similar to: Incoming Caller ID

Hello, We are trying to figure out how to preserve the low-frequency content when we try to convert a .wav file to an .opus file. What settings in Opus that can accomplish this? Sincerely, Vinson Go cid:image001.png at 01C94EE1.4DA09EF0 TBSI - SIMPLICITY in Neuro-Solutions <http://www.tbsi.biz/> Tethered and Wireless Solutions for Neural Recording and Stimulation Vinson

I've looked through the archives - and see questions similar to mine, but no answers. What, if anything, can be done to get the incoming Caller ID to be presented on the Budgetone's Caller ID display? In all other respects the phone+Asterisk seem to be extremely happy with each other.

Running Asterisk Head 1.0.9. Below is a trace of a call delivered to my system which had no caller ID. For some reason, dialparties.agi shoves the incoming provider's IP address into the caller ID so you never have a call that is screened for PrivacyDirector. Is anyone else seeing this issue as well? Have I missed a patch? This call shows on the display with a name of "Unknown"

On incoming calls it seems that * is finding the callerid correctly but my BudgeTone is not showing it in the display. What am I doing wrong? The * console shows: -- Accepting call from '666666666' to '666666666' on channel 0/1, span 1 (numbers changed) but I guess that'c correct?

I have the following setup: Asterisk Server Sip Software Phones and a Wholesale connection At the moment, i can receive and send calls through the wholesale connection to and from the SIP phones. Now i want to collect the CallerID from the wholesale connection, and pass it on to the SIP phones when calls are routed through to them. I've tried a number of things including the line below,

Is it possible to reject all incoming calls that do not have a CID? Could I do something like that (modified version from the book): exten => 123,1,GotoIf($[${CALLERIDNUM} = ]?20:10) exten => 123,10,Dial(Zap/4) exten => 123,20,Playback(abandon-all-hope) exten => 123,21,Hangup( Alternatively, what's the privacy.conf file for? What does it mean for a user to have to chances to

I am aware of how to match a particular caller-id or a caller-id pattern and do something with the call like this: exten => 15554441212/_888NXXXXXX,n,Playback(GoAway) What I am curious about, is the best way to block unknown, private and 000-000-0000 calls. I know I can do this for 000-000-0000 calls: exten => 15554441212/0000000000,n,Playback(GoAway) Is there a better way to catch

I'm using QSIG between an NEC SV8300 and Asterisk (after giving up with CCIS). Things work pretty well with the exception of issues on stations on the SV8300. When I call from Asterisk to a SV8300 station and I send my extension as the caller ID number, it shows up on the SV8300 as "OPERATOR". I've tried a few different TON/NPI values with no difference. If I set callerid to a

I have a bunch of W2K clients on my network and I want to be able to use GPOs, but REFUSE to go the M$ route ;-) Since GPOs are essentially registry entries, it might be somewhat easy to implement a simple program that could give admins that fine-grain control that is missing from NT4-style .pol files. What I would like to propose is a simple Win32 executable that could read from an

I am not that good at regex and it's use in Asterisk. I am running Asterisk 1.4.13 Currently I have this in my extensions.conf for incoming calls on our house phone line: [housemenu] exten => s,1,GotoIF($["${CALLERID(num)}" = "815xxxxxxx"]?s|12); 815xxxxxxx is our home phone number, when caller id fails or is missing that is what is recorded. I want to expand this

Hi, guys, For my server, if i use my handphone to call in the PSTN line by TDM400p card, the server could not receive the caller id correctly. anyone knows the problem? I am currently using asterisk 1.2.14 with freepbx 2.2.1. The CLI is as below, "Caller ID name is 'zap1' number is '4521'" , this 4521 is one of my FXS zap extension created. dialparties.agi: Starting New

I am going to be on site at one of my recent installs tomorrow and I am hoping to fix an issue with the caller id. I would like suggestions for possible problem areas and so I thought I would give as much details as I can. The system has a Sangoma A200D card in it with 4 FXO ports and 2 FXS, The incoming pstn lines are all 3 part of a hunt group and At&t has confirmed the settings for caller

I had the latest wine running under winblows 98 SE for a week or two with no problems, then I installed winblows 2000 and now I cannot get wine to work. It is important for me to get it running again since I run United Devices cancer research agent (http://members.ud.com) 24/7 to help them find a cure for cancer. Is wine compatible with Win2k, or will I need to re-install 98 again? -- Interested

I am trying to have asterisk update the system name in my realtime peers, but it does not seem to be working. Here is what I've done so far. - added systemname => mysystemname in asterisk.conf - set rtsavesysname=yes in sip.conf. - created a table called "sysname" in my peers table in mysql - restarted asterisk - rebooted my phone to force a re-register Is there something

I am wondering if any of the Texas user groups have members in the North West part of the state. I am in the Amarillo area and would like to find some othere in this area, maybe even start a user group in this area. -- Bruce Reeves Nortex Networks -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am getting read to roll out close to 100 polycom phones and wondered if any one knows of a program to take a list of MAC addresses, extensions, and names and generate the configuration files? -- Bruce Nortex Networks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060504/d3bb612a/attachment.htm

> Yes, exactly (and there will be other settings as well, to identify the > type of peer (network, trunk, endpoint) for other reasons). > cool, I really should read the lists more :) > > That's coming too, but in a different way. Actually if your remote peer > can send you Remote-Party-ID headers now, you can set "trustrpid=yes" in > your peer definition

I have compiled 2.4.5 and patched the src rpms for e2fsprogs01.20 and util-linux 2.11b. I can boot with ext3 as my filesystem type in fstab and fsck doesn't seem to mind it or ext3,ext2 but mount from util-linux chokes on having ext3,ext2 in fstab during bootup or later manually. Anyone have any ideas? I am running RedHat 7.1 Thanks, Bill -- Bill Vinson <billvinson@nc.rr.com>

Just to bug you all (feel free to rant at me), a client wants to set his caller*ID number for outbound calls though us to PSTN. the client is using SIP to us, he can set the caller*ID name fine. if he sets his caller*ID number to anything other than his account number (8440101), the call is dropped into the default context (and then hung up by our dial plan). To get around this i

> > To get around this i updated CVS HEAD and changed the sip entity from > > type=user to type=peer (yes peer!) (type=friend works too but im making > > a point) the client now must register to set his outbound caller*ID > Number. > > Yes, that is normal. SIP has difficulty separating the remote party > identification from the authentication identification