Hello,
I am using dsiprouter as a public facing server for pass through
registrations to a fusion pbx box. This setup works great and has for a
long time now. The issue I am having is that we are switching soft phone
clients to copier. Zopier uses a push server. When an incoming call comes
into the freeswitch server, it fires off an invite to the Zopier registered
client. For some reason ONLY in this scenario Kamailio is doing an auth
challenge to Freeswitch preventing to call to be sent to the soft phone.
How can I stop this from happening? Other soft phones and hard phones work
great with inbound calls except for Zopier.
thanks,
Hey all,
I'm trying to find the best way for settings a timeout for the 'ringing' stage of a call - meaning - I would like to wait for 20 seconds between receiving status 180 / 183 / early media from the remote end and 200, if I fail to receive the expected response in this timeframe - call should be dropped. Most of the documented timeouts I saw count between the first invite to any of the aforementioned stages - but I'm actually looking for the opposite.
Edward
Dear Team,
I would like to know how to link Kamailio with openIMSCore?I would like to use Kamailio as an Application server.Please guide me or if there is someone who has worked on this subject, I would like to benefit from his help.
Hi to all,
As I found there isn't any support for this AUID : simservs.ngn.etsi.org in Kamailio.Anyone can confirm that we can't use Kamailio as mmtel for IMS with the current xcap_server module?!!Thank you.
Regards,Hossein
Is it somehow possible that ngrep shows incoming INVITE arriving over
TCP to Kamailio's listening address and port, but there is no debug
trace of the request (e.g. receive_msg(): --- received sip message ...)
in syslog?
-- Juha
Hi,
I am currently using TOPOS to make Kamailio behave more like a B2BUA from
the clients perspective. It's working well, however I have discovered a
scenario where it fails.
To aid with some interoperability I am sending (sl_send_reply) 200 OK to an
in-dialog SUBSCRIBE request during a call. Once this has happened, the
subsequent BYE from the B leg goes to the incorrect (Private) IP. If I
disable this SUBSCRIBE, or relay it to the B leg, the BYE goes to the
expected IP address.
Does anyone know where I am going wrong? I have tried not calling
record_route() for these messages but the result is the same.
I've attached the ladder diagram of the call below. Note the subscribe is
not relayed in this scenario.
Thanks!
[image: image.png]
> Serdar,
>
> Have you tried to clean variables before calling new ds_select_domain(),
> that are using by dispatcher module failover?
>
> Like
> https://kamailio.org/docs/modules/5.3.x/modules/dispatcher.html#dispatcher.… <https://kamailio.org/docs/modules/5.3.x/modules/dispatcher.html#dispatcher.…>
> and so on?
>
> But as I got, you're saying, that calling ds_select_domain() with
> different setid's in a case of fail, not really fails, but using "old"
> available destinations from previous attempt?
>
> Regards,
> Igor
Igor, also thanks for your interest.
> Hello,
>
> delete the xavps based on the names you set via modparams xavp_dst and
> xavp_ctx.
>
> Cheers,
> Daniel
I remove xavp value using "xavp_rm("_dsdst_")", i supposed that my
problem was solved
but in my other tests, i recognised that xavp_rm removed the first index
of "xavp(_dsdst_)" not all previous destinations(indexes).
after researching, i found a post at
https://lists.kamailio.org/pipermail/sr-users/2020-May/109192.html
and i removed all ellements of list as below,
if(defined $xavp(_dsdst_)) {
while($xavp(_dsdst_[0]) != $null) {
xlog("L_INFO", "--- Loaded Dispatchers --- Grp :
$xavp(_dsdst_[0]=>grp)\n");
xlog("L_INFO", "--- Loaded Dispatchers --- Uri :
$xavp(_dsdst_[0]=>uri)\n");
$xavp(_dsdst_[0]) = $null;
}
}
it worked but i am not sure it is a best solution. Is there another
simple way to delete all indexes of xavp?
Best regards,
Serdar
Hi,
I am using an old Cisco 2811 with an FXS card and trying to get it to
convert between POTS and SIP. Mostly for "fun". For the most part it works,
however there are some problems with DTMF in the call.
Because the end device uses pulse dialling it seems that only the Cisco
Proprietary SIP NOTIFY method of sending DTMF is supported. It sends 4
bytes in the message body that indicate the DTMF digit sent. I'd like to
convert it to the more standard SIP INFO method. The docs about the
encoding are here -
https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/sip/configuration/1…
I looked at the transformations but couldn't find anything relevant. Is it
possible to extract the binary from the body and decode it?
Thanks
Matthew
Hello there,
According to ,
https://lists.kamailio.org/pipermail/sr-users/2016-March/092058.html, which
talked about B2BUA (just signalling) in Kamailio.
As i have experienced working with SEMS, freeswitch and Kamailio while
using B2BUA feature, Each of them have pros and cons:
1- The sems is a light sip engine server with several applications (like as
sbc) for using b2bua. All incoming and outgoing calls could go to sems
server for doing b2bua like this:
Incoming<=======>Kamailio<========>Sems<========>Kamailio<=======>outgoing
2- In sems, you could disable rtp realying. It forces sems to work just as
b2bua without anchoring RTP
3- Easy to use different active profiles in routing.
Just a couple of things there are in SEMS. For example, the sems adds
itself (local IP) in Via header, and it couldn't be common in b2bua. like
this:
.
.
.
INVITE sip:200@cloud.domain.com;transport=UDP SIP/2.0
Via: SIP/2.0/UDP
10.10.10.200;branch=z9hG4bK2f53.07a3fd9edaa8c8d609ab2ac6b01a087f.0
*================================>Kamailio
public IP*
*Via: SIP/2.0/UDP
127.0.0.1:5080;received=127.0.0.1;branch=z9hG4bKsPRMNast;rport=5080
===============================>private loopback sems ip*
From: <sip:100@cloud.domain.com
;transport=UDP>;tag=6FFDB493-60EABB3600016ECD-379F9700
To: <sip:200@cloud.domain.com;transport=UDP>
CSeq: 10 INVITE
Call-ID: Y2M1ODQxNTZmMjdkZWZjN2U5MmMyYjBmN2Y2OGY1ODQ._leg2
Route: <sip:mo@127.0.0.1:5060;lr>
Max-Forwards: 69
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE
User-Agent: Z 3.3.25608 r25552
Content-Type: application/sdp
Contact: <sip:127.0.0.1:5080;transport=udp>
Content-Length: 246
.
.
.
For this reason, Is there a way to avoid this issue? I know it is possible
to do this by using other modules like textops and retransformation ?
And why in Kamailio, there is no b2bua module to perform all b2bua
functionality, yet?
Thanks with Best Regards.
--
--Mojtaba Esfandiari.S