similar to: RE: Noise on ZAP channel

brett@websmyths.com wrote: > Also - an outside chance - make sure Tip and Ring > are correct. You could be getting ground loops - depends on the noise. > I am having noise and slip errors between my TE110P and a legacy PBX T1 card. Could this be the same symptom? The connection is made using a 15 pin serial on the T1 Card side to RJ48 on the TE110P side. I can't determine what the

Thanks. I was only loading OSS. I installed the alsa development libraries and then loaded alsa instead of oss and everything is working now. Thanks! -Jonathan -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of brett@websmyths.com Sent: Sunday, October 16, 2005 9:00 PM To: asterisk-users@lists.digium.com

Thanks Gene. Here is my localnet: localnet=172.16.64.0/255.255.240.0 Which matches our subnets network address and subnet mask. Are you recommending that I make it more restrictive? Thanks, Geoff > -----Original Message----- > From: Gene Willingham [mailto:gwillingham@comcast.net] > Sent: Tuesday, June 14, 2005 9:13 PM > To: asterisk-users@lists.digium.com > Cc:

I have a couple SIP phones on a PIII 1Ghz 256MB * server with a TDM01B connected to the PSTN. Calls between SIP phones are clear. Calls to the PSTN are quite noisy. The other person does not hear noise but I hear quite a bit. It is not an annoying sound but definitely much noisier than typical PSTN or even cell phone calls. I believe I have a TDM400P REV H card. I definitely don't

We use SetVar(TIMEOUT(digit)=8) In our dialplan to make sure that the user is done dialing before Asterisk executes the call. I just recently came across the piece I've copied below. It says for new incoming ZAP connections, the default digit timeout is 3 seconds and can only be configured in the source code. Is that true???? ============ How long will Asterisk wait?

I think I found the source of this. Been tracing it for a week. Look in sip.conf. It appears the definition of localnet has a bearing on how some sip devices handle invites and NAT. I had changed the localnet to 192.168.3.0, but did not change the netmask. localnet=192.168.3.0/255.255.0.0; All RFC 1918 addresses are local networks When I changed the netmask to 255.255.255.0 the problem

brett terpstra continues his obsession with markdown... > http://fuckyeahmarkdown.com/ there's also a g-rated version: > http://heckyesmarkdown.com/ brett has accomplished more in the last three months than this listserve accomplished in the last three years. and it looks like he's only just getting started... meanwhile, fletcher is about to send off his rocket, which

I have been getting these for awhile now in my log files. Apr 24 11:02:38 asterisk named[1072]: lame server resolving 'pbx_loopback.so' (in'so'?): 205.166.226.38#53 Apr 24 11:02:38 asterisk named[1072]: lame server resolving 'pbx_dundi.so' (in 'so'?): 205.166.226.38#53 Apr 24 11:02:38 asterisk named[1072]: lame server resolving 'pbx_spool.so' (in

Correction: The hardware is a Wildcard T100P (not a TE110P) Thanks! > -----Original Message----- > From: Geoff Manning [mailto:gmanning@zoom.com] > Sent: Wednesday, May 18, 2005 9:07 AM > To: Asterisk Users (E-mail) > Subject: [Asterisk-Users] Integrating Asterisk into our Legacy PBX > <--Newb > > > I have been successful in setting up asterisk and making >

Say I had a crossover cable that connected a Mitel SX200 to a TE110P and the pinout was done as such: 1 - 4 2 - 5 5 - 1 4 - 2 (the 5 and 4 are transposed on the left side) Instead of the proper way of: 1 - 4 2 - 5 4 - 1 5 - 2 What would the results be? We have had the former as our cabling for a few months and the connection has been fine. Slip errors here and there. But we have had major

On 26/02/15 23:30, Brett Wynkoop wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > On Thu, 26 Feb 2015 22:41:58 +0000 > Rowland Penny <rowlandpenny at googlemail.com> wrote: > >> Try 'samba-tool user add --help' >> All the info you require is there. >> >> Rowland >> > Still fail: > > Used pdbedit to remove the

Eric "ManxPower" Wieling wrote: >> >>> span=1,1,0,d4,ami >>> e&m=1-24 >>> > > Looks like you have told Asterisk to get it's timing from the Mitel. > I'll bet the Mitel is trying to get it's timing from Asterisk. > > Try span=1,0,0,d4,ami and run ztcfg -vvv > I just set this back. It was originally set to your

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On Thu, 26 Feb 2015 22:41:58 +0000 Rowland Penny <rowlandpenny at googlemail.com> wrote: > > Try 'samba-tool user add --help' > All the info you require is there. > > Rowland > Still fail: Used pdbedit to remove the previous user wynkoop used samba-tool to add user wynkoop with specific UID and GID I then

I updated to the latest rc25 and started getting these messages(over and over again). This is on a Solaris 10 box and I tried compiling with gcc and sun c. I backed down to rc23 and the messages went away and everything works. Any body have an idea on what could be wrong? thank you dovecot: Mar 04 21:14:33 Error: child 467 (imap) killed with signal 6 dovecot: Mar 04 21:14:33 Error:

When Asterisk first starts up, it will attempt to "bring up" the B channels on any PRI circuits. If you are using A@H then you can log on to the Asterisk CLI (asterisk -r) and then do "stop now" to stop asterisk. Start up Asterisk again by typing asterisk -cvvv at the Linux command line. You should see a bunch of messages on the terminal and then you'll get the Asterisk

Hi, I'm trying to answer a call on one line and dial out a number on a zaptel x100p fxo, but all I get from the phone I'm dialing is silence after it is picked up, and on the line that's supposed to be dialed out itself, noise. Thanks, Imran

Hello, we have a small Asterisk Network where Siemens PBX's are connected via PRI (Zap) to an Asterisk and the Asterisk's are connected through IAX, so this looks like this: Phone1 --- Siemens PBX --- Asterisk --- (IAX) --- Asterisk --- Siemens PBX --- Phone2 Now, when Phone1 calls Phone2 all wents well until there is silence - then the line seems to be death. My users wanted to have

From: Brett Walker <brett.walker at geometry.com.au> When printing hexadecimal numbers to a fixed width, padded with leading zeros, and also having a leading '0x'; the resultant string can be shortened by up to two characters if any leading zero padding character required is. int hexnum = 0x00001234; printf("%08X", hexnum); // results in 00001234

A call is placed via IAX2 from one asterisk to another, to a TDM400 channel whose extensions.conf entry is exten => 502,1,Dial(${COLIN}) exten => 502,2,Congestion If this channel is already busy when called, the call is instantly hungup, without the caller hearing the congestion tone. The log from the callers asterisk shows: -- Executing Dial("Zap/1-1",

Very very very cool. I have been thinking of this one myself. I thought I tested it and that I came to the conclusion that Samba did not recognize named pipes, and just treated them like normal files (overwriting them when you drag a text file onto them). But I must have made a mistake somewhere. It' perfect. The way I will set it up is like this: a Progress program will sit on a named