Re: [SR-Users] kamailio & vmware

Many thanks for the help, I think I am getting closer, and obviously ;-) the issue is not caused by Kamailio ! under some load (still to confirm); when asterisk receives the relayed invite from K, it generates the reply message (100 trying) but this message does not reach the network, ie, I see

[SR-Users] KEMI basic logging

Hi, Tried just to reject any request with some log message: $ cat kamailio/Dockerfile FROM centos:centos7 RUN yum update -y RUN yum install -y wget RUN wget -O /etc/yum.repos.d/kamailio.repo

Re: [SR-Users] How to extract SIP-I ISUP parameters

Tried to use get_body_part and isup_to_json first, but looks like I can't understand syntax. My config: loadmodule "tm.so" loadmodule "sl.so" loadmodule "xlog.so" loadmodule "textops.so" loadmodule "ss7ops.so" listen=udp:0.0.0.0:5060 route { get_body_part("application/isup", "$var(pbody)");

Re: [SR-Users] 180 and 183 SIP messages

You can absolutely do it: https://kamailio.org/docs/modules/5.1.x/modules/textopsx.html#textopsx.f.change_reply_status You can even remove the embedded SDP if desired, using the various textops functions: https://kamailio.org/docs/modules/5.1.x/modules/textops.html Whether you _should_ do this

Re: [SR-Users] 180 and 183 SIP messages

On Thu, Mar 29, 2018 at 06:44:01PM +0300, Ali Taher wrote: > I'm facing an issue , when I'm getting back the 183 message from the > supplier(B-Party) and forwarding it to the customer(A-Party); the customer > is not hearing the RBT and asking to send 180 rather than 183 . (the > supplier is only

Re: [SR-Users] 180 and 183 SIP messages

Hi, It should be possible with freeswitch as well, google for freeswitch cause code substitution and you will see example. Regards, Arsen Semionov www.eurolan.info cell: +442035198881 On Thu, Mar 29, 2018 at 6:44 PM, Ali Taher wrote: > Hello everyone, > > > > I’m

[SR-Users] 180 and 183 SIP messages

Hello everyone, I'm working on freeswitch , where I have SIP calls coming from A-Party to my freeswitch and I'm routing it to B-Party. I'm facing an issue , when I'm getting back the 183 message from the supplier(B-Party) and forwarding it to the customer(A-Party); the customer is not

Re: [SR-Users] How to extract SIP-I ISUP parameters

As I got it, there is an ISUP parser inside ss7ops module. It may require a bit of C coding to build a function for helping with what you need -- the two functions that are useful for doing it are:   * https://www.kamailio.org/docs/modules/stable/modules/textops.html#textops.f.get_body_part   *

Re: [SR-Users] Kamailio fails to start because of jsonrpl

Do you have jsonrpcs module loaded? Are you running jsonrcp_exec() before trying to access $jsonrpl(key) ? ...This variable gives access to JSONRPC reply after executing jsonrpc_exec(…) in kamailio.cfg On Thu, Mar 29, 2018 at 3:56 AM, Dmitry wrote: > Kamailio

Re: [SR-Users] Kamailio to Asterisk Proxy with NAT

On Thu, Mar 29, 2018 at 11:37:37AM +0200, Benjamin Marty wrote: > I need the scenario for scalability on the asterisk side. > > The scenario I still have in my head is that the clients -could- do the RTP > stream directly with the Asterisk Server. The Asterisk Instances have all > their own NATed

[SR-Users] Kamailio fails to start because of jsonrpl

Kamailio fails to start and I see the following errors: kamailio[2115]: ERROR: [core/pvapi.c:1085]: pv_parse_spec2(): bad tr in pvar name "jsonrpl" kamailio[2115]: ERROR: [core/pvapi.c:]: pv_parse_spec2(): invalid parsing in [$(jsonrpl(body){kz.json, result.NRSETS})] at (4) kamailio[2115]:

Re: [SR-Users] Kamailio to Asterisk Proxy with NAT

I need the scenario for scalability on the asterisk side. The scenario I still have in my head is that the clients -could- do the RTP stream directly with the Asterisk Server. The Asterisk Instances have all their own NATed Public IP (AWS EC2 Instances). But I think I will build more problems

Re: [SR-Users] Kamailio to Asterisk Proxy with NAT

> My Question is now how the RTP Media Stream should/can flow. The clients > are in different other networks. So P2P Media Stream isn't possible. Should > I now run the RTP Stream Client - Asterisk or Client - Kamailio - Asterisk? What do you want to accomplish? Fact is that asterisk has to

[SR-Users] Kamailio to Asterisk Proxy with NAT

Hello, I have multiple Asterisk Servers in a private Network. They also have a Public IP via Destination NAT. Now I want to use a Kamailio Proxy in front of them. The Routing I want to do with the Kamailio Dispatcher module. My Question is now how the RTP Media Stream should/can flow. The

Re: [SR-Users] Missing exit function on KEMI?

Hello, then you do not use the KEMI framework as config interpreter, but native configuration interpreter with inline execution of Python code via python_exec(). KSR module is exported for both cases, KEMI or inline execution, but KSR.x.exit() (or python exit) stops the KEMI execution, not the

Re: [SR-Users] How to extract SIP-I ISUP parameters

Hello, can you provide a pcap (or ngrep output) with such invite in order to see what can be done with existing modules? Maybe it needs just some script operations. Cheers, Daniel On 28.03.18 17:26, Eugene Prokopiev wrote: > Hi, > > I need to extract ISUP parameter 'Redirecting number' from

Re: [SR-Users] Missing exit function on KEMI?

Hi Daniel, it seems that your suggestion is not working, we are not using python for the whole script, we have some functions written in python to manipulate jsons and to do API queries which we keep in python but the core of the routing logic is still written in Kamailio scripting language. We

Re: [SR-Users] Kamailio NAT AWS EC2

I have now put a fix_nated_sdp statement to the route[RELAY]. But the media proxy over the rtpproxy still doesn't work. The SIP Signalling itself works perfectly (I can start a call an the other client receives it). fix_nated_sdp("2", "MYPUBLICIP"); Thanks Benjamin 2018-03-27 12:15 GMT+02:00