similar to: Asterisk not reading fast DTMFs, was: PBX -> PRI -> * -> Telco not working

Hello A user seems unable to type DTMF in our Asterisk IVR menu. Can this be due to their phone or PBX that disables DTMFs when a user is off-hook? Thank you.

Hi, I've just installed DAHDI at two PBXs as follows: *PBX-1 PBX-2* FXO ------------- FXS When I try to send calls from PBX-1 to PBX-2 I just receive the message: "Starting simple switch on 'DAHDI/1-1" It seems like if PBX-1 couldn't send DTMFs, but when I set immediate=yes at chan_dahdi.conf at PBX-2 dialplan is executed at PBX-2 but nothing is heard at

We need to conference together a call center agent, a customer and a third party IVR and send DTMF tones from the agent to the IVR. MeetMe has been eating our DTMFs so we'd like to try app_conference. Has anybody setup such a configuration in app_conference and how did you configure it? -HJC

Hi I've got an Asterisk box connected to a Siemens Hicom 300 using a Digium TDM400. The Hicom provides the calling extension as DTMF at the beginning of the call followed by two *, as in 3425** when 3425 calls my extension, I can hear all 6 tones if I have a handset connected but using Asterisk's Read application straight after Answer() Asterisk usually only gets the last *, sometimes the

OK everyone, so we finally have Mail merged into ActionMailer replacing out TMail This is for the 3.0 release, and not part of the 2.x tree. I am the TMail maintainer and I decided this year to write a ruby email handler, the Mail gem is my solution to this. Mail takes a very object oriented approach to email. It conforms to RFCs as closely as practical and parses the 0.5gb Trec and 2.5Gb

OK everyone, so we finally have Mail merged into ActionMailer replacing out TMail This is for the 3.0 release, and not part of the 2.x tree. I am the TMail maintainer and I decided this year to write a ruby email handler, the Mail gem is my solution to this. Mail takes a very object oriented approach to email. It conforms to RFCs as closely as practical and parses the 0.5gb Trec and 2.5Gb

Hello all. I have an NEC PBX connected via a TE210p E1 line to an asterisk 1.6 box. NEC -> E1 -> TE210P:1 -> * -> TE210P:2 -> E1 -> Telco Incomming calls from the telco to the asterisk box to the NEC work fine with indials and everything. Works sweet. Outbound from the NEC to the Asterisk box fail. Giving an long dial tone that then times out. Ie, pick up NEC handset, dial

Hi all, I have a TDM2400 full of FXS cards and a TDM800 full of FXO cards. If I have a fax machine on the FXS port dialing out through asterisk on the TDM800 FXO, should I be expecting any problems? Or should this just work as expected? (ie, flawlessly with the asterisk box essentially transparent to the whole operation). I am doing it this way to allow many faxes and modems to share a dial

Hi all. If I run rake features or cucumber features/* I get one failing FIT scenario in one of my features. If I then run that feature that contains the FIT table with the failing scenario manually, it passes. I run it again with rake features or cucumber features/* it fails, run individually, it passes. Obviously some state is carrying over between the features. Any idea where i would start

I have a tree of models that represent a book. Looks like this: book sections documents paragraphs index entries glossary entries footnotes index entries glossary entries These are all models with has_many from parent and the child has belongs_to. They also all have dependent => :destroy

On the rails blog this morning: http://weblog.rubyonrails.org/2008/6/1/rails-2-1-time-zones-dirty-caching-gem-dependencies-caching-etc Mikel http://www.lindsaar.net/ruby-on-rails-tips --~--~---------~--~----~------------~-------~--~----~ You received this message because you are subscribed to the Google Groups "Ruby on Rails: Talk" group. To post to this group, send email to

Hi all of both RoR Core and TMail. You may know me better as the maintainer of TMail. However, TMail has been a bit difficult to get working with Ruby 1.9, so in the light of that, I sat down over the past few months and wrote myself an all encompasing mail gem. Mail passes all of its hundreds of specs equally well in Ruby 1.8.6, 1.8.7 and 1.9.1. Mail reads every email in the TMail test suite

Hello, I installed Asterisk 1.6.2.17.3 ( latest as of yesterday ) and had multiple problems with DTMF. I have two machines, we'll call them asterisk and asterisk-pri. Asterisk does IVR and asterisk-pri has a PRI card in it and connects to the PSTN. The two servers communicate via SIP with RFC2833. I setup logger.conf on both machines to display DTMF to the console. Both are built from

I'm looking to build an application which requires a phone to autodial when the handset is lifted off of the hook. Since the system will be available to the general public, I want a solid phone that will withstand abuse, but not be too expensive. I'm leaning towards using a Sipura adapter with an analog outdoor handset. However, I don't nor ever have had a sipura adapter, so I'm

I'd like to setup an autodial that will do the following: 1. First try call my multiple SIP phones at my house - if no answer within n seconds, try my mobile phone. Alternatively call both SIP phones at home + mobile at the same time. 2. Once someone answer in step 1 (on any phone) initiate calling to a specific phone number/extension Currently I have it working except that in step 1 I can

Hei! Does anyone know how to configure this phone to autodial the number after interdigit timeout has passed? Rennes

Hello, I am using asterisk to autodial a phone number once an hour to verify an answering IVR script is working by listening for a beep played by the IVR, and that is working well. The one thing I have not been able to find is a way to trap if the call didn't connect at all, like if the entire IVR is down or the phone line is dead, since it doesnt enter the menu until the call is

Hello, I'm writing a small PHP application that generates calls automatically and tries to store call details on a Mysql Db, using manager API . When making an autodial call, I noticed that I couldn't read $DIALSTATUS values; since I can't evaluate dial status (BUSY, CONGESTION, NOANSWER), I can't understand when a receiver was busy or not. Nobody seems to have solved this

Hi everyone, I am reading through Sangoma Wiki right now. But someone may already and quickly notice this. I have a system that is down since the morning (maybe power intruptions). All seems fine except for "wanrouter status" shows disconnected. Following are the alarms raised. Should I call telco (they have long wait times) or should I just keep searching online for troubleshooting

Hello Please help me, i need transfer a call in asterisk to other telco number and free the channel. Can i do with any q931 function?. Thanks a lot Aris... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090528/2bcd93ae/attachment.htm