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...sage) show <----This is the line to change for user/date/printer showpage %%EOF %%EndDocument count op_count sub {pop} repeat % Clean up dict stack countdictstack dict_count sub {end} repeat b4_Inc_state restore save restore % End of insertion On Mon, Jul 01, 2002 at i06:23:19PM +0200, Roger Brel wrote: > Hi Joel, > > First, I thank you for having made an answer at my E-mail. > > I am agree with you for adding , either a PostScript function in the > original PostScript document (with an underlay, easy generated by a2ps > --underlay) , or merge on each page a PostScrip...

...ream (which the vorbis libs will fill up for us), and our s_callbacks if (ov_open_callbacks (&m_tOggMemoryFile, &m_tOggStream, NULL, 0, s_callbacks) != 0) throw string ("Could not read Ogg file from memory"); m_iStreamStart = ov_pcm_tell (&m_tOggStream); m_bReleaseStream = true; return true; } // ------------- Source.h class GAL_API Source { public: // 1. Set a loaded buffer as current buffer void SetCurrentBuffer (Buffer *pBuf); // 2. operations virtual bool Play (); virtual void Stop () { alSourceStop (m_id); } vir...

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