search for: pedhazur

...747XXXXXXX:YYYYYYY@proxy01.sipphone.com/4321 [proxy01.sipphone.com] type=peer ;auth=md5 secret=YYYYYYY username=1747XXXXXXX fromuser=1747XXXXXXX fromdomain=proxy01.sipphone.com host=proxy01.sipphone.com nat=no qualify=no canreinvite=no disallow=all allow=ulaw ;context=default ;callerid="Hadar Pedhazur" <1747XXXXXXX> (The above has been variously named sipphone, sipphone-out and now proxy01.sipphone.com, all with the same exact result! Also, the above has been tried with auth=md5 uncommented as well, and also no password, and insecure=vary, etc.) Now extensions.conf: ; Dial SIPPhone...

...application. AsteriskHouse*CLI> iax2 debug IAX2 Debugging Enabled Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00001ms SCall: 22150 DCall: 00000 [10.251.1.2:4569] VERSION : 2 CALLING NUMBER : XXX-XXX-XXXX CALLING NAME : Hadar Pedhazur FORMAT : 2 CAPABILITY : 2 USERNAME : hadar CALLED NUMBER : 4444 DNID : 4444 CALLED CONTEXT : from-hadar Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 1655939482ms SCall: 00004 DCall: 22150...

First, many thanks to Greg Boehnlein for his patch to chan_agent.c for adding a "preackannounce" option. I am running CVS HEAD from 2005/07/31, and the patch failed in a few hunks, since the code was refactored to add in some CASE statements where there were compound if statements before. Anyway, I have successfully updated the patch to work against head as of 3 weeks ago, and would

I haven't seen anything this strange, and it's 100% reproducible. I'm hoping that there are some clever ideas out there for what to look for, since I can test to my heart's desire on this one... My Dad lives in Florida, and has a Bellsouth DSL line. Of course, he has a regular POTS line connected on the same line. He has the appropriate filters on every jack that has a phone

Following up on a thread that I started about Agents/Queue and acknowledging calls before bridging them... Greg Boehnlein said that he was putting his efforts into ICD. I downloaded and installed ICD, and I can get simple queue and agent stuff working fine, and see that this new design is much cleaner and more powerful. That said, in the sample conf files, the "acknowledge_call"

...sk, an open source VOIP! Free food! Rob Page, Zope Corporation CEO and President, will present a technical session on Asterisk [1] installation, configuration and operation. A brief discussion of connections to the public telephone network and internet telephony providers will be presented. Hadar Pedhazur, Zope Corporation Chairman of the Board, will present a technical session on call handling and processing using Python extensions to Asterisk. We will also serve delicious fruit, cheese, and soft drinks. We've had a nice group for all the meetings. Please come and bring friends! We also are...

I don't even know where to begin. I run a lot of production Asterisk servers, for a couple of years now, with no real problems. We built a brand new box, CentOS 4.2, and installed Asterisk 1.2.4 from source tarball(s). Built fine, and started up fine. Any attempts to do local audio (e.g. a "Playback(welcome)") results in complete silence. Worse, the Playback command will hang

Hello, did you got your issue solved? I am suffering of the same issue.... On 4/28/07, Hadar Pedhazur <hadar@unorthodox.com> wrote: > > I snipped all of the previous data, as I'm trying to "boil down" > this problem to its essence... > > I turned off the firewall for a few seconds, and still got no > audio. For those that will be suspicious, the commands were:...

I have a system with no Digium hardware in it (two others with 2 X100P cards in each of them as well). I'm interested in using MeetMe in the one without the hardware (it works great in the ones with the hardware). I can't use ztdummy, because the system has usb-ohci drivers, rather than usb-uhci. I have read the little there is about zaprtc, and I am wondering whether there is a

I have been running Asterisk for years on a machine with a public IP. Most recently, I have been running 1.2.17, from the day it came out, with no (noticeable) problems. Yesterday, I switched over to a new server that is on the same public subnet, one higher than the original server. I built 1.2.17 from source on that machine (as I did on the old server). My firewall on the new machine is

...M 'out' to your site. While they are doing this, call from another phone to make sure the test fails, or shows what the problem is. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Hadar Pedhazur > Sent: Tuesday, May 09, 2006 2:18 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] PSTN Incoming call on real line disrupts > VoIPcall over DSL circuit - EXPLAINED > > Juergen K. Zick wrote: > > HI, > > > > well,...

Hi all, After a long time having no time, I have finally done some fresh work on my software fax machine. I have replaced the original carrier tracking with something more robust. I have also added 4800, and 2400 bits per second modes, and cleaned up a few bugs in areas like superfine mode operation. I apologise for this update taking so long. At ftp://ftp.opencall.org/pub/spandsp you will

I am sorry to ask such a simple questions. I have been using Asterisk successfully for well over a year now on three servers. I was using CVS HEAD, and the last time I updated was sometime back in July. I decided to switch to the recent stable 1.0.3. I built zaptel, libpri and asterisk, and installed them in that order. All installations reported success. (I stopped asterisk before installing

Sorry to bug all of you with this, but I have to assume there are a number of Sipura experts here... I have a Sipura SPA 2000 that I've been using for nearly 2 years now. It's flashed up to firmware 3.1.5. On line 1, I no longer get Caller ID (it used to work, and I can't remember when it stopped). On line 2, I always get Caller ID. To my old eyes, _every_ switch on both lines

I have a "support call" AGI script that has been working flawlessly for a couple of years now. It dumps the customer into a MeetMe conference room, then dials a bunch of support engineers, and connects anyone who accepts the call into the conference room. The conference room is recorded. After the support call is over, the recording is emailed to a list for quality control, etc. It