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-------- Forwarded Message -------- From: Michael Baird <mike at tc3net.com> To: Timo Sirainen <tss at iki.fi> Subject: Re: [Dovecot] strip domains Date: Wed, 12 Apr 2006 10:42:43 -0400 On Wed, 2006-04-12 at 17:17 +0300, Timo Sirainen wrote: > On Wed, 2006-04-12 at 09:48 -0400, Michael Baird wrote: > > I don't > > know why there is so much...

I've setup cdr_mysql and am using AGI to authenticate users based on the called-from # (callerid), use the AGI perl module. Looking at the info stored in the caller detail, I see a field called "accountcode", is it possible for me to set this field in AGI? I'd like to tie it to a username, that I pull during my SQL authentication, so I can search the cdr table based on a

I have two identical AIX v4.2 machines. One of them was vendor installed, and I installed the OS on the other one. I am successfully running Samba-1.9.18p4 on the vendor installed OS, but I cannot seem to get any Win95 clients to browse on my other machine (Explorer hangs on the Win95 client). I have performed all of the tests in DIAGNOSIS.txt. All of the tests work except for number 7. I can

Hello, I've been testing dovecot, and it appears to be much faster then courier-imap. I am using the latest .99 version, trying to make the transition from courier to dovecot as smooth as possible. We really only use IMAP via squirrelmail, I think I've got that worked out fine, changing a few settings in squirrelmail as the FAQ instructs and migrating the courier subscriptions to

I saw some discussion on this from the middle of last year. Looking through the archives, there didn't seem to be a full solution using a standard dovecot configuration. Did this issue get resolved within dovecot? Will it strip domains from the login presented by the customer. I currently am using dovecot for imap with a webmail system (I strip the domain there before it hits dovecot), but

I've been using asterisk for a while, only for dialout from a SIP client over a PRI -> PSTN, this works great. Now I have a need to also dialin to asterisk over the PRI/TDM, I've been testing by creating an extension, and essentially playing back a recording on that extension. If I access the extension from a SIP client, it works great, very high quality, no chops are stumbles, If I

I've designed a voice menu, someone calls a certain extension, and I send them to another context via a goto, and play a background message. After playing this message can I provide them dialtone from asterisk again, in order to dial out? Regards MIKE

I'm going to be testing the new realtime stuff further in the next few days, and just wanted some clarification on a couple of things before I start on it. I believe I can store any config file in a external config such as mgcp.conf for example, by adding it to extconfig.conf with the below syntax. mgcp.conf => mysql,asterisk,mgcpchans Doing this will require a reload of asterisk to read

I'm testing my asterisk system and the realtime backend. My Asterisk build is rather aged, 03/18/2005 CVS. I have successfully moved Sip peers and Voicemail boxes to the realtime database backend and this works very well except for MWI. I don't seem to be able to get MWI to work when I store the voicemail information in a database backend, from a flat file it does work fine. I'm using

I'm trying to use RDNIS with asterisk, and I don't appear to be receiving any information (the value is blank). The upstream who provides the PRI says they are passing all the info through, I don't see this value coming across. I've tried it with a Verizon call forward, as well as a Nextel with the same results for both. I'm trying to use this for Voicemail. I'm using

Hello, I'm currently testing Asterisk over a T1 cross connect to a MaxTNT chassis that we have. It is working fine switching the calls through, but there is about a 10 second delay from the time Asterisk initiates the call until the TNT accepts it. It appears to be a ANI issue, I've changed several settings and formatting options on the T1 between the two, as well as turning on/off the

I've recently started PIC'ing some calls into a asterisk box across a feature group D trunk from Verizon. Everything seems to work ok, except for some reason Asterisk doesn't grab the full caller ID from Verizon. I can see that they do send it, but Asterisk drops the first 2 numbers. Looking at the debug log I see. I see that Verizon is sending the digits and the asterisk debug seems

Heh guys, I just received a T400 card, I've been using a T100 for a little while, and it works fine when using a raw channelized T1. I'm relocating my asterisk machine, and PRI's will only be available, haven't found any good config info for PRI's, can someone point me to PRI config info, or let me know what changes I need to make in order to bring them up, I imagine,