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 <SIP/972551000...@mnc001.mcc001.3gppnetwork.org,20>); WIth Regards.Mojtaba
On Mon, Jan 28, 2019 at 1:56 PM Mojtaba <mes...@gmail.com> wrote: > It would be like these lines with afew changes: > mnc001.mcc001.3gppnetwork.org. 1D IN A 10.82.10.56 > mnc001.mcc001.3gppnetwork.org. 1D IN NAPTR 10 50 "s" > "SIP+D2U" "" _sip._udp > 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> wrote: > >> Here is my current zone file: >> >> $ORIGIN 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 1D SRV 0 0 5062 as >> _sip._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" >> <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); >> >> With Regards.Mojtaba >> On Mon, Jan 28, 2019 at 12:34 PM Tsvetan Filev < >> 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); >>> ; 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); >>> >>> 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> 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@lists.kamailio.org >>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>> >>> >>> _______________________________________________ >>> Kamailio (SER) - Users Mailing >>> Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >> >> -- >> --Mojtaba Esfandiari.S >> >> > > -- > --Mojtaba Esfandiari.S > -- --Mojtaba Esfandiari.S
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