search for: pawlowsky

Hi Everybody Any significant changes to CVS HEAD over the last couple of days. I've got two asterisk boxes - both on public IP but one is dynamic. The one on dynamic IP registers at the other one - that part is fine. Calls going from the one with dynamic to the static one goes fine. Call the other way results now in: Failed to authenticate user "1101"

Hi, I've found the Vertical Service Codes / vservices.inc of Julian in the cache of google. It's an very extended extensions include with all the *21 *67 etc services implemented so it is stored to ODBC or if you replace it to Dbget/put etc. I'm wondering if somebody has the macro/agi for using these extensions once stored in the Asterisk db or ODBC. Or am I missing something? And

Hi, I was asked to create a Queue which instead of playing MoH it generates the ringing tone. I had a look around but could find anything, I would welcome and help. Regards Mateusz

Hello folks, i upgraded to the actual CVS head from yesterday (27.5.) but can not get incoming SIP calls from my provider (sipgate). If someone calls my number, my asterisk responds with the following error: May 27 21:30:21 NOTICE[1114606512]: chan_sip.c:6351 handle_request: Failed to authenticate user "<CallerID>" <sip:<CallerID>@217.10.66.11>;tag=as38e9693c I

Hello, I just noticed an error in the current version of Klaus-Peter Junghanns bristuff package, especially in the HFC module. Everytime I try to unload the HFC module with "modprobe -r" I got a kernel panic and the complete server hangs up so I need to do a hard reset. Regards, Julian Pawlowski

Hi-- I've installed icecast2 and ices2 on an OpenBSD box. It all seems sane from the server side, but when I try to listen to my stream from winamp 2.76 with Peter Pawlowski's vorbis plugin (1.12), from a win98se box, nothing plays. Nothing is appended to the ices2 log, and the following is appended to the icescast2 log (when I terminate icecast2): *** [2001-08-02 22:05:16]

I'm having issues with compiling libvirt on OSX 10.12  ~/Development/github/libvirt/ [master] make /Library/Developer/CommandLineTools/usr/bin/make all-recursive Making all in . Making all in gnulib/lib /Library/Developer/CommandLineTools/usr/bin/make all-am make[3]: Nothing to be done for `all-am'. Making all in include/libvirt make[2]: Nothing to be done for `all'. Making all in

> I'd like to create the following: a user picks up the phone > (gets a dial tone), dials '0' for an 'outside' line, gets a > second (different?) dialtone, and is able to enter an > external phone number. Klaus-Peter Junghanns has something like this on his page: http://83.137.99.170/jn/relaunch/asterisk/page19.html It didn't work for me correctly so I

Hi all, I've run into an issue we have with dovecot versions 2.0.12 through 2.0.14 on latest CentOS 6.2 when dealing with Maildirs located on NFS partitions. Although I saw Mr. Egbert's thread, his resolution does not work on our end. When dealing with users with over 500 messages in any of their Maildir folders, they occasionally receive an error from the server: Internal error

We are looking soon at buying a system to deploy asterisk as our company's PBX. We run SuSE here and like it and our asterisk test platform is SuSE 9.0 with the 2.4 kernel. Is anyone running * and the zaptel drivers under SuSE 9.1? With the 2.6 kernel? Is * 64-bit safe (i.e. no 32bit assumptions in the code) so I can run it on an AMD Opteron in 64-bit mode (with whichever kernel is

Hi - Anyone using zaphfc cards (bri-stuff-0.0.2) on a Portugal Telecom BRI service? I am getting loads of errors and the audio is very distorted all the time. System is a Dell PIII/933 MHZ with SCSI. Here in the UK we have perfect audio with a Compaq PIII/500 MHZ SCSI system. Files are identical in both systems as below: Is there anything magic about Portuguese ISDN? The PCI bus in the

I was just wondering if someone has experiences to use Asterisk in an existing Ericsson MD110 environment. Particulary I'd like to know if it is possible to use the MD110's system phones directly connected to Asterisk. I'm not very familiar with it but would it be possible to use ADSI with these phones? Are they more like analog or more like digital phones or is the protocol even more

Hi all ! We have 3 NTBAs which are all going to our existing PBX. Our areacode is 06003 and our DDI enabled number 9141. I want to exchange that PBX with Asterisk, but still struggle to get it working. My CAPI.CONF is currently like this: [general] nationalprefix=0 internationalprefix=00 rxgain=0.800000 txgain=0.800000 [interfaces] context=in-capi msn=60039141 incomingmsn=* controller=1

Hi there! I just try to play with die wake-up function described in http://www.voip-info.org/wiki-Asterisk+tips+wake-up Everything looks fine but there seem to be missing some soundfiles like "wakeup-menu". Where can I get these files in order to make this feature usable? Regards Julian Pawlowski

> > I've installed icecast2 and ices2 on an OpenBSD box. It all > > seems sane from the server side, but when I try to listen to my > > stream from winamp 2.76 with Peter Pawlowski's vorbis plugin > > (1.12), from a win98se box, nothing plays. > > try the newest plugin and/or jorbis. It turned out the audio files I had grabbed from the Net for

I just upgraded my RH9 installation to Fedora 2. To my surprise, imapd seems to be gone from the core. I noticed there are now two IMAP servers docevot and cyrus. After reading brief descriptions of both, dovecot would seem to be more compatible with my setup. Especially since I'm using mbox's. I have 2 questions. It seems like dovecot does not support mailboxes inside of a

Hi there, I just finished work on my current PBX dialplan configuration. I didn't find anything compareable yet so I think it might be interesting for someone else too. Short overview: - seperation of the numbers in blocks (just definded) - error/cause handling - advanced access control (even vor ISDN Zap clients) - access levels - denylist with national number blocks (for Germany) -

Hello, I would like to use my existing HST Saphir V S2M PCI with Asterisk. Unfortunately I could not find any information about the usage with Asterisk. Does someone know if it's possible to use it? Do I need to do something special? Regards Julian Pawlowski

Hi, I have a strange problem, after some time my asterisk server starts using more and more CPU resource and after some time it becomes unusable. I need to kill the asterisk process before it renders the whole system unusable. I have a few SIP phones, IAX trunk to other system and voicemail running on that server. Now, how do I troubleshoot, which part of asterisk is causing the load on CPU ?

Did you use only win95 dlls? If you used dlls from any other version of win, htere could be problems with dcom 95. > > > Ivan Leo wrote: > > To install dcom95 you must have a windows 95 license (The license of no > > other version will do), so you must have win 95, have you tried to use > > the win95 native dlls? > > > > Krzysztof Pawlowski wrote: >