similar to: Asterisk removes ice lines in sdp when calling between webrtc clients

Hello, I'm trying to get calls working between websocket clients and sip clients. For clients I have sip.js based clients on chrome, Zoipers and a Grandstream phone. Challenge here is I'd like to have Kamailio and rtpengine to handle the bridging between different rtp profiles but Asterisk changes them in the sdp bodies along the way. I'm using Asterisk 11.11.0. Is there a way to

Hello, I'd appreciate your comments on the following problem I'm having, please forgive me if this is something obvious, I've been scratching my head on this for a while: I have Asterisk+Kamailio setup where I'm currently testing inbound calls from outside. I have both webrtc and sip clients, where webrtc peers are defined according to sip.js instructions (

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

Greetings, I've noticed a problem that might originate from my Asterisk configuration, could use a hand in sorting it out. Problem is a 488 response from Asterisk whenever it gets RTP/SAVPF profile in the SDP. My current setup has Asterisk Kamailio realtime integration, and Kamailio uses dispatcher to route calls for Asterisk to handle. Now I have only one Asterisk, on the same machine as

Hi All, I have configured WebRTC according to the install document. The clients register correctly. I'm use SIPjs. The clients are able to send messages to the server. The SIP debug shows the messages being received. However I'm stumped for directions on how to route the messages between the clients. Asterisk 11.11.0 Here is my client sip config: [1060] type=friend username=1060 ; The

Dne 11.8.2015 v 12:18 Joshua Colp napsal(a): > Vinicius Fontes wrote: >> I'm having the same issue! The difference in my case is Asterisk server >> has a public IPv4 and the browser is behind a single NAT. >> >> I'm forwarding my configuration below (which I posted previously on >> asterisk-users). >> >> How can we debug ICE negotiation? >

I am having trouble getting Google Chrome to accept a WebRTC call coming from Asterisk, even though Firefox can (now) accept the same call without issue. My setup is as follows: Server: CentOS 7 x86_64 (Elastix 4 RC) with IP: 10.1.0.4 192.168.5.146 asterisk-11.21.0 patched to work around https://issues.asterisk.org/jira/browse/ASTERISK-25659 openssl-1.0.1e-51.el7_2.2.x86_64 [root at elx4 ~]#

I am using Asterisk 11.3.0 and just updated Nightly to 24.0a1 (2013-06017) and get a SIP 488 Not Acceptable Here response. I have no problems using the same Asterisk configuration and the same page to make a call from Chrome. I have seen other people post a similar issue, but I have not seen a solution. If someone with good knowledge of this issue were to respond with "this is a known

hi, is there someone with working asterisk13+chan_sip+SIP.js/sipml5 ? or is chan_pjsip better supported? or the recommended way for asterisk is use respoke.io? my problem with asterisk13+chan_sip+sipml5(the same problem is with SIP.js) chan_sip.c:10496 process_sdp: Can't provide secure audio requested in SDP offer " sip.conf (important parts) [vr1a882] ... nat=force_rport,comedia

Hello after I have upgraded from Asterisk 12 to asterisk-certified-13.8-cert1 none of my realtime SIP peers (saved in MySQL DB) register anymore. [Aug 15 22:03:43] NOTICE[30098]: chan_sip.c:28451 handle_request_register: Registration from '<sip:testacc77 at 178.19.90.240>' failed for '78.119.140.190:5076' - Wrong password [Aug 15 22:04:13] NOTICE[30098]:

I'm having the same issue! The difference in my case is Asterisk server has a public IPv4 and the browser is behind a single NAT. I'm forwarding my configuration below (which I posted previously on asterisk-users). How can we debug ICE negotiation? ---------- Forwarded message ---------- From: Vinicius Fontes <vinicius at aittelecom.com.br> Date: 2015-07-27 13:54 GMT-03:00

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, i'm facing strange problem asterisk13.5 + chan_sip wss transport + SIPML5 1.5.230 person1 to person3 are behind different NATs audio devices double checked call from person1(chrome) to person2(chrome) works call from person1(chrome) to person 3(chrome) - no audio on both side (RTP flowing only in one direction) call from person2(chrome) to person 3(chrome) - no audio on both side

Dear all I solved a problem of customised labels on strips and boxes in bwplot by this construction. > bbb <- bwplot(zavoj ~ typmleti | pu) > bbb$condlevels$pu <- c("Povrchov? ?prava", "Bez PU") > bbb$x.limits <- c("Mleto", "Mleto a s?tov?no", "Nemleto") > bbb but I wonder if some other easy option exist. Let say something

On 05/12/14 16:46, Olli Heiskanen wrote: > INVITE that Asterisk (at port 5070) receives: > PU.BL.IC.IP:5060 > PU.BL.IC.IP:5070: SIP, length: 1046 > INVITE sip:660 at testers.com > <mailto:sip%3A660 at testers.com>;transport=UDP SIP/2.0 > Record-Route: <sip:PU.BL.IC.IP;lr=on;ftag=41030177> > Via: SIP/2.0/UDP >

Dear all, curve(x^3*(1-x)^7, from = 0, to = 1) works as expected but, omitting the "xlim" or the "to" and "from" arguments and calling "curve" more than once: par(mfrow = c(2,2)) for (i in 1:4) curve(x^3*(1-x)^7) gives an expected (al least to me) result. Note also that a "pu" object is returned by curve > pu [1] -0.1802445 1.1802445

hi all, I have a Class, Client, which has_many projects (Project class). the projects depend on a session_id variable, however. So the question is, how can I do the following: has_many :projects, :finder_sql => "SELECT p.* FROM projects p INNER JOIN projects_users pu ON pu.project_id = p.id WHERE pu.user_id = #{session[:user_id]}" The problem is that I need to filter a

R Help on 'chisq.test' states that "if 'simulate.p.value' is 'TRUE', the p-value is computed by Monte Carlo simulation with 'B' replicates. In the contingency table case this is done by random sampling from the set of all contingency tables with given marginals, and works only if the marginals are positive... In the

Hi, I noticed that my Samba 4 DC isn't OK, because the are differences between the data storaged int he Schema on my Windows Server 2008 (isn't R2) DC and Samba 4 DC. This way, I performed several tests on my servers as shown below. Follow the results of command repadmin in the Windows Server 2008: C:\Windows\system32>repadmin /showreps /verbose Default-First-Site-Name\WIN-DC1

Folks, I am having major performance issue with my Erlang application running on openstack KVM hypervisor and after so many test i found something wrong with my KVM guest CPU Topology This is KVM host - http://paste.openstack.org/show/790120/ This is KVM guest - http://paste.openstack.org/show/790121/ If you carefully observe output of both host and guest you can see guest machine threads has