search for: abraxa

Hello all, One of our clients reported that they are experiencing echo in SIP calls (even on internal ones). What do you think could be causing echo in internal SIP calls? We're using Polycom telephones, do you think they could be causing it? Thanks, Alex

I need to make file sharing in my network, the file server sharing have to be in a machine that can be seen from outside. In the begining I wanted to use Samba but now I'm not sure which would be secure and fast, if NFS protocol or Samba. __________________________________________________ Do You Yahoo!? Get email at your own domain with Yahoo! Mail. http://personal.mail.yahoo.com/

Hallo altogether! First of all I want to thank the developers for the great codec! But I have the following problem now: I have built my own Linux from Scratch and also compiled everything that is necessary to have ogg and vorbis and tools for de- and encoding. Just the encoding turned out to have very poor quality independent how high I set the quality level. It has some kind

Hi All, I have one rather annoying problem...my PBX can work great for weeks, when suddenly I start receiving these messages when I try to make a zaptel call: Oct 31 13:52:47 NOTICE[15636] app_dial.c: Unable to create channel of type 'ZAP' (cause 34 - Circuit/channel congestion) Oct 31 13:52:49 NOTICE[15648] channel.c: Unable to request channel ZAP/g1/247 I'm using Sangoma

Hi all, Does anyone know if Asterisk or any available 3rd party add-on for it support "voice recognition" (not "speech recognition") - task of recognizing people from their voices? Thanks, Alex

Hi murf, Speaking as someone who designs and builds billing platforms, this is very exciting. One little thing I have most problems with is the good old fax detection. I know that NVFaxDetect et al do actually answer the call and, therefore, get flagged as ANSWERED in the CDR. But, if the call never gets answered after the initial detection - then, to my customer, it is a missed (NO ANSWER)

Hi guys, I was wondering whether there's anyone who could share his/her experiences with using Microsoft RTC Library. In particular I am wondering what Ethernet capacity should I have in scenario of 30 people using Microsoft RTC Library for SIP communication (PBX is obviously Asterisk :-) ) concurrently (alaw codec being used)? What problems can be expected in such scenario? Would a good 1

Hi all, I need your lights :) There are many hardware provider for E1 cards on the market, what's your exeperience with E1 and what's the preferred provider for Asterisk out of Digium? Olivier

Has anyone in the UK got ANI to work on an inbound call ? Using asterisk 1.4 trunk and zaptel 1.4 trunk, with a Euro-ISDN 30 Julian ______________________________________________________________________ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email

...cross-national approach it has an unique position in the German-speaking area. Besides global players in the IT industry the initiative promotes small and medium business companies that deal with Open Source. Members are amongst others IBM, MySQL, Fujitsu Siemens Computers, Novell, Red Hat, Collax, Abraxas for IT providers, Stuttgarter Versicherung, Federal State of Bavaria, Schweizer Bundesverwaltung and Universities like Linz, Heilbronn, N?rnberg and Mannheim. Further information can be found at http://www.lisog.org/ - -- Greetings Nico Gulden Technical Lead Linux Solutions Group e.V. - LiSoG B...

...cross-national approach it has an unique position in the German-speaking area. Besides global players in the IT industry the initiative promotes small and medium business companies that deal with Open Source. Members are amongst others IBM, MySQL, Fujitsu Siemens Computers, Novell, Red Hat, Collax, Abraxas for IT providers, Stuttgarter Versicherung, Federal State of Bavaria, Schweizer Bundesverwaltung and Universities like Linz, Heilbronn, N?rnberg and Mannheim. Further information can be found at http://www.lisog.org/ - -- Greetings Nico Gulden Technical Lead Linux Solutions Group e.V. - LiSoG B...

I went to put in the new sound files over the weekend, but forgot to backup the custom folder and lost my custom digital receptionist files. I then had to copy the old files back from a duplicate machine. The problem is now though that voicemail just hangs up when I dial it. Other apps work - *70 for example gives me 'call waiting...activated' so I know it's accessing the files

Hi guys, I have noticed that 'logger rotate' command only rotates log files in the /var/log/asterisk directory, but not in the subdirectories. How could I rotate my /var/log/asterisk/cdr-custom/Master.csv log file? Regards, Alex -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, What does 'Got reject for frame...' message really mean, what could be causing it, and how should one start troubleshooting it? Thanks in advance, Alex

Hi guys, I have an IVR configured in my PBX, which callers use to browse thru the list of stores. Once they choose a store, the call gets redirected to that store (obviously using Dial() application). Now, my question is: Each of this calls is logged in the calllog as one entry. How could I configure my dialplan so that that portion of the call, which is in fact just browsing thru the IVR,

Hi guys, Is it possible to allow remote peers to connect to your local DUNDi Asterisk box, even if you don't have them listed in the dundi.conf? Alex

Hi guys, Is there any option to retrieve queue stats (particulary am interested in the time of currently longest waiting caller) from the dialplan? Thank, Alex

Where could I find a manual about administering Samba with Webmin? __________________________________________________ Do You Yahoo!? Yahoo! Auctions - Buy the things you want at great prices. http://auctions.yahoo.com/

Hi guys, I'm using RedHat and am trying to configure my sudo to enable user 'testuser' to run Asterisk. However whenever I try to run 'sudo asterisk' as 'testuser' I get prompted for password. This is the line in my sudoers configuration file that I thought should do the trick, but it doesn't: testuser ALL=NOPASSWD: /usr/sbin/asterisk Does anyone know how to

Hi guys, I have the following configuration: 10 SIP softphones <--> Asterisk <--> PSTN Audio is always good on SIP softphone side, but callers from PSTN side *sometimes* complain that the audio quality is bad (and volume low). The QoS is turned on on the computers where SIP softphone is installed, and the tos setting in sip.conf is set to 0x18. The interesting thing is that usually