similar to: MWI

Hi all- My TDM long distance is provided by MCI. We use account codes where MCI sends a challenge tone after receiving 1NXXNXXXXXX. Anyone have any suggestions of how to accomplish this? I can't get the soft phones to send the DTMF (the other digits go down the d-channel of our PRI). I also have not bee able to get the dial or the outgoing queue command to work. Anyone run into this?

I just inherited a Cisco 2621 with a VWIC-1MFT-T1 card in it. Can I make this thing into MGCP gateway or even a SIP gateway for asterisk? Seems like it should bee useful for something! I'm perfectly happy to do my homework, but also don't feel thee need to reinvent the wheel! So, links with relevant info would be appreciated. If there is a config for a 2621 being used as a gateway

Hi- I have a 7960 using chan_sccp which gives me a busy signal as soon as I dial the 1 in my string of 915552223333. Can't figure out why. I do have a dialplan.xml file: <DIALTEMPLATE> <TEMPLATE MATCH="*" Timeout="5"/> <TEMPLATE MATCH="#..." Timeout="5"/> </DIALTEMPLATE> Anyone have any insights?

As the thread from the other mailing list he sent this to states, it is illegal to share the file(s) he is asking for. Below is the thread from the sccp users mailing list that he sent this to. <sccp mailing list> 2006/2/28, picciuX In fact: the one you mention is not a config file; it is part of the "Locale-Installer for Cisco Call Manager". You need a valid service contract

Hi All- I am trying to create a post call survey application. I would like to: 1. ask the caller if they want to take a survey after their call completes 2. If no, just transfer the call 3. if yes, 4. bridge up another extension 5. wait for that extension to hang-up 6. have the system (not the user) transfer the call to different extension that administers an IVR based survey. Anyone

We've been trying unsuccessfully to connect our Meridian Option 81C to a TE110P via PRI. We've followed the directions in asterisk-meridian-a1.pdf (link on http://www.voip-info.org/wiki/view/Asterisk+legacy+integration), but it doesn't seem to work on our 81C (even though many, many users report it works very well on Option 11's). Has anyone had any success in getting the above

I've been beating myself up over this script but clearly I'm missing something. If I enter an extension like 101 it rings through fine, but if I pick 2 for sales it hangs up with this message: == Spawn extension (sales, s, 1) exited non-zero on `Zap/1-1' Since I'm not sure what that exacly means I cannot take appropriate action. Any help would be appreciated. [default]

Hello, We are looking to adjust the font size of the axis annotation on the graph that results from use of the metaplot() function. Metaplot seems to respond to cex and cex.lab to change those graphical parameters, but it doesn't respond to cex.axis. Is there a way to work around this by creating a customized x-axis, and if so, how? Thanks for all your help. Syntax is below. Best, Dawn

Is there a good Web Conferencing add-on or a compatible package for Asterisk? I know there are web based controls for the audio, but I am looking for PowerPoint or desktop sharing functionality similar to WebEx. Anyone using a package they like?

Dear tinc community, I am using tinc in switch mode. I have three nodes. Two nodes reside on routers, vpn-eth is bridged with internal lan, each router has several machines connected to it's internal lan. Third node is the roadwarrior - "endpoint" linux PC. When the roadwarrior is off - everything works perfectly, machines on both sides can communicate without a problem in any

https://bugzilla.samba.org/show_bug.cgi?id=7757 Summary: with big file, rsync times out out when it should not; the sender is still responsive Product: rsync Version: 3.0.7 Platform: Other OS/Version: Linux Status: NEW Severity: normal Priority: P3 Component: core AssignedTo:

Hi, This is the first time I have setup rsync.conf like, max connections = 20 syslog facility = local3 read only = true hosts allow = 192.168.10.10 [qmail-control] comment = qmail-control path = /var/qmail/control read only = yes list = yes uid = root gid = root Then from 192.168.10.10, I tried, rsync -avz -e ssh

Hi all, I've setup two * servers which are SIP interconnected ala osaka/toronto from the * book (before anyone sugggests using IAX instead, no, I NEED to have them SIP interconnected for verification/test purposes). Then I have a Zoiper connected to one of them via IAX (so that * will not reinvite (?)). As soon as I try to call (via Zoiper) an extension on the other * I get a "Failed to

Hello We recently moved to Asterisk 1.2.4 (from 1.0.x) and our 10 Uniden UIP200 have stopped working ever since. We can make a call with the UIP200 to any other extensions, but it can not receive a call. In fact the UIP200 always appears offline: It does show up in asterisk a few seconds after the UIP200 reboot: -- Saved useragent "Uniden SIP Phone p2 Ver BS4.70" for peer uip200 but

Hi, I would like to have [backup] service hosted on samba machine, resolving each client to different directory (/srv/backup/prod001, or /srv/bacjup/prod001.domain.com). Is this possible? By reading man to smb.conf I see there is %M variable. I have defined following service ## backup share [backup] path = /srv/backups/machines/%M read only = No public = no writable = yes

My NFS server is on 192.168.10.10, with the following setup in /etc/exports: /backup 192.168.10.0/24(rw) Then, form a client (192.168.10.11), I run mount 192.168.10.10:/backup /bck, but get the following error: root at vps01 [~]# mount 192.168.10.10:/backup /bck mount.nfs: Input/output error On the main server, I see this in /var/log/messages: Mar 8 08:42:52 venus kernel: kjournald starting.

Hi! I am studying HTB. I used the topology showed below: 10.1.0.1 ------ 10.1.14.25 (eth1) / 192.168.10.10 (eth0) ----- 192.168.10.11 (host 1) (router) (host 2) All machines use Debian Etch. Has a Iptables masquerading rule to eth1 in the router machine . The NICs are 100 Mb/s. The host 1 has Apache 2 and a file with 670 MB (CD ISO image) to download. When I used

Hello, I got some weird problem with public key authentication using rsa key pair. Let me first of all explain my setup. 1) I got two Solaris 8 x86 boxes uname -a SunOS 5.8 Generic_117351-24 i86pc i386 i86pc <kdc: 192.168.10.11> <---> <module: 192.168.10.10> 2) They're running absolutely identical openssh installations I'm using pkgsrc, so I've builded all

I am working on this app to dial two external numbers. The second is dialed after the first hangs up. I have simplified things down to: exten => 3852,1,Dial(zap/g1/3964,10,g) exten => 3852,2,Wait(2) exten => 3852,3,Dial(zap/g1/7757,10,g) exten => 3852,4,Hangup Here is the debug: -- Accepting call from '0000000000' to '3852' on channel 0/23, span 1 --

I have an adit 600 with one fxo card connected to a Digium single span T1 card. CallerID, disconnect supervision work perfect, however the users complain that they have some sound quality issues, after testing it I realized that whenever one is in a phone call they get like silence between the sounds coming from the other party, almost like a cell phone, in other words if there is no sound coming