search for: pichler

...on(226) ldap_open_connection: connection opened [2003/05/28 16:06:00, 2] passdb/pdb_ldap.c:ldap_connect_system(260) ldap_connect_system: succesful connection to the LDAP server [2003/05/28 16:06:00, 2] passdb/pdb_ldap.c:ldap_search_one_user(272) ldap_search_one_user: searching for:[(&(uid=pichler)(objectclass=sambaAccount))] [2003/05/28 16:06:00, 2] passdb/pdb_ldap.c:get_single_attribute(370) get_single_attribute: [uid] = [pichler] [2003/05/28 16:06:00, 2] passdb/pdb_ldap.c:init_sam_from_ldap(505) Entry found for user: pichler [2003/05/28 16:06:00, 2] passdb/pdb_ldap.c:get_single_attrib...

...00% 1.98MB/s 0:00:04 (xfer#861, to-check=51173/94679) The same - slow - transfer happens at 1GB and 5GB files. Any ideas what can cause this slow transmission? -- Mit freundlichen Gr??en / best regards Ing. Rainer Pietsch ---------------------------------------------------------- PCS - Pichler Computer Systeme Inh. Claudia Pichler-Pietsch Hauptplatz 10 A-2751 Steinabr?ckl ?sterreich / Austria ---------------------------------------------------------- mail: r.pietsch at pcs-at.com web: http://www.pcs-at.com tel.: +43 (2622) 420 19 / 15 mobil: +43 (676) 31 242 69 fax: +43 (2622) 420...

Hi! What is the preferred way to make database lookups from within the dialplan? I only know the MYSQL function from asterisk-addons. Are the other methods too? (e.g. for postgresql, unixodbc) thanks klaus

...he 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

hi all, i am actually trying to get samba 3.01 (on SLES 8.0) working as PDC with the ldap backend. I have already configured nsswitch to also use ldap for groups and passwords (the root user is still in the /etc/passwd file - i can't imagine that putting the root user into openldap is a really good idea). There is one sentence in the howto

...ways (Axel) 3. Re: pri with TE410P not working (Austria) (Peter Svensson) 4. Re: ZAPHFC - only for * 0.7.2? (Holger Schurig) 5. Calling the firefly network? (Martijn van Oosterhout) 6. RE: IAX2 no compatible codecs (Jason Penton) 7. Re: pri with TE410P not working (Austria) (Wolfgang Pichler) 8. Re: embedded Asterisk (Klaus-Peter Junghanns) 9. Re: pri with TE410P not working (Austria) (Michael Bielicki) 10. RE: Cost of IP Phones, or Isn't It Just Software? (Andy Powell) 11. Re: pri with TE410P not working (Austria) (Peter Svensson) --__--__-- Message: 1 From: &qu...

hi all, i have a probably very stupid question/problem. for testing purpose i am trying to get asterisk running with two isdn cards. I'd only like to here the demo sound when i call the number - but nothing works. The output of show channels is not showing any channel - should there be 4 channels ? - capi info shows my two cards perfectly. The ISDN Controller's are attached to an PTMP

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

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, i am trying to get my TE410P (see previous posts) working in Austria (telekom Austria - i am still waiting for an answer for my questions). my /etc/zaptel.conf looks like -------------------------------- span=1,1,0,ccs,hdb3,crc4,yellow span=2,2,0,ccs,hdb3,crc4,yellow bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 loadzone=at defaultzone=at -------------------------------- after

Hi all, at time i am trying to get a better idea of callgroups and pickupgroups (especially within the SIP Channel) A Pickupgroup is relative clear - everyone in the same pickupgroup may pickup a call And a callgroup does what ? - The same ? I thought that a callgroup would act like the ZAP groups - so that you then can dial SIP/g1 - and every SIP Client which is in the callgroup 1 does then

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

Since Ubuntu 20.04 (focal) the rsync option -n (Dry Run) is not working correctly. --- rsync --version rsync? version 3.1.3? protocol version 31 ---- I used this command for about ten years without mayor issues: --- rsync -vaHEiuzn --skip-compress=jpg,JPG,mov,MOV,mp4,MP4,m4v,avi,png --exclude=*.pyc --exclude='somedir/**/.gps.cnf' --exclude='somedir/**/clonezilla' --stats

hi all, i am getting crazy with my TE410P - it won't work now we already think that the only thing which can be wrong is the cable. At our telecom endpoint we have 1-2 tx, 4-5 rx So - which cable do i need to get it working Already tried a cross over cable 1-2 -> 4-5 / 4-5 -> 1-2 (with this cable nothing works -> red alarm on both ends) Also a normal cable 1-2 -> 1-2 / 4-5 ->

hi all, i've now installed a TE410P (Quad T1/E1 Primary Card - Digium) and connected it to a primary line. My telco (eTel) only told me that they are using hdb3 and crc4. So i still don't know which coding i have to use (cas or ccs) - and what timing options i have to use. Have someone already get a card like this up in Austria with a line from eTel or from the telekom Austria ? Another

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

...part i've added to smbldap-passwd changetype: modify replace: pwdMustChange pwdMustChange: $pwdmustchange - changetype: modify replace: pwdLastSet pwdLastSet: $pwdlastset i think i will go to the samba pr?sentation toomorrow in Linz/Vienna and take a look for a samba expert ;-) mfG Wolfgang Pichler

I have a client that wants to put a phone on their web page for customers to call them via their Asterisk server. ) A keypad is needed to enter credit card details. ) "Speed dial" buttons like "Tech Support," "Sales," etc. are a requirement. Actually, passing the SIP address in the HTTP link would work with a bit of arm twisting. ) Free is preferred, but not a

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'd like to know if it would be possible with asterisk (and which hardware would i need) to implement the following (or is it not possible with asterisk - but possible with ...) I'd like to set up something like a "Mobile to Conventionel Network Gateway" - so that users (with there Mobile Phone) which are registered (known Call Number) can Call a Conventionel Network