similar to: How can I make a rotative board?

Rodolfo, I haven't looked up how to do this with sip phones, but the zap channels can be configured in groups that will hunt through the group until a non-busy line is found. http://www.voip-info.org/tiki-index.php?page=Asterisk%20ZAP%20channels#comments Here is a link to PBX hunting with the dial plan. http://www.voip-info.org/tiki-index.php?page=PBX+Hunt+Groups I haven't tried

Hi all. I've being reading posts from the list since yesterday and I feel this question was answered a lot time ago, but the list archives are a mess (yet). I hope some one is willing to help me out. I want to set up this: caller ----- PSTN ---- (SOMETHING1) ------ VoIP --------- (SOMETHING2) ---- PSTN I think this must be a very basic architecture, but I'm not sure wat SOMETHING1

Hi. I have installed a x100p (THE x100p for those who have seen my former post). Now I just want to connect a "normal" phone (not an IP phone) to the card and use it as a sip extension (I have a FWD account)... more clearly: I want to be able to pick up the phone and call any FWD user using my FWD account... receive the FWD calls in that phone, and also to be able to make normal

Hi. I'm getting new lines for using with Asterisk. In my Telco they said I could choose between Analogic lines and RDSI lines... I've already bought a TDM400P with FXO modules. Can you give some hints on the differences between RDSI and normal Analogic lines? Would I have problems for using a RDSI line with the TDM? Any other issue in general? Thanks in advance, RODOLFO --- avast!

Hi. I have a x100p card installed on my asterisk box... my zapata.conf file includes the following lines: [channels] context=default switchtype=national signalling=fxo_ls rxwink=300 echocancel=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 Basically, the zapata.conf file generated by make samples. Then in my extensions.conf I have this: [default] include => demo And demo is

Hi all. I've been reading through Wi-Ki and at the extensions.conf file description (http://www.voip-info.org/wiki-Asterisk+config+extensions.conf) The author says this: "One day, someone is going to write a proper scripting language for Asterisk that can understand a simpler, easier (and more traditional) scripting syntax. All it would need to do is translate the "high

Hi everyone! The following: Any calls coming in on extension 12121212 should get a message telling them to dial the extension of the person they are trying to reach, and then press #. The call should then go to the entered extension. This is as far as I got... *********************************************************** exten => 12121212,1,Wait,1 exten => 12121212,2,Answer exten

How to play GSM files? I want to go through some of them but I'm not sure which player to use. -- #Joseph

Hi list Can someone tell me what the exact CentOS policy for end-of-life is for 5.0? Let me explain: I have been using RHEL for a number of years and had to switch to Fedora for installing some packages only supported under F6. As I find Fedora's policy for release life duration too stringent (and I do no longer need the packages that made me choose F6), I'm considering switching over

Can anyone supply a link on a how-to (or provide intructions) on recompiling a centos 4 kernel? I'm trying my old rhel 3 commands and they arent working.

Hi, Is there any further information on this UPS interface? I have an Upsonic UPS that is reporting the same interface, but I cannot make it work or get any sensible information from it. Thanks Jon --- avast! Antivirus: Outbound message clean. Virus Database (VPS): 000706-2, 25/01/2007 Tested on: 26/01/2007 11:00:14 AM avast! is copyright (c) 2000-2007 ALWIL Software.

Hi, Is there any further information on this UPS interface? I have an Upsonic UPS that is reporting the same interface, but I cannot make it work or get any sensible information from it. Thanks Jon --- avast! Antivirus: Outbound message clean. Virus Database (VPS): 000706-2, 25/01/2007 Tested on: 26/01/2007 11:00:14 AM avast! is copyright (c) 2000-2007 ALWIL Software.

Peter, I have been doing some more testing and I find that if I modify the libusb.c file to set a report size in libusb_get_report to 16 the error messages go away. This is the code as it currently stands: static int libusb_get_report(usb_dev_handle *udev, int ReportId, unsigned char *raw_buf, int ReportSize ) { int ReportSize_test = 16; unsigned char raw_buf_test[ReportSize_test +

Hi. I cant make SIP calls from asterisk. When I start asterisk, I get the following message: What does it means?? Asterisk is not behind NAT or Firewall. ---------------------------------- [chan_sip.so] => (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found Sep 16 09:52:33 WARNING[16384]: chan_sip.c:8477 reload_config: Unable to get IP address for

Hi Alexander, thanks for your good work on the megatec_usb driver. I have made a few more (cosmetic) changes. Is the driver now working stably for your device? In particular, do the instant commands (and particularly "upsdrvctl -k") work properly? In this case, we could hopefully mark the device as "supported" and merge the driver to the main branch. There is still the

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Hi. I'm using asterisk as a PSTN -> SIP gateway, so that you can call to any of the 4 PSTN lines connected to the asterisk box from and dial your number, and asterisk will dial out through one of the 4 sip accounts I have on a SIP -> PSTN provider. I think of something like this in the extensions.conf [incoming] exten => s,1,Wait,1 ; Wait a second, just for

Hi again. I cant get my Budgetone registered in Asterisk, and I cant find what's wrong... uff. This is my config: This fragment is from my sip.conf: [12345] type=user user=12345 username=12345 secret=12345 authuser=12345 qualify=1000 nat=no host=dynamic dtmfmode=rfc2833 reinvite=no canreinvite=no disallow=all allow=g729 allow=ulaw allow=alaw context=sip_default And this is from my

Hi Jon, your patches are malformed because (presumably) your email client wrapped this lines. Could you please send these as a unified diff ("diff -u"), and send them in attachments, rather than as a copy-and-paste? Thanks, -- Peter Jon Gough wrote: > > Peter, > Here are the files I have worked on. I hope I have diff'd them > correctly. I have not touched the

Hi, I've seen that each country has its own PSTN qualities. I would like to know the minimal characteristics needed in PSTN to use Asterisk and also if some body knows which are Spain's PSTN's. Thanks. RODOLFO --- avast! Antivirus: Outbound message clean. Virus Database (VPS): 0436-4, 03/09/2004 Tested on: 07/09/2004 13:57:36 avast! - copyright (c) 2000-2004 ALWIL Software.