search for: hagler

...ace to get a half-height PCI slot bracket for the Digium T100P card? I have an application to fit them in 1U cases for small, deployable PBX's. The T100P itself is short enough to fit in a half-height slot but I need an appropriate mounting bracket to screw it into the case. Thanks, Mark Hagler

...tory from a Win95 PC. I've gone through the DIAGNOSIS.txt file, and everything up to and including stop 7 (smbclient '\\BIGSERVER\TMP') works great... so from what I can tell, its working. However, when I attempt to mount from a PC, I get the error message: c:\\windows> net view \\hagler Error 3545: You cannot start or stop the network from within an MS-DOS window from explorer: trying to mount \\hagler\sflynn error: The following error occurred while trying to connect J: to \\hagler\sflynn The computer or sharename could not be found. Make sure you typed it correctly, a...

Hi, Anybody have lots of experience with PCI ISDN cards and Asterisk? I'm thinking of getting a BRI in my house to deliver more advanced signaling to my PBX (yes, I'm a geek :-)) but I've never played with isdn4linux. Is there any particular BRI card that works better with Asterisk than any other? Also, can the BRI interface cards participate in conference, etc., since they

I have the voicemail setup working in that I get the MWI and it emails the message correctly. When I pressed the MWI button on my SNOM 200, it dials into the voicemail system and prompts me for a mailbox and password. I know there is a way to automatically connect directly into the mailbox via the extension.conf file, but I can not find the documentation I am looking for in reference to variables

Is there a way to get my X100P card to dial "slower" on the line? Mine seems to dial the digits too short/fast for the switch to catch all of the digits and roughly 25% of outbound calls fail to complete. I've monitored on the line with a test set and the audio sounds clean, and I hear the X100P send out digits really fast, but sometimes the switch seems to time out and issue

...'t work. My suggestion would be to place a call from an outside line (or cell phone) through the * to voicemail or the demos to prove that the system works. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Mark Hagler Sent: Sunday, February 01, 2004 10:06 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] how to dial and accept a call with only x100p card on Redhat linux 9.0? Use a soft phone as an endpoint. There are a variety of SIP and IAX softphones you can use to place a call through...

...'t work. My suggestion would be to place a call from an outside line (or cell phone) through the * to voicemail or the demos to prove that the system works. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Mark Hagler Sent: Sunday, February 01, 2004 10:06 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] how to dial and accept a call with only x100p card on Redhat linux 9.0? Use a soft phone as an endpoint. There are a variety of SIP and IAX softphones you can use to place a call through...

I cannot get incoming calls to sip phones behind a PIX to work, outgoing is fine. Asterisk (Public IP) --> Internet --> PIX (NAT) --> Sip Phones I have tried no fixup protocol sip, I have punched a hole in the Pix allowing anything from the Asterisk box into the network, still no incoming. I have done all the Wiki suggests in regarding to NAT. Is their a trick getting the

I need to use R to model a large number of experiments (say, 1000). Each experiment involves the random selection of 5 numbers (without replacement) from a pool of numbers ranging between 1 and 30. What I need to know is what *proportion* of those experiments contains two or more numbers that are consecutive. So, for instance, an experiment that yielded the numbers 2, 28, 31, 4, 27 would be

Hello, I've got a Cisco ATA 186 from Vonage plugged into my Asterisk box with a X100P card. This works great for the most part, but I'm having a disconnect supervision problem. I suspect the Cisco device doesn't provide any sort of analog disconnect supervision when it gets a SIP BYE message indicating the far-end has hung up. This causes Asterisk to leave the channel up

I hope somebody has seen this before... I'm trying to use a Dial command on a inbound call to ring multiple destinations. The calls come in to me from the provider on IAX2, and one of the destinations I try to ring is a IAX2 to call to my cell phone. When I add the IAX2 destination into the Dial command, the setup I am trying to achieve works (i.e. my Zap, SIP, and cell phone all ring) but

Does anybody have a work around for this problem. I create a call file in /var/spool/asterisk/outgoing and Asterisk picks it up and starts placing the call. However if the called channel provides any sort of progress indication (such as a SIP or IAX channel indicating ringing that causes the console to say "SIP/xxxx is ringing") the code in pbx_spool.c indicates a call failure and