similar to: integration with Meridian/Norstar ATA2

Thanks for your suggestion Steve. I have done as you advised and set busypattern=300,200 to match the sample I recorded. This hasn't worked though, asterisk doesn't seem to detect the busy signal. Does asterisk require a the signal to be in a certain power range? The signal I get is very quiet. Thanks for your help Regards Jonathan On Mon, 19 Dec 2005, [ISO-8859-1] Diego Andr?s

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

Is it possible with asterisk to setup aliases for SIP? For example, direct sales@mysipdomain.com to 55544@mysipdomain.com If this isn't possible directly with asterisk, does SER offer anything along those lines? A search of the usual sites didn't turn up anything conclusive. Thanks, Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com

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

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

The AstLinux Team is happy to announce the release of AstLinux 0.7.1. This is a bugfix release which includes updates to Asterisk (1.4.30), Dahdi and several other items as detailed in the Changelog. http://www.astlinux.org/release/071 Existing users can upgrade from the web interface or from the CLI. From the CLI execute the following: upgrade-run-image check

Hi, Does anyone have any experience connecting Asterisk to a Meridian system using an ATA2 and x101p? The basics work -- I can make outbound calls, receive inbound, and use flash to transfer calls, but certain things do not work, specifically with calls from internal extensions. - Does the Meridian/ATA2 pass any kind of callerid info? We do not have external callerid, but I'm not even

Hi. I had successful confiured my Asterix PBX, but now I need to connect it to a Nortel Meridian phone plant, I got a Digium T100P on the machine I got the Asterisk and I already connect it with the phone plant, I need to know what configuration do I have to do so the ip extensions can make calls to the extensions of my plant Nortel. And extensions from the Nortel plant can call ip extensions of

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

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

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

> -----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

Users report that an extent map''s list is still linked when it''s actually going to be freed from cache. The story is that a) when we''re going to drop an extent map and may split this large one into smaller ems, and if this large one is flagged as EXTENT_FLAG_LOGGING which means that it''s on the list to be logged, then the smaller ems split from it will also be

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

On Thu, Apr 25, 2019 at 10:03 PM Pankaj Gupta <pagupta at redhat.com> wrote: > > Dont support 'MAP_SYNC' with non-DAX files and DAX files > with asynchronous dax_device. Virtio pmem provides > asynchronous host page cache flush mechanism. We don't > support 'MAP_SYNC' with virtio pmem and xfs. > > Signed-off-by: Pankaj Gupta <pagupta at

On Thu, Apr 25, 2019 at 10:03 PM Pankaj Gupta <pagupta at redhat.com> wrote: > > Dont support 'MAP_SYNC' with non-DAX files and DAX files > with asynchronous dax_device. Virtio pmem provides > asynchronous host page cache flush mechanism. We don't > support 'MAP_SYNC' with virtio pmem and xfs. > > Signed-off-by: Pankaj Gupta <pagupta at

Digium has released version 3.0.3 of its product registration utility. This is the first version of the registration utility that is compiled against the uClibc C library. A benefit of this transition is that the register binary should run more consistently and reliably across a wider range of Linux distributions. The new versions of 'register' and 'asthostid' can be

Hi Everyone, Knowing that running Asterisk on an embedded board like the Alix2d3 requires some fine tuning. Do you know of any good guides out there that does this from beginning to end? Looking to run this in a small office environment. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

I havent got any mails since 2:42 this morning..usually i get at least the normal 10-15 a hour, if someone gets this can they reply? Thanks! Ron -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060308/c04e3fc9/attachment.htm

Hi All. I have notice a problem with FM feature (screen macros) on Asterisk CVS version. When call goes via IAX and calling part "accept the call" on Dial command with option M, in macros context it's setting MACRO_RESULT=CONTINUE, but anyway it hangups both channels. If anyone faced with such problem please let me know. I need to know whether it's bug or just configuration