similar to: Optimization of call from server 1 to 2 and then back to 1

Greetings, I'm using asterisk to connect our three locations together with a sort of inter-company auto attendant connected like this: PBX (fxs) <-> Sipura 3k (fxo) <-> Asterisk <-IAX-> remote asterisk It works like this: Person picks up their phone and dials a number to get to the auto attendant (I don't have any FXO ports available on our PBX to do it the

On Mon, 27 Mar 2006, Andrew Lenharth wrote: >> Another option is to build an optimized build with assertions on. Do to >> local demand, I added a build option 'make ENABLE_OPTIMIZED=1 >> ENABLE_ASSERTIONS=1' that provides this. > > How does this compare size and performance wise with a debug build or a > release build? I haven't done any scientific

Hi folks, I'm experiencing some problems with sound through phpAGI ... What I'm trying to do is a menu, doing some database lookups and so ... But sometimes the sound become too choppy ... just sometimes .. like 1 of 5 calls ... but is a big percentage ... And I have my current menu on the dialplan that I have no problems with it ... I'm using .gsm for both but different

I would like to setup an option in my digital receptionist that callers can select to hear a read back of their Caller ID. It would be something like, "the number you are calling from is...". I think I can reuse the festival script that is built in, but ideally this could be accomplished without using festival because Allison's voice is so much more pleasant. I'm just a few

Hi. I've searched the wiki and the newsgroup to no avail. We have a couple of grandstream phones and purchased seceral copies of the G729 codec for use with them. We registered the codec (first time) on a testing machine and it worked. We registered it on the production machine (second time) and it worked. Then the production system crashed. After reinstallation we had to

Hi All, for a vicidial server which uses only voip, which is the minimum telephony card which would provide the required clock timing source for conferences to work properly ? Maybe the Digium TDM410PLF card without any daughter card would do the job ? Thank you very much for supporting. Have a nice week-end, Mike

I think using different settings is generally (and this this case too) a bad idea because it makes things more complicated. All developers know how to build LLVM. Most (new/casual) users expect a certain behaviour and they will judge LLVM on subjective grounds. If linking/performance suffers they will continue looking for whatever gives them that extra bit of performance. People like me

Hi all, i have the following setup PSTN -> routing server (asterisk 1.6.2.11) -> IAX -> callcenter asterisk 1.6.2.9 -> SIP -> agent Does work quit fine - then agent does have the abibility to transfer a call to a third party - the agent can initiate the transfer over a web interface - it does generate a asterisk manager atxfer request... So agent does initiate transfer - call

On Mon, 2006-03-27 at 11:47, Chris Lattner wrote: > On Mon, 27 Mar 2006, John Criswell wrote: > > One consideration to weigh is that a debug build of LLVM provides users with > > more diagnostic information to submit with bug reports (since many bugs are > > caught by assertions, which print a readable stack trace). The tradeoff > > seems to be faster and smaller

Hello, Suppose customer A calls attendant. CallerID of A is displayed at the attendant. But, when attendant does a consulted transfer to, let's say, B, the callerID of attendant is displayed at B. When the consulted transfer is succesful, the callerid of attendant is STILL displayed at B. Is it possible to, after a successful transfer change the callerid of the attendant in the callerid of

Hi All, does Digium provide a service support for a compatibility question about their PRI hardware ? Thanks and have a nice day.

On Mon, 27 Mar 2006, Eric van Riet Paap wrote: > I think using different settings is generally (and this this case too) a bad > idea because it makes things more complicated. All developers know how to > build LLVM. Most (new/casual) users expect a certain behaviour and they will > judge LLVM on subjective grounds. If linking/performance suffers they will > continue looking for

Hello all, I was wondering if anyone knew how to construct a multiple line graph on R, where there are 2 (or more) sets of data points plotted against some x axis of data, and you can draw a line on the graph connecting each set of data points. For example: A B C D 0.6566 2.1185 1.2320 5 0.647 2.0865 1.2325 10 0.6532

Hello dear list. I am currently working on a Automatic attendant, and the core things work, but I think the loop function isn't working as expected. I am testing this environment: a sip internal call from 301 to 501. The setup here is when 301 calls 501, and 301 doesn't enter an extension, it will go in loop, 3 times, and then hangup...Can't get that working. Could someone please

Hi, I'm a novice of R. I want to read the following table into R: names mpg cyl disp hp drat Mazda RX4 21.0 6 160.0 110 3.90 Mazda RX4 Wag 21.0 6 160.0 110 3.90 The command I used is: > test <- read.table(file.choose(),header=T) The result is: Error in read.table(file.choose(), header = T) : more columns than column names

On Mon, 27 Mar 2006, John Criswell wrote: > One consideration to weigh is that a debug build of LLVM provides users with > more diagnostic information to submit with bug reports (since many bugs are > caught by assertions, which print a readable stack trace). The tradeoff > seems to be faster and smaller LLVM tools (optimized build) vs. better > diagnostic information for bug

Here is the call flow: [ivr-incoming] exten => s,1,LookupCIDName exten => s,2,DigitTimeout(2) exten => s,3,ResponseTimeout(10) exten => s,4,Wait(1) exten => s,5,Background(custom/ivr-incoming) exten => 1,1,Background(pls-wait-connect-call) exten => 1,2,Dial(${RINGPHONENUMBERS},20,r) exten => 1,3,Voicemail,u${VMBOX} exten => 1,4,Hangup Running * 1.0.5. The calling party

Asterisk 1.6 CentOS 5.0 All - I'd like to offer my users the ability to hangup a call by pressing **. I'm using an attendant, so when ** is dialled I'd like processing to return to the attendant so the user can place a subsequent call. I have setup features.conf to include: [featuremap] disconnect => ** My Dial command looks like this:

I need to avoid MOH on my asterisk box, so i need to have a ringing tone when attendant transfer is made, or a call is on hold.. Is there any way to do that. I did not see a simple way to do that. Regards

One of our clients is using a Grandstream GXP2000 with an attendant console. We have used the same phone with past clients successfully however this particular operator processes around 200 calls a hours and the GXP2000 for sure does not like the quick line shuffling and call volume. We get the following problems randomly: 1. menu stops working 2. transfer key stops working 3. Line 1 LED gets