[SR-Users] Problem with incoming audio from Asterisk

2019-09-14 Thread przeqpiciel
Hi, After a month of fight with SIP signalling I made it but now I have problem with incoming audio from Asterisk. Kamailio and Asterisk are their own public IPs I dont use rtpproxy. When Asterisk sends rtp I'm able to see RTP on my router but it not going to my computer. Maybe somebody have idea

Re: [SR-Users] How to distinguish is from Asterisk or from the client

2019-09-12 Thread przeqpiciel
- treat as > > # "inbound". > > > > ... > >} else { > > # Do authentication challenge and treat as "outbound" > > # from customer softphone. > > > > ... > >} > > > > -- Alex &

[SR-Users] How to distinguish is from Asterisk or from the client

2019-09-11 Thread przeqpiciel
Hi, I looking information how to distinguish whether the packet is directed to the Astersik (my platform) or to the customer (softphone / sip client ). ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/

Re: [SR-Users] Define properly IP in record-route header

2019-08-27 Thread przeqpiciel
Nope, its broken again. I am probably too stupid for this. wt., 27 sie 2019 o 23:12 przeqpiciel napisał(a): > Right now it works. > I added additional network card to my virtual kamailio, then put this NIC > to asterisks network also i configured multiple listen and mhomed=1 > &

Re: [SR-Users] Define properly IP in record-route header

2019-08-27 Thread przeqpiciel
ty and errors. > > On Aug 27, 2019, at 9:43 AM, Sergiu Pojoga wrote: > > Do you have: mhomed=1 ? > > On Tue, Aug 27, 2019 at 8:52 AM przeqpiciel wrote: > >> Thank you for the reply. It little help for me, Right now probably I have >> to make logic which change record

Re: [SR-Users] Define properly IP in record-route header

2019-08-27 Thread przeqpiciel
No, I hear about that first time. But I think i still need program that will rewrite record route depends if it goes to asterisk or to Internet wt., 27 sie 2019 o 15:44 Sergiu Pojoga napisał(a): > Do you have: mhomed=1 ? > > On Tue, Aug 27, 2019 at 8:52 AM przeqpiciel wrote: > >

Re: [SR-Users] Define properly IP in record-route header

2019-08-27 Thread przeqpiciel
t; But the first approach is better and more comprehensive for your > situation, since it also covers Via. > > — Alex > > — > Sent from mobile, with due apologies for brevity and errors. > > On Aug 26, 2019, at 4:12 PM, przeqpiciel wrote: > > Hi, > I have some issue a

[SR-Users] Define properly IP in record-route header

2019-08-26 Thread przeqpiciel
Hi, I have some issue and dont have enought knowledge to fix it. I have simple infrastructure. Internet -> router (NAT) -> kamailio -> asterisk Kamailio and Asterisk are inside the network and Kamailio places his own private address in the "record-route" heade

[SR-Users] Cant finish call

2019-08-23 Thread przeqpiciel
Hi all, I am totally new with Kamailio but i have needs to build small cluster of Asterisk. Right now I could succesfully register to asterisk via kamailio and even make a call to hear tt-weasels but Asterisk cant close call My infrastructure diagram is here http://www.asciidraw.com/#Draw681604162

Re: [SR-Users] How to configure Kamailio 5.1 with Asterisk pjsip

2019-05-05 Thread przeqpiciel
I would like to have an infrastructure where there are two machines with Kamailio installed (K1, K2) and another two with Asterisk installed (A1, A2). Ultimately, I want to be able to make calls through any Asterisk dialplan regardless of which Kamailio the registration took place. When this is ach

Re: [SR-Users] How to configure Kamailio 5.1 with Asterisk pjsip

2019-05-04 Thread przeqpiciel
Villasmil napisał(a): > You’re probably better off doing registration on kamailio only. Depending > on what exactly you’re trying to achieve. > > A trace would help a lot. Check the IP addresses in the SDP actually > belong to the user and the asterisk. > > On Sat, 4 May 20

[SR-Users] How to configure Kamailio 5.1 with Asterisk pjsip

2019-05-04 Thread przeqpiciel
Hi, There is any working examples how to configure dispatcher module of Kamailio? I have working registrations but working in half dialing function.Signalling are come to asterisk and rtp start sending to my IP but i cant hear anything. Asterisk are doing registrar job - i dont know if it is good