search for: hatzistavrou

...lse noticed this? I went through the issue tracker but I couldn't find any relevant bug posted in the past. I am certain that in previous versions I could set the reply message to the desired value, so I wonder if this is a bug in this particular version (1.4.33.1). -- Best regards, Vlasis Hatzistavrou.

Hello, Does anyone know any solution to this? Or is Asterisk-Dev a more suitable list to ask this question? Best regards, Vlasis Hatzistavrou. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Vlasis Hatzistavrou Sent: Thursday, February 16, 2006 3:43 PM To: asterisk-users@lists.digium.com Cc: 'Vlasis Hatzistavrou' Subject: [Asterisk-Users] AGI...

...order to use any other codec, we have to enable only the needed codec and disable all others. Again, this problem did not exist in older * versions, like 0.9.2 and it's limiting the capabilities of Asterisk in H323. Has anyone dealt with this problem successfully? Best regards, -- Vlasis Hatzistavrou.

Hello, I'd like to ask if there are any news about CE certification of the E1 boards. I know that the T1 boards are FCC certified but I'd also like to know what is the status for CE certification. Thanks for any input, Vlasis Hatzistavrou.

...3-0.6.3b/asterisk-oh323- 0.6.3b/asterisk-driver' make: *** [subdirs_all] Error 1 ******************************* I'm not a very experienced Linux user so I can't really figure out what the problem may be in this case. Does anyone have any suggestions? Thank you in advance, Vlasis Hatzistavrou.

...1 > > >>>>cards.... >>>> >>>>Michiel >>>> >>>>-----Original Message----- >>>>From: asterisk-users-admin@lists.digium.com >>>>[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Vlasis >>>>Hatzistavrou >>>>Sent: woensdag 2 april 2003 15:25 >>>>To: asterisk-users@lists.digium.com >>>>Subject: Re: [Asterisk-Users] CE certification for Europe >>>> >>>> >>>>Thank you very much for your reply and for clarifying this point. >&gt...

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...that since Asterisk now supports MGCP some of the people who develop the MGCP channel driver may have such a capture available. I need this for my MSc thesis and unfortunately, I don't have any MGCP compliant hardware... Thanks for any help/input & sorry for the off-topic posting, Vlasis Hatzistavrou.

...ess tones or announcements from the PSTN when we dial ouot through the i4l card. For the moment, if we don't inject the r option in the Dial command, there is only silence during the call negotiation... Using Asterisk RC2 with Eicon passive PCI 2.01 card... Thanks for any help, -- Vlasis Hatzistavrou

I've done quite a bit of googling and haven't found a solution to my problem. I've got the Digium dev kit (wctdm11b) set up and working. I've compiled spandsp and can receieve faxes from eFax (www.efax.com) but the pages are blank. The page count is correct, in that if I fax a two page document, my tiff file has two pages, but they are white blank pages. I found one similar

I'm using get_data in an AGI script and am having a problem when, after a long time in my IVR, when I ask for a 10-digit phone number, the first few tries are always invalid -- the number it reads back is very strange, almost like the DTMF tones from other answers were being cached and then dumped on the call to get_data. Anyone ever experienced this before? I have to do some major

...I would like to know if such an option is available in AGI like an onanswer() function or something equivalent that I can use. Any help would be really appreciated, as I've been searching www.voip-info.org and the Asterisk mailing lists for days now, without any success. Best regards, Vlasis Hatzistavrou.

Hi, A glance at sourceforge.net/projects/openh323 Help Forum made me wonder if this location is the one to use (I got trouble in the past when google pointed to an obsolete site) : some quite old messages remain unanswered. Cheers -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I've been playing with T.38. I observed that mostly but not always, it's the "calling endpoint" that reINVITE the other party to drop current SIP/G711 session and start a new T.38. But sometimes, it's also the callee party that reINVITE the calling party. Which is the "standardized" or most common, way to start a T.38 session ? Shall it come from callee or

...e extension that I use is pretty simple: exten => _X.,1,Dial(SIP/${NUMBER}@x.y.z.w,,r,) Does anyone know if the behavior of 'r' has changed but was not documented? If yes, then how does one inject ringback audio before the call is answered on the called end? -- Best regards, Vlasis Hatzistavrou.

Hello, I use Asterisk CVS-v1-0-12/21/04-11:05:29 and I noticed that when we receive calls from a partner's IP address (who has a static host entry in the sip.conf file) but the RTP comes from a different address than the signaling, our * sends a 403 forbidden message and drops the call. This problem does not llow us to receive calls from SIP proxies. Was this fixed in newer versions of