[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>
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>>>
>>>
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>>
>>
>>
>> --
>> --Mojtaba Esfandiari.S
>
>
>
> --
> --Mojtaba Esfandiari.S
>
>
>
> --
> --Mojtaba Esfandiari.S
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