[Freeswitch-users] Newbie -- Help Needed Transferring Inbound Caller ID to external SIP Gateway URI
Avi Marcus
avi at avimarcus.net
Mon Dec 8 16:21:10 MSK 2014
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
> >> >
> >> >
> >> >
> _________________________________________________________________________
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