similar to: Run script action when Dahdi phone goes off-hook?

On Fri, Jun 19, 2015 at 2:14 PM, asterisk <asterisk at solutionengineers.com> wrote: > > Hi, > > Long story short - I have an ancient Britsh Telecom phone attached to my > Asterisk PBX via Dahdi. It works beautifully, receiving calls, and the call > quality is excellent. However, dialling out is impossible, as Asterisk > consistently mis-reads the number of pulses the

Will the FXS cards that work with asterisk handle rotary? Are there any channel banks that can convert rotary to touch tone (like some sorta bridge)? The goal is to be able to log input from rotary phones. Full PBX functionality would be nice but... (It's for a project, not for serious production). -- // Ethan O'Toole //

Folks, I've got one of those GPO 1950's rotary dial phones that I'm trying to get working in the UK. I've got pretty much everything working with my TDM400, the phone rings and I can receive calls but I cannot dial with the rotary dialer. I have set pulsedial=true or whatever the exact setting is and I can dial from the phone by lifting the receiver and tapping out the number on

I'm trying to use a couple of old Western Electric type 500 phones (desk model, rotary dial). These phones work fine, as tested with telco lines (they dial, receiver/transmitter works fine, etc). I'm running Asterisk 1.6.2.11. I can't get them to dial through Asterisk. They are connected to a Rhino channel bank which is connected to Asterisk via a Sangnoma card (T1 with echo

Hi, I recently got a TDM04B and after installing and getting asterisk up and running I connected a handset to one of the ports. Unfortunately I don't get a dial tone when I lift the handset. What could be the cause of this? Could someone point me in the direction of a proper config for a TDM04B? Thanks.

Ok... I am not a telephone guy... I was born after rotary phones, so forgive my ignorance in this matter. I am trying to get a really old rotary phone up and running with an ATA. Why? Who knows... just thought it would be cool. The problem is that it does not have an RJ11 connector, instead it has three wires (green,yellow,red). Does anyone know what that type of connector is called? Or

On Tue, 2005-01-18 at 09:49 -0600, asterisk-dev-request@lists.digium.com wrote: > > Hi, > > I am Arnaud F?vrier, I teach in a technical university in Marseille. > > I'd like to know if is is possible to connect a very old phone to > asterisk and dial pulses with it? > > Are digium cards pulse dial compatible? > > Is there any specific configuration

Folks, I have a TDM400 with an FXS module. I'm having some trouble ringing some phones attached to the device. I have a real el cheapo handset that cost me about 9UKP. It works fine, I plug it into the Wildcard and call the channel and it rings. I have tried plugging in my BT Diverse 3016 cordless phone but it will not ring, although I can call out from it. I have also tried my GPO 332L

I run a phone in a remote office using the IAX2 protocol. It mostly works fine; except that every 5 mins it loses connection with Asterisk, before reconnecting 30 seconds later; rinse & repeat. Using the IAX2 debugging, I'm seeing this a lot: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00018ms SCall: 04050 DCall: 00000

I've noticed that some callers listen to our main menu and don't press any keys. I have it set up to restart the menu a few times and eventually hang up. I'm wondering if these are wrong numbers (in that case, why don't they hang up) or they really want to speak to someone here but don't understand the menu (what's so hard about "for the operator, press

Hi. Does anyone know about a simple one-fxs ATA with pulse dialing support that can work with Asterisk? A SIP one would be ok. I've been told that the Digium S101i IAXy does support pulse dialing; although it's a iax2-only ata it could be enough. I need a bunch of them to convert some old fashioned rotary phones into VoIP ones (I'd like to disassemble the ATAs to remove the boards

Hi folks, I'm running Asterisk 11 (at the moment - planning to u/grade to v13.7 LTS), I've just configured the voicemail function, and it's mostly working fine... except when I try to leave a voicemail! This crashes asterisk with no entries in the messages log. The system is running on Centos 6 (or maybe 6.5, I'm not sure how to check this). uname -a returns: Linux

Dears; I am facing the following problem: Already we requested from the service provider to enable the auto jumping service for our analoge telephone lines, so because we have 4 telephone lines from the service provider, then if you called line # 1 and it was busy, then the call will be sent to any available line #2 or #3 or #4, and if you call line # 3 and it was busy then the call will be sent

To all Asterisk guru's... Here is my question. 1. Asterisk PBX - 5 Trunks / Incoming lines 2. 1 Building - 3 Companies (sharing phone system) Ok that's the basic layout. Here's the low down - Each company will have one dedicated channel for there company. The other 2 channels they want to be set aside as rollover's (rotary). That is not an issue. Where my concern is, how

Hi, I am just trying to figure out how to setup a TDM04B card for incoming/outgoing calls. I have 4 lines, which are provided as a rotary trunk group, currently hooked into a Nortel system, which asterisk will replace. I have setup a Dell 1800 (Tower) system with the TDM04B card, which seems to work. The question is how do I set it up that all 4 lines are part of a trunk group, such that

Hello All !!! I have 4 PSTN lines in the PBX server 1,2,3,4. First line will be used by only one extension (i.e. for the boss) for incoming and outgoing. This line is dedicated for him only.( The remaining lines will be shared by the employees 1) Group A have access to lines 2 , 3 & 4. 2) Group B have access to lines 3 & 4 3) Group C have access to line 4

When I try to patch the Makefile for asterisk with the Apps_makefile.patch from Spandsp I get the following error. patching file Makefile Hunk #1 FAILED at 47. Hunk #2 FAILED at 76. 2 out of 2 hunks FAILED Has anybody seen this.

Hi I am having the exact same problem. Did anyone every solve it? It occurs on both of my samba servers, one of which is not being used. Jason ----- original message ----- Hi, Hopefully someone can help me with this because its driving me up the wall. I admin a Samba PDC which authenticates through an LDAP backend. Both the samba server and pam authenticate through the entries in the

Hi, If I have a PRI with all channels grouped in group=1, I understand when I want to make an outgoing call that asterisk takes the first channel available. Is there any possiblity to "rotate" the channel taken? I was searching in Wiki but I could not find nothing about. Thanks, Alejandro

Dear all: Does anybody know about label conflicts between xtable and Hmisc? I found a couple of e-mails similar to this problem but is not clear to me how to get around the label problem. The first table(longtable below) is generated with the latex function from Hmisc but for some reason when I try to hyperlink to it,it takes me to the top of the document. The second table created with the xtable