similar to: Echo cancel: chan_misdn vs bristuff? HFC card vs expensive card?

Hi all, I've been having problems with my A20002D lately - callers from the PSTN don't hear me when I answer, but I hear them. Disabling echo cancellation in zapata.conf brings the audio (and echo) back. This used to work fine, until two days ago. The only weird thing in the logs is this: May 12 07:42:53 steerpike wan_ecd: wp1ec: The H100 slave has lost its framing on the bus! May

Quick question: Is there a way to distinguish between calling MSN's when using chan_misdn? More info: I've got my ISDN2 (EuroISDN) up and running here in Romania with 1 base number plus 5 MSN's. Now I want to my * to do different things when receiving a call on from different MSN's (like forwarding the call to my FAX machine or forwarding the call to my mobile). The obvious way

Dear Forum Members, I just purchased two VoiSmart GSM cards. Tried to install one of them on my Fedora Core 5 system, The compilation was not smooth, as expected, but after a small fix, it went through. Then I put two SIM cards in the card's slots. Then I loaded the modules. Then I started the Asterisk. After all I configured the vgsm.conf file according to my settings, that is just changed

Hi, I guess I'd need to run Beronet quad and octo bri cards under bristuff to get zaptel features (echo canceling, timing source).... Am I right or could I achieve this also with chan_misdn - their native driver ? Running bristuff on Beronet cards is unsupported. Has anyone succesfully run Beronet quad BRI cards under bristuff recently ? Do they work ? Regards, Rob.

Hi all, I have a "Cologne Chip Designs GmbH ISDN network controller" and I want to terminate voip calls via this ISDN card. My question is: How I must to wire the ISDN equipment with my ISDN card? With normal cable or crossover? How I can to check if ISDN card is linked with ISDN equipment? In this moment I have 1:1 cable between ISDN's, the mISDN is installed and "misdn

hi does anyone have an idea how it could be possible to do email -> fax gatewaying with asterisk + app_txfax, but still keep track of who sent the fax? i've thought a little about smtp auth, but it doesn't look too easy to integrate smoothly with asterisk.... roy

Hello list, I'd like to share something u all , so that i could understand whats going on into my Asterisk box. i have a setup like this client(ip phone) -----ip network------- [Asterisk]----ip network -------[Service provider] i have configured A2biling in my Asterisk box. so when client call to my Asterisk A2billing's ivr respoce , my client authenticate there pin and call . all

Hi all, I hava SER with many clients (sipura SPA2100). One of these is an Asterisk which have others clients (sipuraSPA2100). I also have a Cisco GW which give me access to the PSTN. I make calls to all IP phones in my network, but I can't call PSTN numbers. After I dial, I hear 2 ringbacks but at the same time Asterisk says: Called pstn_number@SER_ip_address SIP/SER_ip_address-ec75 is

Hi, I tried to find a reference in terms of size but got back a bunch of tech documents and couldn't get the idea of wav49 format. wav49 format is supposed to be half the size of a normal wav right? so, how much disk space takes to save one minute of audio in wav49? I trying to do some capacity planning for a voicemail server. -- ------------------------------------------------------------

Hi List, I would like to know if there is a way to detect gateways which time out (because of network problems or hardware failure for instance) when you send traffic to them. So when you do: Dial(SIP/number@gateway) If a call couldn't get through because the gateway has timed out, i want to do something about it. The idea would be to suspend gateway which time out for 60 minutes,

Hi, Did anyone try to install Asterisk on the Mini-Box M300 with a Versa mini-ITX board 1GHz VIA x86 CPU? The box looks promissing, but I am not sure if Digium cards are compatible with the mother board (Versa mini-ITX) Also I am not sure if the 1GHz VIA processor can handle a Digium 24 port analog board, or an E1 digital board. If anyone had tried the Mini-Box, the processor, of the mother

I'm using avm fritz card usb with kernel 2.6 patch with mISDN. The module is load correctly when I type lsmod I've got the following output: Module Size Used by zaptel 178308 0 avmfritz 21388 0 mISDN_isac 14336 1 avmfritz mISDN_dsp 191424 0 l3udss1 34184 0 mISDN_l2 39040 0

Dear Asterisk Comunity, I'm thinking about developing a callback application based on the following scenario: 1. Customer Calls the outgoing number which is a PSTN line connected to my Zap channel 2. Asterisk captures the Caller ID and calls back the customer. 3. As soon as the customer picks up the phone, asterisk plays a promt to enter the Destination number. 4. Asterisk Connects the

Hi all! I'm getting an error when I try to start asterisk with chan_misdn. I patched my kernel (2.6.12.4), and compiled the whole stuff (kernel, mISDNuser, asterisk, chan_misdn). I got mISDN from http://isdn.jolly.de/download/v3.0/ I'm using a CVS Snapshot of asterisk, which was checked out about 5 hours ago. This is the error: [chan_misdn.so]Aug 15 14:13:29 WARNING[4929]:

Hi group, i'm brand new and i would like to ask about soekris hardware. I read along the web but i have some doubts that i think can be solved here. My question are the following: 1) does the Digium TDM400P fit in the soekris box with a 4801 SBC or a bigger box is needed? Any suggestions about where to pick up another box? 2) does the Digium TDM100P (already discontinued) fits fine in a

Hi, I'm trying to setup a system where incoming faxes are received using SpanDSP and then send on to another (remote) fax machine. The SpanDSP part is working excellently, however I dont seem to be able to get the forwarding part to work. Heres what I put into my extensions.conf: exten => s,4,Answer() exten => s,5,Set(FAXFILE=/tmp/fax-${UNIQUEID}.tif) exten =>

Hello Masters Here i going explain what Iam doing and where i need help .. Iam running Sip Express Router ,Asterisk, on same box (for testing) my Sip express router is working fine and i can accept global register requests with valid account and in front of Sip express router (SER) Iam using Mediaproxy-1.4.2 which is handler to rtp/rtcp streams between nated clients

Hey everyone, A few employees have noticed some problem here and there when trying to make outgoing phone calls. After it happens, they try again, and are able to call through. The dial plan for outbound calling looks like below. Which I know they are getting to the Congestion part (which explains the busy) but what I can't seem to figure out is the cause for why they are getting sent

The place I currently work at has a Panasonic Key system with 9 extensions, and no voicemail. It services 2 PSTN lines. I am hoping to use Asterisk to host voicemail (I would like to use the IVR also, but I don't even know if or how it would work). Do I need to use a PRI between the two, or is there a simple solution? I would like people to be able to answer the phone and

can someone overthere help? the server specs are as follows HP DL380G4 Dual Intel Xeon 3.2GHz processor with 4GB RAM, running fedora core 3 asterisk-1.2.5 ss7-0.8.3d. using sip as advised to receive calls from another gateway in US. using g729 in transcoding way. however, I noticed the call hit the 51 active calls which is 102channels, I run "top" to check the system resources usage