similar to: "o" extension broken?

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

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

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 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, ...)

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 :-)

Dear useRs, Sorry, if this is a FAQ. I just need the direction to dig. I am working on the linux box (its DNS name is raccoon, to be definite), Fedora Core 3. I connect with ssh to another linux box (Debian, dns name chena) and run R on it. When I call a plotting function from the R command line, it produces the file Rplots.ps, containing the required plot. I am running X on the raccoon. When

I have added a sip user in sip.conf. user name is 819,context is c819. and I have added the follows rows in extension.conf. like [c819] exten => 1,1,Answer exten => 1,2,SetVal(voicemail=${exten}) exten => 1,3,Dial(SIP/${voicemail}) It appear a error message(pbx.c:1279 pbx_extension_helper: No application 'SetVal' for extension (c819, 1, 2)) when i dial 1 from 819. The

Hi, I am trying to configure ipsec on asterisk. Have configured /etc/racoon/racoon.conf and /etc/raccoon/psk.txt. Also have policy file in same folder. Have run racoon. Still I can't receive calls. Can anyone please tell if any extra step is needed. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, This issue might be a slightly offtopic, but someone might have experience with it. Thanks for reading this post anyway. I have the following setup: Network 10.227.7.X is connected over a wlan (172.1.1.1 <-> 172.1.1.2) to network 128.1.1.X. This setup works, I have cross-subnet browsing going and I am able to login. When I enable IPSEC (raccoon (linux <-> freebsd)) I am still

there's a regression in the xf86-video-nouveau driver that came up in 1.0.3 that causes the desktop to bounce around in virtualbox whenever using HW acceleration in cinnamon and gnome 3, but not in fallback or cinnamon 2d, only happens in virtualbox and not the host, I've confirmed it was the graphics driver on the host because it happened with the 1.0.3 driver but not the 1.0.2.

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

Hi, Thanks to some pointers from Christian Kratzer, I am now able to join the office VPN from a random WiFi hotspot. With the configuration files changes detailed below, from a public WiFi hotspot I can now use this 3 step procedure to login to the office VPN. 1) While at hotspot, boot up my -STABLE laptop. 2) Insert wireless card. 3) "rsh server" This procedure works for a DHCP

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

Is there some way to remove superscripts from objects returned by html/xmlParse (XML package)? h <- "<html><p>Cat^a</p><p>Dog</p></html>" doc <- htmlParse(h) xpathSApply(doc, "//p", xmlValue) [1] "Cata" "Dog" I could probably remove the <sup> tags from the "h" object above,

Dear All- I am trying to create a legend where the first 9 terms are represented as pch=16 in one column (which I am able to do); but, I would like the last two terms in that column to be 'lines" (which are black and red - just as it looks like in the plot). I have read through many help pages and searched the help listserv, but haven't figured this one out yet. Does anyone have

Is anybody using 1.2 (or 1.2.1) in a production network using Realtime (voicemail, sip or extensions) with 100+ SIP phones? If so, what are your experiences? We've been running 1.0.3 for about a year and it's been rock-solid. We'd like to upgrade to Realtime and 1.2, but I'm afraid of killing our stability. Obviously, we'd do it in stages (upgrade to 1.2, then realtime

I have compiled Spandsp without any problems. I got no errors I have also done the patch without getting any error. I have tried pre4 and pre6 version with same problem. I got no errors with the ./configure make make install. After I add the directory for /var/local/lib into the /etc/ld.so.conf I have ran ldconfig. I am running asterisk as user asterisk on a RH 9 Linux distro. This system has

I was looking at the ILECs' web sites to determine how their users access voicemail. I looked at AT&T, Verizon, Qwest, and Embarq. They supported one or a combination of the following for calling from your phone: *98 #55 Toll free number Your number A varying phone number, based on your number's location. Calling from anywhere else, they supported: Hitting star when