similar to: dtmf Caller-id detection before first ring

Dear all, I am from Saudi Arabiya and I am using digium hardware with asterisk 1.6. When I am executing following AMI originate API. Orginate start to execute extenstion without knowing of PSTN(FXO) channel is ringing. Any one can help me to resolve this issue ? Action: Originate Channel: Dahdi/g0/2923878 Context: outbound-ivr Exten: 1234 Priority: 1 ActionID: ABC45678901234567890

Hi All; If I need to set the opermode to King Saudi Arabia, what the name I have to use? For example, to set it for kuwait then I use opermode=KUWAIT. So what will be for Saudi Arabia? Any advise? Regards Bilal

Hi, I've seen this USB product from Sangoma : http://www.sangoma.com/products_and_solutions/hardware/analog_telephony/usb_fxo.html Is it working ok ? Is it easy to integrate it with Asterisk ? How would you rate your experience with it ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello R gurus, I just spent my first weekend wrestling with R, but so far have come up empty handed. I have a dataset that represents file downloads; it has 4 dimensions: date, filename, email, and country. (sample data below) My first goal is to get an idea of the frequency of repeated downloads. Let me explain that. Some people tend to download multiple times, e.g. if the download fails

I am sure that quite a lot of people would like to have Caller ID working with their X100P and TDM400P cards outside of USA. Judging from previous threads this is just a matter of implementing this support in the software driver! So, I was thinking, if we get together and put few $(USA DOLLARS) into a basket, we could then ask Digium to actually properly implement Caller ID for non USA

Hello, Talking with Soren Sratje about Caller ID in Brazil, we compare ours DTMF tones captured by ztmonitor. wcfxo correctly recognize the "DTMF CLIP" and asterisk shot the AST_STATE_PRERING correctly. But the DTMF tones are not reconized. In the chan_zap.c, the code: if (f->frametype == AST_FRAME_DTMF) { (...) Does not occurs because the frametype is always reconized as voice

Hi all, I am using asterisk-1.6.2.10. I changed say.conf script for customized number reading. In the extension.conf: -------------------------- [number-to-voice] exten => 8765,1,playback(num:344345,say) exten => 8765,n,hangup It executes corresponding say.conf script and produces good results for me. but when I write it in agi does not working. Here is agi debug output from asterisk.

Jose Luis Column `Country` joining factors with different levels, coercing to character vector common_col_names <- intersect(names(sub_kei), names(knowledge)) > common_col_names [1] "Country" "Year" nrow(sub_kei) <- 132 nrow(knowledge) <- 3864 Tiene distinto numero de pais como de año, en el sub_kei aparecen 5 años y en el otro dataset (knowledge) datos anuales

Hello I purchased an AEX800 card to replace the ageing cheap channel bank/T1 card solution a few months ago, assuming that it would be a more robust solution for my small scale phone system. However, it appears to be anything but that. Originally implemented as a XEN dom-u virtual machine on a large server class machine, using PCI passthrough to pass the AEX800 and a small older TDM400, then

Dear R-project team, I have tried but could not install the R statistical package (http://cran.ms.unimelb.edu.au/ ) even after the help of my institute's IT personnel. The setup file could not be downloaded. The latest file R-2.12.2.tar.gz<http://cran.ms.unimelb.edu.au/src/base/R-2/R-2.12.2.tar.gz> does not start installation wizard. Kindly extend the technical support. Best regards.

Hi All, I have installed asterisk 1.6.2.8 Dahdi: 2.4.0 Digium card: Digium, Inc. Wildcard AEX800 8-port analog card I have configured this card properly and it is working for calls too, But there is issue of one way audio on this outbound routes only, My voice is going to outside pstn number but i am not able to get their audio, I have disabled firewall, selinux is also off. If I am applying

The following is a message to be sent to the President of the United States of America. Although we may not be able to do a great deal from where we are, but for the people of America just knowing we care and feel their sadness will help. Please put your name on the following list and send it to all you know and who care. If you are the 100th name and every 100th there on could you please also

I have a AEX800 PCI Express card - sort of a TDM800 with PCI-Express. (or AEX844 - 4FXS & 4FXO) I downloaded Asterisk Now - and have got this loaded on a new motherboard (Intel with 3 PCI, 3 PCI Express - etc). (Downside on PCI-Express is the physical support the express slot gives (very little) compared with an 'old' standard PCI slot!!!) With only this card in the box....

hi group , i am working on dahdi_monitor for measuring voice quality , so i want to know that on which data i can tell that this PRI lines are working properly, is there any measurement on basis of that i can make MOS. i am working from last 2-3 days but i only get idea about making .raw file and making .wav file and visulal mode of RX and TX of PRI line. what i want is measurement of voice

I am running Asterisk 1.16.2.13, dahdi 2.4.0 and libpri 1.4.11.4 on an HP ML110 G6 using Ubuntu Linux 10.04 LTS. I have two Digium TE121 single T1 port cards and a Digium AEX800 8-port FXS card. All PCI Express cards. Co-workers are hearing hissing sounds on some calls, and I am getting IRQ errors when running "dahdi show status". I see that sharing IRQs for Digium cards isn't

maybe usefull for displaying CALLED party name when dialing.... I'm remember, that this feature was planned to add to asterisk, any progress? PJ New and Changed Information Release 8.0(0) includes the following new and enhanced features: ?Remote-Party ID support has been added for incoming INVITE and UPDATE requests and 18x and 200 responses.

Hello, I need to convert the dahdi_monitor output to dBm0, so I can measure Echo Return Loss in dB. I've read a formula that calculate S(k) in ITU-T G.168 recommendation, where S(k) is the signal level in dBm0. Can I use this formula to convert it? If yes, what value should I use to the number of samples if I want to convert a single output from dahdi_monitor? -- Thanks in advance, Gustavo

Hello all, I am running R 1.5.0 on a Pentium 4 1.5 GHz machine running Windows XP Professional. update.packages() doesn't work for me. For a while it goes quiet an then I get the following message: trying URL `http://cran.r-project.org/bin/windows/contrib/PACKAGES' unable to connect to 'cran.r-project.org'. Error in download.file(url = paste(contriburl, "PACKAGES",

Hi, Using 1.4.26.1 & DAHDI 2.2.0.2, FSK VMWI devices off a TDM840 work fine. With 1.6.1.[45] & same DAHDI, instead of the FSK spill I get a line polarity reversal. Stutter dialtone is generated as expected. Has anyone else seen this? Is there anything special I need to do for 1.6.1 to make FSK MWI work? Thanks, --Barry

I am studying about echo cancellation in asterisk and I want to use the numeric information from dahdi_monitor verbose for my research. Unfortunately, I couldn't find anything about the unit of measurement used in this tool. Which unit is used to measure the signal level? -------------- next part -------------- An HTML attachment was scrubbed... URL: