similar to: 2B Channel Transfer on XO-based T1

Ok, so I've made progress on 2BCT (2 B-Channel Transfer). I'm assuming that the debug info below shows that XO doesn't have 2BCT enabled on my line, but if anybody can confirm that'll let me be way more indignant. J -- Native bridging DAHDI/1-1 and DAHDI/3-1 > Protocol Discriminator: Q.931 (8) len=28 > Call Ref: len= 2 (reference 801/0x321) (Terminator) >

Hi all ! We currently have an asterisk box that is rather old (runs Asterisk 1.4.21.2), and it's connected to the PSTN with a sangoma A104d card. Now we have a new PRI at another location, and I use that occasion to build 2 new servers, one to replace our aging one and a new one for this new pri. So I downloaded the lastest libpri / asterisk / wanpipe driver, but the previous version of

Hello, A client has asked for Two B channel Transfer capability (known as TBCT or 2BCT, similar to other features such as ECT, RTL and Q,SIG Path Replacement) in a new Asterisk system and so I researched the capability and came up with quite a few gaps in documentation.

Hello list i have an issue with my dahdi_channels.conf in span 1 when i use it like below i can do my outband calls without issue ; Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER) group=0,11 context=from-pstn switchtype = euroisdn signalling = pri_cpe channel => 17-31 context = default group = 63 but when i add in channel 1-15 like: channel => 1-15,17-31 i receive all

Hi Guys, I am sorry if my issue is not related to this but I think it is. I have a PRI with Bell Canada and when I dial in and have the call transfered to a context to dial out and then have those two channels bridged, the call disconnects with cause 16 just exactly as Jay R. Ashworth shows in his CLI output. Bell Canada support RLT or know as 2BCT or TBCT to some but we have not requested that

Calls that come in on DAHDI FXO ports are routed to [context], extension 's' INSTEAD, I would like to route specific ports to specific extensions, For example: I want DAHDI/1-1 to go to 1234 I want DAHDI/1-2 to go to 2345 I want DAHDI/1-3 to go to 3456 ...etc What is the CLEANEST way to do this? Yes, I can create a private context for each DAHDI channel but that seems messy and

hello everyone, i have question about fax detection on dahdi channels. does dahdi channels detect fax and pass it? if yes, does it detects both types of fax (g711 pass through and T.38)? finally, how can i enable it on dahdi_channels? i set faxdetect=both in chan_dahdi.conf but dahdi can not pass fax(i just wanna pass fax not send or receive it). any comments or hints are really appreciated. SAM

Thanks AJ and David, We were actually using GSM gateways by setting busy forward number on the SIMs and just giving busy signal on every incoming call, telco took care of the forwarding and the line was free within seconds. Now we need to scale up the setup but GSM gateways a very very expensive if we want to scale upto a 1000 DIDs, which means thousand SIMs and a gateway/gateways big enough.

I thought XO was reselling Level 3s (old Genuity assets) network/voip just like Qwest ? -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of jk@bingoconsulting.com Sent: Wednesday, October 11, 2006 3:38 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Load balance Asterisk server,when it is a SIP

I have an XO t-1 line which includes VOIP on 5 lines. When a call comes in, it drops the bandwidth on the t1 and when the call is over bandwidth is restored. The provided a channel bank which does this. My question is, if I use asterisk, then am I losing double bandwidth for each call? For example the only way I guess I can connect to the lines is from the 66block they provided which

Hey Don, How are you? I may be heading your way in the next month or so. Have to meet with a guy in Eden Prairie, and stop off at my brother/sisterm-in-law's as well. Got a question for you - with TBCT, who pays for the call once it is transferred? Still me as the owner of the trunk? Lets say I take a call that was dialled locally (caller believes this is "free"), and I do a

Hello All, Anyone out there using ISDN PRI from either MCI or XO/Allegiance? Gotta make the choice today and the difference per month is only about $25 in favor of MCI. Billing is pretty much the same between the two so I have pretty much no point of reference on which to choose. Any thoughts from anyone experienced with these two compnies would be greatly appreciated! Thanks, Wiley

Hi, I have an odd problem where when a call is made to any landline phone the callerid number shows up on the caller id device with no problems. However when I call a cell phone from Sprint or Verizon it always shows up as unavaliable. I've tried multiple cell phones from each carrier. Though today I called a t-mobile phone and the callerid came through fine. Is this an * level problem

There was recently a question on the R-help list about the eee pc. I had a related question about the XO laptop from OLPC (laptop.org). Has anyone looked at the development environment sufficiently to determine if it would be possible to create an executable image for R? The laptop itself only supports Python, Javascript, etc. but it is running a real Linux operating system.

Hi: I am useing asterisk 11.12. I use G722 as preferred codec for my ip-phone. and my PSTN DAHDI use alaw. G722 is great when ip-phone talks to each other. but when ip-phone dialout to PSTN DAHDI, G722 is not great, since it is need to transcode to alaw. so I try to change the codec when dial from SIP to DAHDI. I tried to use IP_CODEC/SIP_CODEC_OUTBOUND at dialplan. but the SIP

Hello All, I was wondering. What are the primary advantages to using a PRI over a T1? As I understand it, the PRI terminates very fast, meaning you can do immediate answer and dial... This is very handy on the BRI line I have on the asterisk. Can T1 signalling also do immediate answer, or does it just behave like a channelized pots line and ring as usual? I am trying to determine if I should

Almost all DID providers refer to XO and L3. Are they the only sources for DIDs in USA and everybody has to go to them? Could someone explain how exactly it works? Thanks -- Zeeshan A Zakaria -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070531/170b14cb/attachment.htm

As far as video playback is concerned, it seems to mostly come down to how the video is compressed - although Gnash (the open flash player) still has a ways to go to play flash media as well as the closed source flash players. For the Colingo activity and library, we've found it very difficult to find optimum settings for flash and instead have decided to go with Theora. We've managed to

Hello, We currently have a scenario where a large percentage of inbound calls on a telco PRI are intended for professors who aren't currently in their offices/at their desks. My question is, is anyone actively using the Asterisk "RLT" (Release Link Trunking) feature to bounce these sorts of calls back to the telco? The idea being to forward the call to their mobile phone without

Greetings, This may be a bit arcane but does anyone know what the contents of a facility message should be for initiating a TBCT on an NI2 ISDN. I've been trying to get it to work on a DMS100 for the last four months to no avail. The message I am currently sending makes it to the switch but is returned with unknown message. Perhaps someone here has done it before and can help me out.