similar to: 24port or higher fxs

hi guys, do you have any suggestions on how to connect 2 TE410P via E1? (for simulation and testing purposes) asterisk1 --> TE410P ----> ? ---------> ? ---->TE410P -->asterisk2 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030907/698cd499/attachment.htm

I use cisco ATA 186 ( Version: v2.16 ) with sip protocol and asterisk as pbx. I need feature called as 'three way calling' or 'transfer with consultation'. Registering,calling and 'blind transfer' work fine. Is this feature provided by sip clients or by asterisk itself ? What I have to configure in ATA and what keys I have to press on my phones ? Three way calling is

Newbie to Asterisk I've been looking around for a little while, can't seem to find some sample configs for using a Cisco 7206 as a gateway. The below link is an initial plan of an Asterisk solution that may replace our Cisco Call Manager 3.1/ IPCC / IVR setup. We currently have all of the hardware below. Just take a peak and see if there is anything that is off base. I don't know

hi guys, have anybody tried using audiocodes sip fxs against asterisk? how's the device fairing? ~kelvin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030725/ae4b2f25/attachment.htm

I have a question on Asterisk and whether it will work with the following design. Install ASTERISK on the external side of the Network. Purchase an AudioCodes 4/8 port Analog Fx0 gateway. So far everything seems straight forward. Here is the twist. The company currently has Cisco Call Manager 3.3 which does not support SIP Trunking. But it does have a VG248. I would like to place 4 lines

Hello, Did anybody successfully tried upgrade Adtran TSU 600 to firmware which is working properly with T100P and asterisk ? B.

i wrote previous about a setup i thought might work with asterisk and the tsu-600. no one replied, so i thought i would ask if anyone is using a tsu-600 with asterisk and if so how do you have it setup ??

Can I say that RPART is a modified algo of CART and PARTY a modified of CHAID? Thanks. ---- Chua Siang Li Consultant - Operations Research Acceval Pte Ltd Tel: 6297 8740 Email: siang.li.chua at acceval-intl.com Website: www.acceval-intl.com This message and any attachments (the "message"...{{dropped:12}}

Hello there. I have fitted a rpart model. > rpartModel <- rpart(y~., data=data.frame(y=y,x=x),method="class", ....) and can use rpart$where to find out the terminal nodes that each observations belongs. Now, I have a set of new data and used predict.rpart which seems to give only the predicted value with no information similar to rpart$where. May I know how

Thanks Dragos, I assume I will be setting those parameters during initialization of encoder right? Question is, if connection gets too lossy, how will opus adapt to it? Can it automatically shift bitrate down to minimize impact? Mark from IRC suggests that the app has to be aware of the losses and change it on the fly. Has anybody on the list tried this? Kelvin Chua On Wed, Mar 4, 2015 at 5:53

The time has come to review the current Contributor and Core contributor grants for ZFS. Since all of the ZFS core contributors grants are set to expire on 02-24-2009 we need to renew the members that are still contributing at core contributor levels. We should also add some new members to both Contributor and Core contributor levels. First the current list of Core contributors: Bill

hi guys, have any of you guys managed to use msn messenger to authenticate with asterisk using its DNS name? based on my experience with other sip proxies, msn will not authenticate if it sees a different realm than the realm of the client. one workaround i did was to edit the chan_sip.c to send a pre-defined realm, and also edit the Contact: field. after this, asterisk would send a 401 to the

Hello, > I have the following entries, among others, in my smb.conf file: > ===== cut here ===== > logon path = \\%L\profiles\%U > [profiles] > comment = User Profiles > path = /usr/home/profiles > public = yes > writable = yes > browseable = yes > ===== cut here ===== > So, I expect 'net use z: /home' on a win95 machine to map z: to >

Just starting to do the research on this one....I've got a customer who is showing interest in replacing any older Panasonic unit providing service to 96 tip/ring lines from a single PRI. Does anyone have any recent experience with a decent (as in, plays nice with * and has a reasonable per-port cost) channel bank or similar? Mediatrix only goes up to 24 port, as far as I can tell, which

I am using libopus for my implementation. I wonder if anybody in the list have any experience on how to make libopus dynamically adjust its bitrate? On Mar 3, 2015 10:42 PM, "Benjamin Schwartz" <benjamin.m.schwartz at gmail.com> wrote: > It sounds like your software isn't adjusting the opus bitrate in response > to network conditions. For example, many WebRTC

People on the list tend to think you can't make many cards work on a regular desktop. If you're willing to wait a couple of week I might have an answer for you. _____ From: Robert Augustyn [mailto:augustynr@yahoo.com] Sent: Saturday, December 11, 2004 7:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Will Adtran TSU 600 work with *?

Hello there. Is there any function in R that can do cluster on a set of data that has both categorical and numerical variables? thanks. siangli

It is possible to connect ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank to Asterisk ? Somebody offered me that hardware, but I do not know if thats good hardware for Asterisk. rgs, Bartosz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030820/4a9e4608/attachment.htm

I have just received a T100P and an Adtran TSU 600 in the mail. I seem to be having a problem with the T100P card. So far I have done the following: vi zaptel.conf fxoks=1-22 fxsks=23-24 ... vi zapata.conf ... signalling=fxo_ks ... channel => 1-22 ... signalling=fxs_ks ... channel => 23-24 I then run modprobe zaptel modprobe wct1xxp ztcfg -vv There are no errors to report. In

Hi. I need some advises on how to use R to find pi (i is the index) with the following objective function and constraint: max (sum i)[ f(ai, bi, pi) * g(ci, di, pi) * Di ] s.t. (sum i)[ f(ai, bi, pi) * Di * pi] / (sum i)[ f(ai, bi, pi) * Di ] <= constant f and g are diffentiable. So, I am thinking of optim with method = "BFGS"? But wonder how to include the