similar to: One Question:CLI dial cmd

Hello, I am trying to Bridge 2 E1 interfaces over a long distance link exactly the same way Redfone does. How can asterisk be configured to do that? Best regards Arinze Izukanne ___________________________________________________________ Want ideas for reducing your carbon footprint? Visit Yahoo! For Good http://uk.promotions.yahoo.com/forgood/environment.html -------------- next

hi I want to compress and trunk 2E1 capacity over a satellite SCPC link using asterisk. I am nw to asterisk and I need suggestions on how to implement this. Best regards Arinze --------------------------------- ALL-NEW Yahoo! Messenger - all new features - even more fun! -------------- next part -------------- An HTML attachment was scrubbed... URL:

[root@iw-asterisk zaptel-1.2.1]# make gcc -I/lib/modules/2.4.21-4.ELsmp/build/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/lib/modules/2.4.21-4.ELsmp/build/drivers/net -Wall -I. -Wstrict-prototypes -fomit-frame-pointer -I/lib/modules/2.4.21-4.ELsmp/build/drivers/net/wan -I/lib/modules/2.4.21-4.ELsmp/build/include/net -DMODVERSIONS -include

Hello, I'm trying to build a page system using a Dell Desktop PC optiplex 170L, My sound card is working fine under /dev/snd/ exten => s,1,Dial(Console/snd/,20,A(trek)) exten => s,2,Hangup But won't work! I get the following error [Apr 27 11:44:46] WARNING[2950]: chan_oss.c:377 find_desc: could not find <snd> [Apr 27 11:44:46] WARNING[2950]: chan_oss.c:850 oss_request:

I am using a extension to dial the console which has autoanswer enabled. I am getting a strange warning, has anyone seen this before? Nothing on Google, or Voip-Info [Jan 9 13:50:05] WARNING[5009]: chan_oss.c:1048 oss_request: oss_request ty <console> data 0x0xb7851e00 <dsp> << Call to device 'dsp' dnid '(null)' rdnis '(null)' on console from

Hi, I am trying to apply two different functions on on a vector as follow: a<-c(NA,1,2,3,-3,-4,-6) if a>0 I would like to raise it by the power of 2: 2^a and if the a<0 I would like to have the inverse value, i.e., -1/2^a. so I thought of doing it two steps: a[a>0]<-2^[a>0] a[a<0]<-(-1)/2^a[a<0] I got the following error Error: NAs are not allowed in subscripted

On Thu, 3 May 2018 08:40:37 +0100 Rowland Penny via samba <samba at lists.samba.org> wrote: > On Wed, 2 May 2018 19:21:46 -0300 > "Ethy H. Brito" <ethy.brito at inexo.com.br> wrote: > > This is where it is all going wrong, Your PDC isn't using LDAP, so > you will have to rely on the winbind 'rid' backend. The lines below are > wrong in several

Hi,   how to remove levels that have less than a specific number such as 2. i.e..   > f<-as.factor(c("a","b","a")) > f [1] a b a Levels: a b I want to remove level b because level b has less than 2. > f [1] a a Levels: a   [[alternative HTML version deleted]]

Evening folks, I''m new to RoR, and have found some very useful info on this site, thanks. I have a select box that I need to sort. <select name="people[peoplekind_id]"> <% @peoplekinds.each do |peoplekind| %> <option value="<%= peoplekind.id %>" <%= '' selected'' if peoplekind.id == @people.peoplekind_id %>> <%=

Apologies for the novice question, but this is likely beyond my google searching range.. I am trying to create a table from the summaries of a file. The 2 column data table looks like; Harry 335/335 Harry 124/506 Harry 124/506 Dick 133 Tom 335/335 Tom 335/335 with 1000 unique values in col 1, 300 unique in col 2. The 'summary' doesn't seem

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On Mon, 02 Jul 2018 12:38:54 +0200 Bernd Markgraf <bernd.markgraf at med.ovgu.de> wrote: > On Mon, 2018-07-02 at 11:30 +0100, Rowland Penny via samba wrote: > > > Provided that your users have a uidNumber attribute containing a > > unique number inside the '10000-999999' range AND Domain Users has a > > gidNumber attribute containing a number inside the same

Hi, I have a a vector and a data frame with two columns vec = c("C", "A", "B") dataDF = data.frame(A1 = c("B", "A", "C"), A2 = c(1,2,3)) I would like to sort the data frame by column A1 such that the order of elements in A1 is as the same as in vec. After the ordering, the data frame would be A1 A2 C

Please consider the following inputs: PrsnSerialno<-c(735,1147,2019,4131,4131,4217,4629,4822,4822,5979,5979,6128,6128,7004,7004, 7004,7004,7004,7438,7438,9402,9402,9402,10115,10115,11605,12693,12693,12693) PrsnAge<-c(59,48,42,24,24,89,60,43,47,57,56,76,76,66,70,14,7,3,62,62,30,10,7,20,21,50,53,44,29)

I want to split my whole dateset to training set and test set, building model in training set, and validate model using test set. Now, How can I split my dataset to them reasonally. Please give me a hand, It is better to give me some R code. and I see some ways like using SOM to project whole independents to 2-dimensions and find some to be training set and others are test set. like below. I

Can I map 2 tables into one class? Is there any pattern, e.g. using ActiveRecord aggregation? Thanks Szczepan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://wrath.rubyonrails.org/pipermail/rails/attachments/20060103/fa2aed7c/attachment.html

This is probably a really bad question to ask but here goes. Does asterisk work with any of the T1/E1 cards from SBE? I'm sure SBE is a competitor to Digium, but I have access to SBE cards and the linux driver. Just curious more than anything. Thanks. -- Michael J. Lynch What if the hokey pokey IS what it's all about -- author unknown

On Wed, 2 May 2018 21:46:52 +0100 Rowland Penny via samba <samba at lists.samba.org> wrote: > On Wed, 2 May 2018 17:12:30 -0300 > "Ethy H. Brito via samba" <samba at lists.samba.org> wrote: > > > > > Hi all > > > > Background: I am not a samba expert, not even close! > > > > I have a ancient samba 3 (Server role:

I'm trying to create an extension that will connect caller to asterisk sound card. I've followed the example at http://www.voip-info.org/tiki-index.php?page=Asterisk+tips+console. With no luck. What I get is: Jul 13 09:56:45 VERBOSE[1315]: -- Executing Dial("SIP/300-3bd6", "CONSOLE/dsp") in new stack Jul 13 09:56:45 WARNING[1315]: No channel type registered for

Hi, I am calling from a grandstream phone with g723 codec through * to iconnect. Incoming context as well as outgoing context set to g723.1 codec in *. Call get connected and I can talk. However I get the following warning, which scrolls on my screen until I hang-up. [root@asterisk sath]# cat g723.1 - Executing SetCallerID("SIP/-08122ae0", "1001") in new stack --