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(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

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… 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(a)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(a)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(a)inno-networks.com> wrote: _______________________________________________ Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

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" &lt;SIP/972551000002(a)mnc001.mcc001.3gppnetwork.org,20&gt;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(a)mnc001.mcc001.3gppnetwork.org,20); With Regards.Mojtaba On Mon, Jan 28, 2019 at 12:34 PM Tsvetan Filev < tsvetan.filev(a)inno-networks.com&gt; wrote: -- --Mojtaba Esfandiari.S

In another way, you could don't change this file, instead of change your dial plan like below: exten => 972551000002,1,Dial(SIP/972551000002(a)icscf.mnc001.mcc001.3gppnetwork.org,20 <mailto:SIP/972551000002@mnc001.mcc001.3gppnetwork.org,20>); WIth Regards.Mojtaba On Mon, Jan 28, 2019 at 1:56 PM Mojtaba &lt;mespio(a)gmail.com <mailto:mespio@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 &lt;tsvetan.filev(a)inno-networks.com <mailto:tsvetan.filev@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: -- --Mojtaba Esfandiari.S -- --Mojtaba Esfandiari.S