[SR-Users] IMS early media
Mojtaba
mespio at gmail.com
Thu Jan 31 15:35:12 CET 2019
Great,
Using the progress in dialplan is good sounds.
With Regards.Mojtaba
On Thu, 31 Jan 2019, 16:34 Tsvetan Filev, <tsvetan.filev at inno-networks.com>
wrote:
> 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
> <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> 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.filev at 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/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);
>>>
>>> With Regards.Mojtaba
>>> On Mon, Jan 28, 2019 at 12:34 PM Tsvetan Filev <
>>> 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);
>>>> ; 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);
>>>>
>>>> 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> 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
>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>>>
>>>>
>>>> _______________________________________________
>>>> Kamailio (SER) - Users Mailing Listsr-users at 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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