similar to: Asterisk dialplanner

-----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

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, 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 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

Ronald, Grandstream products have a one year warrantee. If you don't have any luck with Pulver, contact us and we can probably get your phones exchanged. Please don't assume that your experience with Grandstream is typical. We sell a lot of these phones and the overwhelming majority of the purchasers are very happy with their units. The quality has improved tremendously over the last

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

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 am trying to connect to a vocal server from an asterisk server. A call is received via iax2 to my asterisk server. I then initiate a SIP connection to the vocal server. everything works great except dtmf doesnt work. A cisco 5300 can connect to this vocal server and do dtmf without a problem. I have my dtmf set to rfc2833 in the general section of the sip.conf . I can confirm that the

-----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

Hi, I was asked about how * could provide redundancy (on the actual PBX Server portion, not the incoming lines), and my immediate thoughts were: Incoming ======== Basically somehow I'd split the incoming T1 line into two parallel lines and connect them to two * servers. -----> * Server 1 | --- T1 --- | -----> * Server 2 Internal

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 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 Looking at the subnet lists produced with kill -USR2 does not show this subnet in the log (but all others, that are configured in the same file). Splitting the subnet in two /9

Hi Ryan, In this case there already is an implementation for this, it's just hard to find being in the internals of the LoopVR pass. I'm planning to pull the multiply and udiv support out of there. Your patch looks good but beyond what Dan mentioned you have a bug calculating NewUpper: the constant ranges are half-open intervals where "[5, 10)" includes the value 9 but not

-----Messaggio originale----- Da: Davide Parise [mailto:d.parise@mhz.it] Inviato: Tuesday, February 26, 2002 23:33 A: Jeremy Allison Oggetto: R: [Samba] PATCH: Samba/Win2K renaming bug I have the same problem, but I also have an explanation. When You click on a multimedia file Windows Media Player opens a preview on the left side of your windows. When this preview is open is impossible to make

Hello, I successfully use tinc for one year in a configuration with several subnets (about 5-7 subnets) and one main server. Sometimes tinc dies with the error message: Error while waiting for input: Bad file descriptor If it happens, this happens immediately, when one subnet has disconnected. tinc (on the main server) tries then to disconnect all currently connected subnets and while