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...sion 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 bum BCU? Also, when performing a ring test from the admin port of the 750, the same behavior is present. Any ideas on this one? Thanks in advance. Scott Bisker

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

Hi all, I installed the estara softphone and had no problem registering it with asterisk. I could make calls to other hardware SIP phones (Cisco 7960) from the softphone, but I couldn't call the softphone from the Cisco 7960s. The asterisk console gave me an error message saying "unable to create channel" to my softphone. What could be the problem? I searched the archive with no

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

Does anyone have any recommendations for overhead paging systems for use with Asterisk? Thanks, Randy Johnson -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031113/82ae09ec/attachment.htm

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

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

I have several phones running Cisco's 6.0 SIP software release at this time. Two of the phones have not shown any abnormal behaviors, but one of them has an unsettling propensity to lock up after several hours, where the softkey labels disappear and the phone stops registering, requiring the standard *-6-settings reboot sequence. Otherwise, the phone seems to work OK except for a slight

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

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

Hi, I have just upgraded my Cisco 7960 phone to SIP firmware today and I have to say it's working great with Asterisk. At work (which uses Cisco Call Manager), when a voicemail is recieved the read light remains lit until the voicemail is retrieved. Is there any way to achieve the same effect with Asterisk ? Thanks, Paul. -------------- next part -------------- An HTML attachment was

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

My PRI is being reset at least once a day with the following errors in the logs. zaptel, zapata, libpri, and asterisk are from CVS this morning.. This has been happening for weeks on all versions (including -stable). the T100P card appears to NOT be sharing an IRQ. xenon# cat /proc/interupts CPU0 0: 1203977 XT-PIC timer 1: 3 XT-PIC keyboard 2:

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,

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