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Hello, Does asterisk supports VXML? Couldn't find much resource on that on google and wiki. Thanks Foong

Hello I see it is possible to buy Flash Disks up to 4GB now. Has anyone any experience of building an Asterisk system with a flash disk as the only storage device? Any brands you recommend? Is 2 or 4GB enough for an Asterisk installation? Typically how many MB is required for voicemail recording files for say a 10 user system? What about voicemail - I suppose files could be emailed and

Hi list! Do anybody has success histories about using the Beronet app_saynumber application? For those that are hearing about for the first time, it?s a piece of software that enables the use of another language in say_number commands in asterisk dialplan or AGI scripts. Link to download: http://www.beronet.com/download/app_saynumber-beta-rc1.tar.gz I?m trying to compile it in

Hi I am trying to get a SNOM 360 to monitor other extensions i.e. when someone makes a call to/from another extension, one of the LED's on the SNOM 360 will change state. I am using 1.0.9/bristuff-8l. I have 2 extensions - 2001 is a SNOM 190, 2002 is a 360 - both are running the latest firmware. (3.60i for 190, 4.0 for 360). I have read all the relevant articles on the wiki on

Hi list! I?m connecting a Brasilian voip (- gvt.com.br -) provider through my asterisk box and using the register command from sip.conf. What I can?t understand is why my unit sends a new registration message every minute! And every time my asterisk box sends a registration, it gots a sucessful response, and shows de message: "Oct 14 16:48:22 NOTICE[4090]: chan_sip.c:8742

Hi list. I have a fine running Ser+Asterisk environment and have just installed ASTCC. It?s working fine either, including its caller-id authentication feature (the one we pass the card-number as CALLERID variable and number-to-dial as EXTEN variable). The issue, a great one, is that when the credit is about one minute to end, the ASTCC prompt gets into the call, says that "you have one

I would like to know if somebody else experienced that: sip show peers will always drop the first character of the 11th line. while sip show peers like [0-9,a-z] will not drop any character. Can anybody test this, please? bye Ronald Wiplinger

hi all, I have a problem with @ 1.2.4 on debian kernel 2.6.8-2-386.: -- Executing Dial("SIP/2003-bbae", "zap/2/03460816149|30|t") in new stack Feb 14 17:25:25 WARNING[4246]: channel.c:2535 ast_request: No channel type registered for 'zap' Feb 14 17:25:25 NOTICE[4246]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'zap' (cause 66 - Channel

I have A@H 2.1, running * 1.2.1. I am trying to put information into the userfield with SetCDRUserField and AppendCDRUserField. However, the field is never populated in the cdr - I've checked the csv files and the MySQL asteriskcdrdb table. The field is defined in the MySQL table, but is always empty. The csv files that get created don't have a userfield at all, that is, there

This is a wierd one. Can't figure it out. I have an SPA-3000 at the house handling my incoming line. It's setup to direct the incoming call to asterisk. Works great 99% of the time. A few times a day, I'm getting calls that ring once internally and are then hungup. I managed to get a detailed log [1] of what's happening today and it looks to me that the SPA is acting wierd.

Hi, everyone: I'm in the processing of deciding what IP phones we should use with our Asterisk server, and I wanted to get feedback from the user community on the quality, reliability and ease of operation of Snom phones. What do you have to say about these phones? Are there other phones you'd suggest along with or instead of Snom? Thanks, -Stephen-

Hello, I am writing a program based on Astersik Manager which needs to put calls on hold and to redirect them to others extensions. I haven't funded any action able to do this. Is there a way to put calls on hold using Asterisk Manager Actions? Amaury ?

Hi, We're using Asterisk with Digium TE110P card for the PSTN E1 interface. Our PRI is enabled with detecting the "connected-party-number" feature. When an OUTBOUND call is made to a phone, the PRI will send back an ISDN messages containing the "connected-number" and we can use that information to validate the extension user is calling the party that he/she is

Hi, I would like to find out; Anyone using Asterisk Business Edition? If yes, could you give us a brief description of what type of solution you use it for? What made you to choose the Business Edition over the Open Source? What advantage/disadvantage you have found? What's the level of stability in terms of number of calls, dropped calls in conversation, echo

I'm running Asterisk CVS-v1-0-06/06/05-17:29:02 built by root@............... on a i686 running Linux. I just spent some time in testing this. I tested the local and IAX2 trunks. Both worked flawlessly. Any comments? Darren Wiebe darren@aleph-com.net

I download and installed ASTCC over the weekend and I am having an issue where the INUSE flag will not get set back to 0 if the user drops a call while the balance is being played. All other times it seems to reset the flag correctly. I have tried both AGI and DeadAGI with the same results. Those of you using it for a while, how did you get around this? Just for fun this is all I am doing in

In my shell environment, I have set a path to R_PROFILE_USER. The file, Rprofile.R, is a collection of small hacks. I want to build rstudio-server from source. Best is to $ unset R_PROFILE_USER before. Unfortunately, this has no effect on my system. ----------------------------------------------- poisonivy at poppy ?? ~ % R *** Successfully loaded .Rprofile *** Welcome back poisonivy working

On Sat, Dec 5, 2015, 9:39 PM peter dalgaard <pdalgd at gmail.com> wrote: > On 05 Dec 2015, at 18:07 , arnaud gaboury <arnaud.gaboury at gmail.com> wrote: > > In my shell environment, I have set a path to R_PROFILE_USER. The > file, Rprofile.R, is a collection of small hacks. > > I want to build rstudio-server from source. Best is to $ unset > R_PROFILE_USER

Hi all I am using * to trunk calls to a number of customers. This is proving unreliable - I set qualify=yes on all the trunks: [customer] type=friend host=customer.dyndns.org username=hewlett secret=******* context=int_calls trunk=yes nat=yes qualify=yes All Internet links are ADSL. I am running 1.0.9. I get a no. of symptoms: - every now and again i get a message that

Paul, Thanks - it worked! For the record, this is exactly what I did: cd/usr/src/asterisk/cdr grep -in "userfield" cdr_csv.c (to find the line that had "#define CSV_LOGUSERFIELD 1" commented out) Opened cdr_csv.c and removed the /* and */ comment marks Saved & exited After shutting down * I went to /usr/src/asterisk and did the usual: Make clean Make Make install I also