search for: yosd

Has anyone tried to compile the current version of MySQLPool from http://www.yosd.at <http://www.yosd.at/> against Asterisk 1.4.4? It seems to not compile... [root@itsentinel res_mysqlpool]# make gcc -pipe -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -I/usr/local/mysql/include/mysql -D_REENTRANT -D_GNU_S...

Hi all, i have the following setup PSTN -> routing server (asterisk 1.6.2.11) -> IAX -> callcenter asterisk 1.6.2.9 -> SIP -> agent Does work quit fine - then agent does have the abibility to transfer a call to a third party - the agent can initiate the transfer over a web interface - it does generate a asterisk manager atxfer request... So agent does initiate transfer - call

hi all, we do have some troubles with zaptel timing source - we have a setup with 3 telco PRI lines, connected to asterisk digium card 0 - asterisk does some handling - calls are leaving on digium card 1 - going to a siemens hipath - there is some call handling - some of the calls then are going from the hipath over a qsig line to a bosch integral PBX - handling the rest of the calls. To be able

Does anyone know of an IAX softphone in Java, whether applet or application? Even the most minimum featureset, just voice and dialing, or even embedded in some other app/let. Preferably GPL. Thanks. -- (C) Matthew Rubenstein

Hi all, i have now created a sourceforge project for the source made public by mexuar - you can find it at http://sourceforge.net/projects/javaiaxphone/ Take a look at http://lists.digium.com/pipermail/asterisk-users/2009-January/224730.html for more information about it. best regards, Wolfgang

Hi all, i know this question is not directly asterisk related - but i have no idea where else to ask. We do have around 50 pieces of LinkSys SPA941 - these phones do have a 2.5mm plug connection - and we do have many many headsets we used with normal PC's before (so 2x3.5mm plug connection). Does anyone here know where i can get an adapter 1x2.5mm -> 2x3.5mm ? Or can anyone here tell me

Hi all, i do have the following setup ISDN BRI Line -> openVOX Card/Asterisk 1.6.2.6/libpri 1.4.11.2 -> Dialplan Dial DAHDI -> ISDN PBX -> ISDN Equipment The user on the ISDN Equipment das enable call forwarding - Teilrufumleitung / Call deflection - so that call will get forwarded by the telco switch - and not using b channels. The forwarding request is coming in on asterisk (i

Hi all, we do have a callcenter system running with 1.4.21.1 - the agents are connected used sip phones. SIP accounts are configured using realtime (sip buddies) - and are configured with call-limit=1. It is operating just fine - but from time to time it does happen that an agent with an active call (inbound or outbound) does start to get a second call offered. I have taken a look at the

Hi all, first off all - sorry for the cross posting - i did already posted this message to asterisk-dev - after that i realized that it isn't really a -dev related question - more a -users questions. So ignore it on -dev .... we have the following setup PSTN 3 PRI Lines <---> Asterisk (1.4.22) <---> Siemens HiCom <---> Bosch Integral The Asterisk Machine

Hi all, i have had sip TLS with an own signed certificate (using the ast_tls_cert script) running on asterisk-1.8.8 - i then have updated to 1.8.9.3 - and now i get the message "FILE * open failed!" I have already recreated the certificates with the script - but still no luck... Does anyone here know the source of the problem ? best regards, Wolfgang Pichler