similar to: Bria softphone registration problems on DNS SRV cluster

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Everybody, I am experiencing some troubles with my Bria iPhone Edition (v. 1.2.8 build 5312, on iOS 4.2.1 iPhone 3G) and Asterisk 1.8.3.2 + TLS/SRTP on LAN (without NAT). With 2 computer clients (Blink, one on Mac, one on Windows/Linux),9i can have a very fine secure conversation in both directions. When I want to do the same with my iPhone,

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've been tasked with finding and implementing a CDR/Queue analyzer to provide information to management about the call center's performance. My Google-fu seems to be returning a lot of things that are more or less abandoned projects. Does anyone have any recommended solutions for a CentOS 6 / Asterisk 10 "vanilla" server? Not opposed to something commercial, provided it

Hello, I am using asterisk 14.2 and PJSIP, with TLS transport. I?m sure I?m doing something wrong here .. In 2 distinct softphone clients (Bria and Groundwire), I am able to register successfully, and place a SIP call, with no certificate warnings. But shortly after I place that first call and hang up, I receive a certificate name mismatch error in the softphone, the error presenting me

I have an Asterisk 11.5 system, using SIP Realtime and operating as a ITSP. One of my customer's endpoints is a NetVanta 7100 PBX system that has a SIP trunk connection to my Asterisk box. The NV 7100 has a public IP on it that doesn't have any NAT between it and my Asterisk system. When the customer transfers a call from one handset to a voicemail box, the NV 7100 sends a RE-INVITE to

Issue is what subject says. Here is the background. Version: 11.11.0 Topology: Asterisk Box at our Data Center behind Cisco Firewall. Everything works fine from remote offices over a VPN. Issue is sales team would like to connect up to our Asterisk box remotely (offnet). Common enough solution, I'm guessing. So, I've opened all the correct holes on the firewall and hammered out

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 >

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

Hello, I've just read in [1] about SIP MESSAGE addition to both chan_pjsip and ConfBridge. It seems very interesting addition as it brings the capability to mix voice, video and text in conferencing. On an other hand, there are some softphones (Zoiper, Bria, ...) that tout voice, video and chat capability. Though Enhanced Messaging solution described in [1] seems more attractive to me in the

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:

We have a recently deployed server with a new TDM2400 card that will not put dialtone or audio on FXS ports after the physical server restarts (though they will ring if called, there's just no audio on the line if the phone at the other end picks up). The symptom can be resolved by stopping Asterisk, restarting DAHDI, and then restarting Asterisk. So far, this has happened on both times the

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

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

I'm trying to set up a way that our users can send an XMPP message to Asterisk (unsolicited) to request information, such as voicemail status or the like. No matter what I set for the dialplan, I'm only seeing Asterisk execute the s,1 priority in the context defined in xmpp.conf for incoming messages, and then the "call" hangs up without executing further instructions. Anything

I've been spending the day trying to get IMAP_STORAGE on my test box, to evaluate for production, but I'm having no luck getting uw-imap to build. I've tried installing it from an upstream package, but Asterisk still isn't finding it to compile -with-imap. My google searches have turned up very little for documentation on dependencies, gotchas, etc for either item, so I'm

Hi, I would be curious to play with an Android phone with Wifi-only capability. My plan is to install Bria on it and see if it could be used within a couple of WiFi access points, as a high-end wireless phone. Which handset would you recommend ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: