similar to: IAX needs a zaptel device?

Good day all I've installed asterisk with sip on my LAN,no special cards,if done sip.conf and extensions.conf and all work 100,I'm using x-lite as a client. I'm trying to do conferencing.What I did was to has out the meetme.conf looks like [rooms] conf => 9876 conf => 2345,9938 and extension.conf exten => 9876,1,MeetMe,9876 When I go onto x-lite and type 9876 it gives me

Hi all. I've made a patch for chan_oss.c to enable callgroups and pickupgroups in it (since wasn't enabled). I needed it for a special use of the console (pickup calls arriving to the console from another phone) btw, If someone is interested, I can submit a patch to the bugtracker. I won't do it until that's usefult for someone... since is a very special features that probably no

Hi, All Is there a provision for "AbsoluteTimeout" application to notify called and calling party of the reason why the call suddenly ended? This way, the parties will be much better informed, hence they will/should not think that their VOIP/telco provider(s) are providing bad service. Ta SJ

So, I have submitted my configurations as public samples, and I should have expected this situation to arise. I changed all the relevant "private" configuration data in my samples to obfuscate or alter IP addresses, passwords, etc. However, I left my email address in voicemail.conf... Let me tell you, it took THREE messages sent by a distinctly British-sounding gentleman leaving

Hi, I would like to test chan_sip with a bigger jitter buffer. Does anybody know where in the code this is defined? I looked through it but could not find where. If anybody else can find it please let me know. Regards, Andres

Hi I have the last firmware for zplex, if you like i send it to you, about the second question 24s means 24 extensions so you can configurate as you wish as fxo or fxs. Att Yelson Vivas

I have a couple of X100P's in my system and while on calls they just randomly hang up for no reason. I have tried messing with the busydetect and callprogress setting them to yes and no same and still random hangups. Is there another setting I should be looking at? My zap config looks like. context = inbound-work include => extensions signalling

Hi, I'm trying to intall zaprtc on my machine (I don't have UHCI_USB nor Zaptel) and I have a strange behaviour during loading. When I do >make load I have : ./zaprtc.o: init_module: Input/ouput error With a quick dmesg got: rtc: I/O port 112 in not free Is someone have any idea to solve this ? I saw on the README file that rtc support should not be enabled in the kernel, but I

I am trying to get the handytone 286 to make a very simple call to * and having problems. It registers with * just fine, but when I place a call (to echo test, for example), the RTP stream seems to have problems opening. Here is there error I get in *: WARNING[98311]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call 20d1c411-e210-5f3d-3f88-19035c8fcb26@192.168.2.6 for

Hi When I call a phone with CAPI if the phone available I hear ringing ok but if the phone is busy I don't hear anything at all. Also, when I call a mobile phone and it is turned off I don't hear the operator voice answer me telling me that the request phone is turned off or unavailable. Any ideas? m

Hi, I've installing a AVM Fritz Card in my ASterisk Box I've configured everything and its running perfectly. The problem is that everybody is allow to call through it. Explaination: All users registered in Asterisk can make a call towards the ISDN network But, everybody from the Internet, knowing the extension of CAPI in the dialplan, can call through my Asterisk to any phone

We find that mpg123 0.59r works best. mpg123 0.59s-mh4 = the devil. What versions does everyone use without problems. 0.59r is PERFECT bkw -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040506/937ae19c/attachment.htm

I wish 'A(x)' was available with AgentCallBackLogin!! :( -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Matteo Brancaleoni Sent: Tuesday, February 03, 2004 6:48 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Playing announcement to called user prior toConfirmation show application

I'd like to play a sound to a user I dial (via SIP) once they answer play the sound and then allow me to talk to them. The new Cisco 7960 SIP code allows to set lines to autoanswer via the speaker phone, I'd like to play a "tone" after it rings through and then talk... Any thoughts on how to do this?

Lots of people seem to want this, so I've stuck it up here: - http://almaw.com/ethereal-iax2-plugin-0.1.zip Note that it currently only does IAX-2. I might expand it to cope with IAX-1 at a later date, but no promises. It's fairly basic - unzip the file and follow the README instructions. Regards, Alastair

Hi I'm using a Teles ISDN passive card configured in modem.conf. when i make call from my sip client (xtex x-pro) to the external world i have more than 1 second of delay and echo very. There is some tuning to do? The performance is better with an active ISDN card or CAPI compatible driver? thanks mark balester

Hello all, As I'm sure is pretty common, I have some extensions that dial mobile numbers after a local timeout. I would like to prompt the caller to record their name after the local timeout and have the recipient be able to hear the name prior to accepting the call. Recording the message is easy enough, so I thought about doing something like dumping them into MeetMe after they record

Hi all. Is it at all possible to have a GS B101 NATed with firmware 1.0.4.40? I've tried both STUN and not STUN. The odds seems best with stun because the phone registers with right ip adress. When the connection is made * sends rtp packets to the right destination AND port, but the phone doesn't accept the packets..... Should I burn my D-LINK 604 or upgrade the GS? /t

Hi my question is: which is the best distribution to work with asterisk? thanks mark

We're proud to announce Astricon 2004 - the first Asterisk user's and developer's conference! * Where? Atlanta, USA * When? September 22-24, 2004 The conference is arranged in partnership with Digium.inc and the keynote speaker is Mark Spencer, lead developer of Asterisk - the Open Source PBX. Among the speakers already signed on are Ed Guy of Pulver.com, John Todd, Jeremy McNamara