search for: brels

Here is a an encapsulated postscript file and some raw, generic postscript commands to precede and follow the EPS file which puts a message on the page. I have tried this out on two postscript files and it works. NOW, you will have to generate this file for each print job with the appropriate parameters, including, I suppose, the right bounding box, the right location of the text, maybe change

The sound file is played correctly for the first time, then when rewind to the initial position. then copy PCM to buffer, the OpenAL report an error. It seems like the OpenAL doesn't recognize the PCM data. The OpenAL error number : AL_INVALID_VALUE 0xA003 void Buffer::PCMData (ALuint id, ALenum eFormat, ALvoid *data, ALsizei size, ALsizei freq) { // Copy

I'm trying to debug a NAT issue and I can't make up my mind if the problem is with my Vigor 2800 or Asterisk 1.6.2. I know the Draytek is alleged to suffer from nat 'issues' but I did not have the issue with 1.6.1 - so I'm wondering if something has changed? The Draytek offers 'NAT & Routed' on a single device - so my Asterisk sits on a Public IP, and I have a

This is a real 'newbie' type question, but I can't get my brain to work today. Is it possible to re-direct an incoming SIP call based on it's CLI? Ideally I would like to check incoming calls against a short whitelist (of say 20 numbers) and redirect to a different extension if there is a match. I would also like to redirect calls that fail to present any CLI (aka

Dear All, How can we know the On board supports echo cancellation I have *Digium, Inc. Wildcard TE410P quad-span T1/E1/J1 card 3.3V (rev 02)*board all working fine but sometimes i got echo when user are calling a PRI. is there any way to know on board echo cancellation . regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL:

Scenario: [Asterisk Server] on routed/public IP \/ /\ \/ /\ \/ /\ \/ /\ \/ [Draytek Router] --> internal IP's \/ /\ \/ /\ \/ /\ \/ /\ \/ [internal Network 192.x.x.x] [IP10s] + [IP10s] + [Softphones] Everything is good as long as only *1* phone is registered from the internal network. The minute another phone goes online registration is dropped and the sip debug complains:

Since upgrading from 1.6.1 to 1.6.2 I get this error on boot: Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory Or if I try to connect to Asterisk: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) If I manually create /var/run/asterisk/ and restart Asterisk I can connect to it, but if the server is rebooted

I have something going on that I don't fully understand after a weekend of looking for answers. I have an iax account with Tesco that works flawlessly with the Zoiper client - but is giving me trouble with inbound calls in Asterisk 1.6. After some playing I have ended up with an iax.conf file that looks like this: [general] calltokenoptional = 77.75.0.0/255.255.248.0 maxcallnumbers = 16382

At the risk of being flamed.... Has anyone had any success get the 'El cheapo' Wildcard W100P clone's (?20 flavour) to work with UK Caller ID? I'm not sure what the status of Asterisk 1.6 is with respect to UK caller ID, being that we have an odd method of sending the FSK ahead of the ring, but I'm guessing I can't be the first to ask this? Keeping in mind that cost is