[SR-Users] IMS early media

Tsvetan Filev tsvetan.filev at inno-networks.com
Thu Jan 31 14:04:51 CET 2019


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/972551000002 at 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/972551000002 at icscf.mnc001.mcc001.3gppnetwork.org,20 
> <mailto:SIP/972551000002 at mnc001.mcc001.3gppnetwork.org,20>);
> WIth Regards.Mojtaba
>
> On Mon, Jan 28, 2019 at 1:56 PM Mojtaba <mespio at gmail.com 
> <mailto:mespio at 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.filev at inno-networks.com
>     <mailto:tsvetan.filev at 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/972551000002 at 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/972551000002 at mnc001.mcc001.3gppnetwork.org,20
>>         <mailto:SIP/972551000002 at mnc001.mcc001.3gppnetwork.org,20>);
>>
>>         With Regards.Mojtaba
>>         On Mon, Jan 28, 2019 at 12:34 PM Tsvetan Filev
>>         <tsvetan.filev at inno-networks.com
>>         <mailto:tsvetan.filev at 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/972551000002 at mnc001.mcc001.3gppnetwork.org,20
>>             <mailto:SIP/972551000002 at 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/972551000002 at scscf.mnc001.mcc001.3gppnetwork.org,20
>>             <mailto:SIP/972551000002 at 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.filev at inno-networks.com
>>>             <mailto:tsvetan.filev at 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.
>>>
>>>
>>>                 _______________________________________________
>>>                 Kamailio (SER) - Users Mailing List
>>>                 sr-users at lists.kamailio.org
>>>                 <mailto:sr-users at lists.kamailio.org>
>>>                 https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>>             _______________________________________________
>>>             Kamailio (SER) - Users Mailing List
>>>             sr-users at lists.kamailio.org  <mailto:sr-users at lists.kamailio.org>
>>>             https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>>
>>         -- 
>>         --Mojtaba Esfandiari.S
>
>
>
>     -- 
>     --Mojtaba Esfandiari.S
>
>
>
> -- 
> --Mojtaba Esfandiari.S
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