similar to: Driving MWI on Norstars (was Maximum tollera ble lag/jitter...)

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

Hello, This is the sort of question that could interess others, so you should have CCed it to the list. As for the question, from the package vignette, section 3.3.6: "writeWorksheetToFile() is a wrapper function, calling loadWorkbook(), createSheet() and saveWorkbook() functions subsequently. It therefore allows for writing data into worksheets of an Excel file in one call." To

Hi- I have an upcoming application that requires use of PRI channels that are primarily used for high-volume incoming traffic, but that are to be used for outbound calling as well. Of course, one option is to have dedicated outbound channels reserved, but this is an inefficient use of channel resources. Normally PBX's are designed to have the CPE yield to an incoming call if a particular

> > vorbisfile.o needs to be removed from the lib/Makefile in LFILES since > > vorbisfile.a is a seperate lib now. As it stands I get unresolved > > symbol errors when linking with them, so this is a fairly important fix. This is an error in the build, yes, but I don't see why you'd get unresolved symbols. *unused*, yes, but not unresolved... > > sharedbook.c

Forwarded, so the list knows about it. I'm making the changes now. Monty > Hey Monty, I have a few fixes that need to get made to the CVS tree to > fix a couple of broken things. None are worth sending you a diff. > > vorbisfile.o needs to be removed from the lib/Makefile in LFILES since > vorbisfile.a is a seperate lib now. As it stands I get unresolved > symbol errors

> -----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

> -----Original Message----- > From: Chris Shaw [mailto:chriss@watertech.com] > Sent: September 7, 2004 4:40 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Maximum tollerable lag/jitter for IAX2 > w/ojitterbuffer enabled? > {clip} > > If you can reproduce it, this smells like a bug... IAX runs over TCP and TCP >

I've noticed a problem with calls to Hangup when talking to my Norstars over channelised T-1 E&M trunk lines - it's been present since I started to fiddle with Asterisk last December and it's still present in 'Asterisk CVS-HEAD-08/13/04-10:37:13'. Specifically, when a call is connected to Asterisk from the Norstar DTI card to my T100p I get the following conditions

Greetings, This may be a bit arcane but does anyone know what the contents of a facility message should be for initiating a TBCT on an NI2 ISDN. I've been trying to get it to work on a DMS100 for the last four months to no avail. The message I am currently sending makes it to the switch but is returned with unknown message. Perhaps someone here has done it before and can help me out.

I am getting the following information and asterisk 1.0.7 is not answering the call. Any ideas? jerry ------------------ Sip read: INVITE sip:2828;phone-context=cdp.udp@qg.com:5060;maddr=161.49.198.102;transport=udp;user=phone;x-nt-redirect=redirect-server SIP/2.0 From: <sip:3173241052;phone-context=+1@qg.com;user=phone>;tag=c22da8c0-13c4-429efa71-657b8da-2ed To:

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

All, I am connecting to a CS 1000 nortel PBX. I can call out, I have limited success with call in. I get debug traffic that a call is coming in but I get the message "Unable to create/find channel". I was expecting that incoming calls over the trunk would be handled from my sip definition and goto the nortel context. It is not. Below is the actual incoming call debug information. I am

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

I'm having a problem with intersite calls over IAX2 being abruptly terminated. Nothing odd shows in any of the logs for Asterisk or the host. The only think I can think it might be is a lag-spike on the site to site connection. How sensitive is IAX2 to lost frames, lag spikes or large variations in jitter with the GSM codec and: bandwidth=low jitterbuffer=no trunkfreq=100 ; Raised from

* I am currently reading in a series of files, applying the same functions to them one at a time, and then merging the resulting data frames e.g.: >MyRows <- c("RowA", "RowB", "RowC")>>File1_DF <- read.delim("\\\\DirectoryToFiles\\File1_Folder\\File1.txt", stringsAsFactors=FALSE, check.names=FALSE)>File1_DF <-

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

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

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

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, 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