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Hello All, I've mostly solved my DID problem from a few days ago. Apparenly the lines weren't configured properly. Now heres the next question. 12 E&M wink lines from telco. I have them all plugging into an Adtran 750 with FXS cards. The Adtran ports are configured DPO. How do I signal this from Zaptel. I have them setup E&M in zaptel.conf and EM_W in zapata.conf. They

Has anyone on the list been able to locate and try out the 1.1.0 firmware? It was released in November, but I have yet to get my hands on it. The Polycom site has way more docs online, but the link to the firmware only brings up the release notes. -sb

Caller*ID used to work as some point, but I can't seem to get it going these days. The card is a x101p. I've tried going up and down the rxgain scale. Can the txgain effect it at all? When I plug in a phone into the line with a splitter it can decode caller id with no problems. Reading through the mailing list archives hasn't given me any move clues. Any ideas?

I just had the same exact problem this morning. The only thing I've done in the last couple of days is update update zaptel. I rolled back my zaptel to 2/11/04 from 3/8/04. And kept my libpri from 3/8/04. I never had this error before updated. I had other issues, but not this one. -sb -----Original Message----- From: asterisk-users-admin@lists.digium.com

Hello All, Quick question here on equipment selection. Is there any benefit to using a bunch of Adtran 850's with RCUs over 850s with BCU L1 to interface with the T400P. I'm looking at building a PBX with 7 channel banks. 6 x 24-port FXS and 1 x 24-port FXO. Would the RCU give me any advantage/functionality over the BCUL1? Thanks, Scott

Hello all, I got the following error compiling h323 support in the latest cvs. Below the error is a diff to the file that I got to make it work. I took an example out of sip as far as the syntax for ast_rtp_new. Not sure if it is correct or not, but it seems to work. Please correct me if I am wrong in the additional 2 arguements. Regards, Scott cc -g -pg -c -o chan_h323.o -march=i686

Hello All, Here is a patch that fixes the problem when forwarding messages with vmail.cgi. Bug submitted with patch on bugs.digium.com. -sb --- /usr/src/asterisk/vmail.cgi.orig 2003-12-17 14:21:47.000000000 -0500 +++ /usr/src/asterisk/vmail.cgi 2003-12-17 15:07:36.000000000 -0500 @@ -672,7 +672,7 @@ sub message_copy() { - my ($mbox, $oldfolder, $old, $newmbox, $new) = @_; + my ($mbox,

Hello All, I am experiencing some intermittent problems with calls coming inbound on my DID trunk. I have 12 DIDs that come into an Adtran 750. From there T-1 to a port on T400P. The problem is that some calls that come in don't seem to bridge properly. Heres what happens. Call comes in on Trunk. Call Routed to correct Zap Channel. Phone Rings. Person Answers phone, but hears nothing but

Hello All, I'm having a weird problem when connecting up to a TA 750 from adtran. The problem I'm seeing is that the third wire on my 66 block is behaving as the tip (or ring) for every extension. Is this indicative of a bad BCU? The only extension that works properly is extension Zap 2. Every other extension is crossed with Zap 2. Very weird. Anyone see this before? Did I get a

I've been trying to get a Win 2000 RAS server working with my asterisk PBX for quite some time, to no avail. I've googled, I've tried loads of configurations, I've rewired phone lines, and still I am not winning the battle. Here's my config. PRI->T400P->Asterisk->T400P->Adtran 750(L36 Firmware)->RAS Server. I have 4 Zap channels signalled FXO_KS to the 750

I take it you are running RedHat 8 (or 9) since this is the most up-to-date kernel. Did you install the kernel-sources and kernel-util rpms as well? You'll need these in order to compile and install zaptel. -sb -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of T. Chan Sent: Friday, January 30, 2004 4:01 PM

I've attached two SIP debugs in reference to bug #116. They are from today's CVS build. 1. pickup.txt is a call from SIP(1) to SIP(2) with SIP(3) picking up the call. After which, SIP(2) rings for about 30 seconds then stops. 2. hangup.txt is a call from SIP(1) to SIP(2) with SIP(1) hanging up before the call is answered. SIP(1&3) are Cisco 7960's and SIP(2) is a Polycom