similar to: Equipment Selection

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'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

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

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

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 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 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

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

Does anyone know where I can get R code for plotting the Brown , Durbin and Evans cumsum procedure ( 1975 ) ? I wrote my own code but I am a little worried that my confiodence bands may not be correct ( I find the formula in the original paper confusing and S+Finmetrics has a formula but that formula implies that there should be 4 lines as far as I can tell ) so I would like to see someone

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 have a Linux 2.6.16 machine with two gigabit ethernet adapters that I would like to use for testing Ethernet equipment---right now just an Ethernet cable, but later switches, bridges, etc. I tried assigning the two adapters IPs of 192.168.50.1 and 192.168.51.1 and then setup the routing table to direct 192.168.51.0/24 traffic to the first adapter, and 192.168.50.0/24 traffic to the second,

On Sun, Jan 18, 2015 at 1:07 PM, Charles Lepple <clepple at gmail.com> wrote: > On Fri, Jan 2, 2015 at 7:31 PM, hyouko at gmail.com <hyouko at gmail.com> wrote: >> Before I merge it (well.. provided that it works..), I'd like to know >> a couple of things more: >> 1. does the load.off command ('S00R0000') work? > > load.off works. > >> 2.

Hi Anyone in australia know of any good suppliers of voip equipment. I cant seem to find anything except the cisco stuff. Thanks heaps

Anyone interested in a lot of gently used IP500's with SIP Image? Also, anyone in need of large quantities of SPA-2000's at an extreme discount? If so, please contact me off list. Thanks, Garrett Smith gsmith@voipsupply.com B2 Technologies/ VoIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 (716) 250-3408 Direct (716) 630-1548 Fax (716) 903-9495 Cell AOL IM:

I have grouped two capi controllers using * as the outgoing msn. when the first two channels are busy normally and I try to use a third channel the channels from the second controller are used. But when a channel is occupied by the capisuite fax and we need a third channel asterisk responds with "every one is busy congested at this time". I think the same behavious would have happened

Hey All, I am looking for some advice on setting up a test lab for Asterisk systems that I plan to install at client sites. Most clients will be using PRI. Does anyone have any suggestions on equipping a lab to be able to test Asterisk servers with T1 cards and SIP phones? I have seen some test equipment around the internet, but I'm not sure what the best is. I have also thought about

Hello Everyone, I am once again wondering about EICON. I have had no success with the Diva Pro or Diva Pro PCI, so my question is, does anyone use an Eicon Server BRI card on Asterisk? Or would I be better off trying to get a split PRI? Regards, Greg