similar to: Phantom (ghost) Calls with Wildcard TDM400P

I've played around with the lightly documented Asterisk voicemail feature whereby a caller can press "*" during the playback of the OGM and be returned to the "a" extension in the context of the voicemail box. No matter what, Asterisk does nothing when you press "*". It does not interrupt the OGM and it certainly does not return to the "a" context.

My apologies for the cross-posting. If you are a business or individual providing Voice over IP services in Canada then we encourage you to read this email carefully otherwise please disregard. ----- As you are most likely aware, the CRTC has undertaken the roll of regulating VoIP services in Canada and is currently conducting hearings with the goal of putting in place regulatory requirements

We are using a Cisco router with a T1 card plugged into a PRI provided by a local telco (Allstream). This Cisco accepts calls and sends them to a couple of servers running Asterisk depending on which number was dialled. But there is a problem. When a call comes in to the Cisco from the PSTN it sends it to the Asterisk server something like this: FROM: 204XXXXXXX@<CISCO IP> TO:

On Tue, 2005-05-31 at 13:38 -0400, Jared Mauch wrote: > On Tue, May 31, 2005 at 12:30:07PM -0500, John Lange wrote: > > I am not a Cisco person; so the question is, is it possible to have one > > of the following: > > > > 1) Have the Cisco authenticate (register) as a SIP client to the > > Asterisk server. This allows me to place the Cisco in its own context. >

Hi All, I have a pretty standard Asterisk PBX setup with 60 SIP Peers, mostly Polycom 501's and a receptionist phone, Polycom IP 601 with 3 attached sidecars and Buddy Watch enabled monitoring all other SIP phones. The problem occurs when a group (all SIP peers) Page is called. Not always but sometimes when the Page is executed, the IP 601 will become unreachable from Asterisk. So when the

Here is the situation that I would like to create: Call comes in Receptionist sees that the caller ID is Jenny <8675309> Receptionist picks up phone and transfers call to Batman Batman looks at his phone and sees that the caller ID is Jenny <8675309> I can't seem to figure out how to forward the caller ID. Is this possible with Asterisk?

Greetings- An odd situation has developed. I use the following code to create .eps files of two very similar graphs: postscript(file='resources.bygt.eps', onefile=FALSE, horizontal=TRUE) barplot(resources.bygt.matrix, beside = TRUE, legend.text=c('narrative','doubt'),

Hi, I''m running Xen 3.0.4 with suse (kernel 2.6.16.46-0.12-xen). I''ve a problem with a "ghost" (paravirt) domain that appear in the output of xm list. As an example: # xm list Name ID Mem VCPUs State Time(s) Domain-0 0 1024 4 r----- 29.0 vm-xyz

Hi, I am trying to do the world's most simple install. I have a Wildcard TDM400P with 3 ports: 1 FXS on port 1 and 2 FXOs on ports 3 and 4. (i'm not using port 3 for now, put want it for expansion purposes) I simply (to start with) am looking to have the FXS phone ring when an FX0 port is dialed. I would also like to be able to place outgoing calls on the FXS through the FXO. Right

Hi, I'm trying to make a legend with a justified list of numbers, so I thought I would use phantom(0) to align the 3-digit numbers properly with the 4-digit ones: legend(x, y, xjust=1, yjust=1, lty=c(1,2,3,4,5), adj=c(0,0.5), legend=expression(phantom(0)*300*plain(K), phantom(0)*550*plain(K), phantom(0)*830*plain(K),

People, My goal is to get asterisk dialing out via my landline (POTS) from a sip softphone. Ive got the phone, The TDM400p is installed and working. (See below) When ever I dial a number that is directed to the outgoing port on my card (fxs/fxo?) I get no ringing, then it claims its been answered. the CLI reports the following: Executing Dial("SIP/301-f97a",

I have a customer who wants their receptionist to input the users' long distance PINs for the because they use each others pins. I am having trouble coming up with a way to do this because of creating a channel between the user and receptionist, dropping the channel and its variables and creating a new one for the actual long distance call. Any advice is really needed. 1. User Dials Long

Hi, guys. I''ve been trying to use the pci-phantom command line options to xen so as to work around the hardware issue with the Marvell 88SE91xx SATA controllers in IOMMU ([Intel:] VT-d) mode, but I cannot seem to get my head around it. From having had a glance here: http://xenbits.xen.org/docs/unstable/misc/xen-command-line.html and in particular the syntax described as such:

Hi all, At our customer site i've installed one asterisk server with 20 Grandstream GXP2000's. Firmware 1.0.1.9. When someone dials the customer, the receptionist picks up, and does an attended transfer (the 'grandstream way') to a collegue. Most of the times this goes ok, but sometimes, when the receptionist puts the call on hold, and tries te reconnect to the caller there's

There's a phantom domain master browser showing up in my Samba nmbd.log file. I keep thinking that maybe it is left over in one of the files that I transferred over from the old server to the new server and it isn't clearing itself out. Is there a way to clear that and is it possible to have a phantom browser fighting over the Domain from a copied over file? I transferred all of the

I'm testing a TDM400P with FXO module to receive incoming calls from an analogue line and send it to a SIP device. To recieve callerid, I need to use cidsignalling=dtmf and cidstart=polarity. The problem is that when a call is finished, the TDM400P seems to require about 20 seconds to prepare for the next incoming call. If a new call comes in within 20 seconds after the previous call was

Hello List, I am currently having a bit of a strange issue with a pair of asterisk servers that we recently set up. For a bit of background, this particular business has two sites in two different towns, about 10 minutes apart. They have 3 analogue PSTN lines connected to the asterisk servers at each location, via a Sangoma A200 (with HEC). They are trying to have just the one receptionist for

I have a Polycom IP600 serving as a receptionist phone. We developed a call manager via c/gtk that runs on a touchpad. It allows them to transfer calls, transfer to voicemail, page, etc. The problem is this: When paging another phone from the touchpad, I have to open a channel to the receptionist phone. This rings the receptionist phone. When she picks up, it then pages the desired person. This is

About once a day I have noticed a phantom incoming call with a caller ID of asterisk@192.168..<cut off>. When I answer the call there is a dial tone and the call is disconnected. Any clues? David Koski david.nospham@kosmosisland.com

I'm having a major problem at one of my branch offices with "Phantom Rings" on their asterisk-based phone system. The system was originally built using 2 X100P cards and was recently upgraded to a Rhino R4FXO-EC card. The upgrade severely increased the frequency of the phantom rings. I've read everything I can find on-line about automatic testing and noise on the line and