Hi Mojtaba.

I managed to get it working in the following way:

1. I set FilterCriteria for INVITE in the user profile
2. In asterisk sip.conf I set outboundproxy (no need to modify DNS)
3. I set a new class in musiconhold.conf
4. I set dial plan in extensions.conf

exten => 972551000002,1,Progress()
exten => 972551000002,n,Dial(SIP/972551000...@mnc001.mcc001.3gppnetwork.org,20,m(mymoh));

Tnx.

On 28.01.19 г. 12:29 ч., Mojtaba wrote:
In another way, you could don't change this file, instead of change your dial plan like below: exten => 972551000002,1,Dial(SIP/972551000...@icscf.mnc001.mcc001.3gppnetwork.org,20 <mailto:SIP/972551000...@mnc001.mcc001.3gppnetwork.org,20>);
WIth Regards.Mojtaba

On Mon, Jan 28, 2019 at 1:56 PM Mojtaba <mes...@gmail.com <mailto:mes...@gmail.com>> wrote:

    It would be like these lines with afew changes:
    mnc001.mcc001.3gppnetwork.org
    <http://mnc001.mcc001.3gppnetwork.org>. 1D IN A           10.82.10.56
    mnc001.mcc001.3gppnetwork.org
    <http://mnc001.mcc001.3gppnetwork.org>. 1D IN NAPTR 10 50 "s"
    "SIP+D2U"    ""    _sip._udp
    mnc001.mcc001.3gppnetwork.org
    <http://mnc001.mcc001.3gppnetwork.org>. 1D IN NAPTR 20 50 "s"
    "SIP+D2T"    ""    _sip._tcp


    On Mon, Jan 28, 2019 at 1:37 PM Tsvetan Filev
    <tsvetan.fi...@inno-networks.com
    <mailto:tsvetan.fi...@inno-networks.com>> wrote:

        Here is my current zone file:

        $ORIGIN mnc001.mcc001.3gppnetwork.org
        <http://mnc001.mcc001.3gppnetwork.org>.
        $TTL 1W
        @                       1D IN SOA       localhost.
        root.localhost. (
        1               ; serial
        3H              ; refresh
        15M             ; retry
        1W              ; expiry
                                                1D )            ; minimum

                                1D IN NS        ns
        ns                      1D IN A         10.82.10.56

        pcscf                   1D IN A         10.82.10.56
        _sip._udp.pcscf         1D SRV 0 0 5060 pcscf
        _sip._tcp.pcscf         1D SRV 0 0 5060 pcscf

        icscf                   1D IN A         10.82.10.56
        _sip._udp               1D SRV 0 0 4060 icscf
        _sip._tcp               1D SRV 0 0 4060 icscf
        _sip._udp.ims           1D SRV 0 0 4060 icscf
        _sip._tcp.ims           1D SRV 0 0 4060 icscf

        scscf                   1D IN A         10.82.10.56
        _sip._udp.scscf         1D SRV 0 0 6060 scscf
        _sip._tcp.scscf         1D SRV 0 0 6060 scscf

        as                      1D IN A         10.82.10.56
        _sip._udp.as <http://udp.as>            1D SRV 0 0 5062 as
        _sip._tcp.as <http://tcp.as>            1D SRV 0 0 5062 as

        hss                     1D IN A         10.82.10.56


        How do I modify it in order to make this work ?

        Tnx.

        On 28.01.19 г. 11:50 ч., Mojtaba wrote:
        Hi Tsvetan,
        Why do you send call back to S-CSCF? You should send call
        back to I-CSCF.  Actually in resolve of domain
        "mnc001.mcc001.3gppnetwork.org"
        <mailto:SIP/972551000...@mnc001.mcc001.3gppnetwork.org,20>,
        The ICSCF's IP should be returned.
         Make sure entry SRV recordd in DNS server are true.
        This kind of call back to IMS is true, But make sure you
        won't have any issue in DNS resolve.
          exten =>
        972551000002,1,Dial(SIP/972551000...@mnc001.mcc001.3gppnetwork.org,20
        <mailto:SIP/972551000...@mnc001.mcc001.3gppnetwork.org,20>);

        With Regards.Mojtaba
        On Mon, Jan 28, 2019 at 12:34 PM Tsvetan Filev
        <tsvetan.fi...@inno-networks.com
        <mailto:tsvetan.fi...@inno-networks.com>> wrote:

            Hi Mojtaba.

            I implemented the AS way and was able to play sound to
            the caller but In order to continue the call and send the
            invite to SCSCF I need to use proxy in the Dial
            application which is a problem (Asterisk is B2BUA not a
            proxy).
            I found this old question here
            
https://community.asterisk.org/t/how-can-i-configure-asterisk-to-act-like-a-sip-proxy-server/18464
            that describes exactly the same issue.
            Here is my dial plan:

            exten => 972551000002,1,Progress()
            exten => 972551000002,n,Playback(vm-starmain, noanswer)
            exten => 972551000002,n,Wait(3)
            exten => 972551000002,n,Hangup()
              ; This will send the call to the pcscf again
              ;  exten =>
            
972551000002,1,Dial(SIP/972551000...@mnc001.mcc001.3gppnetwork.org,20
            <mailto:SIP/972551000...@mnc001.mcc001.3gppnetwork.org,20>);
              ; This will send the call to scscf but it will be
            rejected as domain not supported
              ;  exten =>
            
972551000002,1,Dial(SIP/972551000...@scscf.mnc001.mcc001.3gppnetwork.org,20
            <mailto:SIP/972551000...@scscf.mnc001.mcc001.3gppnetwork.org,20>);

            Can I use kamailio as an AS and implement the same ?

            Regards.

            On 22.12.18 г. 0:06 ч., Mojtaba wrote:
            Hello Tsvetan.
            Actually you could use SIP Early media in AS and also
            with cscf.
            If you choice the first way, i think it is very simple
            and strightforward because you just use early media
            functions on your AS. For example in Astrisk you could
            use Progress application and 'm' option in Dial
            application in your dialplan.
            In second way you should check in Reply-Route block,if
            you got 180 ringing,  you have to use rtpproxy-stream
            funtion for doing sip early.

            Wih Regards.Mojtaba Esfandiari.S

            On Fri, 21 Dec 2018, 16:34 Tsvetan Filev,
            <tsvetan.fi...@inno-networks.com
            <mailto:tsvetan.fi...@inno-networks.com>> wrote:

                Hi all.

                I want to use SIP early media to play music to the
                caller in kamailio
                IMS installation like this:
                http://www.sharetechnote.com/html/IMS_SIP_EarlyMedia.html

                I looked a little bit but didn't find ready
                solution. The information is
                vague on this topic.
                Should this be done through a module or application
                server ?
                May I need to handle ringing in onreply_route and
                send OK with SDP to
                the caller in SCSCF ?

                Regards.


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-- --Mojtaba Esfandiari.S



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--Mojtaba Esfandiari.S
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