similar to: ATA with pulse dialing support over FXS

Hi, Anyone here has any experience with the Nokia E70 and asterisk ? I read on the nokia website this phone is capable of talking SIP and do Presence based on SIP/SIMPLE. Please share your experience, I'm thinking of getting one but want to be sure I can use it with * before I do. Thnx. -- Michiel van Baak michiel@vanbaak.eu http://michiel.vanbaak.eu GnuPG key:

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

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

Hi everybody. I know maybe this question has been posted some time ago, but I need your updated opinion on the subject. I'm replacing our old pbx with asterisk. I have two TE207 dual pri (e1) cards on a clustered system (one on each node). I absolutely need to connect 4/5 analog extensions with modems, they're being used for remote assistance on very old systems which cannot be upgraded

Hello everyone. I know it's a little bit off-topic, but I was just wondering... Has anyone ever had any experience with asterisk, a wi-fi meshed lan (with more than one access point) and wi-fi sip phones? I made some tests but I'm not really satisfied.... Wi-fi phones are a curse (as far as I know even Nokia eSeries -I personally own an e70 model- have their flaws): - random sip

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 //

Hi everybody. I am in the following scenario: 1 Customer "A" calls an asterisk box over a Zap channel on a toll free number during night time 2 The incoming call enters an AGI script on the dialplan 3 The AGI script plays back a welcome message, then starts the music-on-hold stream 4 The AGI script originates a calls to a stand-by operator's cell phone (operator

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

Hi. I'm running some asterisk boxes on different sites, some equipped with a couple of ZapHFC cards, others with Junghanns quadBRI cards. All boxes were compiled with Asterisk 1.2.10 (libpri 1.2.3 zaptel 1.2.6) and bristuff 0.3.0 pre 1s, distribution is Fedora Core 4 with kernel 2.6.17.3 The cards are connected to Telecom Italia's NT1/NT1+ S/T lines; some of them are point-to-point,

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

I am currently being defeated by grep. I am attempting to determine the value of the last letter of a character string. An example of my data set is shown below. Regarding the codes, I would like to identify the value of the last character and then take the appropriate action, e.g. If the value is L then label UL rating XXX It the value is F then label UL rating YYY ... I assume it will be

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'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 useR's, I have an image plot I want to overlay a map of the United States on, via map('usa'). The image is basically a large rectangular of various colors. When I overlay the United States map, the full rectangle of the image plot is visible, but I only want to display the image WITHIN the United States boundary. Does anyone know a way to do this? Thanks in advance, dxc13 --

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

When i made internal call into my LAN using x-lite sip phone client I retrive in askterisk CLI : ----------- ERROR ---------- Verbosity is at least 6 -- Remote UNIX connection -- Executing Dial("SIP/201-979d", "SIP/201|60|t") in new stack -- Called 201 May 29 18:09:28 WARNING[6082]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call

Hi, My collector supports only base pointers as stack roots. This wasn't a problem until I tried to run some optimizations before RS4GC, which introduced (interior) derived pointers. The statepoint documentation mentions that these collectors exist, but doesn't mention whether and how this is currently supported. What could I do to make it work? -Manuel

Thanks for your suggestion. I have compiled according to http://www.voip-info.org/wiki/index.php?page=Asterisk+zaptel+pulse +dialing dialing at 10 pps works fine with Asterisk with the newly compiled wctdm. but when I dial at 20 pps, the pulses cannot be decoded correctly. I tried changing the make/break ratio but dialing at 20 pps still has the decoding problem. Does anyone have any

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