Re: [SR-Users] pseudo variables in strings

Daniel-Constantin Mierla writes: > Dialplan can evaluate variables, is it a special case you need to return > variables and then evaluate after dp_translate() again? I made a test rule: match_exp subst_exp repl_exp ^\#11\#$ (empty)$fu and dp_translate(id) on sip:#11#@test.tutpro.com pr

Re: [SR-Users] Kamailio 4.3.4: receive_msg(): no via found in reply

Hi Daniel, You mentioned below about this issue not being complaint with the RFC. Our supplier is telling us this is normal behavior and that they went through the RFC and found this was normal behavior. If it's not too much work, could you tell me in which part of the RFC this is described? R

Re: [SR-Users] Kamailio 4.3.4: receive_msg(): no via found in reply

> On 04 Apr 2016, at 10:02, Grant Bagdasarian wrote: > > Hi Daniel, > > You mentioned below about this issue not being complaint with the RFC. > Our supplier is telling us this is normal behavior and that they went through > the RFC and found this was normal behavior. > > If it’s not too mu

Re: [SR-Users] Multiple SIP-servers with SRV-records and authentication secrets

> On 03 Apr 2016, at 18:09, Alfred E. Heggestad wrote: > > Dear SIP-experts and DNS-SRV gurus; > > > I have some questions to the deployers of SER/Kamailio and > best current practice for multiple SIP-servers with SRV-records > and authentication. This is not a question about Kamailio itself >

Re: [SR-Users] Kamailio 4.3.4: receive_msg(): no via found in reply

Thank you Olle! I’ll discuss this with our supplier. From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Olle E. Johansson Sent: Monday, April 4, 2016 10:16 AM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio 4.3.4: receive_msg(): no via found in re

[SR-Users] kamailio number translation and route

hi all; I want to use kamailio achieve: First phase: 1) inbound caller number translation; 2) outbound caller number translation; 3) inbound callee number translation; 4) outbound callee number tanslation; Example : Call Flow backend server -

Re: [SR-Users] Kamailio 4.3.4: receive_msg(): no via found in reply

> On 04 Apr 2016, at 10:02, Grant Bagdasarian wrote: > > Hi Daniel, > > You mentioned below about this issue not being complaint with the RFC. > Our supplier is telling us this is normal behavior and that they went through > the RFC and found this was normal behavior. > > If it’s not too mu

Re: [SR-Users] Kamailio as a presence server

Hi! Thanks, but actually with debug=3(4) in syslog last message I get is messages about hash tables in tm module. 2016-04-04 8:45 GMT+03:00 Daniel-Constantin Mierla : > Hello, > > can you run with debug=3 in config file and see if you get some hints from > syslog messages about what happens ? >

[SR-Users] Kamailio setting the reqeust URI

Hello,I'm using Asterisk to originate a call via Kamailio.Request-URI in SIP INVITE coming from Asterisk looks like thisBut my objective is to use Kamailio to forward the call to a remote endpoint. What header should I put in the SIP INVITE from Asterisk to Kamailio to conveythat Kamailio should

Re: [SR-Users] Load balancing traffic based on SIP URI

Thanks guys for your ideas, I finally think that I have an idea on how to do it. Nitesh Date: Fri, 1 Apr 2016 16:28:02 +0200 From: alberto.sagr...@avanzada7.com To: sr-users@lists.sip-router.org Subject: Re: [SR-Users] Load balancing traffic based on SIP URI Hi Federico. In mi case i have to forc

Re: [SR-Users] Kamailio as a presence server

Hi Igor, If Im understanding this correctly you have a separate Kamailio just to handle Presence. In this case you mention that a SUBscribe has been acknowledged by this Presence Kamailio but it does not do anything further!! What are the chances that this new Presence kamailio is separate from the

Re: [SR-Users] Load balancing traffic based on SIP URI

Great. You can share with us how did you did it :) Others would thank you also BR 2016-04-04 15:22 GMT+02:00 NITESH BANSAL : > Thanks guys for your ideas, I finally think that I have an idea on how to > do it. > > Nitesh > > -- > Date: Fri, 1 Apr 2016 16:28:02 +0200

Re: [SR-Users] kamailio number translation and route

Take a look to LCR module and DIALPLAN module BR 2016-04-04 6:43 GMT+02:00 li...@neulinx.com : > hi all; > > I want to use kamailio achieve: > > First phase: > > 1) inbound caller number translation; > 2) outbound caller number translation; > 3) inbound callee number tran

