similar to: E1s ISDN PRI & CPC

Has anyone worked out a way to transfer the Calling Party's Category codes to Asterisk through E1 / T1 connections? I know this is normally available on SS7 interconnects but is it also available to asterisk on the ISDN signalling channels? (I kind of doubt that it is......) Thanks, Derek -- Derek Conniffe Rivertower Ltd DDI: (Local Ireland) 01 201 0146 (International) +353 1 201

Hi everyone, Sorry for forwarding and top-posting this email again but its as if my TDM40b has keeled over yesterday. After a few hours last night and swapping the card to another asterisk server (with exactly the same result) I needed to have the FXS ports working ASAP this morning so I have repaced the functionality of the TDM40b with some Grandstream handytones which I already had in

Derek, could you give me some details regarding the solar power supply you're using for your installation? Thanks! J?rg > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Derek Conniffe > Sent: Thursday, September 15, 2005 12:28 PM > To: Asterisk Users Mailing List -

Hi Derek, I'm not sure your recording will match with my needs. I wanna be able to do this myself with our currency here. Can you just tell me what to use and how to use it ?? Thanks. Daniel. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Derek Conniffe Sent: lundi 7 f?vrier 2005 11:59 To: Asterisk

Hello all, Can anyone help me out with this issue ?? I got ASTCC running, but the audios doesn't match my needs (currency, etc.). is there any way to create my own audios and replace the current one?? Thanks. Daniel. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Someone on the list a while back suggested that if you were having problems with call disconnects, to look into a product from Viking TellecomSolutions called cpc-disconnect: http://www.vikingtelecomsolutions.com/catalog/model_CPC-1.htm I received my unit on Friday and put it into place Saturday afternoon (SBC in this area doesn't supply call disconnect supervision). The unit was acting

Hi everyone, I decided to have a look at SIP & NAT again and I've been at it for a [quite a] few hours but typically nothing is working for me. Actually I'm not sure if SIP and NAT can ever work but some emails on this list do suggest that someone has got it working, once, maybe. I'm experimenting with a ZyXEL 2000W [WiFi Sip phone] which supports "Outbound Proxy",

Hi, Finally, the long promised CPU performance counter provider has arrived into Nevada build 109! For those that need an injection of excitement at the start of the week, you can check out some usage examples on my blog: http://blogs.sun.com/jonh/entry/finally_dtrace_meets_the_cpu Also, you can find a new section for the provider in the DTrace docs:

Hello, I want to use Asterisk PBX in front of my old, legacy PBX. The legacy PBX can be outfitted with caller-ID and is already able to handle Calling Party Control Signal Detection (this is a Panasonic KX-TD1232 Super Hybrid PBX. My question is how would one enable Asterisk to control the TDM400P/FXS port to provide to the /FXO CO port on the legacy PBX, support for proper answer supervision/CPC

A newbie on Zaptel asks: Is there any way I can force CPC on a Zap channel - Digium TDM four port FXS card? /O

A plea to all! Has anyone had any success with two or more avm fritz pci cards with either misdn, chan_misdn, or chan_capi, and any version of linux 2.6.x? I have managed to get misdn to load under 2.6.13 and detect two cards using misdn-capi and chan-capi (using capiinfo and capi info under asterisk) - but the second card/controller doesn't answer or dial calls. But if I try misdn

Hi Ray, I was wondering if the "qualify" option is used [in sip.conf] to keep a connection (from the SIP phone inside the firewall to the Asterisk server outside the firewall) open then would the firewall not allow two way communication without incoming port mapping/NAT (providing that the SIP phone started "talking" first)? I'm not sure about that - I'm being

Hi everyone, I've just upgraded from Asterisk 1.0.X to 1.2.5 (and the matching latest libpri & zaptel from www.asterisk.org). All compiled fine but now I've got a weird problem with my EuroISDN lines connected to a Digium quad E1 card: - Asterisk suggests its answering calls and playing voice prompts to the incoming call but, in fact, its not really - the caller hears a couple

Tracing Fans, I know it''s been a long time in coming but the CPU Performance Counter (CPC) provider is almost here! The code is currently in for review and a proposed architecture document is attached here for review. Any and all feedback/questions on the proposed implementation is welcome. Thanks. Jon. -------------- next part -------------- An embedded and charset-unspecified text

I'm wondering if anyone can recommend one over the other. I'm mostly interested in running open source solutions, so I would prefer if your recommendations are within the open source arena. Basically, I contemplated the idea of using SER as a NAT Helper and possibly as a SIP server for a portion of our user base. We prefer to have Asterisk in the mix because of the additional

Hi everyone, I'm a bit unclear about how PRI voice channels use CPU usage in an asterisk box - it is like a codec conversion or does a PRI channel have a generic codec itself? I am wondering what the CPU requirements are to, for example, handle all 120 channels on a TE405p and then trunk all the conversations elsewhere over an IAX2 link. Thanks for any advice, Derek Conniffe ---

Hi, I would like to realize a voip testbed that should simulate the scenario in which two companies have an asterisk PBX connected through a PRI-ISDN to the Telco operator. I have no experience of T1/E1 connection but I think that the above could be relized with 3 asterisk boxes equipped with Digium TE405P cards. One of the box will represent the Telco, the other two, the two companies PBX. I

> > > > I would be grateful if anybody could tell me what I should > > tell Verizon > > in NJ so they would enable "disconnect supervision" for my lines. > > > > Apparently "remote hangup notification" or "disconnect > supervision" or > > "calling party control" is NOT the magic phrase for them. Although >

Hi, I have not been successful in working out how to dial a # within a phone number. EG: exten => _12345,1,Dial(Zap/1/0868563823#,5,t) or exten => _08XXXXXXXX,1,Dial(Zap/1/${EXTEN}#) I'm trying to append a # character so that I can use a cellsocket (mobile phone to pots adapter) connected to an x100p. I think that asterisk is simply ignoring the # character. The docs on

I try to install astcc. Make install gives me: DBI connect('database=astcc;host=localhost','astcc',...) failed: Unknown database 'astcc' at ./astcc-admin.cgi line 67 DBI connect('database=;host=','',...) failed: Access denied for user: 'root@localhost' (Using password: NO) at ./astcc-admin.cgi line 60 Ignoring that (!!! ;-) ) and going to the