similar to: Streaming Audio Player for SBC environment?

Hello, I would like to know the difference between encrypt the rtp and signaling between two asterisks, or putting an SBC in front of each Asterisk pbx. I'm trying to understand whether an SBC could fit an Asterisk deployment Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

*Hi All, I have starting to reading About SBC and found one artical reagding SBC and they gives a solutions like this. i want to know is this true in realtime sceanario while we think of an big implementation and is it possible with cloud computing. i have found from http://www.smartvox.co.uk/products_gateways_explained.htm Asterisk as a Session Border Controller* Equip the Asterisk server

Can anyone help me with the term that SBC uses to refer to disconnect supervision? I have an Adit 600 channel bank which has helped improve the disconnect detection time down to about 8 seconds. This is still causing some issues in particular with call progress on we are having a few disconnects while calls are in session. I have talked both to some local phone contractors and SBC directly and

Hi all, I have been using asterisk to do such kind of thing, But I must admitt, this is not 100 % conveniant (Mainly because Asterisk isn't a SIP Proxy). I just wanted to know if you knew/used some kind of SBC or packages which would deal both with SIP AND RTP ! SER/OpenSER woulc be a good SIP Proxy ... but then how to deal with RTP ? Any tip, info greatly welcome ! Thanks, JM

Hi, I am new to the world of Asterisk PBX. I have been given the task of coming up with a solution to our office phone situation. From what I have been reading, Asterisk sounds like it could be ideal. However, most of the information I am finding is focused on the VoIP aspects, which is something we want to use for our remote employees, but the first hurdle is integrating Asterisk with the

Hi, I'm trying to get my * server connected to a softswitch through an SBC. I get the following error when * trys to register. Got SIP response 423 "Interval Too Brief" back from xxx.xxx.xxx.xxx Jan 20 12:43:54 NOTICE[2138]: chan_sip.c:5473 sip_reg_timeout: -- Registration for '1234561234@xxx.xxx.xxx.xxx ' timed out, trying again (Attempt #9) Is there something I can

I have what my SBC/ATT representative calls a "Supertrunk," but he can't tell me the specifics I need to know to configure Asterisk to work with it. By fiddling about, I've come up with the "almost working" configuration below. This works except that it takes about 4 seconds from when the console says the line is answered until it plays my prompt. Watching the

Hi, This is a bit OT. If I have a phone behind NAT and the phone registers via an SBC can I set NAT=NO and canreinvite=yes in asterisk or will I still have issues ? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080313/41d639ee/attachment.htm

This is a really strange one, IMO. I just bought an SBC 900Mhz two-line portable phone. My intent was to use it with one of my TDM20 lines with asterisk, and have the other line be a PSTN line. But it appears as if for some reason I can't intermingle the two types of lines. If I hook *both* lines from the TDM20 to the unit it works just fine. If I hook *both* of my PSTN lines to the

Anyone out there have Verizon or SBC as the local exchange company in your area? I'm in the process of negotiations for additional interconnect agreements with these carriers (we're a wireless carrier). I'm looking for Asterisk folks who want a T1 (or several) for a PSTN gateway and would like to "network" their Asterisk boxes. This can take place at little or no cost

SBC installed a T1 ADTSe (Digital Trunking Service Enhanced) e&m wink start with 24 1 way trunks. The CO says they dial pulse DP the seven digit dnis number. The channels work now but take long time to answer and get these messages repeating until I guess the CO stops Pulse dialing the number. Nov 23 19:08:58 WARNING[1827865]: chan_zap.c:4718 ss_thread: getdtmf on channel 8: Operation now

Hi David, If cost is an issue then, yes you can do all that (the speex at least) with the eZDSP USB stick. Only limitation is a slow emulator that is integrated with USB, and so there is no JTAG output to use with other emulators. If you don't have an emulator anyway then this is an advantage. Cheers, M -------------- next part -------------- An HTML attachment was scrubbed... URL:

Can anyone help me with the term that SBC uses to refer to disconnect supervision? I have an Adit 600 channel bank which has helped improve the disconnect detection time down to about 8 seconds. This is still causing some issues in particular with call progress enabled in * we are having a few disconnects while calls are in session (about 2 reported in first 5 days of use). I have talked both to

On Wed, Aug 28, 2013 at 10:00 AM, Lucas Stach <dev at lynxeye.de> wrote: > MSIs were only problematic on some old, broken chipsets. But now that we > already see systems where PCI legacy interrupts are somewhat flaky, it's > really time to move to MSIs. > > Signed-off-by: Lucas Stach <dev at lynxeye.de> > --- > drivers/gpu/drm/nouveau/core/include/subdev/mc.h

I want to shade the area under the curve of the standard normal density. Specifically color to the left of -2 and on. How might i go about doing this? Thanks -- View this message in context: http://n4.nabble.com/Shade-area-under-curve-tp1586439p1586439.html Sent from the R help mailing list archive at Nabble.com.

Am Dienstag, den 23.12.2014, 18:39 +0800 schrieb Vince Hsu: > The Tegra124 and later Tegra SoCs have a sepatate rail gating register > to enable/disable the clamp. The original function > tegra_powergate_remove_clamping() is not sufficient for the enable > function. So add a new function which is dedicated to the GPU rail > gating. Also don't refer to the powergate ID since the

Greetings: I would like comments on this example and after fixing it up, I need help from someone who has access to insert this in R's help page for plotmath. I uploaded a drawing http://pj.freefaculty.org/R/Normal-2009.pdf that is created by the following code http://pj.freefaculty.org/R/Normal1_2009_plotmathExample.R This will be a good addition to the plotmath help page/example.

From: Deepak Goyal <dgoyal at nvidia.com> - Maps PMU firmware into PMU virtual memory. - Copy bootloader into PMU memory and start it. - Allow the PMU to interact with HOST via interrupts. PMU after successful configurations (to follow after this patch) will: 1.Autonomously power gate graphics engine when not in use.It will save us a lot of power. 2.Provide better way to scale frequencies

I am trying to get to the bottom of audio clicks, pops, dropouts with my Asterisk server. These occur even when the system is under minimal load (e.g. 1 Zap device in a queue being played music on hold) and occurs with both Zap and Sip devices so isn't network related. The audio problems occur at the same time on all channels and seems to be when Asterisk "gets busy" and uses

Hi.. I am about to replace my aging Nortel Venture system with an Asterisk system and 6 Polycom IP 501 phones, and a couple sipura 841's for less used areas. We have 3 phone lines here. One is SBC, one Vonage, and one Voipjet... One hangup is that I can't figure out how to pass through a voicemail waiting indication from SBC. This is important because my wife and her family all