similar to: PRI dropped calls w/ asterisk dropped between pstn & norstar

> -----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

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)

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

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'm attempting to use an asterisk box with a Digium TDM01B as voicemail for an existing Meridian/Norstar PBX with an ATA-2 adapter. We're having problems where hangup is not always (but sometimes) detected. It's not detected probably 70% of the time or more. (The users transfer callers to an extension--caller then has to navigate a menu to get to the appropriate user). After

Hi there! We have an old legacy norstar phone system m8x24-ds ( dr5 ) and a couple of m0x16. It has 5 external analog lines. It has no auto attendant, and no voicemail. So every incoming call is forwarded to a operator, she pick up the phone, talks to the caller and transfer the call to the right extension. We are in Argentina, so buying a star talk is out of the question, there is no selling of

Hi, I have a legacy Norstar system that I'm looking into integrating with my Asterisk setup. My first attempts have worked, which involves a Wildcard X100P FXO card in the * box connected to the Internal ATA (FXS port) on the Norstar system. Calling from SIP -> Norstar works fine, since the SIP caller initiated the call and generally will be sane enough to hangup the phone when

Hello all, I have Asterisk 1.2.1 and a TE110P connected to a Nortel Meridian Option 81C system. The PRI line is currently setup as DMS100. Here are the relevant lines from zaptel.conf and zapata.conf: zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone = us zapata.conf: [channels] language=en context=from-internal musiconhold=default switchtype=dms100

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

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

Does anyone know which signalling protocol works when connecting the output of a NT8B90AL Norstar ATA2 analog terminal adapter to an FXO card? (fxsks, fxsgs, fxsls?) I'm trying to add an Asterisk branch to my Norstar PBX as outlined at http://www.voip-info.org/wiki-Asterisk+Nortel. - Mike

Good day all How do i set a connection between 2 asterisk servers via PRI In Bri I would set one to NT and TE How shoud the zapata.conf and zaptel.conf look And how should the cable be? All I got on the web was to set one to "pri_net"...this cant be all? And the cable > pin1 <--> pin4> pin2 <--> pin5> pin3 <--> pin6> pin4 <--> pin1> pin5 <-->

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

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

----- 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 have Asterisk setup with two PRI's one going to my telco and the other going to a Norstar Meridian system. The Norstar Meridian is sending a BTN number that needs to be passed to the Telco. Is there a way to pass the BTN as a variable in the dial plan? Like CallerID(num)? What is the variable for BTN if so? Many Thanks. -- *** Forrest Beck IAXTEL: 17002871718 jonforrest.beck@gmail.com

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

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

I have not been able to get two B-channel transfer to work on DMS100 PRI. I consistently get the following errors: [Nov 6 11:12:49] ERROR[2774]: chan_zap.c:8178 zt_pri_error: ROSE RETURN ERROR: [Nov 6 11:12:49] ERROR[2774]: chan_zap.c:8178 zt_pri_error: OPERATION: RLT_OPERATION_IND [Nov 6 11:12:49] ERROR[2774]: chan_zap.c:8178 zt_pri_error: ERROR: RLT Not Allowed I have tried on two

Hello, I'm using Asterisk (latest CVS head) to perform outbound call as robot/testing tool for an IVR platform, with a Wildcard T100P configure as ISDN Pri. For develop the exten context script I was using a real PSTN ISDN Megalink (DMS100) to reach the platform and my script was able to correctly detected the DTMF tone send back by the platform to synchronize the script. But for load test,