similar to: primuxisdn capi

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 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

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, sometimes my capi_channel stop works - e.g. when i try to call number which does not exist ( typo error ) and i must restart asterisk. following lines appears in the log files : ERROR[393234]: File chan_capi.c, Line 1050 (capi_request): no free channel on controller 1! will continue searching. ERROR[393234]: File chan_capi.c, Line 1050 (capi_request): no free b channel on controller 1!

Hello List, I'm trying to get a PrimuX Card (www.primuxisdn.de) working. The Manufacturer says that chan_capi (the older one) used to work. Now I'm trying with chan_capi-cm and have got the following problems: Outgoing calls "ausgehend_ueber_ntba_primux.txt" the other phone rings but when I answer the call asterisk crashes. Incoming: first tried with immediate=no

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/

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, do i need some other software than asterisk to use database commands - dbput and dbget in asterisk ? 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, can anyone help me ho to simply connect 2 asterisk boxes, which are in different loactions e.g. with iax protocol ? some simple config files.. 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

hello, i change my isdn line to ptp mode and i have some trouble to configure chan_capi in that way. What is the correct setup - e.g. if numbers are 123456x - where x can be from 0 to 9 ? 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, got anybody succesfully setup asterisk with three avm fritz pci cards - using the howto described in http://www.quiss.org/caiviar/Two-Fritzcards-HOWTO i already have asterisk working with 2 cards, by when i add third card and compile driver ( see capiinit debug below ) asterisk freeze on capi initialization.... does anybody know how to solve this ? regards Marian ------------------

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/

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:

Hello, i have asterisk with chan_capi ( AVM Fritz card ) working, isdn line BRI, setup to point-to-multipoint. working fine, but when both channel on interfaces are used and somebody wants to call in - the response is "ha-la-li" - sounds like msn or called number does not exists. i think that right response should be - line busy tone... right ? can anybody tell me what is wrong ?

hello, i try to compile * in uclibc enviroment (uclibc 0.9.19 ), but still getting following error does anyone know how to solve it ? regards Marian --------- gcc -g -o asterisk -rdynamic io.o sched.o logger.o frame.o loader.o config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o callerid.o fskmodem.o image.o app.o cdr.o tdd.o acl.o rtp.o manager.o asterisk.o

Hello, is it possible to change how are caller id on incoming call from isdn, capi lines displayed od sip phones ? ( e.g. SNOM ) standard is 1234567@domain.net. I just want only 1234567 to be displayed. is it possible ? 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, 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 #

Hello, i have asterisk with 2 internal isdn cards - handled by isdn4linux and i need to setup whol system like this route some call beggins with 0 or 00 - long distance through first card, route calls to mobile network via second card ( tehere is isdn gsm gateway connected).how i can do this using only isdn4linux (/dev/ttyi) ? regards Marian -- SUNTEQ s. r. o. Bojnicka cesta 35 # Prievidza