Greetings,
I'm building some testing procedures for my Kamailio proxy. In order to do
that i have a SipTester (Acting as Calling Number) from where i want to
originate calls to Kamailio. After Kamailio processes the call, i want it
to route it to SipTester again so it can act as Called Number.
This should work with only one SipTester in one machine, so i created two
SipTester GW's on Kamailio, belonging to different users and GW groups.
Also, the two GWs use different Kamailio addresses to communicate with the
proxy.
However, in this combination, when routing i get this error " ERROR: dialog
[dlg_handlers.c:666]: pre_match_parse(): bad request or missing CALLID/TO
hdr :-/". If i change the destination or the origin of the INVITE the calls
work without a problem.
Does this happen because both GW's have the same address?
Best Regards
Hello Kamailions,
I have been able to get the presence configured, subscribers see the status of their subscriptions, this is working fine, for internal, incoming external and external calls. This was solved by enabling use_pubruri_avps in the pua_dialoginfo module.
Now I want to be able to route calls based on the presence state.
Using the presentity table could be used by doing a query based on the userpart of the ruri, which works fine when the call is still active, but it can take a while for the record to be removed from the presentity table, as that is dependend on the expires value. So when a query is made after a call is finished, a record would still be present in the presentity table till clean_period timer has hit again and the expires value is expired.
So I should get the status from memory, but I have not been able to find the information on how to do that.
Can someone enlighten me on where to find that information?
Rgds,
Gertjan
hi i m using kamailio
where i m tring to hold the call bitween switch A and Switch B
PSTN -> Switch A -> SWITCHB -> SwitchC -> DESTINATION
When PSTN hangs up, I need the call to stay up between SwitchA and B for an
arbitrary number of seconds or minute
plz help
thank you
--
*Regards:*
Gaurav Kumar
Hey support,
Today I ran into this error on my latest 5.2.3 Kamailio box:
Jul 18 15:05:28 sjomaintpsg51 /usr/sbin/kamailio[5525]: CRITICAL: tls
[tls_init.c:671]: init_tls_h(): installed openssl library version is too
different from the library the kamailio tls module was compiled with:
installed “OpenSSL 1.1.1 11 Sep 2018” (0x1010100f), compiled “OpenSSL
1.1.0g 2 Nov 2017" (0x1010007f).#012 Please make sure a compatible version
is used (tls_force_run in kamailio.cfg will override this check)
There's not support for openssl1.1.1 ?
Thanks
--
Andy Chen
Sr. Telephony Lead Engineer
achen@ <achen(a)thinkingphones.com>fuze.com
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I am configuring kamailio as registar and asterisk for media services.
in kamailio + asterisk integration how two SIP clients RTP traffic
will flow, are they go via Asterisk or directly from kamailio ?
Hello list,
if KSR.is_OPTIONS() and KSR.is_myself_ruri():
KSR.x.modf("options_reply");
exit();
with a valid URI (no username) gives the subj output (opt_reply(): called for non-OPTIONS request)
How could that be?
Python KEMI, 5.2.3, docker.
________________________________
Regards,
Alexandru Covalschi
VoIP Engineer and System Administrator
tel: +37367367850
I'm trying to connect a VoLTE smartphone to an IP PBX using Kamailio IMS.
I've managed to get the smartphone to connect to Kamailio IMS and make calls with our devices registered with Kamailio.
The problem is that I need the smartphones to register with the IP PBX which is a VigorBX 2000n. This is so they can use an internal extension number and make calls to other local extensions with hard wired desk phones.
Should I be using Kamailio IMS to "proxy" the smartphone SIP connection to the VigorBX PBX?
Or would it be better to connect the smartphone directly to the VigorBX using SIP? If so, how?
I've read absolutely every how-to document I could lay my hands on but none of them describe integrating IMS with a PBX.
Thanks for any help / advice you can offer.
Dave
