similar to: Asterisks integration with pre-existing PBXs

Are there any other codecs that can be used with the 7960 and the ATA-186? I have been using the default gsm codec and wanted to see if I could make use of something a little less bandwidth intensive. Kim Callis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030723/34e950e0/attachment.htm

Hi! Sure, just look for: Wonder Shaper. It's a HTB based shaper configuration wich have some very good features, I use a variation of that here at my College. http://lartc.org/wondershaper/ It is the page (a simple google search). Also make sure to uncomment the line tos=lowdelay in every config file of asterisk that have it. Hope it is usefull, sincerely, Ildefonso Camargo

Hi bkw Yep, which is going to be a huge problem since it's only taking a line and not doing any transmittal until after you get a line out, the line of course is being rejected before I can even get there :( Of course I can't even establish connectivity to the telco whilst having it peered to the PBX too due to the D channel issue :( Lee From: "brian" <brian@bkw.org>

Hi all OK this may sound like a good one but maybe someone can tell me. Simple context is - I want to unplug my existing conventional PBX from the Telco and place * with it's TE410P in between. Now the difficult part, the existing connection is E1 PRI (Q.931) with 6 B-channels. I need to be able to trigger a D-channel to the old PBX and a D-Channel to the Telco (Not BT!). Next I can put

The networkmag@cmp.com appears to be broken. I dug around the magazine contacts and found Doug Allen, senior editor, you can send comments to dallen2@cmp.com . I didn't get a bounce back from that e-mail so I assume it made it to the editor. JR

* Brethren, It's a sad day in our community. Please join me in a moment of silence for the death of responsible journalism. Silence.....................good enough. This article goes on to tell about Pingtel's announcement of forming the "first open source community aimed at creating SIP based servers".

Hi all, Look at the following "stupid" function that is created just to show the problem: outlayer <- function(x) { inlayer <- function(z){ e1 <- parent.frame() eval(expression(y<-log(z)),e1) } inlayer(10) y } I have a compilation error when running this. My purpose is to create "y" variable equal to log(z) in the frame of "outlay". How can I

Hi All, I am new to both Asterisk and PBX stuff. I have 3 Tohiba PBXs in 3 separate offices as follows, Toshiba Strata dk28 Toshiba Strata dk280 Toshiba Strata dk8 I need to install 3 Asterisk servers in these 3 locations and integrate them with each of the Toshiba PBX s. This is to give IP Phones/soft phones to the users and to route these VOIP calls through the PBX to POTS. What are the

Hi, > I've never run against a commercial PBX that didn't need > maintenance. Acknowledged. > VM hard > drives fail, > ... > Asterisk is > every bit as stable > as the old-gen KSUs and PBXs. There are big differences. As I know of no other PBX that uses 'consumer' hardware, asterisk is also struggling with problems in the underlying Hardware. And

Bicom Systems releases outCALL, an Asterisk open source Outlook integration LONDON, UK (11th April 2007) - Bicom Systems announced today it has released outCALL, an open source desktop application allowing integration Microsoft Outlook. OutCALL allows users an easy way for placing and receiving phone calls integrated with users Outlook contacts. "The open source PBX market needed

Hi all, I'm just curios and would like some input from the community on this one. We're busy budgeting for a couple of new servers and I thought it would be good to try out the Core i7 CPU's, but see the majority of them don't offer VT-d, but just VT-x. Looking at the LGA1366 range, only the "Intel lga1366 i7 980XE" (from the list of what our suppliers stock) have VT-d,

I noticed that there is a soft button for transfer when you initiate a call. I pressed it, and it actually put the call on hold, although I was able to call another extension. Is that soft button functional? And if so, how do you make use of it? And if not, how does one transfer a call? Kim C. Callis -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi all, i need an asterisk board that can support up to 30 FXS ports. i found the following on digium: TDM400P. It only has 4 port, does it mean that i have to buy 8 of those to have my 30 FXS line? Or is there any channel bank solution that digium or other providers can provide us with in order to do that. Experienced asterisk user, how do you manage to make PBXs for 30 to 100 regular phones?

Is it just me or do others have a problem with the ATA-186 de-registering? Every couple of hours, if I don't make use of the ATA connected line, I find that I have to unplug and let the ATA reboot. After that it is good to go for awhile, but eventually I have to repeat the process. My ATA sits behind a NATd firewall, any ideas what might cause the de-registration? Kim C. Callis

Hi List, Merlin Magix hardware v02 I'm trying to get asterisk to act as a voicemail server for a lucent merlin magix PBX that we purchased used. We have 4 FXO channels between the two PBXs on a Sangoma A200 card. The 770 dialgroup is working properly, in that calls to 770 are answered by Asterisk. The magix is sending mode codes in the format #XX#XXX#, where the 2nd block of digits

We're getting ready to ditch our hosted virtual PBX for an Asterisk solution and I've got a couple questions that probably come from experience; we're looking to host two PBXs on the same box if the PBXs are identified by different DID numbers, and I was curious if: 1. Is it possible to have duplicate extensions between the two PBXs? Eg: 555-1111x100 and 555-2222x100 on same

As I was working on my extensions.conf file, I started to segment calling privileges. For the everyday workers, I don't free reign to LD access unless it is business related. So I was wondering if there was a way to implement some type of accounting code to be entered before accessing LD, which of course would be noted in the CDR (however it is implemented, either comma delimited or MySQL).

Yesterday, I started to experiment with Cisco to Cisco SIP calls using the g729 codec. According to the documentation, both the ATA-186 and 7960 are able to make use of the g729. >From an earlier e-mail, I made a change to the configuration of the ATA, changing the values: LBRCodec:3 RxCodec: 3 TxCodec: 3 The first thing I noticed was that when I did a sip show channels, the format had

Dear, I have two Asterisk PBXs with an E1 interface/RJ-45 port in both boxes. I need to connect both PBXs with E1/R2 and UTP cable. What are the requirements to deploy the UTP cable ??? Straight-through or crossover ??? What are the pinouts in both peers ??? Thanks a lot, Alejandro

Alright, I am basically cheap, and I have a cellular plan which allows for free incoming calls (Nextel). I was wondering if there was any way to do sort of a dialback trick in the extensions.conf. I call into the system from my cell phone (maybe via DISA), I dial an internal extension, and dial a phone number. Then * sends to my cellphone the number dialed thus giving me a in call on the cell. Or