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Does anyone have a working way of having a Cisco 7960 reload its config remotely. I have tried some of the scripts that I have found on the web, but to no avail. Thanks for the help. B. J. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040116/aa4eda3c/attachment.htm

I have a quick question for the list. For what reason would you have multiple instances of asterisk running on a single box? I can maybe see it if you have multiple IP addresses, but other than that I am drawing a blank. Thanks, B. J.

Hello all. We have an asterisk system set up, and we are seeing a lot of multiple DTMF digits being read by asterisk. In digging through the archives the only answer I have seen is to put in the statement relaxdtmf=yes in the zapata.conf file. Since we are not using any zapata devices, I have tried to put that statement in my sip.conf file to no avail. Any help would be appreciated as my end

Due to a packaging error, the tarball released on Friday night did not have a version number embedded in it, which results in various strange build errors and other odd behavior. The tarball on the FTP servers has been updated to correct this situation. Sorry for the inconvenience :-)

I am having a problem with Polycom auto-answer. I have the auto-answer working between PhoneA and PhoneB, but when I try to use the intercom between more then one phone I start having problems. PhoneA dials *3 which calls PhoneB, PhoneC, and PhoneD. All the phones ring, but only one will pick up, the rest will hang up and I get this error on Asterisk: Got SIP response 500 "Internal Server

Hello all. I know * doesn't directly support recording mp3 files, but I was wondering if anyone has created an AGI to do it indirectly. Thanks in advance. B. J. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040105/22d999c4/attachment.htm

Hello all. I just found out that I am no longer able to exit out of voicemail properly by hitting the 0 key, but the * key works. Asterisk comes back and says "I'm sorry, I did not understand that response" and goes on in the context. Is this a new "feature" or bug? Is anyone else having this problem? I am using Asterisk 1.0.3, and have tried it on two separate

Hello all. Has anyone had any success using ChanIsAvail with only SIP channels? Is there another, better way to check if an extension is busy without dialing it? Thanks, B. J. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040109/48ac2c3e/attachment.htm

Hello, I'm having problem with a wildcard TE110P. As soon as I load the module (wcte11xp for kernel 2.6.10), it spawns a yellow error with or without an E1 plugged-in. Any one managed to set it up in France? Here are my files: zaptel.conf: span=1,0,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 zapata.conf: [channels] language=fr context=default switchtype=euroisdn pridialplan=unknown

Hello asterisk community, I have successfully set up asterisk as a SIP PBX and now would like to connect to the outside world using a Cisco 2600 with VIC-BRI as an ISDN gateway. This works already in the lab, but I have security concerns before conecting the gateway to the internet. I currently don't know exactly what VoIP services the Cisco runs by default besides SIP (H.323, MGCP, ...)