search for: jsci

I know a few ppl have those CAC Adit 600's with t400 I can't seem to get my second span up on the T400 connected to the second spand on the adit (A:2) A:1 seems ok Can someone post they zaptel.conf span defintions And maybe a "print config" from the adit 600 cli I think my issue is timing srcs the coding, framing. bld out are all matched thx -------------- next part

Can anyone recommend some phone sets that are adsi compliant and work well with asterisk?

digium support answered, for anyone's future reference, they should be left unlocked (aka set to all zero's). Thanks. -----Original Message----- From: Joe Antkowiak <joe@jsci.net> Date: To: asterisk-dev@lists.digium.com <asterisk-dev@lists.digium.com>; asterisk-users@lists.digium.com <asterisk-users@lists.digium.com> Subject: [Asterisk-Dev] adsi phone codes? Does anyone know off hand what codes should be used with the adsi phones in order to work with d...

...3) Suspend Lock Flag 4) Delete lock bcus in my experience only the mfg aastra can do this >digium support answered, for anyone's future reference, they should be left unlocked (aka set to all zero's). > >Thanks. > >-----Original Message----- >From: Joe Antkowiak <joe@jsci.net> >Date: >To: asterisk-dev@lists.digium.com <asterisk-dev@lists.digium.com>; asterisk-users@lists.digium.com <asterisk-users@lists.digium.com> >Subject: [Asterisk-Dev] adsi phone codes? > >Does anyone know off hand what codes should be used with the adsi phones in o...

FYI, asterisk DOES now support g723, but you have to pay for it: http://store.yahoo.com/asteriskpbx/asteriskg729.html -----Original Message----- From: Dan Fernandez <danfernandez00@hotmail.com> Date: Mon, 5 May 2003 17:33:05 -0300 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Has anyone gotten SIP_CODEC= to work? Basically, since I?d like to use g723 for sip

So I have an X100P, and a TDM10B both working (at least I think they are). The drivers have been loaded and ztcfg -vv shows no errors in the configuration of two channels. When I run asterisk -vvvc and pick up my phone (plugged into TDM10B), I don't gear a dialtone. in phone.conf, I have [interfaces] mode=dialtone format=slinear ... Shouldn't that produce a dialtone when I pick up the

Hi all; I have been searching for an answer to a question that a customer asked me and I have only found a few older answers. So, wanting to find out if anyone has any experience with this issue and can help provide me with some advice. I have a customer which is strongly interested in using Asterisk as a PBX. One of the core requirements, however, is that the system MUST be able to

I've tried the precompiled version of gastman, but it does'n work properly under windows, so I would like to try to compile it under windows, so maybe it begins to work. I've tried to compile gastman with mingw, but the db3.1 libraries request the cygwin include files and libraries. Then I've copied the cygwin's include files and libraries to mingw's directory, but the make

Hi, I haven't been able to find it in the docs, but I believe I remember some people talking about a web interface to the asterisk voicemail application. Does this cool beast exist? And if so, where might I find some docs on it? Also, I noticed there was some talk about a gnome interface to asterisk, does that exist as well? If so, same question... Thanks in advance, and thank you to

Just an FYI, I noticed some people not being able to use the CLI of their adit 600's because of a set password. You can erase the configuration of the adit 600 by setting the RST dip switch to on, powercycling the box, waiting until it finishes coming up, setting it back to off, and powercycling again. -Joe

Can anyone tell me what I need to do to tell the voicemail and voicemail2 applications to download to a different slot? The only problem I have remaining is this one... I dial the voicemailmain extension from a PT480, and the display tells me "services is full, download refused", but asterisk PBX only occupies the first slot (I changed this)... Any help would be greatly appreciated.

Hi, I have a server coming, which consists of an ASUS SK8N motherboard, an AMD Opteron 1.4g cpu, and 512m of dual channel memory. If you're not familiar with this, the amd opteron is a 64bit cpu that does hardware 32bit emulation. It will be running suse enterprise server for AMD64. I will not be running * on this box, but in the future it may be a good performer. So, before it goes into

Here's our config: cisco 7960's running 6.3 sip code latest cvs of * t100p zaptel card adit 600 channel bank 7 pots lines and 2 fax machines on the adit 600 dialing out from the cisco phones gets sent out via the zap channels, but I'm having some serious echo problems. I currently have the adit set to +3 rxgain and -6 txgain, with my zapata.conf containing: echocancel=128

I have this in extensions.conf: exten => 1,1,Dial,Zap/g9/4439568899/|24 exten => 1,2,VoiceMail(u8004) and this happens: -- Playing 'js-joe-trvm' -- Executing Dial("Zap/2-1", "Zap/g9/4439568899/|24") in new stack -- Called g9/4439568899/ -- Zap/1-1 is ringing -- Zap/1-1 is ringing -- Nobody picked up in 24000 ms -- Hungup

I noticed a few other messages posted about this problem, but I couldn't find an answer... I'm having a problem with SIP echo when calls are received into asterisk via an x100p and bridged with a sip extension (back to the pstn with iconnecthere). the person calling in to asterisk has no echo problems, but the recipient of the pstn call, everything they say, they hear back about 1 second

Ok, so I figured out my problem with my pt480s. But, now I have a few more. 1. When I dial into the voicemailmain or voicemailmain2 application, the phone and * start talking adsi, but then the phone tells me "programming download canceled, services is full.", but my services list isn't full, only "Asterisk PBX" occupies slot 2, slots 1, 3 and 4 are available. Any ideas?

Hi, I am using a channel bank and zaptel hardware. I have a credit card machine on one of the channels that appears to be dialing "too soon" for asterisk, every complete number recognized by asterisk is missing the first 1-4 numbers. This is a serious problem for me, anyone have any ideas on whats going on? The pstn picks up on the dtmf tones just fine. I was able to get it to

Hi, I have a 1 port T1 card in an asus p4p800-vm board with a 2.0g Celeron, and 512m of 266mhz ram (256 on each channel). This board has video, Ethernet, and serial ata all on-board, I got it because of that, there wouldn't be anything else on the pci bus that would mess with the zaptel card(s). So, here's my problem. I'm running redhat 9 with kernel 2.4.20-20.9. I can load the

I bought an Aastra PT480 from digium, but I wanted to see if I could get some more help with this before Monday. Any help would be appreciated. I have the phone connected to the TDM400P card, and I also have the T100P and the X100P in the same box. My problem is, it appears as if the phone and asterisk can't understand each other. The port the phone is connected to always remains

Hi, One of my pbx's seems to be having some new issues. crackling interference on the zap channels running through the channel bank, and I noticed that these happen when I hit an "irq miss" in zttool: Current Alarms: No alarms. Sync Source: Digium Wildcard T100P T1/PRI C IRQ Misses: 82695 Bipolar Viol: 0 Tx/Rx Levels: 0/8 3 Total/Conf/Act: