similar to: Parallel T1 cable?

-----Original Message----- From: Chad Brown Sent: Tuesday, December 21, 2004 8:02 PM To: 'el_flynn@lanvik-icu.com' Subject: RE: [Asterisk-Users] Dialplan help - Can dial any user but not thePSTN Flynn, Yes, that makes sense. However, in my case I have incoming calls arriving on an IAX channel from a PSTN gateway. I think the concept is the same. That said, if incoming calls have access

Hi all, Just wondering if anyone's come across this issue, and what might be a fix for it: We've got several BT-101's deployed, and upgraded to firmware v.1.0.5.16. The phone can do proper supervised transfer, but _only_ once. If the user attempts to transfer a second time, it won't work. any suggestions/hints/tips are welcome.. Flynn

Hi all, I used to have an OpenLine4 card, but decided against using it due to some problems with hangup detect. Does anyone on the list actively use Voicetronix's OpenSwitch12? What are your opinions on the card? Cheers, Flynn

Hi all, I'm just wondering about these VoIP services -- do you have to sign up one account -per- client that will be using the service? I've got multiple extensions behind my Asterisk box, and I want to be able to allow all my staff to place calls via the provider. So if I sign up for one account, will multiple users behind my Asterisk box be able to make calls, using that same

hi all, i've got a proposed setup that i was wondering if you guys could comment on. the client wants * and a couple of SIP phones to be on a separate network than the rest of the office, so that in case their primary network crashes for some reason the PBX won't be affected. one other factor: the client may at some later point set up SIP UAs sitting on the primary network that will

Hi there, I've got an installation where there's 12 POTS line incoming into *, and am trying to get some insight as to which VoIP hard phone would be most suitable for this scenario. Most of the VoIP phones I've looked at only have 4-6 line presentations; is anyone aware of one that has more? I tried to get some info about Snom's Keypad 220 since it has loads of programmable

Hi, I'm running two boxes side by side, identical specs and setup but with differing dialplans. Both are on ast/zap/libpri versions 1.0.9. Both boxes share the same folder for voicemail, exported via NFS from another file server. Everything was working fine for an extended period of time, until just recently when someone rebooted Box A. Now when I dial an extension associated with a SIP

I just started poking around with writing a python module to interface to the Manager API, and it suddenly hit me... how the heck are you supposed to program this thing? All the events seem to be dumped to all the open connections. If I send a command, such as a login, there seems to me to be no way to determine which response are intended for me, and which may be intended for another open

Hi all, I was wondering if anyone had any pointers on how to determine whether or not a module has gone wonky on the TDM400P? I have a 2 FXO (channels 3 and 4) and 2 FXS unit (channels 1 and 2). The bad (?) module in question is the FXO module on channel 3. I can't dial in to or out of that channel; dialing in gives a busy signal, dialing out just shows * hanging around after attempting a

Hello, I am very new to asterisk and voip in general and so far have managed to get the FXO card and a few sip phones working fine. My question is where does the Sipura SPA 2000 come in the picture? Can it be used as an extension (i.e FXS) ? Or is it to be used as a line (i.e FXO)? Or it can be used as both? My understanding is that its just like another ATA186. Is that true? I guess what I

I am currently developing a asterisk ncurses interface using the manager API. The project is currently awaiting sourceforge's approval but I have a beta online at http://sig.lange.googlepages.com/assman . The projects real home will be assman.sf.net. This project really consists of two parts, libassman is a C manager API and assman is the ncurses portion. It's still beta but I have been

Hello all, I'd like to mention that we've put together a simple Java-based application that provides a somewhat point-and-click interface to create an Asterisk dialplan. You can get to the dialplanner at http://www.lanvik-icu.com/asterisk/dialplanner/index.php You can create contexts and extensions, then select the appropriate command from a list. Then you'll be prompted to enter the

Hi there, I'm trying to get hold of an evaluation IP-enabled FXS channel bank. I'm not sure if it's more a channel bank, or should be called a multiport-ATA. Oh well. On the surface it looks quite nice - 16 FXS ports, and since it's connected to the network I can do away with an E1/T1 connection required of the normal channel banks (if it can be called that :) Here are some

Consider this dialplan fragment, where the call is being dialed into [macro-process-routing] over an iax2 channel from another (same build) Asterisk server: [macro-process-routing] ; This is the entrypoint of the debug call but is also refered to by Macro(process-routing) elsewhere in the dialplan ; XXX-NNN-6800 exten => _6800,1,Macro(6800-interceptor) ; This is matched when 8 is

I was once told by a lineman that the cables they use didn't have that many twists in them because it wasn't needed, and that the extra twists would effectively use more cable and thus cost and weigh more than triple what they do now. He told me that with the number of twists in the Cat 5 cable it would cancel out any interference, but he also stated that the effective length was

Can't anyone stop self-promotion and tell the poor guy what he needs. A T1/E1 X-over cable using an RJ-45 (8-cond.) is pinned out as follows: 1 - 4 2 - 5 3 - NU 4 - 1 5 - 2 6 - NU 7 - NU 8 - NU NU = Not Used I have not in my experience seen any problems with using a Good Quality Cat5 vs. Cat 3 (telco standard) cable for X-connects. YMMV, but you should be fine. As far as the shielding

Hi all, I've got a TE410p connected via T1 to four channel banks (2 FXO and 2 FXS), no PRIs. Some users are complaining that they hear clicks and pops on the FXS lines, generally when they pick up the phone it's noisy. This happens only after a while, e.g. after a fresh restart of everything, all is fine but after some time these noises start appearing. From what I've read on the

I'm connecting a T1 PCI card to a Nortel Option 61 switch T1 card. My Sangoma A102D shipped with 2 T1 cables - which I assume are straight through. Do I need to make crossover cables for this scenario? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071026/9cea5e74/attachment.htm

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Am 06.02.2015 um 21:36 schrieb Guus Sliepen: > On Fri, Feb 06, 2015 at 09:01:21PM +0100, Flynn Marquardt wrote: > >> Analyzing a routing problem in a tinc net I found, that a declaration of >> a class-A subnet (x.0.0.0/8) in a host configuration file is ignored, e.g. >> >> Subnet = x.0.0.0/8 > [...] >> Splitting

Is there any technical difference between a T1 cable and a cat5e patch cable as far as using them with Digium T1/E1 cards? Can PRI circuits terminating at a smart jack connect successfully to Digium cards using straight through CAT5e cables? If so, are they using all of the pins in the cable? Thanks in advance