similar to: Location with PRI / Analog lines

Hi there, I am running a PBX under asterisk 1.6. I have few FXO analogue lines connecting to PSTN. These lines are in a hunt group. I trying to make my extensions to dial 91, but this is a bit scary, I mean if somebody make an emergency call after hours and without completing call is not able to tell his/her location. How can I make 911 call center to know the exact location of my extension. I

Hi Shahnawaz Have you considered how you are going to address location issue for Mobile users calling 911. You should think of SS7 MAP/TCAP to atleast know their Cell ID Regards Sam > Thanks very much everybody who contributed their thoughts. I would try > to get some DID's so that each physical location can be identified by > 911 call Center. > > Regards > > Shahnawaz

Hi, I am trying to implement 911 funtionality in my PBX. A call should drop if all lines are busy. Here is my context nineoneone from extensions.conf [nineoneone] exten => s,1,Set(SET_EMERG_FLAG=0) exten => s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK}) exten => s,n,Set(EMERGENCY=1,g) exten => s,n,Set(SET_EMERG_FLAG=1) exten => s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM})

hello list, i have a problem regarding the synchronisity (clock source) when using multiple cards. e.g. when having connected one PRI port of our TE410P to the telco, i need to have the analog card like the TDM400P or a B410P synchronous to the clock of our telco provider. otherwise faxing on the analog cards does not work or i get cracking noise or even hangups on my BRI lines, due to bit slips.

Hello All, I'm trying to connect Asterisk to an Executone phone system with an analog DID card and I'm hoping someone can help me figure out what I'm doing wrong. The Executone DID card provides battery to the telco, when the telco wishes to dial a DID it goes off-hook, waits for a wink from the Executone and then dials the last three digits on the number with pulse (as opposed

I've some DID's that I'm using for in-bound faxing, but I'm having some trouble with getting that working perfectly on my T1. So I'm thinking of pointing them to an analog line. Will the DID's simply come in over the analog, presumably sending the DID digits via DTMF? Or is that not something that'll work? Thanks, -Ken

I am using Verizon FIOS to my home. I subscribe to a 5 MB down 2 MB up data package. I continue to pay for a standard voice line in addition to the broadband connection only for local calling, fax and emergency 911 use. The way it works is that the fiber optic connection is terminated on the house via an ONT (optical network termination). The ONT can provide connectivity for three types of

Perfect, thanks very much hth. I just set it to unknown, but it doesn't work. Have I to use also prilocaldialplan ? Thanks again Giordano ________________________________ Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Colin Anderson Inviato: gioved? 29 settembre 2005 16.22 A: 'Asterisk Users Mailing List -

I am a Linux sysadmin who has been tasked with developing the phone system for our nonprofit's new US headquarters building. We cannot bring our legacy phone system with us, so I am building this completely from scratch. I have already read "Asterisk: The Future of Telephony" and done a fair amount of googling. I am completely sold on Asterisk, and the new building's

> -----Original Message----- > From: john lawler [mailto:maillist@tgice.com] > Sent: Wednesday, January 07, 2004 1:38 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] yet another question on DID trunks > > Hey Steven, > > Sorry to bother you yet again w/ a question on my seemingly endless > quest to get DID trunks setup for a customer. > >

Hi guys, I'm new to the telco game and still pretty new to Asterisk, although I've been using it for a couple of months now and like most of what I see. At my office, we've got a small two extension setup w/ two Digium cards for a single FXO line and three FXS extensions, but I'm also working on designing a larger installation for a customer which will involve ~16 analog

I have a couple of DID's with LiveVoip and am having major DTMF issues on incoming calls. I am connecting to them through IAX using ULAW. When someone dials one of these DD's (from a landline) they are for the most part unable to navigate the IVR menu successfuly. I would say the failure rate is greater than 80%. For example if the caller presses 5 sometimes * will see the DTMF as 55 or

Just turned up a new PRI with DID's in the US. I'm receiving 5 digits of the DID numbers as I requested. Assuming I have 100 DID numbers but only define 50 of those in extensions.conf, is there an easy way to send the incoming calls for the 20 undefined numbers to a common resource (ivr, operator, or canned message) without having to define each one?

Hi, I followed the how-to from http://www.alcatelunleashed.com/viewtopic.php?f=44&t=840 All works fine except for Asterisk->Alcatel calls. Actually, calls from Asterisk to analog extensions on the Alcatel work. However, calls from Aserisk to digital extensions on the Alcatel 4400 do NOT work. I get this error in the Asterisk log: -- Executing Dial("SIP/4053-0823dd48",

Anyone have experience with T.38 passthru in Asterisk 1.4 to a MAX TNT Media Gateway? We're experiencing sporadic results... Topology is described below... Thanks in advance.. -Joe Traditional Fax <-Analog Line-> ATA <-SIP-> Ast1.4T.38Passthru <-SIP-> MAX TNT <-PRI-> PSTN -------------- next part -------------- An HTML attachment was scrubbed... URL:

This depends on what you mean by ?not involving the service provider.? If you are literally forwarding calls that come in on the PRI back out on the PRI, the most efficient way is with Two B-Channel Transfer (TBCT). Check it out in the wiki. You need to make sure your carrier supports the feature. When you want to do a ?transfer,? you have an incoming call alerting or answered, you

Following the successful deployment of asterisk servers at several of our branch offices, in the near future, I'll likely be implmenting an asterisk server at our HQ. We currently have a T1 PRI terminated on a legacy PBX. I'll be doing a stepped deployment in which, via a dual T1 linecard, the asterisk server will initially pass all incoming/outgoing calls directly through to the PBX.

Is it possible to send CLID NAME on a PRI? The numbers we send out are being received by telco and propagated, but the names we send out are not showing up. Is this a feature in PRI? Do we need to set PRI_NET instead of PRI_CPE? Is this just not possible? Is this a telco config issue? Thanks for your help... I've read voip-info, and various other sources, and search engines, and google...

For outgoing calls made on our PRI circuit we are setting the Caller ID using the format Exten => _9XXXXXXX,1,SetCallerID(1601XXXXXXX) The monitor shows that the CallerID is being set to the specified number, but yet when the call is received on the user end the ID is always the base number of our DID. For example we have 8600-8650 as DID's but the callerid is always 8600 regardless of

Is there any known reason that the DISA() routine should behave differently than WaitExten() as far as recognizing DTMF tones? If not, I suspect there's a bug here. Try it yourself--two DID's on our PRI, numbers below let you test each routine: It is my observation that some setups/phones DO and some DO NOT express this variance. --I could not show any variance on a sprint mobile phone