similar to: Two outbound calls at once

Hi all, I'm having a problem with * being very finicky about the length of DTMF key-presses during menus, voicemail, etc. Basically, short (<100 ms) tones are ignored, anything between 100ms (or so) and about 300ms is correctly detected, and anything >300ms is interpreted as multiple presses of the same key. This is terrible for callers who are trying to get to the correct

Hi, I recently set up the following in a production system (2.8 GHZ Xeon, 1 Gig Memory, Dell 2650). Telco - PRI - Asterisk - T1 - PBX I am getting an occasional noticeable echo on some of the phone lines (random inbound and outbound). Everyone I ask keeps telling me that I can't be having echo since I am on a PRI, which is a digital circuit. Ok, so I can't be having echo, but I am!

I'm trying to make the MWI indicators on my client's Vodavi Starplus DHS phones work. The actual signalling - in-band DTMF from the ZAP interfaces directly to the PBX system - works fine. I can manually tell asterisk to send "#9610" as DTMF and voila, the MWI on extension 10 lights (or goes out.) The question is, how is this integrated with voicemail, i.e. so that the MWI

Does anyone on this list know of any Linux based apps that will work with Dialogic or Brooktrout that provides voice-mail/Autoattendant only?? It seems that Panasonic, Avaya, and Mitel all use Unix/linux based OS on their firmware for their proprietary voice mails. My wish list would be; A software that provides all of the drivers for a dialogic or brooktrout board Voice Mail Messages in WAV

We've got an Asterisk 1.0.9 machine with a Sangoma T1 card installed being used to accept incoming calls on a SIP trunk and hand them off to a Vodavi XTS-IP PBX via a PRI running on the Sangoma card. Outbound calls simply run the process in reverse - the call comes in on the PRI from the Vodavi, and is sent out via the SIP trunk. The problem is that we're getting reports of dropped calls

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Hello - I have a non-PRI T1 setup and have been making outgoing calls for several months no problem. I have Zapata.conf setup for fxs_ks on these channels. How do I take incoming calls on these same channels? Do I need to change the signaling? Any help is appreciated. Thank you, Dave -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I'm having a problem with an asterisk server being unable to call certain cellphones and answering machines. Anytime the person answers the phone call, everything works well. But when the call goes to voicemail or an answering machine, I get the error message below: ===================================== *CLI> -- Attempting call on IAX2/u92613106/15551234567 for

I recently inherited a Vertical Xcelerator IP system with IP2007 phones. I would like to use the phones with an Asterisk system instead, but there doesn't seem to be much information on it on Google. Is it even possible? These phones claim that they are SIP phones. Thanks! Kevin -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, does anyone have the setup for go2call ? I have digium boards and quicknet linejacks and phonejacks. The cards work fine in asterisk without the g729 or g723.1 for the phonejack. I will like to do SIP origination using the codec in the phonejack and linejack g729 or g723 and send the calls to go2call. Anyone has the setup for this ? Or similar setup to a SIP provider using g729 or g723

Hi, I'm going to forward SIP request to special outbound SIP proxy with none SIP port. I have this in my sip.conf [sip_proxy-out] type=peer ; we only want to call out, not be called username=408 host=192.168.0.95 outboundproxy=192.168.0.74 port=9097 I want a To: 408 at 192.168.0.95 by proxy 192.168.0.74:9097 but it turns out the "To" also has the port To: 408 at

I am posting this in case someone need help.. ========================================================= YOU THA MAN!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! No sure how I will repay you, but anything you need, just let me know!!!!!!!!!!!!!!!!!!!!!!!!!!!!! Thank you, thank you, thank you!!!!!!!! -- Executing GotoIf("SIP/2007-12c7", "0?4") in new stack

Hi, I have been reading the lartc docs and I am a bit confused. I am trying to do the following: I have 3 computers. A linux router and 2 clients. On one of the clients I run linux and I have a ftp server using passive mode. I want to run traffic shaping on the linux client. It has 1 nic. I want to shape the outbound ftp traffic to 40 KBytes because my cable connection is 45K. I have

Hi, I just wanted to share my script with the list. I have been trying to shape outbound passive and active ftp traffic without affecting inbound and lan transfers. I have tried to do this for a long time and it seems that I have finally figured it out. Feel free to comment on the below script if there is anything that can be improved. It seems to work flawlessly so far. #!/bin/bash

Hi, I am trying to use the following script to limit my passive ftp traffic to 35KBytes. Problem is, it kill''s the entire connection on that computer. The script is running on the same machine as the ftp server. I was hoping to limit the ftp traffic, and only the ftp traffic, leaving the computer. It seems to limit everything, i tried transfering a file with samba and the whole

Hi, I am using the following script to limit my outbound traffic. This scipt runs on a box behind my firewall. It limits my outbound passive ftp traffic to 39K perfectly....just like i want. However, i just noticed that it is also limiting uploads coming to my server. Is there something I can change to make it not limit uploads to my server? #!/bin/bash #shaping passive ftp traffic #

> -----Original Message----- > From: Tom Eastep [mailto:teastep@shorewall.net] > Sent: Monday, May 13, 2002 9:05 AM > To: Steve Herber > Cc: Shorewall Development > Subject: Re: [Shorewall-devel] RE: [Shorewall-users] SMTP outbound > problem (fwd) > > > On Mon, 13 May 2002, Steve Herber wrote: > > > I think we should add an FAQ entry for tcp_ecn. > >

> > Hi All, > > > > Customer is requesting 1 incoming toll free #, that dial out to 4 > > different terminating numbers, not ring all at once but ring #1, then > > #2, then #3, then #4, then back to #1 consecutively on inbound calls, > > regardless if someone is on #1. So this is not like a hunt group, > > more like an agent queue set with a round-robin

I''ve just started to recently look at CBQ and HTB, but I''ve been wondering whether or not something like the following is possible: I''ve got two external links: +--+ | |-----ISP1 |FW| | |-----ISP2 +--+ One link has a bandwidth cap, and the other doesn''t. What I want to do is, punt all outbound traffic out via ISP1

Any vitelity customers with pbxinaflash boxes? I'm able to call in-house, but failing to make outbound calls. My assigned server at vitelity is not reachable. I can ping to my ISP OK. Any help appreciated. Such as actually how to make email contact with support at vitelity. They're not responding. Thanks, Tom