similar to: existing ISDN PBX <-> asterisk <-> 2xBRI for IVR and SIP

Hi list, I have a "Wildcard TDM400P REV I Board 1" with 4 FXO modules and * 1.0.9 up-and-running. Only 2 FXO ports are used for 2 analog phones and are doing fine. I now wanted to use the 3rd and 4th port, but when I insert an analog phone, take it off hook, I do not get a dial tone. With my 1st and 2nd port, I get messages like: -- Starting simple switch on 'Zap/13-1'

I have a running * with a TDM40B board in it. I have 3 analog phones that works (rings) perfectly when connected to a Telco POTS line. When connected to the Digium TDM40B (with FXS port), I have problems with 'ringing': 1 phone 'ringes' normally 1 phone 'ringes' a bit cripled (instead of ring-ring... ring-ring..., it does 'ring-ri... ri.... ring... ri...) and the 3rd

I have a serious problem that repeats very often, after 30 - 50 calls and I can only solve it by stopped and restarting * :-( After a while, * seems to loose track of something. When an ISDN call from PBX needs to go to the Telco, I get 'Ring requested on unconfigured channel 255/255 span 2' It's always channel 255/255 ??, the 'span' number is random Setup: * between Local

Hi, I am new to Asterisk and have some general questions _before_ I start buying equipent to install and get everything up-and-running. (this means I have no running Asterisk (yet)). I have read already a lot of doc, but some things are not clear to me, since I'am inexperienced in Asterisk and PBX. The goal: Make PBX using Asterisk and Cisco 79xx equipment for 25 phones (1 x 7970, 4 x 7960,

Hi, I have just started to setup and configure my asterisk box and am having trouble with it. I have a dlink nic in the box as well as the digium card. When the nic is in there by itself it works, but when I put the tdm400p in, there seems to be some sort of conflict with the network card. The tdm appears as another type of network card, which I don't think it should, should it? It shows

Hi all, How can I make a X100P ZAP channel not answering to any incoming calls? Thanks. Newbie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050912/73c47a3f/attachment.htm

Hi, I wanted to configure the following : 1. VPN + some other special connections (TCP 82,8282,23,22 and ICMP) to have priority over the rest 2. special upstream for our updating system on port 4000 3. within the VPN tunnels citrix traffic ( TCP 1494, 2598) + icmp has priority I need this for both incoming and outgoing traffic as it is the bandwith managment config on a central system from

Hi, We are using kernel 2.4.24, tcng version 10b. I''m trying to do some policing in the ingress queue of the internet device. Until now we had configured some filters dividing traffic into queues, and on these queues the Double Leaky bucket meter was applied. The idea is to have a minimum of bandwidth assigned per class (the cir values) and a maximum (pir values), just as with the

Hi, People here often work on 2-3 places (office 1, office 2 and home). I would like to give them 1 extension (XXX) and to ask them to 'register' the phone they use at a certain moment. The idea is that, when you need someone, just dial XXX and the phone near him (in Office 1, Office 2 or at Home), will ring. This will keep my queue system and other tricks intact, where I always use the

Hi, We are using Asterisk 1.2.1 with Cisco 7940 and 7960 phones. Most things are running fine ;-) But, when you are calling and you want to Transfer, you need to press first on the 'more' button (4th), then you have the label 'Trnsfr' to Transfer. these are the lables on the softkeys when having a phone call: "Holt / EndCall / Confrn / more" press more and you get

I have a very odd problem connecting to some websites from my CentOS 5 box Target websites: www.connecttech.com www.3ware.com (two of my HW vendors) I can usually get some kind of response, but if the content (download or page itself) is larger in size (downloads never pass 100K), then it hangs... When I fire-up wireshark, I get a lot of ougoing highlighted Checksum Errored packets but I

I'm able to receive but not make calls with zaptel using an X101P connecting to Asterisk with an Xlite client. My client has context = flat in sip.conf and extensions number 8919 In extensions.conf I've got: [home] ; Line 1 ; exten => 8919,1,Dial(${PHONES1},20,Ttm) exten => 8919,2,Macro(vmessage,${PHONES1VM}) exten => 8919,3,Hangup [outgoing] exten =>

Hi, I am an experienced libvirt user on Fedora versions from F15 to F17. I have developped scripts to route trafic from outside on multiple interfaces/multiples IPs to multiple VMs, and back to affect each VM the required external IP address. I have servers with more than hundreds external IPs, and up to 4 VMs, each of them route trafic on different external IPs. I have servers with Fedora

Does anyone have any experience with an Asterisk compatible softphone application which meets the following criteria: 1) Is able to use touch screen rather than mouse for on-screen functions. 2) Has an API which can be used to export Caller ID info to another App on the same compuer. Thanks Bill

Hi, I'm experiencing some voice delay (2-3 sec) after outgoing call is setup. It means during the first 2-3 secs, audio is very choppy or nothing. So usually I can't hear the 'Hello". I use IAX2 for my ougoing calls with Grandstream phone as a client. Any hints to prevent this? Thanks, David

Hi, I have a running asterisk 1.2.1 (bristuffed) with a TDM400 Board. Connected to 1 port is am ordanary Fax Machine. Everything 'seems' to work, however receiving faxes is very unreliable. Sometimes I receive a normal page, without problems. Sometimes half of a page and the rest is scrambled, but most often, I receive nothing and the other site reports a Fax error... The Fax machine

Hello, I searched the archives mailing list of LARTC. Everyone discussed about marking outbound ftp traffic . I could not find any thread discussed about marking inbound ftp traffic. With inbound ftp traffic , we don''t know the random ports specified by ftp servers in passive mode ? So marking inbound ftp traffic is impossible ? If it is possible, can you tell me, Thanks in

Perfect, thanks very much hth. I just set it to unknown, but it doesn't work. Have I to use also prilocaldialplan ? Thanks again Giordano ________________________________ Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Colin Anderson Inviato: gioved? 29 settembre 2005 16.22 A: 'Asterisk Users Mailing List -

Hello, I have som problems with my BRI/capi setup. I manage to call in to the system (some rows below). ---------------- -- Executing Dial("CAPI[contr1/16453]", "SIP/BYEXTENSION@janm|10") in new stack -- Called s@janm -- SIP/janm-63f5 is ringing -- SIP/janm-63f5 is ringing -- SIP/janm-63f5 is ringing ---------------- But I can't make outgoing calls from

Hi, we are using asterisk with Junghanns QuadBri and some sip phones. 2 channels are configured in NT mode (ISDN PBX connected, internal ) and 2 channels are connected to the public ISDN network (bri-cpe). We use Bristuff 0.2.0 RC8C from Junghanns. When a call comes in from the public phone for a specific extension (Hotline Number), we initiate a parallelcall to some SIP phones and also to our