similar to: Zaptel problem after upgrading to 1.2.16

I have had a TDM400 with 2 FXO and 2 FXS working for ages (>12 months). It has stopped working. All four green lights are still lit. I have rebuilt zaptel and asterisk and restarted but the problem persists. /sbin/ztcfg -vvvv Zaptel Configuration ====================== Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03:

Some light relief .... SNOM say "Please note that you will not be able to reach us by phone." http://www.theregister.co.uk/2008/03/13/dont_call_us/ regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com

For some unfathomable reason, Siemens USA doesn't offer the Gigaset IP range in the U.S. I'm particularly interested in the Gigaset S685 IP. Since it's DECT 6.0, and there's an English (UK) version, I'm thinking it should work just fine, after dealing with the walwart issue (and maybe caller ID signalling). Anyone imported one from the UK and using it in the US? for how

> get.hist.quote(instrument="INR/USD", provider="oanda", start="2005-10-20") trying URL 'http://www.oanda.com/convert/fxhistory?lang=en&date1=10%2F20%2F2005&date=11%2F01%2F2005&date_fmt=us&exch=INR&exch2=&expr=USD&expr2=&margin_fixed=0&&SUBMIT=Get+Table&format=ASCII&redirected=1' Content type

Hi, I am having some issues with a new server install in Singapore. Outbound calls work fine. Inbound calls are not picked up by Asterisk. Zaptel 1.2.25 and Asterisk 1.2.28 both built from source. libpri installed wctdm and zaptel load without error Jun 6 23:34:03 fs01 kernel: [211138.372933] Zapata Telephony Interface Registered on major 196 Jun 6 23:34:03 fs01 kernel: [211138.372937]

Hi, Our operator frequently gets requests to transfer a call directly to voicemail in order for the caller to leave a message without disturbing the callee. Basicly, I'm looking for a blindxfer to vm. My first thought was to prepend a digit (eg 7) to the extension but this does not fit well with our dialplan. According to an article on voip-info.org Asterisk@Home appears to implement

I have some phones (and an ATA) that are shared between two users who each have separate voicemail but they are not behaving as desired nor expected. Incoming calls show up on the correct lines. Calls originating from the device are seen, at the terminating device, as coming from the account listed last in sip.conf, regardless of the line selected. This creates three main issues I would like

*Good morning! * *I verified that the trixbox version Trixbox 2.6.1.13 has support for OpenR2, I verified in the repository that has to libraries of the project openR2, but I don't manage to do to work in the trixbox, when I type the command (it colors show channeltypes)ele no demostra the support to MFC+R2, they could help finding out which package is necessary of the trixbox and which the

Hi, We have a few Aastra 480ci phones and we've noticed that in order to get the phone to receive a call, qualify must be = no. Apparently the Aastras do not respond to the qualify message (or respond in a way Asterisk doesn't understand) and Asterisk thinks the phone is unreachable. However, this now prevents MWI from working properly on the phones. Does anyone know how to get MWI

I was on vacation the last week and wrote some code to run a 500-day correlation between the Nasdaq tracking stock (QQQ) and 191 currency pairs for 500 days. The initial run took 9 hours(!) and I'd like to make it faster. So, I'm including my code below, in hopes that somebody will be able to figure out how to make it faster, either through parallelisation, or by making changes. I've

Hello, 1. Is it possible to install several SIP softphones on the same PC, have them registered to the same Asterisk server and attribute to each softphone a specific extension, ringtones or call forwarding rules ? 2. Is possible to do the same with SIP hardphones ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I installed s fresh system yesterday and it seems that zttool did not install!! ztcfg is there.. Anyone else had this problem or is it just me? -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Hello, I have an Asterisk 1.8.11 installation. When I built up this Asterisk, I didn't install DAHDI channel, if I issue command connect*CLI> core show channeltypes I would have response like: connect*CLI> core show channeltypes Type Description Devicestate Indications Transfer ---------- -----------

Hello; I am trying following but getting a warning message : Warning in rbind.zoo(...) : column names differ, no matter whatever I do. Also I do not want to specify column names manually, since I am just writing a wrapper function around getSymbols to get chunks of data from various sources - oanda, dividends etc. I tried giving col.names = T/F, header = T/F and skip = 1 but no help. I think

Hi, I'd set the daylight saving option to yes on all the GXP-2000 phones, but apparantly it doesn't move it an hour back on last sunday of October. So now I am stuck will all the phones showing the wrong time. Isn't there an option so that it'll automatically update daylight savings? Thanks -- Zeeshan A Zakaria -------------- next part -------------- An HTML attachment was

Grandstream GXP-2000 with firmware 1.1.4.18 (beta) fixes an issue where the phone did not send rtp keepalives when on mute (resulting in disconnect from tech support hold and concalls) A side effect seems to be that Asterisk pops the following warning on the console... Jul 26 14:06:35 WARNING[31654]: rtp.c:463 ast_rtp_read: RTP Read too short Grandstream say they are not sure what it is but

Hi I need a small PBX for use on the move. This means that outbound calls will need to be made over the cell phone network. Assuming a small hardware PBX with a spare mini-PCI slot or a USB slot then what hardware options do I have to get an outbound cellular channel? Options need to be rock solid, so no bluetooth to a cell phone kind of solutions need apply. Can any of the 3G usb

Dear all, I'm using Asterisk 1.4 as my SIP server of my LAN users. The SIP clients are using different operating systems such Debian, Gentoo and Windows XP so they use different SIP softphones like SJPhone, Twinkle and X-Lite. In order to let SIP clients to see the presence status to each other, do I have to establish any special setting in Asterisk 1.4 ??? Or the presence status (online,

Ok, I have spent all night trying to figure this out, and hopefully somebody has a similar experience. I have a very basic asterisk config. Sample configs, with the only addition being by SIP phone, and my incoming voip. Last week I got everything setup, calls were working, etc.,. I cam across a tutorial for voicemail, followed it, and it worked. When I call my phone and dont answer, it goes

I am trying to compile zapata under a 2.6 kernel (Suse 9.1 all patches installed). I am getting the error bellow: Any ideas? Anybody able to successfully compile this in Suse 9.1? 500@suse91lx:~/dl/pbx/zaptel-1.0-RC2> make linux26 cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -c -o zttool.o zttool.c cc -o zttool zttool.o -lnewt make -C /usr/src/linux-2.6