similar to: SIP over 3G Mobile Network using NAT

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

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

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

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:

Hi, Is there any softphone or TAPI plug-in that allows one to dial from a web page? As you may know, Skype has a mechanism that converts phone numbers on a web page to a click-to-dial application. I'd like to use this but on a normal softphone (Bria, Xlite, other). Regards, Mike -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I'm trying to use presence with PJSIP and I have a "issue". I created correctly hint priorities like: exten => 1000,hint,PJSIP/1000 exten => 1001,hint,PJSIP/1001 Extension 1000 can subscribe extension 1001 y vice-versa. The problem is when the extension 1000 make or receive a call. In the softphone where the extension is present on buddy list, the extension appear

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

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

Hi all, my name is Arnau Bria and I''m a sys admin in a center where we must deal with hundred hosts. We''re currently working with quattor, but it''s too complex for our purposes, so I''m looking for new admin tool. I''ve been playing with CFengine for few days (2 or 3) and I''ve seen some limitations that makes me thing that CFE is not our

What are the recommended settings to successfully implement VoIP over 3G/4G data connection. Assume we are talking about using Polycom phones, and the 3G/4G data connection comes from a Cradlepoint router that is plugged in with AC power and has high gain antennas. The device will be stationary, so we will not have to worry about tower handoff?s breaking the connection. This will be for fixed

> Could you please describe what is the benefit of that? Currently there are two ways to provide DISPFlagDefinition, via bool and SPFlag, I would like to make it only via SPFlags, it will be NFC and it will make the changes in parser simpler for moving five flags from from DIFlags to DISPFlags. Currently parser checks the presence of SPFlags to see if the definition is present in bool or spflag

Hello, Thanks for prompt reply. Motivation, In case of debugging of bi-endian dwarf and value known at runtime some of the raw values needs to be byte swapped to change the endianity. We have a llvm based compiler which generates such dwarf, TAG_variable marked as big endian and debugging on little endian host using lldb. The listed operator on dwarf issues does that, one operator instead of

Yes, removing the support for isLocal, isDefinition fields completely from ll files, currently the LLParser still parses it. I want to remove it and update the all the ll files which still uses it. Also the metadata read will support old format, no changes in that. so if ll file has isLocal and isDefinition it will result in parser error. But the bitcode read will work as usual. - Chirag.

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

Hello Paul, Thanks for the reply. Yes, I am only looking for dwarf support at the moment and planning to support both PLI/COBOL decimal types. Also thanks for the suggestion, you are right as it is going to be rare cases, so it will be better to implement a separate subclass to avoid memory overhead for others. >> (Somebody is actively working on scaled binary operations, although they

could anyone help explaining what does this error mean? i get this error when make a video/ audio call from X-lite to Bria prof. phone rtp.c:1739 ast_rtp_read: Unknown RTP codec 126 received from '10.0.0.26' Gres -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, Trying to do my first WebRTC. Using stock asterisk 1.13.0. I setup the asterisk according to the recipe on the wiki, but cannot get it to work. Dialing from sipml5 on chrome I get no sound, regular bria on standard sip works. My network setup by the way: I am working from a cable modem, I created the test setup at digital ocean. From my laptop I also have a direct VPN connection to the

Hi, I have an error when i use rake test, plz help akyen@ubuntu:~/Desktop/p$ rake test WARNING: Global access to Rake DSL methods is deprecated. Please include ... Rake::DSL into classes and modules which use the Rake DSL methods. WARNING: DSL method Pictographs::Application#task called at /usr/local/ lib/ruby/gems/1.9.1/gems/railties-3.0.7/lib/rails/application.rb: 215:in

Hello all, There is one enhancement request open for dwarfv5, http://dwarfstd.org/ShowIssue.php?issue=191107.1 The request is for addition of dwarf expression operator to swap the top of the dwarf stack, the response seems positive but it may take some time till v6. I like to add that operator as vendor extension but I am not sure how to go about it for llvm/lldb. Currently I am using it as

Hello, In regard to the review request https://reviews.llvm.org/D74470, I am trying to move five of the DIFlags to DISPFlag for the moment namely DIFlagExplicit, DIFlagPrototyped, DIFlagNoReturn, DIFlagThunk, DIFlagAllCallsDescribed. The llvm ir format for DISubprogram currently has backword compatibility where the isLocal, isDefinition, virtuality, isOptimized and SPFlags are mutually exclusive.