search for: 12167

Hi, ? I run the program testcall with one E1, it works fine; I receive DNIS and ANI for making calls and answering calls. ? When I start the Asterisk I receive call from outside correctly including DNIS and ANI, and receive the following messages: ? Sep? 7 10:29:59 WARNING[12167]: Answer Call Sep? 7 10:29:59 WARNING[12167]: MFC/R2 UniCall/2 Call control(5) Sep? 7 10:29:59 WARNING[12167]: MFC/R2 UniCall/2 Answer call Sep? 7 10:29:59 WARNING[12167]: MFC/R2 UniCall/2 0101? ->????? [1/????? 20/Group B?????? /Accepted Paid] Sep? 7 10:29:59 WARNING[12167]: Unicall/2 event Ans...

I know this has come up before but... Has anyone done anything to get an IAX client built on Windows? I thought someone had started one, but I haven't heard anything about it since - and that was months ago? Anyone have any idea what the status is? -- Ron Gage - Saginaw, Michigan I am looking for work - resume at http://www.rongage.org/resume.doc Electrical Engineering, Linux Programming,

Trying to restart postfix installed from yum. Restart fails, I get: type=AVC msg=audit(1430429813.721:12167): avc: denied { unlink } for pid=31624 comm="master" name="defer" dev="dm-0" ino=981632 scontext=system_u:system_r:postfix_master_t:s0 tcontext=system_u:object_r:postfix_spool_maildrop_t:s0 tclass=sock_file I guess it needs to remove the /var/spool/postfix/defer soc...

Hi I want to set up * as a conference bridge. I would like to be able to conference is SIP calls (up to 12) I am looking through all available documentation for * to get info on how it is done. No luck so far. Can somebody direct me to the info in this subject? Thank you Serge _________________________________________________________________ Protect your PC - get McAfee.com VirusScan Online

for today we only have experience with BRI applications together with asterisk. is the following scenario possible and stable enough for production? FYI : We want to build a unified messaging application integrated with SIP. We have an E1 connection in Belgium with 100 msn's We would think about having 2 servers : Server A : Asterisk PRI card (Digium TE410P) Server B : Fax

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

Hi, If anyone is looking for a small Asterisk installation I have managed to get it down to 296MB (If you remove the kernel source code.. could probably be made smaller if some of the devel packages and asterisk source is removed as well.) To do it I used Trustix Secure Linux 2.0 (http://www.trustix.net), did a minimum install with ssh support(92MB) and then added the required packages

Hi, Can any one please explain why the following code doesn't work? Or can anyone suggest an alternative. Suppose       x<-c(23,67,2,87,9,63,8,2,35,6,91,41,22,3)        mat<-0;        for(j in 1:length(x))        {              for(i in 1:p)                      mat[i,j]<-x[j]^i;        }    Actually I want to have a matrix with p columns such that each column will have the

...sed on >>their software and GPLed hardware design doesnt sound good to me ..... >>rather sounds like M$ style to me. >> >>regards >>kapejod >> >>-- >>Klaus-Peter Junghanns >> >>CEO,CTO >>Junghanns.NET GmbH >>Breite Strasse 13 - 12167 Berlin - Germany >>fon: +49 30 79705390 >>fax: +49 30 79705391 >>iaxtel: 1-700-157-8753 >>email: kpj@junghanns.net >> >> >>Am Don, 2003-04-03 um 17.15 schrieb d hinton: >> >> >>>you know, i asked diguim guys for this info and...

I'm currently using the SJPhone softphone with asterisk for remote SIP. When I dial into the voicemail, and attempt to pass the extension, I "hear" the sounds, but asterisk is not receiving any DTMF signals. If I use the Estera softphone, asterisk does receive the DTMF signals. Normally, I'd just say "Use the Estera" softphone to myself, but that's not an option,

hi all talking to this friend if mine just now, I realised I had no 'micky mouse' sound as eaerlier with chan_capi and AVM C2. We talked in 1 hour and 18 minutes without encountering any problems whatsoever. This was him calling me, but I don't know if that's important. roy -- Roy Sigurd Karlsbakk, Datavaktmester ProntoTV AS - http://www.pronto.tv/ Tel: +47 9801 3356

I have an old A1 ISDN card that we used to use in the mid 1990's for the delivery of data to our ISP I have seen some users have been using them, with respect to doing web searches. Can someone please let me know what drivers I require to use this card under debian and with asterix, and what the limitations are of using this card. I dont want to make calls but do inbound using it. The

hi all is it hard/possible to move the following from chan_capi_pvt.h into a setting (preferably global) in capi.conf? #define AST_CAPI_NATIONAL_PREF "0" #define AST_CAPI_INTERNAT_PREF "00" and ... Is it hard to move or copy the txgain and rxgain to [global], either as a given 'default' if nothing's set in the interfaces, or as a overall

I have been using chan_capi and a Fritz AVM (passive) card for a little while now.. So far so good.. There are still a couple of things I am not totally sure about.. First.. What is the difference between "softdtmf=0" and "softdtmf=1"?? Which is the preffered/recommended option?? Second.. What is Early B3 all about?? Should I be using it in my dial string?? Which is the

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. Although I can hear the demo etc. now, I notice during asterisk launch I get :- [chan_oss.so] => (OSS Console Channel Driver) == Console is full duplex == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found WARNING[98311]: File chan_oss.c, Line 232 (sound_thread): Read error on sound device: Resource

...e NT mode will be supported. We will also have a 4 port BRI card available in mid/late november that works in TE and NT mode (onboard termination). Check out www.junghanns.net/asterisk/ during the next week. regards kapejod -- Klaus-Peter Junghanns CEO,CTO Junghanns.NET GmbH Breite Strasse 13 - 12167 Berlin - Germany fon: +49 30 79705392 fax: +49 30 79705391 iaxtel: 1-700-157-8753 email: kpj@junghanns.net http://www.junghanns.net/asterisk P.S. i have no idea where i should put the 18" cheese ;-) if it was 19" i could bring it into the colo... _________________________________________...

hello! I'd like to use my asterisk-box on a p2p-line from my alcatel pbx. isdn4linux and capi works fine on a s0-line from the pbx, but i need more then 8 msns, cause i'd like to implement a voice-mail system and an extension to our "old" pbx with voip-phones. had anyone success using isdn4linux on a t0 (p2p) line (of course with an inexpensive passive card like my avm

Just tried compiling chan_capi 0.3.4 under CVS Head and get the following errors. chan_capi.c:60: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) chan_capi.c:61: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) chan_capi.c:62:

Does anyone know of a company that provides German DIDs (preferably Berlin) and termination of calls to Germany at reasonable rates? Thanks, Eric jacksch@tenebris.ca