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Good day, The unusual thing about this problem is that it doesn't occur just during a CAPI call, or just during an IAX/SIP call. Only during IAX/CAPI I'm having some trouble with the CAPI interface and it only occurs when a call comes in on an IAX channel and goes out the CAPI interface. The capi debug in the asterisk console is below as well as the relevent parts of .conf files from

Hi Gurus, We have configured asterisk to trunk with avaya with ooh323 channel driver. The sip phone registered on asterisk can dial the extensions registered on avaya via this trunk , and vice versa works too. Even we can make the avaya branch to dial asterisk?s extension and then this extension dial back to another avaya?s extension. But if we dial the external DID number via this trunk from

...heir initrd is pretty standard. If I'm missing something obvious I appologise but I'm honestly stuck here. If needed I'll provide any files you may want to look at. Ciao A.J. -- A.J Venter Lead Developer, DireqLearn 082 726 5103 http://www.direqlearn.org http://www.direqlearn.net/olce http://silentcoder.co.za

Hello, I think I have the same problem as Martin Bene mentioned in http://lists.digium.com/pipermail/asterisk-users/2004-January/034521.html Since I found no further information about this I'd like to ask wether you know what the reason for this problem is and how one can get around this. * is registered to the innovaphone gatekeeper. Trunk connection is done with an AVM-B1 and chan_capi.

Hi everybody ! I'll try to explain first what I would like to do with the functionalities of tap and bridge interfaces, qemu, some isos(I used slax(slackware live cd)) and the forwarding mechanisms of the Linux kernel. I'd like to simulate such a network: Internet | (eth0:@public address) * host * | br0 (@192.168.0.254)

Yo! I changed callprogress to no, and in wcfxo.c source around line 334 i changed the value 32000 and -32000 to 10000 and -10000 because it had something to do with the DC voltage when it was ringing. I found reference here (http://www.voipuser.org/forum_topic_1791.html) with an interesting diagram of wiring that was incorrect for sending voltage to a phone or something like that. So put it

We are proud to a announce another alpha release of Samba 4. What's new in Samba 4 alpha7 ============================ Samba 4 is the ambitious next version of the Samba suite that is being developed in parallel to the stable 3.0 series. The main emphasis in this branch is support for the Active Directory logon protocols used by Windows 2000 and above. Samba4 alpha7 follows on from the

Hi all, I have VOIP network built with MGCP endpoints.The manufacturer of endpoints is ASKEY. I downloaded latest Asterisk software and found it very useful for me. I configured it and it seems taht everything works OK when I am testing it with one or two endpoints. After that I tried to move Asterisk to working network and replace existing call manager. It starts working and calls are

[mod: Forwarded by Jeff Uphoff. I tried to mangle the headers that it appears as the original post: with an invalid return address. -- REW] A few months back, a very alpha version of bliss got posted. That shouldn''t have happened, but, it was pretty much ignored so I didn''t worry about it. But now it seems there''s a bit of a fuss about this. I''ll post the

<-----------------------------------------------------------------------> Changes to user passwords are captured by a special DLL, which traps and then stores the password changes in encrypted form in a private area. On each synchronization schedule, the synchronization service first examines the SAM file for changes, and then checks this private area for passwords to be synchronized. Once

Morning, I've installed asterisk on FC4 and had a few problems with zaptel stuff. Having installed it on SuSE I was able to check a few things that were different. When installed with the RPMS, the udev stuff gets put into the normal 50-udev file under /etc/udev. On SuSE, it worked when the necessary zaptel section was in a separate file name 55-zaptel.udev (the name is arbitrary,

A very good day to you all, We can't get the phones to pick up on an incoming call on analog trunks. We're using the digium products in the box, with snom phones internally. This is the output from the asterisk console: linux*CLI> zap show channels Chan Extension Context Language MusicOnHold pseudo pstn-incoming en default 1 pstn-incoming