similar to: Fedora Core 2 firewall rules - NO NAT!

How can i control the codec for the calls. For example I have 3 SIP phones registered to asterisk The firs two are in the local area network (behind nat)- I want to use g711 between them and to connect directly (canreinvite=yes) and the third is in internet - want all calls to it and from it to use g729 and media to go through asterisk. So if Phone 1 calls Phone 2 the codec to be g711, but when

HI ALL: I am newbie to IAX, my iax.conf is as follows: [general] port=5036 ..... but I donot why it doesnot listen on UDP PROT 5035, instead it listens on 4569 Asterisk CLI debug says: [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2)) == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': Found Nov 30 11:52:12 WARNING[1076220544]:

Hi. I've got a few questions on moving from X100P's to a single T100P. I've installed the cards, they're recognized by the OS<FC 1 box> as well as asterisk. When placing a call I am getting the following: Nov 24 14:17:15 VERBOSE[-1101243472]: Asterisk Ready. -- Accepting AUTHENTICATED call from 192.168.220.10, requested format = 256, actual format = 256 Nov 24

Hello all, I found a LSI161SCREV2 Dialogic board in one of my drawers, and i was wondering if by any luck, i could make some magic happen with asterisk ... If asterisk does not support it, is there any PSTN to H323 or PSTN to SIP gateway that support this dialogic card and that can be connected to an Asterisk Box? Digium, I PROMISE that I will buy my cardf rom you once my tests are conclusive

On how to even search for this "feature" as I have no idea on what it can be. I've got a meridian linked to * (by EuroISDN) which is linked to a ISDN30. I can make calls from the meridian, and receive calls into the meridian. Great stuff. However, if someone dials an invalid number, then instead of hearing a "three tone", the line just drops and goes dead. The console

Hi List, I've got * randomly hanging up on inbound or outbound calls on zap channels. I use a Digitnetworks X100P clone card. Any idea of what might be happening? Cheers, Jean-Michel.

We are in the process of testing * for company wide deployment. We are using Polycom 300 phones, the only problem that I am running into is when I call an 800 number that has an IVR I get disconnected after about 60 seconds. Here are the logs from asterisk. I am not sure if this is a problem with asterisk timing out or if it is the phone. To me this looks like asterisk is timing out.

I'm setting up a Samba 3 server on Ubuntu 10. The server will have five local shares, which it will provide to the local network (let's call that network 1.2.3.0/24). The samba server is a slave to the local Windows AD domain -- that is, the samba server does not do its own authentication but just passes along such requests to one of several local domain controllers that actually deal

r-devel from this morning says hothorn@www:~ > R R : Copyright 2001, The R Development Core Team Version 1.4.0 Under development (unstable) (2001-12-09) R is free software and comes with ABSOLUTELY NO WARRANTY. You are welcome to redistribute it under certain conditions. Type icense()' or icence()' for distribution details. R is a collaborative project with many contributors. Type

I admit that this might be some very basic question... How do I obtain Asterisk 1.0.3 from CVS? Does '-r v1-0' get me 1.0 or 1.0.3? Thanks, Adi

When I make a call from a SIP phone to a speaking extension on *, such as one that speaks digits or similar, when I monitor * in very verbose mode I can see it running through the routine associated with the extension, but I am getting no RTP data stream back to the phone. Does the machine housing * need a sound card? Does it need OSS or ALSA modules installed? What actually generates the RTP

I've configured an analog/single line off my mitel sx-2000 to a x100p card in my *, however when the remote caller drops the line I get a dialtone back from my pbx and the x100p doesn't detect the end of the call. Any way I can tell * to drop? I've tried call progress but it doesn't seem to work...

Hello, I've 3 * boxes connected with IAX2. Everything was working correctly and since few days, after upgrading them to Asterisk 1.02, i started seeing IAX2 notify messages excahnges on port number 1024 and 1040 in addition to the specific 4569 iax2 port, is that normal ? All call between * boxes are rejected with "NO SUCH EXTENSION or CONTEXT" message, I've checked my

Hi, I compiled * and chan_alsa.so is loaded. But I can't hear any busy signal messages when calls cannot connect. Do I need to record my own message, or does * use some default ones? May I ask where can I find them? Regards, Norman Zhang

Hey gang, Getting ready to run some test bills for customers. Most SIP phones have both an extension and a DID. If a person calls a DID asterisk redirects the call to the right extension: exten => 8005551212,1,Goto(companyA-internal,3022,1) The problem is, that if someone calls 8005551212, the CDR shows the DST number as 3022. Is there a way around this? I understand that 3022 is the

Hello I try to set up voicemails for extension. When VoicemailMain gets called, it prompts for mailbox and password. It seems not able to read from the phone. So the authentication always fails. I desparately need help to understand what is wrong. Here is a part of my extensions.conf: exten => _8500, 1, Wait(2) exten => _8500, 2, VoicemailMain(${CALLERIDNUM}) exten => _8500, 3, Hangup

HI; I have an Asterisk with 10 "SIP" ip-phones, our pbx features are now: Voicemail and Call Transfer. How can I serve both "Call Waiting / 3 way calling" for our SIP Phones.?/ Appreciate Any Help Mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL:

OK. I just downloaded asterisk-1.0.3.tar.gz and did a 'cvs co -r v1-0 asterisk' into 2 seperate directories. I then did 'diff -ur asterisk-cvs/ asterisk-1.0.3/' and there were source code line differences between the two. Some code that was in asterisk-cvs wasn't in asterisk-1.0.3 and vice versa. Which of those is "the most recent"? If someone wants to use cvs to

Okay, now I am really confused. I have two PRI's coming in from two different Carriers (QWEST and ELI), both of them are supposed to be setup to pass name and number on incoming calls. Problem that I am having is that I am not receiving inbound caller id name on either PRI, the only thing that both carriers have in common is that I am terminating into a DMS switch at the carrier.

If i leave my asterisk server running for a long time then try to dial outbound on the zaptel channel i get this high pitch static noise and won't dial out. This behavior is happening over two different servers i am using. Rebooting asterisk does not sovle the problem. I rmmod the zaptel driver then reload and that solved the problem. But i cannot continue to do that. Also sip to sip