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Dears Any one knows a standalone voip transcoder software name,not an ip pbx. What I want is to transcode the incoming sip calls from g711 to g723 or ilbc or g729 ..... and forward it to a media gateway .. Regards Khaled chehab ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by

Hi, anyone knows an open OMA DM tool that would be able to configure Nokia phones (mainly the sip-stuff of the e-series) for use with asterisk? Anything open i could find was the device manager from funambol, which was last updated in 2006 :-( Regards, Jay -------------- next part -------------- An HTML attachment was scrubbed... URL:

...ernel), would be greatly appreciated. There are many reasons why using libguestfs is a better solution for us in the long run. Thanks. -Jeff ---------------- Original Message --------------- Subject: Re: [Libguestfs] getting guestfs_rsync_out to work From: "Jeff Brower" <jbrower@signalogic.com> Date: Mon, February 9, 2015 3:39 pm To: "Richard W.M. Jones" <rjones@redhat.com> Cc: libguestfs@redhat.com ------------------------------------------------- Richard- > On Mon, Feb 09, 2015 at 08:27:22AM -0600, Jeff Brower wrote: >> Yes we know that...

All- Recently an Asterisk server we host was hacked and used to route some unauthorized calls. We have since improved our security measures, including installation of fail2ban. The interesting thing is the way in which this was discovered. The unauthorized calls were occurring intermittently last Thurs evening thru Sat morning. On Sat morning, some of our employees were attempting to log-in

I have asked a GSM operator in my country if he can route a number or a short code to my asterisk server via SIP (since they dont give DIDs in my country) the operator said they do not support SIP, they have no way of converting GSM calls to SIP to then send them to me. I would like to know what is needed from the operator side to do this, what kind of material is needed, or what can be done from

We're having an odd issue with codec negotiation from one of our SIP providers. Here's the basic situation. We receive an invite from them advertising support for G711, G729, and G723. In our response, we send back that we support G711 and G729. In about half the cases, this results in no problems, with audio being encoded with G711. The other half of the time, they send us a second

Hello, http://www.voip-info.org/wiki/view/Asterisk+Letting+SIP+clients+connect+directly The link above indicates that it is possible to setup RTP streams to directly flow between endpoints and completely bypass Asterisk. I would like to know if this configuration would work when, a) both endpoints are behind NAT, and/or b) both endpoints don't support same codecs with media flowing

All- My apologies in advance if this is an obvious question and I've missed it on Asterisk FAQs and how-to's... Can Asterisk operate with just an FXO card? By that I mean, no network connection (none, no local network). I want to build some type of user interface to go off-hook, route FXO port voice lines to a headset (or speaker and microphone), perform an automated conversation (e.g.

Hello folks, I am trying to add HDLC module support in the menuconfig. However, I am facing difficulties in rebuilding the kernel and modules. My main objective is to get the HDLC supported in kernel 2.6.9. I have a CentOS 4.4 with kernel 2.6.9.42. I need help in finding the best way to add the HDLC support whether through getting an RPM package for it or refering me to the right procedure

Richard- > On Mon, Feb 09, 2015 at 08:27:22AM -0600, Jeff Brower wrote: >> Yes we know that it doesn't know. We'd be ok to re-launch the image >> to pick up changes except for the time it takes (if we could get >> into 1 to 10 msec range it would be ok). > > There's no feasible way to start a Linux kernel in under about 1-2 > seconds. It just takes

Hello, We are in the process of debugging a voice quality issue for a client of ours that is a VoIP services provider. The client uses a softphone that runs on a pjsip stack. When placing a call using the softphone, it negotiates the use of G729 codec with the remote endpoint (ptime = 20ms). The endpoint transmits RTP packets with encoded G729 payload. VAD/DTX is enabled. We see that the last

Hello, ------------- -------- --- -------- |Sip Softphone|-------|Internet|--------|F.W|-----|Asterisk| ------------- -------- --- -------- IP addresses: a.b.c.d q.w.e.r The SIP softphone(x-lite) is configured to register with the asterisk server through port 9090 (Domain q.w.e.r:9090).Firewall(F.W) is setup as the

Richard- >>> Per this page >>> >>> https://rwmj.wordpress.com/2013/04/22/using-rsync-with-libguestfs/ >>> >>> We are trying to get the libguestfs API guestfs_rsync_out to work. We’ve tried combinations similar to: >>> >>> guestfs_rsync_out(g, “/HostShared”, “rsync://root@10.0.1.90:2999/backup/ -av”, -1) >>> >>>

Richard- > On Mon, Feb 09, 2015 at 08:01:31AM -0600, Jeff Brower wrote: >> Richard- >> >> > On Sun, Feb 08, 2015 at 12:11:37PM -0600, Jeff Brower wrote: >> >> With continuous loop testing, what we found is that we have to shut >> >> down and re-launch the image handle to see changes on the Win7 live >> >> guest. Unfortunately image

Richard- > On Sun, Feb 08, 2015 at 12:11:37PM -0600, Jeff Brower wrote: >> With continuous loop testing, what we found is that we have to shut >> down and re-launch the image handle to see changes on the Win7 live >> guest. Unfortunately image re-launch takes time, 3-5 sec (the image >> size is 50 GByte). I'm assuming this is because libguestfs makes an >>

All- Per this page https://rwmj.wordpress.com/2013/04/22/using-rsync-with-libguestfs/ We are trying to get the libguestfs API guestfs_rsync_out to work. We’ve tried combinations similar to: guestfs_rsync_out(g, “/HostShared”, “rsync://root@10.0.1.90:2999/backup/ -av”, -1) surrounding it with APIs as in the recipe examples, for example replacing libguestfs_cat(), with no luck. In the