similar to: Data calls through *

> > Yes... I used todays CVS. > > And on another note, my system having weird issues. > Some phones would just stop working. 3-5 at a time. > A restart would fix it temporarily. > > So I back tracked to my previous version 06/06/2003, > this has been my most stable version ever. > > John > > > On Monday, August 25, 2003, at 03:23 PM, TC wrote: >

I currently have two fax machines on my system. Both of them seem to send and receive very slowly. My end users are complaining; saying it was faster before we moved to * (Straight Analog Lines) Any help would be great. PS: I already have the d option on the Dial line. Both fax machines are in their own context: [faxes] exten => _9NXXXXXX,1,StripMSD,1 exten =>

I was having problems getting a Linksys PAP2T-NA to work with Pitney Bowes mailing station so it could use its modem to dial home and download postage/software updates. After scowering the web, I couldn't seem to find a definite how to article on what settings were needed. I finally came up some settings by combining the information from various places around the 'net. I have typed out

Here's one last plea for help from the list before I give up on the totally cool software FAX modem concept in complete despair (for now, at least, until I have more time to dig into it).... I am simply not able to successfully receive a FAX no matter what device I transmit from or how I bring the call into Asterisk. My lastest effort was a call from a Pitney-Bowes FAX machine into an X100P

Has anybody out there tried to use two different iconnecthere accounts with Asterisk? What I want to do is use a second account if the first is busy. I have tried the following: exten=>_91NXXNXXXXXX,1,StripMSD,1 exten=>_1NXXNXXXXXX,2,Dial,SIP/BYEXTENSION@iconnect ;iconnect is the first account exten=>_1NXXNXXXXXX,3,Dial,SIP/BYEXTENSION@iconnect2 ;iconnect2 is the second account But that

When I place a call using Iconnecthere as my sip provider, I hear no ringback when making a call. Does anyone else have this problem or offer any suggestions? Thanks, Kevin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031003/b048f72f/attachment.htm

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I've been seeing some strange issues recently as it relates to our Asterisk phone system. Let me give a little background on things: We have a T1 provided by Knology. We recently had an outage of Internet and phones. The following problems (except for the log messages) seemed to have cropped up after the outage. I was prompted by a friend of mine

On Mon, Apr 27, 2009 at 7:00 AM, Andrew Premdas <apremdas at gmail.com> wrote: > Currently I have a very simple constants implementation being loaded as a > Rails initialiser > > module MVOR > ? module Postage > ??? THRESHOLD = BigDecimal.new(''6.99'') > ??? RATE = BigDecimal.new(''30.00'') > ? end > end > > > My scenarios

Hi folks, The obligatory newbie disclaimer: "Hi, I'm new to Asterisk and I have a couple questions..." OK, now that that's over with: I've just started working for a small CLEC, and I'm trying to sell * to my boss as a replacement for some expensive/inflexible/closed-source software he's been using to provide residential dialtone with for a couple years now.

In the past we have used Adtran Atlas 550's to break out FXS ports for devices like modems. The great thing about the 550 is that internally it is all TDM so there is absolutely zero latency. We are able to use ATA's for faxes and analog phones but devices that use modems, they fail 99.99% of the time when using an ATA. We tried to migrate to TA908 devices; they have FXS ports built

I have a linejack and a phone jack in my asterisk server working well between the SIP phones and the phonejack. what I cannot get to work is the outbound linejack Phone/phone0 trunk line? how can I get a SIP or Phone/phone1 phonejack phone to dial 9 then outside number and pickup Phone/phone0 and dial it? right now it accepts a 95551212 but busy's on the last digit 2. no outside dial.

hi guys, I have about 80 CentOS T-Shirts, ranging from Medium to 3XL in size. These are the grey T-shirts we can see Ralph, Garry and the guys from hostdime modeling for us at: http://www.karan.org/pics/centos/images/002-IMG_2571.JPG If you would like one, please send me an email on kbsingh at the centos.org domain, and let me know your address and what size you would want, I would be happy

Hi all, I'm working on a setup for a small office. I'd like to use SIP/iconnecthere most of the time, because they're cheap. But they only allow a single call. When the single iconnecthere line is in use, I'd like to use IAX2/voicepulse instead: exten => _1NXXNXXXXXX,1,Dial,SIP/BYEXTENSION@iconnect exten => _1NXXNXXXXXX,2,Dial,IAX2/npI42VoD38@voicepulse/${EXTEN} Well,

Hey I need another pair of eyes on this! I would like to add phones numbers to the blacklist from any handset so I did this: exten => _*66XXXXXXXXXX,1,StripMSD,3 exten => _XXXXXXXXXX,2,DBPut,blacklist/BYEXTENSION/1 exten => _XXXXXXXXXX,3,Hangup However what I get in the database is: /blacklist/BYEXTENSION : 1 And BYEXTENSION is not replaced with the actual number

http://bugs.freedesktop.org/show_bug.cgi?id=15564 Summary: New account request Product: swfdec Version: unspecified Platform: Other OS/Version: All Status: NEW Severity: normal Priority: medium Component: plugin AssignedTo: swfdec at lists.freedesktop.org ReportedBy: jbowes at

Hello-- I'd like to do a little processing on external phone numbers from within the asterisk pbx. Fairly simple stuff, but... devilishly hard to make it work so far! 1. I'd like to dial 9 to get an outside line. 2. If the number dialed after the 9 is 754XXXX, I'd like it to go thru unmodified. It's the only local number available here. 3. I'd like all 1 XXX XXX XXXX numbers

Does anyone know what this means? It is in DMESG, and we have people complaining about echo. Disabled echo canceller because of tone (rx) John

This patch adds a 'hairpin' (also called 'reflective relay') mode port configuration to the Linux Ethernet bridge kernel module. A bridge supporting hairpin forwarding mode can send frames back out through the port the frame was received on. Hairpin mode is required to support basic VEPA (Virtual Ethernet Port Aggregator) capabilities. You can find additional information on VEPA

This patch adds a 'hairpin' (also called 'reflective relay') mode port configuration to the Linux Ethernet bridge kernel module. A bridge supporting hairpin forwarding mode can send frames back out through the port the frame was received on. Hairpin mode is required to support basic VEPA (Virtual Ethernet Port Aggregator) capabilities. You can find additional information on VEPA

This patch adds a 'hairpin' (also called 'reflective relay') mode port configuration to the Linux Ethernet bridge kernel module. A bridge supporting hairpin forwarding mode can send frames back out through the port the frame was received on. Hairpin mode is required to support basic VEPA (Virtual Ethernet Port Aggregator) capabilities. You can find additional information on VEPA