Re: [SR-Users] dispatcher [dispatch.c:1416]: ds_load_remove(): cannot find load for (from asterisk call-id)

Hi Are you using ds_select_dst("1", "10"); somewhere ? You have to check Asterisk is responding to OPTIONS as dispatcher considered UP when 200 OK from Asterisk or SIP Gateway is received.. and not if not received. I have used dispatcher but not with 10 option on algo... BR 2016-04-04 8:07

Re: [SR-Users] Using 302 redirects with Dispatcher Module

Just a point about using 302 redirect. It is expected to be handled by any SIP device.. but maybe you could find some (I found them on Spectralink DECT ) that could not handle 302 redirect and had to change way to use dispatching for that devices.. BR 2016-03-28 22:16 GMT+02:00 Marrold : > Well

Re: [SR-Users] dispatcher [dispatch.c:1416]: ds_load_remove(): cannot find load for (from asterisk call-id)

Hi Alberto, thanks for your reply yeah, i am using ds_select_dst("0","10") in this case because i use setid=0 in the dispatcher table as you can see in this route block route[TOASTERISK] { #!ifdef WITH_LOADBALANCE xlog("L_INFO","UBICAR: Forwarding non-registrar requests to Asterisk \n");

Re: [SR-Users] dispatcher [dispatch.c:1416]: ds_load_remove(): cannot find load for (from asterisk call-id)

hello. how can i block SIP messages by message content type. for exlamle : text/plain tanks in advance! From: sr-users [sr-users-boun...@lists.sip-router.org] on behalf of TEG AMJG [tega...@gmail.com] Sent: Tuesday, April 05, 2016 1:35 AM To: Kamailio (SER) - User

[SR-Users] Detecting calls with missing ACK (Lazy SIP scanners)

Hi, I have been running a couple of Asterisk honey pots to get a better understanding of the tools and methods potential hackers are using to exploit SIP servers. I have observed many attacks from the 'sipcli' user agent that don't send ACKs. At this stage I'm not sure what they're trying to ach

Re: [SR-Users] Detecting calls with missing ACK (Lazy SIP scanners)

I am assuming you're adding Record-Route to your initial INVITEs? If so, in the case of the answered call, it might be a defective UA that doesn't follow Record-Route for in-dialog messages (which include e2e ACK). In the case of the negative replies, a scanner might not waste additional reso

Re: [SR-Users] get_redirects with t_next_contacts

To not leave this thread forever unanswered, it turns out the above method using t_next_contacts() was working as it should be, but as every contact in the 302 redirect response was returning a 401, it tried all available contacts and then ran out of options which resulted in t_reply("408", "Reque

Re: [SR-Users] Detecting calls with missing ACK (Lazy SIP scanners)

Thanks for the speedy response as always. The current scenarios are taken from Asterisk only, Kamailio is not yet in front of it - so no record route headers. I imagine the ACKs aren't being sent due to it being a waste of resources as you suggest, but I'm interested in detecting these in Kamaili

Re: [SR-Users] dispatcher [dispatch.c:1416]: ds_load_remove(): cannot find load for (from asterisk call-id)

Hi again There is something that i am starting to realize and i think is the source of the problem. Since i am not sure, I think someone can clarify this to me. As i said the problem rises when i do a 2 call leg conversation. In which, since Asterisk is a B2BUA and creates another channel (which

Re: [SR-Users] pseudo variables in strings

I added pv_evalx() function in pv module which evaluates twice a string containing variables. No time to test it, so any feedback is appreciated. It will be also good to stress test it and check if there is a memory leak for pkg, as the result for first evaluation needs to be parsed again for other

Re: [SR-Users] pseudo variables in strings

Daniel-Constantin Mierla writes: > I added pv_evalx() function in pv module which evaluates twice a string > containing variables. No time to test it, so any feedback is > appreciated. It will be also good to stress test it and check if there > is a memory leak for pkg, as the result for first eva

Re: [SR-Users] Kamailio setting the reqeust URI

NITESH BANSAL wrote > Hello,I'm using Asterisk to originate a call via Kamailio.Request-URI in > SIP INVITE coming from Asterisk looks like this kamailio@.x > >But my objective is to use Kamailio to forward the call to a remote > endpoint. > What header should I put in the SIP INVITE from

[SR-Users] Kamailio Updates on VUC 588 - this week on Friday

Hello, The weekly VoIP Users Conference (VUC) #588 moderated by Randy Resnick will host a session for Kamailio Updates. Everyone from community is invited to join and actively participate to discussions. There are many ways to connect, from SIP to classic PSTN dial in as well as listening or watch