similar to: Network design question

Hello. Second question. Should I be asking this on the dev list (that's not the question by the way). Q. - there are several mentions on the list that asterisk :- "can interoperate with almost all standards-based telephony equipment" "interconnection with digital and analog telephony equipment" "visual message waiting indicator" etc. etc, That seems

Hi, all. I've got an old Telrad PBX with an Emagen(?) voicemail box. The VM box, itself, is beginning to show its age. Big-time. We're thinking it might be time to look for a replacement. I'd love to install Asterisk with an FXO card or something, but I don't think it supports whatever protocol legacy PBX's used to speak to VM systems. If someone can tell me I'm

Hi all, I've seen a couple of posts recently from people who are doing something with Asterisk and a T100P and a Nortel PBX. However it's not clear exactly what they are doing. Does anyone know if it's possible to use a T100P or T400P as a DSX-1 interface to connect to a T1/PRI CO module in a Nortel PBX? We have a Nortel Norstar Modular ICS PBX and I'd like to plug its CO

I am trying to integrate a Nortel Norstar system with an Asterisk service using a TDM04B card (4 fxo). So far everyting works from Asterisk to Nortel. The problem is when someone dials from the Nortel PBX to the Asterisk server. Asterisk answers the call and provides a dialtone (with DISA) but appartently the DTMF tones are not passed to asterisk and the call cannot proceed. This only

I'm looking to build a VoIP solution for 100+ service vehicles that have WiFi hot spots installed (with cellular uplinks). Currently we are trying out Skype wireless handselts and Majick Jack. I'd also like to consider an Open Source solution that can bring the calls back to our data center [possibly integrated without our existing BCM 3.x VoIP PBX]. For hardware someone on the IRC

Hi, Some time ago, there was a discussion about the inability of nortel meridian pbx to dial analog tones thru an meridian ATA, and the work arround was to enable 808 function that makes the dtmf tones long for the current call. The nortel meridian is connected via a nortel ATA to a TDM400 to a FXO port. Anyone can say me who to actually use that function (you dial something or is pbx

Hey gang, I'm looking at escaping from a Nortel Meridian CISC system to Asterisk/Digium/SIP phones. I'm currently in the testing and proof of concept phase. I'm going to need a SIP phone and don't want to re-purchase and have "orphans" around. We currently run Nortel 7310 phones and they work great. I'm sort of overwhelmed by all of the different IP phones. I

Dear Harry, Thx for the explanation. My team manage building's PBX that use Asterisk 13.x. We use Asterisk PBX for this buildings that have apartment and office customer. >From my Asterisk PBX, we connect to IPPhone (yealink) or ATA Converter (cisco SPA112). Others are using PBX like panasonic analog, audiocodes SBC, etc, and we use ATA Converter to convert from SIP to Analog (CO Line)

Hi, Is there anyone here who has experience connecting Asterisk (ver 13.8) with PBX Panasonic KX-TDA600 ? The architecture more less like this : Telco Sip Trunk ---> Asterisk 13 ---> Panasonic KX-TDA600 ---> Phone / Fax Thanks in advance, Regards, Ikka - Jakarta, Indonesia -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm trying to find a high port count ATA device. We have a lot (up to 110) analog devices that we need to bridge to IP. Rather than go out and buy 110 ATA's, it would make more sense to buy a single chassis type device with some number of ports and blades. Anyone know if such a device exists? Thanks, Doug. -------------- next part -------------- An HTML attachment was scrubbed...

Dear All, I have a setup where three errision PBX's (head office & 2 branch offices ) are @ three different location. Head of has a errision BP250 where the branch officers has Errision BP50's. I wana connect all three PBX's through IP link's. My idea is to connect BP250 ( head office ) to asterisk. This has to be done using analog interfacing coz there is no E1 / or T1

Hello, Our university has to upgrade soon its old Nortel PBX's which holds around 10,000 extensions tied to 5 PBXes. Up to now we thought about commercial solutions but now there is a window openning for open source solution. However, I need examples to convince that this solution is feasible, and preferably at other universities. Are there any pointers for such installations?

Hi, I'm having trouble getting caller*id to appear on my phone connected to an ATA186, and being called from Asterisk. Does anyone out there successfully see callerid on their ata186-connected phone? The "From:" header in the INVITE to the ATA seems to have the "right stuff" - eg From: "Study phone" <sip:6002@195.217.255.45:5062>;tag=as412db061 But

Hi, back in the old analog telephony days there was "digital" PBX-es and digital "system" phonesets. This phonesets have had many individual illuminatable buttons connected with extensions. The PBX can show on the buttons if some extension is ringing (blinks) or busy (constant light), and the user can transfer the call with one touch (pressing one of this button). I search

Thanks everyone for the input. After pricing everything we need out, it's not worth trying to get our old system to work, so I've pitched ditching everything and starting over. I'm very excited and hoping they'll go for it. Regardless, I'm going to throw a box together for my house, we have no home phone (just cell phones) so this'll be a great way of testing. All

Has anyone integrated Asterisk for Voicemail with a Nortel Meridian phone system or know if this could be done? Thanks, John Haigh

Hello. I am going to be putting together my first * system using FXO/FXS interfaces. All the systems I have set up thus far have been pure VoIP setups. The system I need to set up should have 3 FXO interfaces and 1 FXS interface, as well as several SIP phones. I have noticed people complaining about Digium's TDM cards - are these isolated incidents or are these cards unreliable? I intend to

Hi, I know this is slightly off topic but I figured the knowlege here is probably the best on the subject.. I want to setup remote offices with 4 to 6 SIP phones (SNOM 200) using ADSL and the internet to connect to the Asterisk box.. These phone will be behind an ADSL router using NAT... I don't want to setup another Asterisk system in each office so IAX is not an option.. I could use

Dear list, I've got the following setup : [FAX-ATA]--[PBX LAN]--[Firewall]--[PBX WAN]-----[upstream SIP] On the PBX's we run Asterisk 1.4.33 with t38pt_udptl=yes in [general]. The FAX ATA is a Teles VoIPBox with T.38 support (that works). On the PBX WAN, i see the following in udptl debug : Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 184, len 32) Got UDPTL packet from

We would like to present a toll free CallerID when making outbound toll calls. In the past, when our PRIs were directly connected to a Nortel CS1000 we could do this, without issue. Now that the PRIs are front ended by a mediagateway facing asterisk, we can no longer do this. Is it possible to set the billing number via a SIP header and set what should be presented as callerid as another header