similar to: Initial Chirp while dialing

Displaying 20 results from an estimated 5000 matches similar to: "Initial Chirp while dialing"

2005 Jun 28
1
TDM400
Hi, I have an TDM400 4 FXO module setup on my dual Celeron server running asterisk 1.0.2 and I have had to restart the asterisk process multiple times lately. I was wondering if anyone else has to restart the asterisk process after storms roll through their area. Before I restart I usually have sound quality issues and weird pickup problems. But, once I restart the asterisk process everything
2003 Jun 25
1
indication tones and callwaiting chirp too loud
I am wondering if anyone could help me figure out how to turn down the volume on all the dial tones, indications, etc.. and especially the call-waiting CHIRP! I don't want to change the txgain and rxgain because they are working at levels that I would like. However, when voice conversations and voicemail recordings are at good levels then the dial tones, busy tones, etc are way too loud.
2004 Dec 03
1
PolyCom MWI Chirp issue
Has anyone had an issue with the polycom's not discontinuing the mwi chirp even after the message has been acknowledged? -- James M. Milne Nuvio Corporation CCNA - CCNP - CIPTSS - CCIE milne.james@gmail.com
2004 Dec 03
4
Polycom 500, won't ring??
Hi, I have was testing some of the different ring types with my polycom 500, and the ALERT_INFO settings. Now when my phone receives a call it won't ring. All the other phones ring fine, and my phone wasn't the only one I rebooted with the changed sip.conf and impd.conf. I have reverted back to a standard sip.conf and impd.conf and I still can not get my phone to ring for any incoming
2005 Jan 04
1
Re: Polycom Buddy Feature
I'm still trying to work this out. I've got this in my sip.conf [1003polycom] type=peer secret=abc123 host=dynamic defaultip=192.168.1.215 context=default mailbox=1003 subscribecontext=phonestatus [1004polycom] type=peer secret=abc123 host=dynamic defaultip=192.168.1.214 context=default mailbox=1004 subscribecontext=phonestatus And this in my extensions.conf [phonestatus] exten =>
2004 Dec 08
2
Dropping Calls, irregular interval no logs
Has anyone seen an issue with SIP phone (polycom 500) dropping calls at irregular intervals with no errors in the asterisk log files? I am having this issue as described and it is a complete pain in my rear to trouble shoot because when I call my cell phone I can get a call to last over 30 minutes yet when I call another office that uses a standard pbx I can't get past 10 minutes. For some
2004 Dec 07
3
Continuance on Polycom issue, not ringing
Ok, so I emailed the list earlier about my polycom phone not ringing when anyone called in. Well, polycom support said that is impossible that this could happen because of a change in a configuration file. However the new phone arrived today (a refurb.) and it also would not ring. So I obviously got rather frustrated and blasted away all of my configuration files from the FTP server. I then copied
2004 Dec 23
1
Polycom 600 problem
Andrei, Do you have X-Windows running on the linux box? I had a similar issue that was eliminated when I stopped this process and samba from running. Now samba is only allowed to come up during non-business hours, for changing BG music. Also, make sure your registration period in either (polycom) ipmd.cfg or sip.cfg is set to be at least the default 3600 time period. I also removed the
2007 Dec 11
1
Using predict()?
I'm trying to solve a homework problem using R. The problem gives a list of cricket chirps per second and corresponding temperature, and asks to give the equation for the linear model and then predict the temperature to produce 18 chirps per second. So far, I have: > # Homework 11.2.1 and 11.3.3 > chirps <- scan() 1: 20 2: 16 3: 19.8 4: 18.4 5: 17.1 6: 15.5 7: 14.7 8: 17.1 9: 15.4
2009 Jun 12
1
Resampler saturation
Hi Jean-Marc, I use the resampler to convert various sampling frequencies to 48 kHz on my Blackfin platform (fixed-point) 48K -> 16K speex -> 48K chain does not sound very good compared to plain 16K. But the main issue is when processing loud signals, I have truncation (and not clipping/saturation) I could hear it and see it with various music and speech messages. See example.png. I also
2005 Feb 04
1
Polycom Auto-Answer and Call Transfers
I have my * and polycom system setup to do Auto-Answer for internal SIP/Staff calls, and I am running into an issue with this and the polycom call transfer feature. * is seeing a new call come through from the polycom and is then transferring the call over. I need to know if there is some way I can grab a message from the SIP header or something to determine if I should not set the ALERT_INFO tag
2005 Oct 11
0
Echo on SIP Side?
