similar to: Sudden appearance of SIP/2.0 401 Unauthorized

Hi, Pretty sure I'm missing something simple, but I've seen references to this feature but not found documentation for it: I have a queue set up so that many people are contacted (ringall) when a call comes in. I would like the answering party to confirm that he is a human being rather than cellphone voicemaill by pressing a digit. This is somewhat similar to the 2nd macro example

I've noticed that some products, like Fonality's HUD, allow live monitoring of a VoIP call (not just Zap Barge). The Asterisk {client | manager} command set only seems to allow recording to a file without the use of a meetme room. Does anyone have a good solution for this? What I'd like to implement, ideally, is that once an incoming call is transferred to a particular operator,

One of my user is praising Skype!!! I cannot figure out anymore what I can improve! This users sip show peers is jumping from 65 msec to 1800 all the time. Of course his voice quality is like a morse code with dashes or dots of connection time. The next minute he calls me via Skype and it works fine !!!! What indicates that there is no fault on his Internet connection!!! He is using his

Hi! I am currently working on setting up an Asterisk system, and I was wondering if anyone has worked on an alternate mp3 player to mpg123. We have a library of MP3 files that we would like people to be able to select and play over the phone -- and this will require pause & resume, as well as fast forward / reverse (jump forward / jump back). It doesn't seem like mpg123 can do this. Is

All, I'm having a problem with an Innomedia 6328-2Re (old Sunrocket Gizmo). It keeps losing registration after a period of time ranging from a few minutes to a few hours. It seems that right before it loses registration, it fails to send a second register (after the 401 unauthorized). Here's a transcript from wireshark (at the end). The last message is all that's received and

We've been running an Asterisk-based phone system here in our office for a year and a half, and it's pretty much been running smoothly. One employee who works out of the office has a problem that she can't make outgoing calls on a temporary basis every so often (a few times a day). No one else has this problem, her settings are fine, and she regains the capability spontaneously with no

Hi, I'm running asterisk 1.4.22 on a debian server. I have php5 installed and it works correctly command line. When trying to run a php script via AGI, I get messages such as: GI Tx >> I> AGI Rx << #!/usr/bin/php5 -q AGI Tx >> 510 Invalid or unknown command The scripts are completely executable and owned by asterisk -rwxr-xr-x 1 asterisk asterisk Googling is not helping

The options are not seperated by commas. exten => s,1,Dial(SIP/50,23,r,d) should be exten => s,1,Dial(SIP/50,23,rd) -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of John Klimek Sent: Thursday, June 22, 2006 2:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users]

I'm trying to sync from 3.0.5 to 2.5.5 (SCO, *sigh*) The source machine errors out b/c the --remove-source-files (or --remove-sent-files) options don't exist on 2.5.5. But, since these option are only acted on by the sending machine (which is where I'm typing the command line), it would be so nice if the source machine would NOT pass the option on to the receiver (or at least, not

I have 5 or 6 :-( different copies of a filesystem on various Linux boxen, all backups taken at different times, with different exclusions, and squirreled away. Now's the time to clean up my attic. I'd like to merge them all into one big filesystem. When there are different copies of the same files, I'd like to keep the newest; I don't know what else to do. My plan (assuming

Dear sir, I got trouble on InnoMedia video phone with Asterisk-1.2.4. If InnoMedia video phone as a caller, then the call will be a success, no any problem. The problem happens: If InnoMedia video phone to be a callee, the call can not make successfully. For instance, Caller 23267668 dialed to callee 23267663, callee 23267663 was ringing. If I picked up the callee's phone, then abnormal

(Shorewall 1.4.4b; running the Mandrake edition.) Occasionally, usually during a zone transfer, I get unusual Shorewall messages, like this: Sep 30 20:30:08 yoreach kernel: Shorewall:FORWARD:REJECT:IN=eth1 OUT=eth1 SRC=10.1.1.1 DST=10.1.1.230 LEN=54 TOS=0x00 PREC=0x00 TTL=63 ID=21332 DF PROTO=UDP SPT=4778 DPT=53 LEN=34 where the src is the DNS master, and the DST is the slave server.

Hello everyone, I am trying to fit the following model All X. variables are continuous, while the conditions are categoricals. model <- lm(X2

Hi, we're running Asterisk 1.2.10 and have been randomly being left blank voicemails with long messages that we can't hear. I've searched and searched but cannot find a solution. This is what happens: Internal Server runs Asterisk 1.2.10 where our mailboxes are Incoming Server (behind a firewall) runs Asterisk 1.2.13 and calls are bridged between this server and our internal server.

Hallo, All, I have a question about changing the height or scale of the y axis. When I use following two R codes, I can get two plots. Please look at the y axes, the number of indices (x1, x2, ?) on the y axis in the first plot is smaller than that in the second plot, and hence the space between any two indices in the first plot is wider than that in the second plot. As the number of indices

Hi everyone, I run MCollective 1.2.1 together with ActiveMQ 5.5 under Scientific Linux 6.1 on Amazon EC2. Overall it works like a charm, but sometimes (eg. 1/30) discovery fails. Still the exit-code of mco will be 0, which is a problem for me as I use MCollective e.g. to trigger deployments from Jenkins. I would like to ask for some feedback on the following ideas, that could fix this problem.

We are testing the innomedia ATA's to possibly replace our current line up of ATA's that we are using. Has anyone used their product? What is their track record on stability, voice quality, DTMF talkoff, T.38 Thanks Bryant ---------------------------------------- From: "Zeeshan Zakaria" <zishanov at gmail.com> Sent: Wednesday, October 13, 2010 10:41 AM To:

Hi All, I have a query about time based sequences. I know such questions have been asked a lot on forums, but I couldnt find the exact thing that I was looking for. I want to create a time-based sequence which will mimic the trading window AND would span multiple days. Something like below: "2011-01-03 09:15:00 IST" "2011-01-03 09:15:01 IST" .... .... .... "2011-01-03

Regarding this message, I've actually been told one caller who has consistently had this problem was using Vonage, but calling from his Verizon line, it worked. This skewed my survey. Therefore I do believe it's the same callers having the issue, and in which case, I think Vonage is to blame. I found this thread:

I currently us Linksys/Ciscio, Grandstream and AudioCodes ata's. none of the three perform well in all enviroments. Between stablity issues, T38 and DTMF talkoff all three suffer some combination of issues. I am looking at Patton and Innomedia. Has any one tried either brand and what is your experience with them. Which would be the base for stability, audio quality, provisioning, DTMF