similar to: Asterisk with FXO card only, no network

libswfdec/swfdec_text_field.c | 288 ---------------------------------- libswfdec/swfdec_text_field.h | 14 - libswfdec/swfdec_text_field_movie.c | 297 +++++++++++++++++++++++++++++++++++- 3 files changed, 291 insertions(+), 308 deletions(-) New commits: commit 2e79c953066b7f6857e88e7efaaed4d8846e9609 Author: Pekka Lampila <pekka.lampila at iki.fi> Date: Wed Oct 17

Update: Using mount (read-only) and unmount commands in a script that loops continuously, we are able to see near instantaneous file modifications (less than 0.5 sec) performed on a Win7 live guest. We have to use the sync.exe utility (technet.microsoft.com/en-us/sysinternals/bb897438.aspx), otherwise changes are not visible for 10-20 sec. We tested sync.exe with our libguestfs test program

libswfdec/swfdec_text_field_movie.c | 135 +++++++++++++++++++++++++++---- libswfdec/swfdec_text_field_movie.h | 2 libswfdec/swfdec_text_field_movie_html.c | 1 3 files changed, 123 insertions(+), 15 deletions(-) New commits: commit 9ca546dd0eaab842f879ed3373ccceb7183688ac Author: Pekka Lampila <pekka.lampila at iki.fi> Date: Sun Nov 4 16:13:30 2007 +0200 More

libswfdec/swfdec_text_field.c | 18 ++++++++++++++++-- 1 file changed, 16 insertions(+), 2 deletions(-) New commits: commit 13a1f11d9aebddc8d2e24c94878547589c485832 Author: Pekka Lampila <pekka.lampila at iki.fi> Date: Sun Oct 14 18:22:17 2007 +0300 Implement aligning TextField lines even when wordWrap is off diff --git a/libswfdec/swfdec_text_field.c

Hello all, I am using a headset and the X-lite softphone (sometimes I use IAXComm, but I'm having difficulties using OSS emulation with it) to connect via uLaw to my internal Asterisk server, which is a Pentium II 400 with 128 megs of RAM. After getting this headset, most or all of the echo people on the other line were complaining about is now gone, according to them. However, every

>The call always has echo on it. The Asterisk sip extension >hears them selves echoing. The remote party does not notice any >difference. >I have also notices that the Asterisk >levels are very hot from T1-PRI to Sip. I have had good success fiddling with the txgain and rxgain values in zapata.conf on my PRI. In my setup, cranking the gain down a LOT eliminated most of the

Dear ALL: I find a program named "asterisk_b2bua" on http://developer.berlios.de/projects/b2bua/ And I also download them(two components) and try to test it. But I have not enough knowledge about asterisk. It seems a Software PBX. Does asterisk_b2bua work? Does anybody ever try it? I have questions about my scenario. |======================> UA2

Hello, all. Alas, another convoluted question. All the simple things are, well, simple so I suppose we only need to trouble the list with squirrely problems! We've noticed a call history problem when using Asterisk where the call history on the Snom phones (with which we are very pleased) reflects the number of the PBX extension used by the B2BUA to dial the end point. I assume the same

All- Recently an Asterisk server we host was hacked and used to route some unauthorized calls. We have since improved our security measures, including installation of fail2ban. The interesting thing is the way in which this was discovered. The unauthorized calls were occurring intermittently last Thurs evening thru Sat morning. On Sat morning, some of our employees were attempting to log-in

Hello all! It's my try to make b2bua from asterisk. It's patched asterisk and some AGI script for it. What it support? Full vovida's b2bua radius emulation, radius failover, LCR, Call failover, Codec based routing, Session-Timeout and much other things that can be useful. Any suggestions and critics welcome! http://b2bua.berlios.de Best regards, Mike

Hello all! It's my try to make b2bua from asterisk. It's patched asterisk and some AGI script for it. What it support? Full vovida's b2bua radius emulation, radius failover, LCR, Call failover, Codec based routing, Session-Timeout and much other things that can be useful. Any suggestions welcome! http://b2bua.berlios.de Best regards, Mike

Hello List, I've searched the internet and I haven't been able to figure this out. I hope someone here can point me in the right direction. I have a MacOSX 10.3 machine and a WinXP machine that are being used to produce large video clips (@1gig). After they are rendered and ready for playback, they are copied to the playout machines using a system that uses watch folders shared out over

I have asked a GSM operator in my country if he can route a number or a short code to my asterisk server via SIP (since they dont give DIDs in my country) the operator said they do not support SIP, they have no way of converting GSM calls to SIP to then send them to me. I would like to know what is needed from the operator side to do this, what kind of material is needed, or what can be done from

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, Is the asterisk B2BUA patches useful anymore?? I'm trying to set a prepaid SIP network and the only way seems to get through a patched asterisk with B2BUA functions.. The patches failed, Hunk + problems: I have repaired them, but is it very useful?? Thanks -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (Darwin) Comment: Using GnuPG

David Baelde wrote: > You didn't describe much of your bugs or the specific features of your > playout system... hmmm, I thought I did describe the playout system, but I'm happy to elaborate if you want more details. But you're right about the bugs. The critical (i.e. urgent) ones in no particular order: 1. PRS will disconnect from Icecast after a certain amount of time

Hi Thomas, > As this is historical data I don't see any reason why you'd try to get > this through the web interface. The use case is when there is an Icecast service, but there might not be shell access to that server. So we have a stats interface in Airtime that only needs the admin password to create a listener count graph against time. > Chances are, that there you'd

Hi everyone, I thought long and hard (ok, not *that* long) about whether or not to crosspost this to icecast-dev. In the end I decided not to since this isn't about an Icecast or Xiph project. My name is Geoff Shang and I've been an enthusiastic user of Icecast for a long time. I am also the chief tech guy for a project called ACB Radio (http://www.acbradio.org), which is run by

Hello. LCR means least cost routing, and it's billing system problem where to route a call, not b2bua's. But currently I dunno any free billing system that support it, so i moved this logic to b2bua. On Sat, 12 Feb 2005 07:05:39 +0330, mohammad <mohammad@mirzaee.net> wrote: > Hi Mike; > Thanks for your new application, but I think it would be better if you put >

Hello list. I'm having some problems with the CDR Radius in my Asterisk 1.4. I'm using two TC400B cards for transcoding. When I reach nearly 100 simmultaneous calls, the CDR radius packets are being duplicated and I'm getting this message in the asterisk console : cdr_radius.c:227 radius_log: Failed to record Radius CDR record! I'm also using the radiusclient-ng 0.5.6

> > Different brand/model, but similar as they are both el cheapo, > entry-level headsets. I tried using them on a laptop, and I get > marginally better microphone output, even with its volume cranked all > the way up + automatic gain control enabled. > > I guess those on-board soundcards by Realtek aren't as good as a > quality microphones. I'll get a USB headset