similar to: asterisk -> my cell phone's voicemail sound problems

Hi, I am attempting to implement Answering Machine Detect and have also played with using BackgroundDetect instead. Does anyone recommend one over the other? Here is the code I am using for the BackgroundDetect method (from voip-info.org). Thanks. [detect] exten => s,1,Set(MACHINE=0) exten => s,2,Answer exten => s,3,BackgroundDetect(silence/5, 1000, 50) exten =>

I'm looking for an application that can monitor a channel for voice input and then proceed on. The closest thing I've found is BackgroundDetect, which expects DTMF. Here's what I'm doing: -Call file generated which calls someone and connects them to an extension. -Extension plays stuff, etc. etc. etc (not important) With digital or VoIP termination, this works fine, because *

Hi, It is possible to create a dummy account (SIP or IAX type) in order to be used in a "dummy" extension? I want to be able to use it as a normal extension (as an IP phone connected to it), but without the need to answer or call from that extension. I want that when I call that extension to hear the ring, and after the defined period of time to enter in the Voicemail system. I

Hi, There seems to be no solution for call progress on analog lines and using outgoing spool call files . My wave file starts playing before the person has answered the phone so the first part of the message is missed. Can the backgrounddetect app be used for this. I have tried but the message still plays before I answer. I generated 60 seconds wave file. [callprogress] exten =>

Hi all, I hate to ask such a simple question, but it has stumped me over the past couple of days. I have 2 asterisk servers connected to the house lan and also via a crossover ethernet cable. The original purpose of the crossover was to create a private lan for TDMoE. I have a TE410P in each machine using PRI. I also have setup SIP and IAX2 between both machines. What I'm doing is

2 questions on the code of mini-os (the one in XEN 3.0.3): 1. In xen-3.0.3/extras/mini-os/mm.c:52 is a declaration of stack extern char *stack; but in file xen-3.0.3/extras/mini-os/arch/x86/setup.c:48, there is also a decl of stack: char stack[8192]; If one dereferences the variable stack, which is apparently not the case yet, it gives surely a segfault! 2. In file

Hello, To detect an answering machine I have found following two commands, BackgroundDetect (comes with asterisk) MachineDetect (asterisk add-ons) First question, does BackgroundDetect works well with g729? I havn't try MachineDetect yet, what is the benefit of MachineDetect over BackgroundDetect. If anybody used any of this command successfully, please help me. If possible, please let me

Un saludo a todos. Estoy haciendo un ACC y utilizo el paquete Ade4. Cuando quiero realizar un plot del resultado utilizo plot(). el resultado son 6 gráficos: loadings, correlation, inertia axes, scores and predictions, eigenvalues y species. Lo que quería saber es cómo representar en una sola ventana (los 6 gráficos a parecen en la misma) el gráfico correspondiente a Species. Gracias por vuestra

Can't find a way to control the dtmf mode on a per session basis with pjsip, used to use SIPDtmfMode from the dialplan with chan_sip. Any hints on how to do this?

Why is there an argument returned in the function below (of hvm.h), knowing that there is void? static inline void hvm_init_ap_context(struct vcpu_guest_context *ctxt, int vcpuid, int trampoline_vector) { return hvm_funcs.init_ap_context(ctxt, vcpuid, trampoline_vector); } _______________________________________________ Xen-devel mailing list Xen-devel@lists.xensource.com

To whom it might concern, The Gesko Ikarus 1200S analog telephone has advanced callerid capabilities. When used with an ATA186, it show the username and the phonenumber of the caller. (or whatever you let * tell it) http://www.gesko.be/idgg004.htm Price is 77 euro something and available with Telec. (NL) Met vriendelijke groet, Pauline Middelink -- GPG Key fingerprint = 2D5B 87A7

Hi everybody, I'ld to use virsh over ssh in order to connect to my xen hypervisor from one of my DomU. I tried: virsh -c xen+ssh://root at 192.168.0.67:22 list --all And it correctly listed the started virtual computers. And when I tried virsh connect xen+ssh://root at 192.168.0.67:22 And I got. error: no connection driver available for xen:/// error: failed to connect to the hypervisor

Incoming FWD calls from other FWD users, iaxtel, or via ipkall, need to have dtmfmode=rfc2833. However, incoming FWD calls from the dialup access numbers (such as libretel) need to have dtmfmode=inband. To solve this problem, I created a second FWD account and configured sip.conf as follows, in order to match the incoming number to the proper dtmfmode: [fwd-rfc] type=friend secret=*****

Hi, I've been having some problems with my internet connection, it cuts out for aprox 30 seconds at a time and after that i have to do a reload in asterix for it to re-register my sip account with broadvoice otherwise it won't accept any connection till i reload, is there a way for it to automatically re-register or am I missing something else ? Thx. Armand

Hello openssh developers, I had built openssh-4.7p1 on arm-apple-darwin9 platform, but I get this error when I load sshd using launchd: root# launchctl load /Library/LaunchDaemons/com.openssh.sshd.plist errors on getaddrinfo(): nodename nor servname provided, or not known When I execute sshd directly, I don't get any warnings or error messages, but the sshd is not listening on the intended

Hey. How would I set up my dialplan if a user wants to call its voicemail from an external phone? I'm thinking of getting the user to enter its mailbox number. Something like this: 1. User calls the dedicated voicemail number. 2. Phone prompts for mailbox number. 3. Voicemail(${mailboxnr}@context) Thanks.

Hello, I've seen various suggestions thrown around for hardware when people ask, but can we all agree on some basic hardware recommendations for a few basic setups(and post them on a website) to make it easier for new people to avoid some of the hardware/software pitfalls when they are setting up their first systems. Something like this: (THIS IS JUST A PROPOSED LAYOUT SO PLEASE BE GENTLE)

Hi, Since I'm using ubuntu 8.04.1 and asterisk 1.4.22 I cannot have color in console. Do I miss a package or compilation option ? Thanks

Hello Asteriskers, :-) We're trying to set up an outbound notification calling for system alerts with Asterisk 1.4.0. We generate a call file in /var/spool/asterisk/outgoing and the outbound call is originated through Zap/1 (Sangoma A200D to a Canadian POTS line). The problem is that Asterisk does not wait for the other side to answer before it starts playing the message. So the

I am now at a total loss. Using Sipura spa-2000s connected to *, I get DTMF working just fine for internal extensions, voicemail, etc. If making an outgoing call like this spa --> * --> Cisco AS5350 --> PSTN, I get no dial tone. I am working unsuccessfully with Cisco right now on this, but they cant find anything wrong. I have tried all suggestions I can find from the list and elsewhere.