similar to: Interface analogue exchange line to VOIP phone?

I've just been doing some tests using the manager API to originate an outgoing call via a X100P and connect the call to an extension: Action: Originate Channel: Zap/1/01234567890 Context: local-extensions Exten: 6000 Priority: 1 I've noticed that extension is getting called as soon as the outgoing call has been placed, rather than when it is answered. Is the X100P capable of detecting

Does anyone here know anything about GEN-GEN analogue circuits, also known as Manual Ring-Down (MRD)? Apparently they are widely used in Hoot'n'Holler systems for financial dealer-boards. I have been asked to try and interface to such circuits, and have been having great difficulty locating any specifications for the interface. Apparently, they are always-on 2-wire analogue circuits with

I have a client who wants a Meetme box with 12 FXO ports, to connect to Analogue lines coming from an Ericsson PBX. It looks like I could do this with four different hardware configurations: a) three TDM04B cards (based on TDM400P) b) one TDM04B and one TDM808B c) one TDM804B (or TDM854B?) and one TDP808B d) one TDM2403B (half filled TDM2400P) Apart from considerations of cost and PCI slot

Hi, all I am trying to figure out how to forward incoming PSTN call on SPA3000 to VoIP extension(s). Basically, I have converted my home to VoIP. I have normal phone (connected to SPA3000) and couple of IP phones. All call coming from VoIP DID do ring all phones (analogue via SPA3000 and IP ones). Now I need to do same thing for incoming PSTN calls. I have enabled gateway function in SPA3000 and

On Mon, Nov 17, 2008 at 10:20 AM, Tony Mountifield <tony at softins.clara.co.uk > wrote: > > If I do this from an NEC digital extension I get 141496920000, but if I > do > > it from an NEC POTS extension I get 1942124000 > > That looks like when you pick up the analogue phone and dial 9, it > immediately opens the outgoing line and sends the 141 acces code, but >

I am a newbee to asterisk. I recently installed asterisk@home. Everything went well and my set up is running fine with soft phones, such as kphone and XtenLite. Now, i want to be able to connect my analogue phones to my asterisk pbx box and use it as if i make a regular Phone call (I do have my PSTN gateway account with broadvoice.com and already configured to route through it). I do NOT have a

Hi All, I am using a Sipura SPA-3000 as an FXO gateway to bring calls in and out of Asterisk. I am using "PSTN Ring Thru Line 1" (on the "PSTN Line" tab) so Asterisk answers the call rather than the SPA-3000. It is all working perfectly except I can't get the SPA-3000 to pass caller ID to Asterisk. It passes "Display Name", "User ID" and any "PSTN

I have today reported a bug with the latest channel.c (1.134) that affects music-on-hold for the first user in a MeetMe room when calling from a Grandstream BT102. The music is broken up about 5-10 times a second. It doesn't happen when calling from Firefly. It is also fine on both clients with 1.133 of channel.c. I am using the ALAW codec. Mark at Digium can't reproduce the problem,

Hello; I am facing a trouble with A2Billing when using analogue lines because the channels are not closing properly when dialing happen through A2Billing (it seems the dialing scenario including the hangup is not handled properly through A2Billing but I do not have control on this). But when I do dialing from asterisk and using analogue lines, I do not face a trouble because I can write the

Is there anyone using * in AU that has successfully extracted the CLID from an incoming analogue PSTN phone call, and would like to spread the word? Note - I am only interested in analogue, not ISDN phones. -- Howard. LANNet Computing Associates; Your Linux people <http://www.lannetlinux.com> ------------------------------------------ "When you just want a system that works, you

So made my first forray into 1.4 and DAHDI and hit a problem. (Not convinced this is a DAHDI issue though...) Testing an analogue line and asterisk sees the caller ID being passed, but then fails to detect ringing. A plain old analogue phone plugged in rings just fine. Console output: == Starting post polarity CID detection on channel 4 -- Starting simple switch on

Hi, Can anyone recommend a USB phone that can be used under Linux, either interfacing directly with Asterisk in some way, or using a soft phone program on Linux that doesn't need screen interaction (only using the phone's keypad)? The idea is to be able to plug it into the USB port of an Asterisk box in a rack, where screen, kbd and mouse may not be available. Thanks in advance! Tony --

In article <mfbt6f$9rt$1 at softins.softins.co.uk>, Tony Mountifield <tony at softins.co.uk> wrote: > I am trying to debug a SIP issue, between an Asterisk 1.2.32 system that > is behind a network device to which I don't have ready access, which is > performing NAT with possibly some kind of SIP ALG, and an Asterisk 11 > system on a public IP. > > My question is

These allow and disallow work with NuFone for me disallow=all allow=ulaw allow=alaw allow=gsm Jeff Message: 11 Date: Fri, 11 Mar 2005 11:15:51 +0100 From: "Edward Banfa" <edward@radform.com> Subject: [Asterisk-Users] NuFone Configuration [problem] To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com>

I try to perform a clustering using an existing dissimilarity matrix that I calculated using distance (analogue) I tried two different things. One of them worked and one not and I don`t understand why. Here the code: not working example library(cluster) library(analogue) iris2<-as.data.frame(iris) str(iris2) 'data.frame': 150 obs. of 5 variables: $ Sepal.Length: num 5.1 4.9 4.7

In article <n009u2$85v$1 at softins.softins.co.uk>, Tony Mountifield <tony at softins.co.uk> wrote: > Apologies, this is slightly off-topic being to do with an EPEL package, > although it's running on CentOS6, so I thought others here might have come > across this issue. > > I have five CentOS 6 systems running fail2ban from EPEL, and this > package was updated

Has anyone else experienced a problem with the LED for span 1 on a TE410P or TE405P? I had a TE410P on which the span 1 LED would not light red, but once the span was connected, it did correctly light green. I RMAed the board to our UK distrbutor and received a replacement. However, the replacement board displayed the same problem! Wondering if it was related to the computer I was putting it

In article <20151125133008.6369360.14455.17239 at gmail.com>, Israel Gottlieb <isrlgb at gmail.com> wrote: > Try putting progress instead of answer Yes, I tried Progress already, and it didn't help. But thanks for the suggestion! Tony > I have a puzzling situation, and would be grateful for any insight. > > I have a dialplan that forwards an incoming call out to

In article <ml1jnh$afr$1 at softins.softins.co.uk>, Tony Mountifield <tony at softins.co.uk> wrote: > When I started using CentOS 6 instead of CentOS 5, I discovered that > "less" no longer understood \< and \>, which I had been used to using > since almost forever. > > Eventually research revealed that in the Fedora version on which > RHEL 6 was

Thanks for the suggestion Tony, I installed each codec for MoH, core sounds, and extra sound packages. Unfortunately the tests produce the same results. [Sep 1 20:36:45] ERROR[10081][C-00007fe5]: frame.c:343 ast_frdup: FRACK!, Failed assertion Excessive refcount 100000 reached on ao2 object 0x20380b0 ( continuously for a while followed by a [Sep 1 20:36:46] WARNING[7761][C-0000770d]: