similar to: Asterisk on amd SERVER

How do I route all the outgoing calls through a SIP gateway, this should send more than one outgoing call to the sip gateway at once. please show me the sample configurations on how to do this. my SIP gatway can accecpt direct IP traffic or SIP proxy traffc. Thank You Kanishka -------------- next part -------------- An HTML attachment was scrubbed... URL:

Who mentioned fax? :-) Native CAPI isn't an option for us (unfortunately), as our service is a point-to-point service, which the AVM CAPI drivers don't support. We've now got another problem -- we're now able to make calls -- once. The kernel panics as soon as someone terminates a call (and, so it would seem, at various other times too). This has only been occuring since mISDN

hi has any one used OOH323C i tried this it is installed but do not know how to configure has any one used this, what is the best h323 addon to use with asterisk

Hi I want to create 2 groups of extensions, for example group 1 can't make outgoing calls they can only call other extensions and extensions of group 2. group 2 can call any of the extensions + they can make out going calls using our SIP server. Please let me know how to do this. I was going through the docs and I sae that I have to specify a group in zapta.conf , this is not clear please

Hi Guys Just a quick question regarding on the 501, has anyone been able to configure the transfer button and messaging buttons to work with asterisk? Can you share a configuration to do this? Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060301/df504eef/attachment.htm

why can't we compile the asterisk coading in windows, it's done in c++ so it should work in windows as well

Hi Guys I am just looking around for a small billing program but can't really find what I am looking for. It needs to bill straight off the CDR. It should grab all the CDR records from the asteriskcdrdb mysql database then have a rates table to that it calculate a bill from. Is there any open source packages or commercial packages that will account for billing say only 5 extensions?

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello everyone, As you know, we released Asterisk 1.0.4 earlier this week. However, there was a callerid bug in chan_zap that has caused us to go ahead and make another release. Asterisk 1.0.5 is available at all of the usual locations. I'm sorry for any inconvenience this may cause. Russell Bryant -----BEGIN PGP SIGNATURE----- Version: GnuPG

has any one installed this, i just tried this on a test server, i get voice but it's corrupted, i do not get the natural voice any idea why -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050315/af5a3bb2/attachment.htm

Off topic. Anyone know where I can get Polycom SIP software v1.6.6, unofficially? Not that Polycom is analy retentive, or anything like that... Doug

Hi, I've been having problems getting Asterisk to work with a fritz card (in ptp mode). We've managed to get everything configured, however, as soon as the call is connected we get "Unhandled message: prim 281 len -22 from addr 1000000, dinfo 0 on this port" I checked, and mISDN_dsp is loaded. Any help would be much appreciated! Regards, Chris Jones // Network Administrator

Hi how do i set an SIP users to make outgoing calls that is worth only $5. if they exceed $5 they can't make any calls. what i need is not a calling card, but to limit outgoing calls for SIP users depedning on a value i give. I use realtime asterisk. Thank You Kanishka -------------- next part -------------- An HTML attachment was scrubbed... URL:

has any one implemented asterisk with 723 and 729 codecs, what is the cheapest way. is there a limitation in the open 723 implementation ?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050225/d5daf369/attachment.htm

I've got the same problem. MusicOnHold works if I use something like: Exten => 1111,1,MusicOnHold() but if I try to answer a call and then transfer or put on hold the call, I get no music. Does anyone have any idea? Bye, Gianluca. _____ Da: Kanishka Somaratne [mailto:kani@technoportal.biz] Inviato: gioved? 17 marzo 2005 5.53 A: asterisk-users@lists.digium.com

I am trying oh323(version 0.67) , make call from sip UA to h323 gateway, can't get Call-id pass from sip UA to h323 gateway, h323 always gets call-ID sent from Asterisk as *. are there any configure to pass the correct call-id from sip UA to h323 gateway? or this is a bug in oh323 0.67? how about oh323 0.73 ? Mario On 9/29/05, Kanishka Somaratne <kani@technoportal.biz>

Hi, Im trying to thumbnail a swf file based on this found at http://www.mail-archive.com/swfdec at lists.freedesktop.org/msg00821.html. #include <stdlib.h> #include <stdio.h> #include <swfdec/swfdec.h> #include <cairo.h> void swfdec_player_save (SwfdecPlayer *player, guint width, guint height, const char *filename) { cairo_surface_t *surface; cairo_t *cr;

Hi List! any body use www.simpletelecom.com? I subscribe to www.simpletelecom.com for A-Z termination and paid US$15.00 and US$70.00 via credit card in two days, but my account has US$15.00 only. I checked my credit card from the bank and they said me the payment already paid to merchant. I've lost US$70.00 :( so anyone here has experience with them? are they a SCAM? Thanks! </Madhawa>

Hello all, Can I execute linux command like(ls, mkdir) through the Manager interface? I can't seem to access the manual at digium.com. I keep getting 'Forbidden' error. Looks like they are upgrading or something. CF

I have an interesting problem. I am attempting to install a payphone utilizing a Cisco ATA-188. The payphone actually works, but there are some timing issues. What happens is you pick up the payphone and the ATA grabs a line and goes offhook. While you monkey with putting money in and dialing the number, you are eating up the time before you get the offhook reorder tones (or howler tones

Hi list, Does anyone know the legalities of connecting an Asterisk box to the PSTN in Manila or where I can find this info out? I know it is illegal in some countries. thanks -Matt