similar to: incoming calls dropout on PRI over TE110p

Hi, I am currently trying to find a replacement for a dinosaur PBX and want to replace it with a VoIP solution. We have just moved our lines over to an Optus Multiline from a Telstra ISDN Onramp 30 service with 100 lines. My question for you good people is what sort of hardware do I need to interface Asterix into the Optus Multiline? The Optus service is terminated in my office to a SHDSL

I still can't get this thing to work. Outgoing calls are fine, incoming just hangup straight away. I get a lot of & characters in my logs, I'm wondering whether they're caused by some sort of bad data ? Mar 3 16:03:01 VERBOSE[10437]:

Hi, I have an asterisk installation with 2 E1 cards Software version is Asterisk 1.2.6 Libpri 1.2.2 Zaptel 1.2.5 I'm having problem with fax transmission, let me explain better my setup: My fist TE110P E1 card is connected to the telco line the second TE110P E1 one to an Nexspan PBX so the server is basically sitting between the line, and the pbx. every call coming from the line is

Dear users. My current setup uses a euroISDN E1 with 8 cahnnels for incoming and outgoing calls from a digium te110p. Currently all phones use SIP. However, I need to add some faxes lines and some POS credit card machines. These will require POTS lines with a fixed DDI. I have purchased the tdm400p and 4 FXS modules. My problem is with the zaptel.conf and zapata.conf. I am a little confused as

I just managed to put in a TE410 card in an Asterisk box to work with OnRamp 20(E1 downunder). I am able to dial in but was not able to dial out. Can anyone offer me some advice please? In my extensions.conf, I just put in: [default] ... exten => 0,1,Dial(Zap/g1) and I get this on the console when I dialled 0. -- Executing [0 at default:1] Dial("SIP/5166-b76004f8",

Hello all. I have here a TE110P (configured as E1) and a TDM400P (with four X100P - FXS). Both boards are recognized by the operating system as showed above: 0000:08:00.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b1d9:0003 Flags: bus master, medium devsel, latency 64, IRQ 169 I/O ports at e800 [size=256]

I am trying to solve the fax problem by installing an E1 channelbank (Megaplex MP-104) It's a box that has 8 x FXS ports and a single E1 port. The plan was to use one of my 4 E1 ports to connect to the Telstra onramp and one to the MP104. I have since discovered however that the MP104 only supports CAS signalling and I am having trouble getting asterisk to work in this mode. Currently we

Hey everybody, I have a situation where we have 2 Asterisk (CVS as of 08/25/2005) connected via IAX. On the corporate side, we have 1 TE110P connecting to a Definity G3R and it's connecting to a TN464F card, giving a 23 channel connection. I have echocancel=yes, echotraining=yes and echocancelwhenbridged=yes. One the remote office side, they a Adit 600 channel bank for 10 outside

Hi All, I have configure my Asterisk as follow (using Asterisk@home): [zaptel.conf] span=1,1,0,ccs,hdb3,yellow bchan=1-15,17-31 dchan=16 loadzone = uk defaultzone=uk [zapata.conf] [channels] switchtype=euroisdn pridialplan=local prilocaldialplan=local internationalprefix=00 nationalprefix=0 usecallingpres=yes busydetect=no ; not need on pri callprogress=no ; was yes but wiki says experimatley

Dear R users,   Please assist me with the following problem. I have a dataset that looks like the following:   dat<-data.frame(   'id'=rep(c(1,2,3),each=3),   'time'=rep(c(1,2,3),3),   'y'= c(2,2,NA,2,NA,NA,2,5,7) )   I wish to create a variable for dropout time in dataframe 'dat' such that the dropout time is the time to drop out by the subject as follows:    

I am attempting to install Dahdi on a virtual machine running Centos 5.5 and having various problems. yum install kernel-devel gcc make gcc-c++ libxml2-devel Loaded plugins: fastestmirror Loading mirror speeds from cached hostfile * base: mirror.optus.net * extras: mirror.optus.net * rpmforge: fr2.rpmfind.net * updates: mirror.optus.net Setting up Install Process Package

Hi, now that these cards have approval in Australia, has anyone had any luck in connecting them to a Telstra Onramp 10 service? Are these configured as a PRI, bchan=1-10, dchan=16 in zaptel.conf and switchtype=euroisdn in zapata.conf? Better yet anyone have any of the above .conf files they could share? Is an onramp 10 what is referred to as a 'channel bank'? Craig

Hi, I need some help again. :-) I have "Too much recursion error" prototype.js line 1288, I heve this error when I try to use Effect.dropout (script.aculo.us) if I remove table ID, there is no error, and the effect works fine (remove the tr from table) But I need this ID assign to this table, as table ID is used by other effect (Builder.node) Please help .... Thanks YUAN Here is

Hello all, Over the weekend, I setup and linked an Asterisk box at another site to the Asterisk box here. The phones here are a mixture of Cisco 7940/7960 and Grandstream BT-100 phones. The phones at the other end are Grandstream BT-100 SIP phones. The Cisco phones run SIP 7.1 (upgraded last Friday from 6.1), the Grandstream phones run 1.0.4.68. Both Asterisk boxes are running stable CVS

Hi! I am a new user of Debian and am having an issue loading RStudio Server, which may or may not be related to a previous 'Release file expiration error' when trying to perform apt-get update prior to installing r-base into my remote Debian system. E: Release file for http://mirror.optus.net/debian/dists/jessie-updates/InRelease is expired (invalid since 13d 17h 49min 2s). Updates for

Hi, I have an MGE Ellipse 1000 connected to a FreeBSD 7.1 system and it works well except that if I yank the cable it doesn't detect a problem.. It seems to quite merrily read the old data (upsc reports the same values). There is nothing logged by NUT to indicate comms is lost (usbhid-ups is still running). In the attached log I yanked the able at the 28 second mark and plugged it back

Hello All, For some reason my asterisk server is not registering a port number with my VSPs. This is causing problems where people are not able to dial in from any of my SIP or IAX VSPs. I do have one VSP that has hard coded my IP and port so I can get incoming calls but this still leaves a problem with my other VSPs. Hose can I get asterisk to register my IP and port? I have been

Hi everybody, I'm facing a strange problem after upgrading Asterisk from 1.0.9 to 1.2.4. Sometimes, when receiving an incoming call from pstn, although my sip phones ring correctly (I've got both softphones and hardware phones), noone can pick up the call. Asterisk CLI shows me that the phones are ringing, then nothing happens, so there's no problem _after_ someone picked up, simply

I have a phone (a panasonic globalrange phone) which always sends a fully qualified phone number. That is, for a local Canadian number, even if I key in 6135551212 it actually sends to asterisk 01116135551212. This means of course, along with "normal" phones I end up having twice as many extensions for outdialed numbers. Is there any way I could canonicalize this down to the more

Deat all, I am in middle of integrate Asterisk with Toshiba astrata legacy pbx. Following is my setup *Asterisk <-> Digium TE110P <-> E1 card in toshiba pbx <-> Toshiba PBX* A =============================================> B C <============================================ D Asterisk PBX and strata PBX connected using back to back E1 cross cable. Physicall connectivity