similar to: VoIP over 3G/4G Data

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:

Has anyone been able to configure Asterisk to work over 3G? I bought Nokia Cell Phones just for that purpose and they register fine over WiFi and 3G but the quality is just not good enough and sometimes the call just disconnects. I have Allow as: ilbc gsm ulaw alaw -- Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336 Phone/Fax (855) 760-COOP (2667)

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

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

From: Jean-Christian de Rivaz <jc at eclis.ch> I suspect that some instructions about how to use the diag/mbr was not updated when the source file was renamed to handoff.S. Here is a simple proposition to fix that. Jean-Christian de Rivaz --- diag/mbr/README | 4 ++-- diag/mbr/handoff.S | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/diag/mbr/README

Hi all, I''ve noticed that facter version superior from epel do not display lsbmajdistrelease fact: # facter lsbmajdistrelease 5 # rpm -qa|grep facter facter-1.5.5-1.el5 # cat /etc/redhat-release Red Hat Enterprise Linux Server release 5.5 (Tikanga) # facter lsbmajdistrelease # cat /etc/redhat-release Red Hat Enterprise Linux Server release 5.5 (Tikanga) # rpm -qa|grep facter

I recently compiled asterisk 13.2.0 on an RK3288 , I am suspecting problems with the format H264, Asterisk 12.8.1 compiled on the same hardware is behaving very well for the same format H264 Problem of asterisk 12.8.1 is IAX2 trunk bad voice quality. Could someone investigate the problem of Asterisk 13 with video support on H264 ? Thank you. -------------- next part -------------- An

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, 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:

I've got a dilema. For months, I've been planning and implementing a large project involving vorbis, ices2, icecast2 and wifi networking. The plan is to stream live audio from 10 Austin nightclubs during the SXSW music festival this March. Well yesterday, it was made plain to me that I must also provide mp3 streams, since "it's an mp3 world." I'm upset about this

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

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:

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

Hi Everyone, Which paid or unpaid commercial plugin is available out there for Asterisk that would do Outlook contacts pop-up that is proven to work great with MS Outlook 2007 and Asterisk 1.6. It would be a bonus to do Dialout as well through the Outlook. Thanks, Bruce -------------- next part -------------- An HTML attachment was scrubbed... URL:

On Thu, Feb 2, 2017 at 4:43 PM, Phil Turmel <philip at turmel.org> wrote: > On 02/01/2017 10:40 AM, Phil Turmel wrote: >> Hi All, >> >> I've been running Gentoo on a ZBook with great success for a couple years, >> but I've been stymied in my attempts to implement SecureBoot by an >> apparent problem with efifb to nouveaufb handoff, but only when

Hi all, from http://docs.puppetlabs.com/guides/language_tutorial.html this syntax should be ok: file { [ ''foo'', ''bar'', ''foobar'' ]: owner => root, group => root, mode => 600, } I''ve this in my code: file { [ ''/software/atlas'' , ''/software/cms'' ,

Is ROR a good fit if I want to edit and expand on a developed script like RoundCube( http://roundcube.net/ )? I''m planing to edit it for my business as an order/transcript application. If anyone is interested this link is what I want to make. http://handoff.wordpress.com/

Hi I have a new ASUS A6Tc laptop with 1.6 GHz AMD Turion dual core processor. I get a strange error: <4>0000:00:0b.0 OHCI: BIOS handoff failed (BIOS bug ?) 000007b4 <4>0000:00:0b.1 EHCI: BIOS handoff failed (BIOS bug ?) 01010001 This only occurs when booting into the xen kernel, not without xen. Can someone tell me why and how it can be fixed? Art

I want to collocate an * box somewhere, where better than where voicepulse chose to put their servers? They probably did their homework and selected someplace where good handoff to the pstn can be found, right/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050520/9f5975b8/attachment.htm