similar to: ICE

Hello, you can have a look on Homer http://sipcapture.org/ regards On 27/09/2016 10:39, Gholamreza Sabery wrote: > Hello, > > For service monitoring you can use tools like sipsak in combination > with Zabix or Zenoss. Also using Zenoss or Zabix you can monitor the > health of your servers. This way you have both top-down and bottom-up > monitoring. For monitoring call

Hello all, The question isn't directly related to Asterisk, but I'm looking for recommendations for a monitoring tool to monitor the health of Asterisk instances running in Production. Ideally, the tool should be able to generate monitoring traffic (OPTIONS ping or INVITE), use the response/no response from Asterisk to store the health of an Asterisk instance running somewhere in the DB.

Hi, I am getting *Can't provide secure audio requested in SDP offer* with sipml5 client hosted on my local system [1060] ; This will be WebRTC client type=friend username=1060 ; The Auth user for SIP.js host=dynamic ; Allows any host to register secret=sameer ; The SIP Password for SIP.js encryption=yes ; Tell Asterisk to use encryption for this peer avpf=yes ; Tell Asterisk to use AVPF

Hi, I have a large dataframe than I'm writing functions to explore, and to reduce cut and paste I'm trying to write a function that does a subset and then a plot. Firstly, I can write a wrapper around a plot: plotwithfits <- function(formula, data, xylabels=c('','')) { xyplot(formula, data, panel = function(x,y, ...) { panel.xyplot(x,y,

Hello all! I want to estimate parameters in a MIMIC model. I have one latent variable (ksi), four reflexive indicators (y1, y2, y3 and y4) and four formative indicators (x1, x2, x3, x4). Is there a way to do it in R? I know there is the SEM library, but it seems not to be possible to specify formative indicators, that is, observed exogenous variables which causes the latent variable. Thanks,

Hi, I've tested wine 1.0rc5 today with fully up-to-date Debian testing and latest NVIDIA Driver and Guld Wars and here my report : -Incredibly slow (run over-smoothly on the same system with windows XP) -Absolutlly no shaders displayed (char over-bright, horrible effects, ect...) -Texture problem in loggin screen (i can see "junction" of texture in the background) I've tested

I would like to discuss the following commit: https://github.com/gsliepen/tinc/commit/4a0b9981513059755b9fd15b38fc198f46a0d6f2 ("Determine peer's reflexive address and port when exchanging keys") This is a great feature as it basically allows peers to do UDP Hole Punching (via MTU probes) even when both are having their source ports rewritten by a NAT, which is extremely useful.

Hello, I have a problem connecting to one of my computers using tinc 1.0.x on Windows. It used to work, now it suddenly stopped (and nothing changed :) ) We have a server with a known ip address and port forwarding set. All computer connect to this server. I can ping from my computer (laptophenk) to the server and some other computers but not to jeffrey2015. When I set tincd to -D -d4 I get ( I

Full_Name: Elliott Forney Version: 2.9.2 OS: Linux, Fedora 10 Submission from: (NULL) (129.82.47.235) The following code creates a list of functions that are lexically closed over a single argument. If a print statement is included then each function in the list evaluates to a different value. If the print statement is not included then each function evaluates to something different, as

Hello, I noticed a strange thing while testing my Asterisk-Kamailio Realtime setup. In an INVITE the From and To headers contain the same number when calling through a Realtime integration setup. This happens when the INVITE leaves Asterisk. Can you guys tell me what might be causing this? I have 660 at testers.com as a websocket client and 700 at testers.com (caller) using a Zoiper client (db

I have configured support for ice in sip.conf, and made a connection with motif to jingle, but does not work for me [Jul 15 12:03:32] ERROR[21758]: chan_motif.c:1955 jingle_interpret_ice_udp_transport: Received ICE-UDP transport information on session '8b4hdffbt37vg' but ICE support not available -- Executing [s at xmpp-in:1] NoOp("Motif/allan-ce76", " llamada de

Hello list when trying to set up webRTC communications with sipjs client package (tried 0.7.0, 0.10.0 and 0.19.0), I see in the asterisk debug log-file the following : DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP c=IN IP4 99.88.77.66... OK. DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtcp:9 IN IP4 0.0.0.0... UNSUPPORTED OR FAILED.

Hello, Where can I find doc about PJSIP's ice_support parameter ? Do you need to configure things elsewhere in Asterisk config files (rtp.conf, PJSIP transport sections, ...) to make ICE work properly ? I'm asking because, if I'm not mistaken, STUN requires setting a STUN server so I think ICE most probably, should also require settings some public resources. Best regards

Hi asterisk users, I am in need of information about how to configure the sip.conf and extension.conf for subscribers to support the dialog event package rfc 4235. I am using asterisk 1.6.0.1 version. The below are the configuration of sip.conf and extension.conf files which I have done. I have three subscribers as one from my application(App) and other are x-lite1 and

Hi, I want to be able to bridge two sip channels using direct RTP between my endpoints (Audio IP : not local) but without using reinvites. So I set up my asterisk sip endpoints as follows: [test1] type=friend host=dynamic username=test1 dtmfmode=info context=test_rtp allow=all canreinvite=no directrtpsetup=yes [test2] type=friend host=dynamic username=test2 dtmfmode=info context=test_rtp

Thanks Joshua for replying ! What would you advise : - leaving STUN address empty, in rtp.conf, as "STUN is not required for ICE" - configure it with one public STUN (I'm using stun.voip.ovh.net for this but I don't know how this server really works) Cheers Le mar. 27 oct. 2020 à 09:53, Joshua C. Colp <jcolp at sangoma.com> a écrit : > On Tue, Oct 27, 2020 at 5:35 AM

Hi asterisk users, I am in need of information about how to configure the sip.conf and extension.conf for subscribers to support the dialog event package rfc 4235. I am using asterisk 1.6.0.1 version. The below are the configuration of sip.conf and extension.conf files which I have done. I have three subscribers as one from my application(App) and other are x-lite1 and

As an update to this: http://connect.microsoft.com/VisualStudio/feedback/details/769222/cl-exe-ice-when-building-llvm-trunk-at-o2 Microsoft has reproduced the ICE, given a workaround, and is planning a fix for a future MSVC release. I know not a lot of people are building with VS, but it's nice to know. The workaround involves marking a single function with attribute((noinline)) and is

Hi All, I've got a strange problem with Centos 5.9 64bit running Ice Cast, soon as the server hit 1019 connections it dies. Set the fs-file-max to 65535 and added in hard & soft no files to 655535 yet still not joy. (Ice Cast listeners are set to over 5K connections) Has anyone come across this problem? Kind Regards, Mike

Hello, I have a problem with a call between 2 webrtc clients. Asterisk removes the ice-related lines from the sdp when it sends the INVITE out, and the called webrtc client rejects the INVITE due to the missing ice lines. Both webrtc clients are defined exactly the same way, same values in all fields except the number of the peer. There's probably something I've changed that causes this