similar to: Play an announcement while receiving DTMF?

I have Asterisk running voip-only on my colo server, and have subscribed to the Voiptalk PSTN->SIP service in the UK. All has been working fine while I have had incoming calls going straight to an phone extension. I am now trying to put in a simple IVR Welcome script. I have found that I cannot receive DTMF keypresses from the incoming PSTN caller. Nothing registers on the Asterisk server, and

I'm not sure if this is a -users or a -dev question, since the answer probably comes down to something in the code. I'm running the latest CVS-STABLE, and am subscribed to PSTN service using IAX2 via Voiptalk in the UK. I've just been alerted by a customer that the sending of DTMF from my asterisk box to a remote PSTN user doesn't work, although it used to. To test it, I have

I am using Asterisk 1.2 with a TE410P connected to E1 PRI trunks. The application relies on a DTMF digit string sent by the phone after the call has connected. This DTMF is detected by Asterisk under the control of WAIT FOR DIGIT commands send from an AGI processor over a FastAGI connection. Usually the DTMF is detected without error, but on a significant minority of calls, Asterisk is missing

worked came to one ring only now. Thank you very much. If I use TE410 or TE405 instead of X100P. do it make that first ring disappear? Shakil -----Original Message----- From: tony@softins.clara.co.uk [mailto:tony@softins.clara.co.uk] Sent: Tuesday, April 20, 2004 12:27 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Auto Answering PSTN --> Asterisk using X100PCard In

I have an Asterisk system running on T1 PRI trunks using a TE405P. It seems to be running ok, but one thing puzzles me. Every so often I get a raft of messages like this: -- B-channel 0/1 successfully restarted on span 1 -- B-channel 0/2 successfully restarted on span 1 ....... -- B-channel 0/22 successfully restarted on span 1 -- B-channel 0/23 successfully restarted on span 1 I could

Has anyone else experienced a problem with the LED for span 1 on a TE410P or TE405P? I had a TE410P on which the span 1 LED would not light red, but once the span was connected, it did correctly light green. I RMAed the board to our UK distrbutor and received a replacement. However, the replacement board displayed the same problem! Wondering if it was related to the computer I was putting it

Does anyone know how a program, script or shell user can best determine whether the machine is running on bare metal or is a VMware guest? Cheers Tony -- Tony Mountifield Work: tony at softins.co.uk - http://www.softins.co.uk Play: tony at mountifield.org - http://tony.mountifield.org

In article <20151125133008.6369360.14455.17239 at gmail.com>, Israel Gottlieb <isrlgb at gmail.com> wrote: > Try putting progress instead of answer Yes, I tried Progress already, and it didn't help. But thanks for the suggestion! Tony > I have a puzzling situation, and would be grateful for any insight. > > I have a dialplan that forwards an incoming call out to

I want to use fail2ban on CentOS 6 to monitor Apache with the standard default logfile format ("combined"). Has anyone here succeeded in doing so? The format has the IP at the start of the line, followed by two dashes (if no authentication) and THEN the timestamp. What I've read on the fail2ban wiki seems to say that the timestamp must ALWAYS be at the start of the line, followed by

Has anyone here done any work on clustering Meetme conferences over multiple Asterisk boxes? The scenario I am thinking of is where there are two or more boxes connected to a set of PRIs that all answer to the same PSTN number, and where it's not possible to know in advance on which box a call would arrive. So it would be possible to have some calls on one box and some on another, that should

[apologies if this comes through twice - the original doesn't seem to have shown up even after 16 hours] In the handling of agents, when using AgentCallbackLogin, a call placed to an agent needs to be accepted by the agent pressing the '#' key. I'm trying to replicate that kind of operation in a non-agent scenario: I want to call Dial() from my dialplan, play an announcement to

I am seeing some curious behaviour with a 1.2.32 system, which I do not understand and so can't work out how to fix it. I have a PRI routed to context default. Here is the complete default context: [default] exten => _9X.,1,Dial(IAX2/m1peer/${EXTEN:1}) exten => _20XX,1,Dial(IAX2/sipeer/${EXTEN}) exten => _X.,1,Dial(IAX2/m1peer/${EXTEN}) exten =>

I'm just setting up a SuperMicro system which has twin SATA disks on an Intel IHC7 RAID-capable controller. The system came with Fedora 5 pre-installed, which I will be removing and replacing with CentOS 4.5. But before doing so, I've been having a look at how the original vendor configured it. When I've built systems previously, I've disabled any RAID controller and used kernel

Hi, I have a project coming up which will need to interface Asterisk to E1 trunks in the UK. I have a couple of questions which I hope someone can answer, or give me some pointers: 1. If I want two E1 trunks, is there anything to choose, performance-wise, between using two ports on a single TE405P, and using two E100P cards? 2. How can I test the E1 operation in the lab, which doesn't

I have been writing a lot of AGI programs in C with good success. I would like somehow to have an AGI program continue in the background while the pbx execution returns to the dialplan and continues. Is this possible? I was thinking that perhaps I could fork or create another thread within the AGI prog. The reason I want to do so is in order to monitor external information (e.g. credit limit and

Thanks a lot Tony and Dhaval for your much appreciable suggestions. Regards, Rajib Rajib Deka SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. www.siemens.com Mob: +91-9176780669 | E-Mail: rajib.deka at siemens.com Date: Wed, 20 Apr 2011 13:55:25 +0530 From: DHAVAL INDRODIYA <dhaval.it01034 at gmail.com>

Roberto estoy en uruguay en estoos momentos. Recien lllego el miercoles El mar 31, 2013 1:59 p.m., <asterisk-users-request at lists.digium.com> escribi?: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users

The background: trying out direct peer-to-peer SIP between a Grandstream BT100 and a Sipura SPA-3000 with its FXO port connected to a PSTN line. I found initially that the Sipura was very unreliable at detecting the digits dialled by the Grandstream. I have had all my GS phones set to DTMF=SIP-INFO. I found that by changing to RFC2833, the Sipura would then detect digits reliably. I'm

Does anyone here know anything about GEN-GEN analogue circuits, also known as Manual Ring-Down (MRD)? Apparently they are widely used in Hoot'n'Holler systems for financial dealer-boards. I have been asked to try and interface to such circuits, and have been having great difficulty locating any specifications for the interface. Apparently, they are always-on 2-wire analogue circuits with

I have a puzzling situation, and would be grateful for any insight. I have a dialplan that forwards an incoming call out to another number via the same SIP trunk as it came in on. e.g. [from-siptrunk] exten => 0123456789,1,NoOp exten => 0123456789,n,Dial(SIP/siptrunk/0987654321) Now, if I use a different SIP trunk for the outbound call, than the inbound call came on, the call is set up