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/972551000...@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/972551000...@icscf.mnc001.mcc001.3gppnetwork.org,20
<mailto:SIP/972551000...@mnc001.mcc001.3gppnetwork.org,20>);
WIth Regards.Mojtaba
On Mon, Jan 28, 2019 at 1:56 PM Mojtaba <mes...@gmail.com
<mailto:mes...@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.fi...@inno-networks.com
<mailto:tsvetan.fi...@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/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
<mailto: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
<mailto: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
<mailto: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
<mailto: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
<mailto: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.
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--Mojtaba Esfandiari.S
--
--Mojtaba Esfandiari.S
--
--Mojtaba Esfandiari.S
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