipgw.com;+234diallednumber SIP/2.0
>>         Via: SIP/2.0/UDP
>> my.public.ip.address:5080;rport;branch=z9hG4bK2S6KjD1jXFj1Q
>>         Max-Forwards: 5
>>         From: "234diallednumber"
>> <sip:+234diallednumber at my.public.ip.address>;tag=04tDF6pXQU4Ng
>>         To:
>> <sip:sina.nowahala at sip.othersipgw.com;+234diallednumber>;tag=8c8DZK1aK9mKN
>>         Call-ID: d1ef2fdb-f8a8-1232-80a6-525400ecad09
>>         CSeq: 68640179 ACK
>>         Content-Length: 0
>>
>>
>> So you recommend a transfer on fail wav to be played? I have been
>> tearing my hair out trying to play sounds before the call is
>> transferred to them, so that the dialler hears something.
>>
>> On Sun, Dec 7, 2014 at 4:44 AM, Avi Marcus <avi at avimarcus.net> wrote:
>> > How long is the delay? I assume they return a normal_clearing or fail
>> > 4XX?
>> > If the delay is short and they have a "real" status return, then.. for
>> > success, you can just queue a "success.wav" to play after.
>> > And for fails, you can use
>> > https://wiki.freeswitch.org/wiki/Variable_transfer_on_fail and transfer
>> > to
>> > an extension that plays an error message.
>> >
>> > This SMS sending sounds like a normal API call.. which if it was, could
>> > be
>> > done with curl or inside a lua/js script.
>> >
>> > -Avi
>> >
>> > On Fri, Dec 5, 2014 at 6:47 PM, Notify Me <notify.sina at gmail.com> wrote:
>> >>
>> >> Hi Avi!
>> >>
>> >> Thanks for the reply, I did figure it out, thanks! This is what I did,
>> >>
>> >> <include>
>> >>  <extension name="firstsipgw-inbound">
>> >>   <condition field="destination_number"
>> >> expression="^0123450(\d{2})$"/>
>> >>   <condition field="caller_id_number"  expression="^0(\d{10})$"
>> >> require-nested="true">
>> >>      <action application="set"
>> >> data="effective_caller_id_number=+234$1"/>
>> >>      <action application="set"
>> >> data="caller_id_number=+234${effective_caller_id_number}"/>
>> >>      <action application="set" data="transfer_ringback=$${us-ring}"/>
>> >>      <action application="set"
>> >> data="transfer_ringback=file_string://$${hold_music}"/>
>> >>      <action application="set"
>> >> data="transfer_ringback=local_stream://connecting"/>
>> >>        <action application="bridge"
>> >>
>> >>
>> >> data="sofia/external/sip:user.name at sip.othersipgw.com;${effective_caller_id_number}"/>
>> >>   </condition>
>> >>  </extension>
>> >> </include>
>> >>
>> >> I can see from the logs that dials the othersipgw.com URI, and I can
>> >> see tcpdumps that corresponds to the traffic being sent,  and the
>> >> other side gets it. The other side immediately drops the call and does
>> >> some processing that returns an SMS to the dialler.
>> >>
>> >> I dont know if I can trouble you for more help.. there is no ringing
>> >> when the transaction happens, and the dialler is unsure what is
>> >> happening, if the call actually connected. As you might have guessed
>> >> from the above I am trying to make it ring for a few seconds before
>> >> the call drops. Is this possible? I would like for the dialler to be
>> >> able to hear it ring a few times before the call is cut.
>> >>
>> >> Any help gratefully accepted.
>> >>
>> >> On Tue, Dec 2, 2014 at 2:22 PM, Avi Marcus <avi at avimarcus.net> wrote:
>> >> > Hi - did you figure this out yet?
>> >> >
>> >> > One comment:
>> >> >  <action application="bridge"
>> >> > data="sofia/gateway/othersipgw/+234$1"/>
>> >> > -- I
>> >> > don't know if $1 is available anymore. You might want to just set
>> >> > that
>> >> > as
>> >> > part of the actual number to route, e.g. add a 1212 prefix and match
>> >> > it
>> >> > again or set it as a channel variable. You can see in the logs if the
>> >> > $1
>> >> > is
>> >> > resolving correctly.
>> >> >
>> >> > -Avi
>> >> >
>> >> >
>> >> >
>> >> > _________________________________________________________________________
>> >> > 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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>> >> >
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>> >
>> >
>
>

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