similar to: Caller Id problem

Dear all, a newbie question... I have two external lines (PSTN & SIP) and two internal lines. When receiving an incoming call, I correctly get the CID, but it's not propagated to the internal lines. My analog phones shows "External call" instead of the CID. My analog device is a TDM400P (2 FXO + 2 FXS) with two analog phones attached to it (Siemens Gigaset and a wired

I'd like to use the VoIP service from Free with Asterisk, but am having a couple of problems. Here are some details: ADSL from Free T?l?com comes bundled with VoIP and TV services. Most users access the VoIP via the supplied Freebox, which is an integrated DSL modem, router, ATA, and media player. It is of course possible to connect the Freebox to Asterisk via an X100P or other FXO

Sam> In France, the second most important ADSL provider (named "Free") Sam> offers a phone line (which uses VoIP but can only be used as a FXS) Sam> with unlimited free calls to landlines. I also have Free ADSL in Paris, and would very much like to get their VoIP working natively with Asterisk. Free assigns each user both a public (for Internet access) and a private (for VoIP

When I reload the config, I see this error in the CLI. However, I don't see what I have done wrong: == Parsing '/etc/asterisk/zapata.conf': Found May 30 16:38:42 WARNING[12630]: chan_zap.c:10088 setup_zap: Ignoring signalling -- Reconfigured channel 1, FXO Kewlstart signalling May 30 16:38:42 WARNING[12630]: chan_zap.c:10088 setup_zap: Ignoring signalling -- Reconfigured

I got many answers to the question I asked below , and I thank you all. Several of you told me to use ->> but told also that "it is not a recommendable way of items manipulating in R". I don't really understand what it exactly means : 1) does it mean it's not a very good way of programming , a dangerous way of programming because the variable a can be modified ? , etc Or

Internet telephony (VoIP): Regulators and industry debate 'irreversible' trend The Commission is weighing up its options on how to regulate internet telephony. Major telecoms operators are already proposing services to avoid being squeezed out of the market. More at: http://www.euractiv.com/cgi-bin/cgint.exe/1?204&OIDN=1507828&-tt= Cheers, Philipp

I'm having some trouble dialing phone numbers in Argentina with Digium Zaptel cards. Does anyone have some sample configuration that works with Digium TDM04B cards in Argentina? I'm mainly referring to the /etc/zaptel.conf and /etc/asterisk/zapata.conf. I have two Zaptel cards: the first one has 1 FXS port and 3 FXO ports; the second one has 4 FXO ports. My current configuration is

Dear R users, I would like to write a script to launch R commands from a Unix prompt but I do not have any idea how to do it. Can someone bring me help please? Thanks in advance Denis Choquet -.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.- r-help mailing list -- Read http://www.ci.tuwien.ac.at/~hornik/R/R-FAQ.html Send "info", "help", or

I know that it is the right way to use this ML but as nobody answered me, i repost this help ... in fact, it is quite hurry, and i want to understand ... Why do the mount -t smbfs just halfWork ?? can see some directories, but see no file ! :o ... Hello everyone, I'm having such a strange problem and i hope you'll understand the situation. I want to access files that are shared by

Hello... I have a big problem with asterisk. Every time i make a call asterisk does not bridge the zap channels. The zap channel from which i'm calling remains in state:ring and applicaton:dial and the zap channel with the external line configured remains in state:dialling an Application:AppDial. Zap/20-1 agentie s 1 Dialing AppDial (Outgoing Line) 09399 (None) Zap/9-1 int_omg 09399 5 Ring

I have a Digium TDM400p (1 FXO, 1 FXS) with the FXS module connected to a standard analog handset with caller id display (US caller ID). Although it appears that caller id information is coming into asterisk (it shows up in voicemail), I can not get it to display on the analog handset. Is there anything special I need to do to send the caller id info out the FXS port? I've tried a few

Hello, I have a system with two cards: a HFC-PCI ISDN and a TDM21B (2 FXO and 1 FXS), running Asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1l with freePBX beta5 dialplan. I have connected an analog phone to TDM FXS port, but when I pickup the phone to make a call, Asterisk "hangs up" the call. Let me explain: In another system, when I pickup the phone, Asterisk give me tone to dial: >---

Hi, all I have installed TDM400 with 1 FXS and 1 FXP ports. Now I am goig through documentation on how to configure it. It mentions 3 protocols: Loopstart, Groundstart and Koolstart. Which one do I use? Can someone send me sample zaptel.conf file for Australia? This will save me some time and will be used as a working example. Thanks, Rudolf

Hello Group I have a Asterisk server running with 2 Digium T1 cards installed. 1 card connects to Telco via a PRI. The 2nd card is connected to a fax server via Digi DataFire RAS 24 PT1 Adapter (Digi0001). The idea is to have Asterisk route the calls based on DID or FAX tones. Everything is working great so far. The only problem is the Fax server does not see the DID. How can I tell if Asterisk

FYI, I posted a blog a few days ago about a DTrace provider for NFS that is currently in development: http://blogs.sun.com/roller/page/samf?entry=a_dtrace_provider_for_nfs Let''s discuss any questions, comments, etc. here. I also advertised this on nfs-discuss at opensolaris.org. Naturally, I would expect the discussion here to be more on the specifics of DTrace, and the

I was just looking through the Wiki for some info on how to retain the original caller's callerid when make transfers to internal extensions, and I came across the parameter below: useincomingcalleridonzaptransfer=yes There is nothing in the zapata.conf file from vers. 1.2.7 so I am wondering if this is still a valid parameter. If not, does anyone know how I can do this? Thanks, Joe

Hi all, I have TDM2406E with 24FXO ports connecting to 10 POTS line sitting in my office. the out going calls symptom like when called party pickup the phone but the calling party still hearing the ring tone from the IP phone. Please light me up. it been many sleepless night by googling around trying to get the right answers. The digium card running on Intel 915G chipset. Below are my zaptel

I don't see that anywhere. Here is my zapata.conf This is only happing on my 7970 all other phone are working without trouble. [channels] context=pri signalling=pri_cpe switchtype=dms100 group=1 usecallerid=yes hidecallerid=no restrictcid=no usecallingpres=no useincomingcalleridonzaptransfer=yes callerid=asreceived faxdetect=incoming musiconhold=default echocancel=yes

I've tried to google this issue with no resolution. I'm having the same issue as this person: http://lists.digium.com/pipermail/asterisk-users/2004-August/058280.html Basically, anytime I try to dial out on my TDM400P w/ FXO, I get "we're sorry, but your call cannot be completed as dialed." When I "debug channel Zap/x-x", I get a whole bunch of this: [ TYPE:

I currently have Asterisk 1.2.7.1 and the Sangoma A200 w/ 6 FXO ports and HW Echo canceller. I have outgoing calling setup to use a group so that if one channel is busy it goes to one of the other channels. What's weird is that when I dial an outside number, sometimes it goes through and other times I get "You have reached an invalid pager number MCLL327." I have no idea what that