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

Sun Dec 7 14:47:24 MSK 2014

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
>> >
>> >
>> > _________________________________________________________________________
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>> >
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>> >
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>
>