similar to: Seperate Incoming calls on TDM02?

Hello All, I am trying to set up faxing using Asterisk@home 0.6. I have followed the instructions to the best of my knowledge. When a fax comes in, the system seems to detect OK but does ot manage to make the fax to pdf to email leap. Here is what I saw in the CLI when I tested. Any help would be appreciated. Thanks! Wiley -- Starting simple switch on 'Zap/2-1' -- Executing

I'm setting up a new AAH 1.0 box to replace my AAH 0.6 box. Though on the new box, I've installed a generic ebay X100P. I don't have my livevoip or voicepulse accounts set up yet on the new box (can both boxes be registered at the same time?). I've set up one IP phone (SPA841) with the new box. I have my SBC POTS line plugged into the fxo card. I set up everything in AMP.

Okay, first a little background - I've been with Packet8 since a month after they started. I found that we were outgrowing their services and decided to move to an asterisk box in the office. I found a service provider that offered me a reasonable rate. After a fair ammount of testing I decided to stick with their services and port my 3 primary DID's from Packet8 to the new service.

Here's something I haven't been able to discover as of yet - I need to set up a "direct link" from my Asterisk box to an external line... basically I need to be able to pick up an internal extension and have it call a local phone number. This is call forwarding, I know - the question that I have is how do I set it up so that the extension always forwards. There will never be a

I ditched the idea of using Asterisk straight through my router, there was too much to set up in too little time for me. I've found a spare machine and installed Asterisk@Home on it. Things run smoothly except for connecting to my IAX provider; It doesn't even look like packets are going out at all. Here's my config: ANY Cable ------> Firewall/Router

This is probably a stupid question.. How do I login to voicemail from the PSTN? I can dial *98 from within the system, but when dialing from the PSTN I have it set up to ring a dial group, then to an extensions vmail. During the extensions vmail prompts, I dial *98 and it sends me to the directory. -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi all, I wanted to call my asterisk on the Zapchannel with mainnumber+DID number: it's ok for calls from handy and from sip i get the right extension(DID) on phone, but when i call from an analog telephone the DID number is not mentioned by asterisk. in my zapata.conf: overlapdial = yes immediate = no That should take care of the waiting for DID numbers, which it doesn't. Asterisk how

Hi, I have a couple of questions about MeetMe and call queues. I'm still pretty new to Asterisk, but already having to write a Service Center call manager for it (which I might add, our director has agreed to make open source!). MeetMe: How can I get MeetMe (does it even do this) to ask the user to speak their name first, and play that as the new member announcement. It seems

Hi, I'm having trouble getting an answer to the following problem on -questions - I hope someone here has done something similar and can help. I'd like to compile support for FreeBSD OPIE into sshd. Presently I have to use PAM to achieve one-time password support. On a 4.x system I have in /etc/ssh/sshd_config ChallengeResponseAuthentication yes and in /etc/pam.conf sshd auth

I just signed up for the BroadVoice service a few hours ago, but for the life of me I can't get any incoming voice. The incoming connection is fine as it rings my extension from outside, but I can't hear anyone talking. Outgoing voice is working fine though. I've been looking through the archives, but I haven't found a solution to the problem yet. I even tried another router

I am trying to configure MeetMe so that external callers can enter the conference rooms after an IVR menu. I have created Conf rooms for all internal Ext's with a prefix of 8. When I call into the system from my vonage trunck the IVR picks up but will not let me dial a conf room. It tells me it is a invalid extension. Can anyone help with a sample conf on this? Thanks, RC

I am having problems with * not hanging up an incoming PSTN line, if that line is not answered before the person calling in hangs up. The line hangs in various states, it has hung with a busy tone, with no tone at all. I am running *@home and have a digium 4port line card. This was configured by the genzaptel command I then added trunks for each line. I also have a Pulver WiSip phone which I

DISA is currently disconnecting when I dial 8888 to access DISA. Below is my extensions.conf file from A@H and some lines which shows the disconnect. Should DISA be loaded as a module in modules.conf? When I do a 'show applications' i see that DISA is there. Help! -------------------------------------- ;Asterisk CLI as I placed a call from cell into the system. Playing

Hi all, I just installed an TDM02B. My system is a dell pc with linux 2.6.12-1.1456_FC4 asterisk-1.2.0-beta1 zaptel-1.2.0-beta1 libpri-1.2.0-beta1 in /etc/zaptel.conf I have (all others are default): fxsks=3-4 <--- I saw light in the ports channels=1-2 <--- change it to 3-4 has same result but... [root@nmsd0 asterisk]# /etc/rc.d/init.d/zaptel

I am trying to reconfigure an asterisk box that was using an HFC-S card with bristuff but is now using 2 analog lines therefore I want to use the TDM02B to connect to two POTS lines. The TDM02B has 2 red modules. I have this in /etc/zaptel.conf loadzone=nl defaultzone=nl fxsks=1-2 I have /etc/asterisk/zapata.conf signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes echotraining=400

Hi, FYI, Red Hat released an advisory today about a vulnerability in Ruby. So far it doesn't appear in the VuXML, but am I correct in presuming it will soon? https://rhn.redhat.com/errata/RHSA-2006-0604.html http://cve.mitre.org/cgi-bin/cvename.cgi?name=CVE-2006-3694 cheers, -- Joel Hatton -- Infrastructure Manager | Hotline: +61 7 3365 4417 AusCERT - Australia's national

Hi everyone, I just installed a TDM02B and surprisingly, I had really no problems except one. If I place an outbound call on the Zap line (Zap/3), everything works fine except when the called party hangups before I do. I do get congestion, but that is expected. However, when I try to make another outbound call using that Zap line, the CLI shows that the call is being dialed, but nothing

Hello, I have started having a strange problem. Asterisk is connected via 4 analog lines to PSTN and we have SIP phones internally. All was working fine but recently each time a user calls from PSTN and when he is leaving a voicemail for someone, the caller gets disconnected after 30 secs. We have AMP installed. This is reproducible and is happening always. It seems that Asterisk is disconnecting

Hi all, I'm trying to figure out a way to create a data graphic that I haven't ever seen an example of before, but hopefully there's an R package out there for it. The idea is to essentially create a heatmap, but to allow each column and/or row to be a different width, rather than having uniform column and row height. This is sort of like a Marimekko chart in appearance, except that

Hi, On behalf of one of our users, I installed R 1.8.1 on HP-UX11.0 by compiling the source using the gnu compilers and all was fine with make check On 1.9.0, I get an error from 'make check' when running reg-tests-1.R which gives error exit code 1. Commands used are: ./configure --prefix=/apps/global/Gnu_R/R-1.9.0 --with-x R_BROWSER=mozilla (all on one line, of course) make make