similar to: ICE Candidate collision on dualstack hosts?

On 13 Oct 2015, at 22:31, Heiko Schlittermann <hs at schlittermann.de> wrote: > > Hi, > > still using 2.2.9, I've two directors, and these directors > use both IPv4/IPv6 addresses. > > `host directors.<domain>` returns one A and AAA for each > of the two directors: > > directors.<domain> has address 149.x.y.96 (director1)

Hi, still using 2.2.9, I've two directors, and these directors use both IPv4/IPv6 addresses. `host directors.<domain>` returns one A and AAA for each of the two directors: directors.<domain> has address 149.x.y.96 (director1) directors.<domain> has address 149.x.y.97 (director2) directors.<domain> has IPv6 address

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 ~]#

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 (

Hey i have an interesting topic to discuss here. The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support . the problems that i faced with this is the following and i hope i could get an advise here. asterisk 13 vanilla version has some issues marking the video packets this complain

Hi, At my provider (xs4all) I've got an ipv6 tunnel working. Now I would like to distribute ipv6 via the tinc tunnel. My tinc.conf: ------------ Name=server AddressFamily=ipv4 Device=/dev/net/tun PrivateKeyFile=/etc/tinc/fvhglobalnet/rsa_key.priv GraphDumpFile=|/usr/bin/dot -Tpng -o /var/www/htdocs.keetweej.vanheusden.com/stats/tinc-fvh-network-graph.png Mode=switch KeyExpire=299

Hi, I'm using Asterisk 13.x and have defined a pjsip TCP IPv6 transport: [transport-tcp-ipv6] type=transport protocol=tcp bind=[2001:1234:5678:abcd::2]:5060 I also have an IPv4 version of that: [transport-tcp-ipv4] type=transport protocol=tcp bind=10.75.22.8:5060 I've then configured an endpoint to use it: [outgoing] type = endpoint context = default dtmf_mode = none disallow = all

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 thank you for your answer. I don't understand how there are many tutorials and examples on the web where every time the outcome is a working setup. Very strange I feel now after my personal experience with Asterisk 11 and webRTC. You also say Asterisk 13. How about Asterisk 12 then ?? Kind regards. On 10-08-16 21:53, Matt Fredrickson wrote: > I don't see an ice-ufrag or

For those that were interested I have attached the kamailio.cfg which we have working with Kamailio 4.2.1 and Asterisk 1.8.23/32. Specifically, the following yum packages: kamailio.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms kamailio-auth-ephemeral.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms kamailio-bdb.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms

Can anyone recommend a good WebRTC phone to use with Asterisk? I do not mind if it is commercial or open source. Customers are starting to ask for web solutions and we need to start testing. -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez +52 (55)9116-91161

On 10.05.2016 18:57, ????????? ???????? wrote: >> this seems to be relevant in chroot environments; >> >> as I noticed when configuring the DDNS-feature, that this is a little >> bit >> weired, when running in a chroot environment; I saw the >> recommendation not >> to use a chroot in the man-page and removed bind-chroot and then the >> zone

For the client: JSSIP and Sipml5. If you are going to be coding something up yourself I like the JSSIP 0.5.x javascript interfaces. If you are simply going to use a pre-canned one then sipml5 works pretty well and remembers your settings in localstorage. I haven't used any closed source versions since the above works really well for us. For the server: If you are using Asterisk 1.8

On Wed, Mar 4, 2015 at 12:47 AM, Jarrod Cuzens <jarrod at mogl.com> wrote: > For those that were interested I have attached the kamailio.cfg which we > have working with Kamailio 4.2.1 and Asterisk 1.8.23/32. Specifically, the > following yum packages: > > kamailio.x86_64 4.2.1-4.1 > @home_kamailio_v4.2.x-rpms > kamailio-auth-ephemeral.x86_64

On 10.05.2016 21:36, ????????? ???????? wrote: >>> I'm also using ddns and have my zone files in >>> /var/named/chroot/var/named/dynamic. >> are you using DDNS in DualStack (IPv4 and IPv6 together) or do you >> have only DHCP or DHCPv6 and not both? > > IPv4 only. > if a host has IPv4 only or IPv6 only this works fine, but when a host has both -

[It seems the last time this came up was in march, under the heading of "Sony Cybershot should be in hardware notes". This message is intended mainly to document what I've managed to track down.] The m500s still will not sync with pilot-link 0.11.7 in -STABLE. An easily triggerable panic is another issue [1]. The pilot-link code first opens /dev/ugenX and then switches to

>>> TL;DR: please try the patch out and report if it causes "Did not receive >>> identification string" log messages. I believe it does not. Aw crap. My homegrown anti-dos tool for ssh looks for either DNRIS or if logging is verbose enough a connection that didn't result in a login. I give the attacker a few tries and whitelist any successful candidate so I

> > Asterisk needs urgently to push the RTP engine to the Kernel, away from > userland, like professional and commercial softwares do. I measured the > cost of passing call from a public IP to a private IP, like typically a > Session Border Controller may do. In Asterisk, ulaw, no transcoding, it > takes 1.7% of a 3 Ghz core. If the packets where flowing through the > kernel,

Hello I am trying to set up webRTC video calls from my Chrome webbrowser (Fedora) to my Chrome webbrowser (Windows 10). There is local video input (I can see myself), but never video on the receiving side. This is the case in both directions (so it makes no difference which peer is calling which peer). Both webRTC SIP peers have opus and H264 codec in their peer definition :   Video

I am developing apps using ARI which need suppression of DTMF tones in the audio, and I have been told (back in December) that asterisk depends on SIP providers to suppress DTMF tones in the audio stream. Having sorted out my ARI code to suppress DTMF as I wanted, it turns out that SIP providers are not very good at doing that suppression (leaving audible clicks, or failing to suppress the tones