similar to: PRI stopped accepting calls

I have a two port TE205P Digium card. I have set everything up to create a native zap bridge between the two spans. Everything works perfectly except one thing. Our telco has a "password" that has to be entered as soon as a long distance call is made. So if I dial a long distance call from my meridian system, asterisk bridges the call between two channels, my telco picks up and gives

All good information, thanks. However this is private network between Asterisk and a Norstar MICS about six feet away. So I'm holding both ends of the link. :-) > -----Original Message----- > From: David Troy [mailto:dave@popvox.com] > Sent: September 16, 2004 4:57 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] No Caller Name

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Gary Reuter > Sent: Thursday, August 11, 2005 12:59 PM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] PRI dropped calls w/ asterisk dropped > betweenpstn & norstar > > > I poured over my logs most of

Hi! I have the chance to play with a couple of E400P cards, each installed in a IBM e330 XSeries servers (2 x 1GHz P-III CPU 2 Gb RAM, 36Gb SCSI HDD with RH8.0 2.4.18-smp kernel), and I'm trying to test/benchmark this e330/E400P combo generating calls thru /var/spool/asterisk/outgoing One e400P if doing the carrier work making calls and the other just receives the calls: Server#1

Ok Guys, I've done a tonne of hunting around on this problem, but can't find much help. I'm running: asterisk 1.4.19.1 libpri 1.4.3 and zaptel 1.4.9.2 which I believe has been modified by RedFone to add the ztd-ethmf module. My interface is a RedFone foneBridge2 4 Span; and I'm connecting to a BT E1 PRI / ISDN30e with 15 lines on span 1, and a legacy Panasonic PBX on span 4. Upon

Hi, I dropped an asterisk server with a TE405P between a Norstar Meridian PBX and it's PRI PSTN connection. Everything seemed to work fine using a pass-thru-type dialplan configuration... except now we've realised that outbound calls to celphones get dropped upon connect, but not on every call (almost always the first try, but not the second). All other calls to non-celphones had no

Hi, Looking through the source code, I see two functions in the 'libpri' code for message-waiting: pri_mwi_activate, and pri_mwi_deactivate, but they are not used by anything anywhere. Is there a sample application somewhere which makes use of these functions? I believe that the ISDN MWI is used when a Norstar MICS has an external voicemail provider. I don't know exactly how this

Hello, Quick Diagram: Telco-PRI -> Asterisk <- Norstar PRI -> Norstar PBX (DMS100) (TE405P) (DMS100) | | V Cisco 7960G (SIP) I'm trying to change the Origination Number on my outgoing PRI, and running into a weird problem. If I make a call from a SIP extension off asterisk using the following context:

I have a PRI link up and running between Asterisk and a Nortel Norstar MICS v4.1 . I'm having a problem getting the textual Caller Name across the link from Ast to Ns, however numeric Caller ID arrives and displays fine. From Ns to Ast both elements come through fine. I'm forcing dummy values for testing using: exten => s,1,SetCIDName(Test) exten => s,2,SetCallerID(1234561234)

To All; My current issues is a 5 second delay for call that is being transferred from the Norstar units to the Asterisk servers VIA a PRI. Is their anything that can be done to speed up the transfer on the Norstar. Below is my current phone config. < Norstar1 >----PRI----< Asterisk-1 >----IP-WAN----< Asterisk-2 >---PRI---< Norstar2> The Norstars are MICS 0x32 4.1

----- Original Message ----- > Message: 3 > Date: Thu, 16 Sep 2004 07:57:15 -0400 (EDT) > From: David Troy <dave@popvox.com> > Subject: Re: [Asterisk-Users] No Caller Name sent from Asterisk over > National or DMS100 PRI to a Norstar MICS? > snip> > > I have a PRI link up and running between Asterisk and a Nortel Norstar MICS > > v4.1 . I'm having a

I am currently working on an Asterisk test system, and will be presenting a demo to the Board of Directors tomorrow night. I want to make sure I have all of my ducks in a row. The Asterisk system will be used to replace a Norstar MICS. The location has two PRI's coming in, with a few hundred DIDs. I know how to make * use the DIDs incoming, and I know how Nortel uses the DIDs. Now for the

Out of the blue i started receiving the following error on my PRI line which connects my asterisk server to a Norstar 0x32 key system. The asterisk zaptel.conf file was configure as follows and this config worked for 6 months until friday. Nothing was changed on either system prior to friday. here is teh zaptel.conf span=1,0,0,esf,b8zs bchan=1-23 dchan=24 span=2,0,0,esf,b8zs bchan=25-47 dchan=48

The last post I saw regarding this was June 2003 (before my time with *, I started in September): http://lists.digium.com/pipermail/asterisk-users/2003-June/013324.html Is there any support for Q.SIG in * at all? The Norstar MICS uses a MCDN (Meridian Customer Defined Network) key to enable the "SL1" protocol which from everything I've been reading is just Q.SIG. This protocol

Hi, Is it possible to set the pridialplan (or prilocaldialplan) on a call-by-call basis, or for a particular PRI channel-group? >From my experimentation so far, it appears to me that the pridialplan and prilocaldialplan are global in zapata.conf, unlike other options such as switchtype, signalling, and context. I currently have two PRIs from different providers (and different

I have a 2 port T1 card doing PRI passthrough, Span 1 answers from Telco, Span 2 sends to my existing phone system(Nortel). My Span1 gets sent to the context from-pri, detailed here: [from-pri] exten => _49XX,1,Set(CALLERID(all)=${CALLERID(all)}) exten => _49XX,2,Dial(Zap/g2/${EXTEN},,twk) exten => _49XX,3,Congestion() exten => _49XX,4,Set(CALLERID(all)="") exten =>

I have been investigating more tight integration between * and the Nortel MICS... it appears that it is at least theoretically possible to have * store voicemail and log which stations call where. Both require a T1 card. The T1 card requires either a clocking module or the 6-port fiber module to provide T1 timing. Naturally a T100P or TE405P is required on the * side. To log which

Unfortunatly no on both counts. The arrangement right now has: PSTN Trunks & Stations <-> Nortel Norstar#1 <-CT1-> Asterisk#1 <-IAX2-> Asterisk#2 <-CT1-> Nortel Nortstar#2 <-> Stations The Asterisk boxes provide Voicemail to their sites Norstars and intersite calls over IAX. Local Voicemail works flawlessly at each site but there have been reports of PSTN calls

I am displaying a collection of expenses in a form the view in which the fom is displayed : ... = simple_form_for(@project, :html => {:class => ''form-stacked'' }) do |f| - unless @project.expenses.count > 0 No expenses - else #expenses = f.fields_for :expenses do |f| = render "backoffice/projects/expense_fields", :f

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