similar to: Incoming SRTP call not working with Bria iPhone Edition

I have a pair of Asterisk 11.5.1 servers operating as a load balanced cluster, with DNS SRV records set up to weight them 60/40 relative to each other (both at priority 0). The back-end is MySQL Realtime, and everything works pretty well with the Cisco SPA phones & ATAs that represent the majority of my endpoints. I recently tried to add an iPhone with the Bria softphone application, to

Hi! I use Bria and eyebeam and it seems that asterisk doesn't send RCTP keepalives when a SIP channel is on hold. This is a known issue as is described here: http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+Bria This gets very annoying because very often people are put on hold longer than 30 seconds (the phone's default.) In a company with more than 100 soft phones

Hello! I tried using sendtext() in the Asterisk dialplan to send a SIP MESSAGE to Bria or a recent Eyebeam on my mac. I know it used to work, but right now I get "100 trying" and nothing else from the softphone. Anyone that knows what's going on here? Thanks, /O

Hi, In the aftermath of Digium's and Counterpath's Bria for Asterisk announcement, we're happy to chat with Todd Carothers, Counterpath Product Manager today at 1 PM EST. For more info, http://vuc.me Join us on IRC #vuc on Freenode.net or use the web client at http://vuc.me/irc Call in starting at around 12 Noon EST: sip:200901 at login.zipdx.com Hear you there! /r

I don't think that will fix my problem. The hints on the individual user extensions (101, 102, 103 and 104 below) are working just fine. sccp.conf example of 1 user: [devices] type = 7970 description = User1 tzoffset = -6 autologin = 101,401 speeddial = 102,User2,102@wct-internal speeddial = 103,User3,103@wct-internal speeddial = 104,User4,104@wct-internal device => SEP000F90CEF9D3

Hi all, We're testing TLS and SRTP on Asterisk 1.8.10.0 and have it working with a commerical (not self-sign) AlphaSSL wildcard (GlobalSign) using Blink Lite 1.6.2 as per https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial We've tested with Bria on an iPhone and that doesn't recognised the commercial CA (GlobalSign Root CA). On a Yealink 28P with V60/V61 is registers

On Tue, Mar 10, 2015 at 5:22 PM, Toufic Khreish (Gmail) <toufic.khreish at gmail.com> wrote: > Thank you, I needed a starting point to start my post. > > 1. Asterisk 12.8.1 (IAX2 voice issues) no video issues. > Voice issues on IAX2 Trunks, All extensions are SIP. > The IAX2 trunks on Asterisk 12.8.1 produces only one error out of : iax2 > set debug trunk on >

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi list, I am trying to setup a DomU with linux-2.6.18-xen.hg in order to test Remus on a Debian Squeeze 6.0 / Xen 4.0.1 (2.6.32-5-xen-amd64) Dom0. After applying several patches to the Debian Squeeze Xen packages, I have built the linux-2.6.18-xen.hg, but when I start my DomU, I have got this error message : Error: (2, ''Invalid

We have 4 employees and we're running Cisco 7970 phones. Each phone has a unique SCCP line configured (in the autologin area of the sccp.conf file) for each employee. We have hints set up in the extension.conf file like the following: exten => 101,hint,SCCP/101 exten => 102,hint,SCCP/102 exten => 103,hint,SCCP/103 exten => 104,hint,SCCP/104 We have speeddial= lines set

Hello Matthew, I have compiled Asterisk 13.2 with the following compiler Flags enabled: DON'T_OPTIMIZE DEBUG THREADS BETTER_BACKTRACES My asterisk is running with the asterisk_script: root 24048 39.4 2.4 128564 50640 pts/1 Sl 00:02 2:21 /usr/sbin/asterisk -f -vvvg -c core show locks ======================================================================= === 13.2.0 ===

Thank you, I needed a starting point to start my post. 1. Asterisk 12.8.1 (IAX2 voice issues) no video issues. Voice issues on IAX2 Trunks, All extensions are SIP. The IAX2 trunks on Asterisk 12.8.1 produces only one error out of : iax2 set debug trunk on [2015-03-10 16:28:42] WARNING[9614][C-0000000b]: chan_iax2.c:1793 compress_subclass: Can't compress subclass 2097217 On the box running

Hello, When I receive a call to my CounterPath Bria from Asterisk 1.4.18.1 and I place the call on hold, the call is dropped after 30 seconds. It looks like there is no RTCP/RTP sent to the client from Asterisk while on hold (music on hold playing to caller) thus client disconnects the call. During this time, I get the following messages in the CLI: NOTICE[24194] rtp.c: Unknown RTP codec 126

Hi, I am running asterisk 1.8.5.0 and have compiled in the srtp module All but Snom phones are working. I have set the srtp tag on the snoms to 80 and RTP/SAVP to mandatory and they worked for a few hours. This morning all snoms are reporting this when trying to make a call (this is snom calling snom). ---------snip------------------ == Using SIP RTP CoS mark 5 -- Executing [10000 at

Dear, Kindly guide with the 2 issues mentioned below *#1* - *Host unreachable 0 last qualify 0 (only in 3G**)* I am trying to use SIP client over 3G. It registers and call can be initiated from the client but it can't receive call; cause *asterisk sever *marks it as unreachable immediately after registration. "[2014-10-08 14:32:47] NOTICE[1610]: chan_sip.c:29596 sip_poke_noanswer:

31.08.2023 20:10, Aaron de Bruyn via samba wrote: > It doesn't appear to be affecting anything in our environment, but a vendor recently insisted on "seeing" the security properties on a share. > > When we right-click on a particular share, go to properties and then click on the security tab, explorer crashes and reloads. > > I tested on a few Windows 10 and 11 boxes

Thanks--I didn't notice the Debian packages were so far behind. I'll try building a newer package and I'll test it out. Thanks, -A On Wed, Nov 11, 2020 at 7:30 PM Andrew Bartlett <abartlet at samba.org> wrote: > On Wed, 2020-11-11 at 19:19 -0800, Aaron C. de Bruyn via samba wrote: > > I wanted to do a little playtesting with Samba as a domain > > controller.

Thanks Andrew, but we checked for that. Firing up dnsmgmt.msc shows no entries with those computer names. -A On Mon, Nov 6, 2023 at 11:34?AM Andrew Bartlett <abartlet at samba.org> wrote: > On Mon, 2023-11-06 at 10:02 -0800, Aaron C. de Bruyn via samba wrote: > > DNS is suddenly not working properly for some machines. > > > > > > > > We had a bunch of

Hi, I've been doing some testing on moving from 1.2 to 1.4 and one issue I've encountered is re-transmits whenever an INVITE is cancelled. I have a stateless SIP proxy in fron of my asterisk servers (all it does is direct requests to one asteisk server or another) and the re-transmits do not occur on 1.2.17 which is the current verion I have in use on my production servers. The

On Mon, 2023-11-06 at 10:02 -0800, Aaron C. de Bruyn via samba wrote: > DNS is suddenly not working properly for some machines. > > > > We had a bunch of machines that were joined to the domain, but the > computer > > name was wrong. > > > > To fix this, we unjoined the machines and deleted the computer > accounts out > > of Samba (because

Hi list! I'm trying to make a SIP test call from Bria and/or 3CXPhone from a PC behind NAT. From Bria/3CXPhone I connect to an Asterisk 11.25.0 server on the internet at 100.100.94.210 with a SIP account "3333" created in sip.conf: [3333] type=friend secret=something host=dynamic nat=yes qualify=no disallow=all allow=alaw allow=ulaw canreinvite=no context=voipin I dial +1234