similar to: Monitored outbound dialing via Zap interface ?

I'm using a T100p to interface to a channel bank and from there to analog PSTN lines. Because of my particular setup I have to do post-connect inband DTMF dialing - which takes up to 5 seconds for a 10 digit number, assuming 0.5/sec per digit (ie. using "zap/g1/31|5|D(6045551212)". Even with an 'outside transfer' voice prompt before commencing dialing my users are getting

Thanks a lot for all the suggestions! Unfortunately, it still gives problems. Most common error message is "ast_realaudio_callback Failed to write frame" after "paying the beep". Then it says "User disconnected". Also, it doesn't react to any extension entered and doesn't do any forwarding (as it should in "exten =>

Just wondering if there is a reason PPP support is compiled into zapata by *default*: # Uncomment for Generic PPP support (i.e. ZapRAS) # KFLAGS+=-DCONFIG_ZAPATA_PPP Especially since the comments imply that it should be commented out by default... The main reason I ask is because I usually try to re-compile the kernel to only include the bits that I need, and so I don't include PPP...

Hi all, I've just updated to current CVS, and have 2 polycom IP phones, one is a IP600 and the other is a IP300. The IP600 shows the status of the IP300 and a ZAP line quite nicely, but the IP300 won't show the status of the IP600.... Is there any additional debug apart from "show hints" to see why this might not be working ?? -= Registered Asterisk Dial Plan Hints =-

Hi, I was just wondering if it is possible to cause a script to run (configured in the domu.cfg file) each time "xm create domu.cfg" is run, but before the machine is actually started? ie, I''d like to "setup" the disks for the VM before xen/qemu tries to use the devices and allocate to the domu. I''m using xen 4.1.3 on Debian testing, and using the xm

If anyone is looking at upgrading their Digium hardware, or has an excess of any of these cards, please let me know. I am not interested in non-Digium hardware, but thanks for asking :) E100P E400P TE410P I only need one card, in that preference order. Would also prefer if the card is already in Australia. I'm in Sydney/Australia if that matters to you. Regards, Adam -- Adam Goryachev

I seem to have tripped over a problem that just seems plain weird to me... I have the TDM card with 1 FXS and 3 FXO interfaces. The FXS connects to a modem and EFTPOS terminal, which is working fine. The 3 FXO ports are connecting to another PBX as internal extensions. Usually, everything works fine, however, yesterday incoming calls stopped working. When an extension on the PBX calls one of

[This is being re-sent because the previous one got caught in the 'moderators' mailbox. It would seem that the moderator never approves those emails though? It would be nice if that would happen.] [Original Message] Tonight I thought I might start hacking on gastman to extend the features in the directions that I would like. Except then I got stuck, because some information is not

Couldn't see the forrest for all the fascinating tree-like applications that are out there: For future reference, see: http://www.voip-info.org/wiki-Asterisk+call+parking :-) -----Original Message----- From: Kris Boutilier [mailto:Kris.Boutilier@scrd.bc.ca] Sent: August 11, 2004 1:10 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Inband announcement of parking slot from

I have hired someone to do some work on asterisk for me over the next couple of months, before they start, I would like to get a better idea of what other people need, which I can then dovetail into what I think my own 'requirements' are... Initially, a largish area that I think needs to be worked on is some sort of gastman type interface which will run on MS Windows, and be usable for a

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I''ve been using xen for a number of years now without issues, but now have a problem. First, the problem: I have a Windows Server 2008 Standard 64bit DomU, I''ve installed a number of applications, and it works quite well and runs stable for weeks at a time so far (ie, has never crashed yet). I tried to install QuickBooks

----- Original Message ----- > Subject: Re: [Asterisk-Users] Analog Phones with Status Light Indicators > From: Adam Goryachev <mailinglists@websitemanagers.com.au> > To: asterisk-users@lists.digium.com > Organization: Website Managers > Date: Thu, 12 Aug 2004 14:53:02 +1000 > Reply-To: asterisk-users@lists.digium.com > > On Wed, 2004-08-11 at 20:42, Steven

I''ve been trying to migrate a win nt 4 machine to a xen domu for the past few months with no success. However, on my current attempt, the original hardware no longer boots, so I''m trying to resolve the issues with xen properly, or else take a long holiday... Anyway, the physical machine had a 9G drive (OS drive), a 147 G drive (not in use) and a 300G drive (all SCSI Ultra320 on

The originating PRI system passes the entire dialed number in the d-channel setup frame, thus the concept of a wait time for additional digits is meaningless. Progressive digit gathering implies that the signalling is occuring 'in-band' as would be the case with DTMF signalling on analog lines. You need to look in the Ascom and find the configuration table that lays out the dialplan for

Hello, I am looking for information on setting up digium FXO card for use as a PSTN Gateway (H323-PSTN) to work with GNUGk. I am basically looking for the setup and it would be great if anyone can share his experiences with the same. Also, if there are any limitations in going for such a setup and problems that may arise/things that I should keep in consideration. Thanks & Regards, Deepak

Well, I'm writing yet another queue_log analyser program in PHP, and I have noticed the following entry in my queue_log file from today: 1104796626|1104796618.532|queue|NONE|ENTERQUEUE||no 1104796664|1104796618.532|queue|NONE|EXITWITHTIMEOUT|1 So, pretty sure that I didn't make someone wait 30 minutes in my queue. extensions.conf snippet: [remote-oldnum] exten => s,1,Answer exten =>

Unfortunatly no on both counts. The arrangement right now has: PSTN Trunks & Stations <-> Nortel Norstar#1 <-CT1-> Asterisk#1 <-IAX2-> Asterisk#2 <-CT1-> Nortel Nortstar#2 <-> Stations The Asterisk boxes provide Voicemail to their sites Norstars and intersite calls over IAX. Local Voicemail works flawlessly at each site but there have been reports of PSTN calls

I'm having a problem with intersite calls over IAX2 being abruptly terminated. Nothing odd shows in any of the logs for Asterisk or the host. The only think I can think it might be is a lag-spike on the site to site connection. How sensitive is IAX2 to lost frames, lag spikes or large variations in jitter with the GSM codec and: bandwidth=low jitterbuffer=no trunkfreq=100 ; Raised from

Hello all, I updated from CVS 3 days ago and now my IAX2 gateway is dropping calls without warning. It happens right in the middle of a conversation with no pattern. I never had this Problem before and am usually talking 2-3 hours a day. Is their a bug? Should I rollback? Cheers, Paul Seniuk -------------- next part -------------- A non-text attachment was scrubbed... Name: Paul

To answer my own question for the record: The relevant timing parameters in zaptel.h are #define ZT_MINPULSETIME (15 * 8) /* 15 ms minimum */ #define ZT_MAXPULSETIME (100 * 8) /* 150 ms maximum default, lowered to 100ms */ #define ZT_PULSETIMEOUT ((ZT_MAXPULSETIME / 8) + 50) And the pulse detecion loop that consumes these parameters begins at line 4866 of zaptel.c The