similar to: Hardware spec comparison

All, Since I have now (at least partially) got my 7961G phones working with Asterisk, I have temporarily moved on to try to get the expansion modules working. There doesn't seem to be much in the way of documentation here either. Does anyone have this combination working (or any 79X1) here? My goal is ultimately to do the monitoring approach. I have Google'd around, but come up

All, I have run into some CallerID issues. It seems to have happened as a result of just moving my config from 1.2.12 to 1.4.4 (although I am not sure of this). Therefore I am sure its just a misconfiguration somewhere, I just don't know where. I have throughout the office either Cisco 7961G or Polycomm Soundpoint SIP 430 IP phones. The problem is with CallerID showing up in some

hi, I am facing some issues while using MixMonitor. My extensions logic is attached below: exten => s,1,MixMonitor(${CALLERID(number)}-${TIMESTAMP}-${UNIQUEID}.gsm,b) in this extensions TIMESTAMP is not working in Asterisk 1.4. can any help me why TIMESTAMP is not working in Asterisk 1.4. regards, Asif

I am getting this error [Aug 6 15:28:26] WARNING[24452]: chan_sip.c:1920 retrans_pkt: Maximum retries exceeded on transmission 8f68421-22821e1e at localhost for seqno 102 (Critical Response) [Aug 6 15:28:26] WARNING[24452]: chan_sip.c:1944 retrans_pkt: Hanging up call 8f68421-22821e1e at localhost - no reply to our critical packet. any Ideas? Jason

I have installed Hmisc and Design. When I use ols, I get the following error message: Error in if (!length(fname) || !any(fname == zname)) { : missing value where TRUE/FALSE needed The model that I am running is: > ecools <- ols(eco$exp ~ eco$age + eco$own + eco$inc + inc2, x=TRUE) I have tried several other combinations of arguments that take TRUE/ FALSE values, but no luck.

Hi, Has anyone here ever used a Polycom IP 4000 Soundstation SIP Conference Phone with asterisk? If so, how well does it work and how does it sound?

I posted this yesterday to r-help and Ben Bolker suggested reposting it here... Dickison, Daniel <ddickison <at> carnegielearning.com> writes: > > The documentation for agrep says it uses the Levenshtein edit distance, > but it seems to get this wrong in certain cases when there is a > combination of deletions and substitutions. For example: > > >

I am running Asterisk 1.4.5 and addons 1.4.1 in a CentOS 5 Server. My PBX has experienced several core dumps the last couple of days and I am not sure if this is what's causing it, but it always seems to happen when a particular extension on a grandstream phone uses ChanSpy SIP group. I have not been able to locate where the core dump file is being saved. I can't find it in my

Hi folks, A user here has asked if we can display the current voicemail message's envelope (date/time/caller id of message) in the LCD of the Polycom phones we use (430 & 501). I realize this is somewhat like the many caller-id-after-the-fact threads, but I figured maybe someone had solved this a different way. Has anyone been able to do this, via caller ID, messaging, the mini-browser

Hi all, When the Dahdi driver detects DTMF, it seems it's not muting the first 5-15 ms and sometimes the last 2-10 ms of the DTMF tone. This shows up in recorded voicemail greetings -- you hear a very short DTMF '#', or sometimes two blips, at the end of the recording. I have a Mitel SX-200 connected to Asterisk 1.6.0.1 by a couple of Digium cards: a TE420 w/Octasic and pri_net

The bar is getting raised yet again http://www.voipmonitor.net/2007/06/05/New+Features+And+Services+For+Pack et8+Virtual+Office.aspx I personally use Snapanumber $30 or there abouts (after trialing a few other TAPI solutions and finding them sub-par) and think it's a great product but interesting to see how more people are expecting desktop/phone integration applications. Does anyone

This may not be the right place to ask, but I have an annoying issue with a Sipura/SPA1000-2.0.10(e) ATA device connected to an Asterisk box. (The system is remote to me, so I've only been able to observe this by dialling into a VoIP phone on-site, then run commands on the box remotely!) First of all it's all working fine connected to an Asterisk box and the user can make/take calls

Strange issue.... when I record a file from a phone to the asterisk system I get a blip in the recording every 30 seconds. It's a very small blip, but it is there. It seems like it's only if I'm recording, not when I'm playing back that the issue happens. My SATA drives, ETH0, and my Sangoma card are all on seperate IRQs. Any thoughts on what might be causing this and how to

Hey guys, Earlier in this thread Oddsock said that a player recieving metadata it did not expect would most likely produce blips in the output. Has anyone checked any of these network dumps to see if any metadata is being sent? A while back I reported difficulty with playing Icecast2 streams through my TiVo. I recently upgraded to Beta 1 and the problem still exists. I don't want to

Using simple authentication I have been able to tie FDS to Samba 3.x.24. Knowing that the unix passwd and smb passwd are different, dare I ask how difficult it would be to have them sync? Most of my users are using netatalk w/ posix user info and MD5 password. I would like to swing this over to samba without the worries of two passwords per user. I have seen blips on this but not directly related

I am looking for help from anyone to try and get this program running. I recognize that this program is not listed. Thats fine. That does not mean it can't be made to work. What should I be looking at to find out why it isnt running? How should I be monitoring it? The program is free at http://www.karaoke5.com/welcome-uk.htm. I am running 1.3.9 on Ubuntu 10.04. Please do NOT mention

Hello, We use MS Access 2000 (I know, we're migrating away from it) as an application to automatically dial phone numbers. The old phone system we have allowed the call representative using the application to take their phone off hook, push a button in the app, and the app would send the phone number to the phone system and dial the number. We are moving to Asterisk for our main phone

Hey folks, So I'm trying to get Festival() working on 1.2.17. I'm trying to use app_festival: Here's the show dialplan output from that extension: '3378' => 1. Answer() [pbx_config] 2. Festival(Hello Asterisk caller. How is your day?) [pbx_config] 3. Playback(vm-goodbye) [pbx_config] 4. Hangup()

Hi, I have lookedaround and don't see this anywhere. Is there a way to tell the ip500 to not make the aural MWI blips?

I've just done some rudimentary testing to see how Vorbis degrades at absurdly low bitrates without downsampling. In summary, don't hope for anything decent below -q 0 for now. I tried oggenc -b <bitrate> -M <bitrate> for the below and a few in between: 24k - spectral energy "floor" captured decently, but many pure-tone blips (think old computer movie sound effects)