similar to: isdn config

hello, please where i can get channel modem dsp patch - for correct use with i4l and patch to solve problem when the calling party can hear voices from last call, for a little moment regards Marian -- SUNTEQ s. r. o. Bojnicka cesta 35 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/

Hi, For the project I've used the Eicon DIVA card. It has 8 BRI ports, and for about 25% of the time there are 7 or 8 in use. So we want to replace it with an E1 card. Only issue is, replace it with what? The idea we have been playing with was to get a Digium E1 card (we already have bought lot of Quad E1 cards :-) and then just put it back to back against Asterisk server. And instead of

Hi! I have some (short) question about the Eicon DIVA (V-)4BRI cards I want to have (short) answered before buying the DIVA card. I know there are several Eicon guys active on the list, thus I ask on the list instead of directly to Eicon so that all other will benefit as well. 1. Do Eicon DIVA (V-)4BRI cards support TE and NT mode? 2. Can the clock (master/slave) be configured independent

Hello, I'm having trouble working out how to send DTMF tones to an external IVR. My system has an analog phone connected to a TDM400P card, a SIP software phone (Zultys LIPZ4) and is connected to a BRI in Australia with a NETjet-S card. I'm using ISDN4Linux and a 2.4.25 kernel patched with the ISDN audio patch from Traverse (which allows the card to do voice). DTMF works fine between

Can anybody recommend an ISDN2e card that is a reasonable price. Also, does anybody know if the AVM B1 V4 card has the echo problems, I believe I saw somebody mention they had echo problems on the V3 card. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030507/c7cc9e15/attachment.htm

What is this? chan_iax2.c line 4695 (build_peer): Unable to support trunking on peer 'lamas-tigris' without zaptel timing codec_g729b.c Line 413 (load_module): Unable to initialize va stuff: -1 This is why I can't start asterisk in the background -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, i 've written a little Callgen script for generating calls through the outgoing spool directory. The calls goes over 8 ttyI devices to another pbx and come in through other 8 ttyI devices. But when i generate the calls, sometimes * register the calls but never initiate them. Especially when the files come to fast into the outgoing dir. What can be wrong ? Is it possible that the

does anonybody know what is the status of gsm codec in snom phones ? they were some issuses in archives, some problems so i would like to know what is the actual status. best regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ majo at sunteq dot sk

Hello! I have isntalled Settlers 4 Platin Edition and I ve managed to run the game (wine 0.9.23) - the problem is, that I cannot enable hardware accaleration ingame and some the whol game is very very slow - whenever I try to enable hardware accaleration I get: xxx@neo:~/.wine/drive_c/BlueByte/Die Siedler IV$ fixme:d3d:IWineD3DDeviceImpl_GetAvailableTextureMem (0x12

Hi, I''m new to using Prototype and script.aculo.us. I''m trying to use prototype to adjust an elements top margin based on the height of the screen. I''m also using script.aculo.us to make the element move negatively off the left of the window to create a scrolling effect. It works fine in Firefox, but I keep getting an Invalid argument error in IE6. Here my margin code

hello, does anybody have some agi script that can do following : when extension didn't pickup phone call, system send mail notify ( via sendmail ) to user mailbox with date, time and caller id ? ( something like missed call ) please can somebody help me with whis ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax:

Hi all, i tried to write a call generator script. It generate i file like sample.call in ast src tree. But when i set the callerid and make a call over Modem[i4l] then the caller id is not set for the outgoing call. Is it impossible to set the callerid at this time for ttI devices? Thanks for help, Thomas. ******************************************* beroNet technologies GmbH Dipl.- Ing.

Hello, does anybody suceesfully setup snom phones with sip firmware with asterisk to redirect call when phone is set to redirect if busy/ or allways redirect ? My console says : chan_sip.c Line 3000 (handle response) Dunno anything about a 302 Moved Temporarily from SIP... regards Marian -- SUNTEQ s. r. o. Bojnicka cesta 35 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 #

ok i have another problem - howto run asterisk as a daemon ( fork ) in uclibc enviroment ? uClinux can only do vfork() a i think this is problem... does anybody know how to solve this ? regards Marian -- SUNTEQ s. r. o. Bojnicka cesta 35 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/

hello, does anybody know how to setup avm fritz pci card in p2p mode ? regards marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ ------------------------------------------------------------ A mind is like a parachute... it only works when it's open.

hello, i have problem with oh323 channel driver (tryied differnet versions). when i try to make call between oh323 - sip, oh323-isdn, oh323-capi asterisk crash with segmentation fault. Channel driver was compiled with pwlib 1.5.0 and openh323 1.12.0 libs. Does anybody know solution ? WrapH323Connection::WrapH323Connection: WrapH323Connection created. -- Executing Dial("H323:31119",

hello, i found in list archives some notes about grandstream sip voip phones. Does anybody succesfuly tested those phones with asterisk ? Mark ? What about the prices ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ ------------------------------------------------------------ A mind

hallo, does anybody successfully managed to get swissvoice ip10s with h323 firmware work with asterisk ? mgcp firmware works fine, but with h323 i'm still getting one way audio. regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/

Hello, can i use different audio codecs when i calling between sip devices ( snom phones ) and different when i making call from isdn to sip or from sip to isdn ? best regards Marian -- SUNTEQ s. r. o. Bojnicka cesta 35 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ ------------------------------------------------------------ A

Hi, I try to use ISDN4Linux drivers with Asterisk. In modem.conf i put /dev/ttyIO. Everything is OK when i lauch asterisk but, when i call Asterisk nothing happen. Someone can help me ? Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030407/13e78d8e/attachment.htm