search for: kleis

Hi, I'm trying to set up an Asterisk pbx based on a Fedora Core 1 Linux box (customized kernel version 2.4.24). I want calls from my SIP soft-phones to simply be dumped onto the PSTN line via a BRI (EuroISDN). I have a cheap HFC-S based PCI ISDN card connected to the NT1+ interface, so I need zaphfc. I've read everything I've found at www.voip-info.org, then I've downloaded the

Anyone had Realtime working with IAX2 and RSA authentication to connect two PBXs, please? It seems that inkeys/outkey fields are not read at all and the following warning is logged when dialing: Jun 2 02:41:36 WARNING[6299] chan_iax2.c: I don't know how to authenticate ******** to XXX.XXX.XXX.XXX Using iax.conf it perfectly works. Maybe a bug in Realtime? TIA, Alex

Hi all! So far I've always used plaintext passwords for SIP, but now I've decided to use MD5 encryption. For each client I edited its section as follows, then: auth=md5 md5secret=hashed_passwd ;secret=plaintext_passwd where hashed_passwd is the output of echo -n "user:realm:plaintext_passwd" | md5sum When the first SIP clients registers with Asterisk after a "sip

Finally I decided to rewrite my dialplans according to the right sintax, that is exten => someexten,priority,application(arg1,arg2,...) should be exten => someexten,priority,application,arg1|arg2... Isn't there anybody skilled enough in regular expressions that could write a quick Search 'n' Replace vi command, please? TIA, Alex

Hi, to register my Asterisk with a SIP provider I use the following syntax, as shown in the default sip.conf: register => 2345:password@sip_proxy where [sip_proxy] type=peer context=from-messagenet host=sip.messagenet.it port=5061 <------------- please note this one!!! 5061 is provider's port I have to register to. This also would work for me: register =>

Hy, I have currently a big problem with Win95 clients generating files with invalid inode types. This only happens with Quark documents created from scratch. Example: ?rwxrwxrwt 1 falkenbe av 35638891 Jul 29 16:02 /vol04/pc_03/Deerberg/ctp_Deerberg/test07 after copying the file it really is a file !! -rwxrwxrwx 1 schoenhe av 35638891 Jul 29 16:00

Hello all, It''s my first letter on this list, and, my English is not very well. Please take me indulgence for grammar/syntax and over erorrs :)) I have trouble for acl''s of ip range. But, acl for one host (with ip adress) work fine. Please help me for make work acl/find erorr in acl. Becouse I''m new shorewall user, I maked test configuration on Virtual Mashine

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users- > admin@lists.digium.com] On Behalf Of Matteo Brancaleoni > Sent: Monday, July 26, 2004 5:22 PM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Can't dial SIP<->EuroISDN (HFC-S based > PCIISDN card): Unable to create channel of type 'Zap'

Hi all, finally yesterday I was able to set up my Asterisk machine and now it works greatly. Assuming I use Asterisk also to manage a BRI via a HFC based ISDN card, I can dial outgoing calls via ISDN but I still can't answer incoming calls from that BRI. I read that special extension 's' is for analog only. What do I have to put in extensions.conf to make Asterisk answer incoming

Hi all, I'm playing with Asterisk and I've already configured all needed .conf files. It works quite well, but now I need your help to tune the system: when I place a call from a softphone to the PSTN, I can't hear directly Telco's tones and I can't use its services, e.g. a mobile's answering machine. I don't know if I have to modify the dialplan or if it depends on my

Hi all! Was anybody able to install kapejod's zaphfc drivers together with RealTime application? I'm in big trouble because bristuff relay on STABLE version, while RealTime is included in the CVS-HEAD. I found this hint, "Installing zaphfc with CVS-Head" at http://voip-info.org/wiki-Asterisk+zaphfc+install, but it was written many months ago: may it be still useful? TIA, Alex

Hi all! We've released VoiceOne 0.4.0, a web-based and open source solution which allows to fully manage an Asterisk service hosted on a LAMP server. We focused on an charming and overall user-friendly interface. Thanks to the authentication based on roles, once configured by a super user, the PBX may be easily maintained even by an Asterisk unskilled users. From a technical point of