It appears that I am getting echo only on the SIP side of a SIP -> TDM call. I am using polycom IP500's with a 4 port fxo TDM E/F model. I get this echo after about 2-5 minutes and after another 2-5 minutes the echo disappears. It does not appear that the other side (TDM) can hear the echo only myself. Does anyone have any recommendations on what to check, or what this might be? Thanks,
2006 Feb 06
4
DO NOT REPLY [Bug 3488] New: writefd_unbuffered failed to write 4096 bytes: phase "unknown" [generator]: Broken pipe (32)
https://bugzilla.samba.org/show_bug.cgi?id=3488 Summary: writefd_unbuffered failed to write 4096 bytes: phase "unknown" [generator]: Broken pipe (32) Product: rsync Version: 2.6.6 Platform: Sparc OS/Version: Solaris Status: NEW Severity: major Priority: P3 Component: core
2005 Sep 09
4
Huge Echo
asterisk-users-bounces@lists.digium.com wrote: > In the following setup: > call coming from a pstn line -> into FXO card -> asterisk -> SIP > phone > > i get an incredible loud echo in the SIP phone (about 0,5-1s) > (everything i speak into SIP phone microphone i hear in its > speaker). The person calling from PSTN is not getting any echo. Make sure you're not
2005 Jan 31
2
Dialing out on TDM400p 4 port FXO
Hi, I have two small companies that are going to be sharing a * box. I have 2 TDM400's with 4 fxo ports each. Each company has its own sales person and they would like the sales people to always show their own caller id and have their own lines ring directly to their phones. Company 1 sales person uses the 1 port on the tdm400 and company 2 sales person uses the 2nd port of the tdm400.
2004 Nov 19
2
app_sms: problems sending a sms
Hello, i try to send out a sms, but with no success. The trunk is a E100P, and the sms should go out to the Telekom SM-SC. What i want to to at the first run is, sending out a sms when a certain number is dialed. I tried: In extensions.conf: exten => 35953,1,SMS(${TRUNK}/9350193010,,0179NUMBER,"Hi there") exten => 35953,2,SMS(${TRUNK}/9350193010) exten => 35953,3,Hangup
2008 May 31
0
FFT Resampler spectrograms
Using the following MATLAB snippet: fnames={'chirp','perfect','block','filter'} for k=1:length(fnames) fn=fnames{k}; myfile=fopen([fn, '.fl'], 'r', 'ieee-le'); x=fread(myfile, Inf, 'float32', 0, 'ieee-le'); fclose(myfile); X=specgram(x,2048,1,kaiser(2048,14)); spectrogram_floor=-96;
2010 Aug 02
3
OT -- apcupsd messages
Does anyone here have any "feel" for the "Battery disconnected" and "Battery reattached" log entries? The rebooting came as a result of me turning off modems, router, external drives, monitor, cordless phone and finally, the computer, trying to locate a quiet but annoying "chirp". The "chirps" stopped when I tilted the UPS to look at the front
2008 Nov 19
3
TDM400 (?) zap hangup
And if that ain't confusing I don't know what would be. I bought a TDM400 with two modules (FXO, FXS) a couple or so years ago and ended up never using it. Passed it along to a friend who is having some problems with it. (He isn't on this list.) We've both tried searches using Google but haven't been able to find anything that helps. So this is more a question of
2005 Mar 11
0
Sipura 2100 and Asterisk and Fax
I've just made an interesting observation that I'd like to share with you all: the popular Sipura SPA-2100 just doesn't seem to be as great as I'd hoped. I've been trying to get inbound AND outbound faxing working via Asterisk and at least one of my termination services: Voicepulse or Sixtel. In general, inbound has been working flawlessly but outbound has been pretty