similar to: TE110p and pri_cpe signalling not recognized

I have had a bit of a problem getting consistency in filenames between NTFS and HPUX-Samba shares. Every time files are moved or copied to the samba-shares they convert all cases to lower. I have put the following in to smb.conf case sensitivity = yes preserve case = yes short preserve case = yes which I put in the global section and in to several of the shares sections also, but the

Hi guys, I'm using SIP phones with Asterisk 1.2 and going fine, most of the time. However, when the duration of a call is longer than 20minutes I often stop hearing the other party, but that one keeps most of the time hearing me. Does any of you know of this or similar problems? Thing is that I'm connected to the Asterisk over VPN tunnel, but how sensitive is the SIP protocol to a short

Anyone have any idea what this is talking about. Here is my zapata.conf [channels] switchtype=5ess signalling=pri_cpe echocancel=yes echocancelwhenbridged=yes echotraining=400 callerid=asreceived group=1 context=default musiconhold=default faxdetect=incoming channel => 1-23 Here is my zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 # set this to 1-15,17-31 for E1 dchan=24 # set this to 16 for

hi after some testing with A@H, i've decided to install my asterisk server on a slackware (because it's my favourite distro and it is still suggested here http://www.voip-info.org/wiki-Asterisk+Linux+Slackware) , so, i've installed the last 10.2 release, and i've recompiled the 2.6.15.4 kernel. then i've downloaded asterisk1.2.4 and zaptel1.2.4 i load module zaptel and

Hello, Avaya has a nice feature that allows you to a) ring both a cellphone and a desktop phone at the same time b) transfer calls (and access other PBX features) from the cellphone that recieved the call, as long as the call is bridged through the PBX c) while talking on the cellphone, pick up the handset on your desktop phone and the call is automatically moved there, hanging up the cellphone

Hello. I have just installed Asterisk on my HP DL140 Fedora Core 1 Server. The server has two interface cards active. I have been unable to fix this error although all DNS lookup works fine. my /etc/host.conf contains order bind,hosts /etc/hosts contains # Do not remove the following line, or various programs # that require network functionality will fail. 127.0.0.1

Good day all We got a Wildcard TE110P I installed linux,zaptel,libpti and asterisk I coped over my zaptel.conf and zapata.conf from a previous E100P config But when I try to start asterisk it gives error not bying able to load zap channles: == Parsing '/etc/asterisk/zapata.conf': Found Jan 10 08:17:18 WARNING[-1084595552]: chan_zap.c:9308 setup_zap: Ignoring switchtype Jan 10 08:17:18

Hi, I wonder if anyone is using Digium's TE110P card on an E1 connection. I have been try to, but so far it wasn't much of a success. It only works more or less in EuroISDN as PRI CPE. And even that config gives me some trouble with channel negotiation. My current config: *zaptel.conf:* span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone=be defaultzone=be *zapata.conf:* [trunkgroups]

When I reload the asterisk I get the following warnings: Oct 13 08:29:17 WARNING[10170]: chan_zap.c:10874 setup_zap: Ignoring switchtype Oct 13 08:29:17 WARNING[10170]: chan_zap.c:10874 setup_zap: Ignoring signalling Oct 13 08:29:17 WARNING[10170]: chan_zap.c:10874 setup_zap: Ignoring rxwink Everything works fine as far as I know, I can dial and complete calls. So why am I getting this

Full_Name: Olafur Arnar Ingolfsson Version: 1.9.1 OS: Win XP Submission from: (NULL) (213.236.225.194) Copying lattice graphics only works with copy as bitmap. Printing the graphic window doesn't work either. I have the same problem with 1.9.0 and 1.9.1, just updatet packages If I have done som (obvious) mistake, I do apologize. x.val <- rnorm(100,50,3) library(grid);library(lattice)

Hi there, I'm looking into developing an in-house switchboard application. Does anyone here know of a way to control a hard-phone from such an application. For example, the attendant forwards a call with another one in queue. Once the first call has been forwarded (by keyboard shortcuts or dragging-n-dropping) - she presses a button (on the computer) to answer the waiting call. Now, if the

I reloaded my asterisk and found some red lines flushing by. When I stopped it I see: WARNING[21481]: cahn_zap.c:9773 setup_zap: Ignoring signalling WARNING[21481]: cahn_zap.c:9773 setup_zap: Ignoring echocancelwhenbridge WARNING[21481]: cahn_zap.c:9773 setup_zap: Ignoring echotraining Reconfigure channel 1, FXO Kewlstart signalling Reconfigure channel 2, FXO Kewlstart signalling

Hi, I've just got a TE110P card and installed at Asterisk. I configured zapata.conf, according to www.digium.com/index.php?menu=configuration, but the following error is happening: ... ... ... [chan_phone.so] => (Linux Telephony API Support) == Parsing '/etc/asterisk/phone.conf': Found == Registered channel type 'Phone' (Standard Linux Telephony API Driver)

Good eye! Its getting late maybe I should just stop now Thank again! -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kevin Bockman Sent: Monday, March 13, 2006 8:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Spam? Re: [Asterisk-Users] Unknown signalling method

We have integrated an Asterisk (TE110P) and a Mitel SX200. We usually get over 500 frame errors and over a 500 slip errors per hour. When the errors reach 1000 per hour the Mitel will take it's T1 card offline. At that point no calls can be routed from the Asterisk server to the Mitel and the TE110P reports a Yellow alarm. What can be causing all these Frame and Slip errors? We have been

Hi list, Ive got a server with 6 ports on it (4+2 port card) we have a DS3 delivering all voice DS1's to us. Carrier has a trunkgroup for the first 8 span (we only have the first 6 plugged in right now). Everything works fine until we fail the primary D channel (D's are on 24,48) the secondary then picks up and outbound calls do not work, if we reboot Asterisk the D on 48 comes up and it

Hi all i need to put on one graph 2 functions who's x axis is the same and y not. I mean on horizontal the time, and on vertical left: pressure, on vertical right: rpm of a motor, is R able to do that? i've found this that i could adapt maybe (i don't need time series really?) :/ : (http://tolstoy.newcastle.edu.au/R/help/04/03/1456.html) ## ## Description: A simple function which

Is there any dump utility that exists for vorbis streams? What I am intersted in is something that will do a break down like: How many bits are used for encoding _each_ codebooks, how many bits are used for the residue, how much is used for the lpc coefficints. A selective dump of the codebooks themselves would be nice too of course. I'm wondering if anyone has such a little dump utility.

Hi Since Im using the mISDN drivers and no zaptel stuff, I had to install ztdummy to get MeetMe to work. Well, that was the plan. Now, after getting the latest zaptel version over CVS (Im using Kernel 2.6), uncommenting all the modules except ztdummy in zaptel.sysconfig file and compiling this by "make", "make install" and "make linux26", I rebooted and

I am running Asterisk 1.6.2.17.2 with a Openvox A400 card with 2FXO/2FXS modules. I'm trying to set-up things to route analog fax calls from a FXO port to an analog fax machine on a FXS port on the same card. Outgoing faxes work just fine. But incoming faces are routed to the right DAHDI extension, but the call dropped right as the fax machine rings for the first time. The fax machine