similar to: Confirmation key to answer -- for a queue

Yesterday we moved one of our servers to a new IP. We updated DNS and various adapters configured to register to that server registered to the new IP correctly. All seemed to be well. This evening I discovered that with one exception, all of the adapters are getting a SIP/2.0 401 Unauthorized message back from asterisk. The exception is an Innomedia adapter -- Linksys PAP2's and (I

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

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

(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.

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.

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.

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:

Anyone from puppetmanaged.org listening to this list ? The web page to create an account <http://www.puppetmanaged.org/user/register> is busted. The CAPTCHA does not show up and you cannot register without it. So I tried their mailing list -- <http://www.puppetmanaged.org/mailman/listinfo> I was able to join, but then my attempt to post to the list bounced:

----------------------------------------------------------------------------- Shorewall 1.2.12 # uname -a Linux yoreach 2.4.18 #1 Sun Apr 21 12:50:34 CEST 2002 i686 unknown # ip addr show 1: lo: <LOOPBACK,UP> mtu 16436 qdisc noqueue link/loopback 00:00:00:00:00:00 brd 00:00:00:00:00:00 inet 127.0.0.1/8 brd 127.255.255.255 scope host lo 2: dummy0: <BROADCAST,NOARP> mtu 1500

John (to a serial querulant): ...but with such a sweeping lack of information from you, don't congratulate yourself if you get a helpful answer. It wasn't your fault. David Winsemius Alameda, CA, USA 'Any technology distinguishable from magic is insufficiently advanced.' -Gehm's Corollary to Clarke's Third Law

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

If I understand this correctly, this is a user outside your firewall dialing in to your office over the Internet. Always, inbound calls work, but sometimes, outbound calls do not work. So if you have replaced the hardware totally, and you still have the same problem, it could be a routing issue with an upstream ISP. The way to test for this is to do a traceroute from her LAN to your office. Then,

Hello, > I am setting-up a system to place outgoing calls for a certain > number of minutes (as allowed per the customer's account). I would > like to "break into" the long distance channel to announce "1 minute > left", etc. What asterisk command can I use to do this? > > Thank you in advance for your help. > > Chaya Rosenberg >

Hi, I am using Polycom 501 and I came across a problem. As soon as I have incominglimit=1 in sip.conf, which is necessary for buddy watching, I cannot transfer calls. On the console it tells me: Call from user '3052' rejected due to usage limit of 1. Can someone please tell me how to get around this problem? (I don't know if this is relevant, but in the phone.cfg file, I have