similar to: Pattern matching - how to ignore numbers after 10 digits

All: I've starting building an asterisk system for our company, which has about 60 users. I am new to asterisk, so thank you for your patience. I've stripped the sip.conf and the extensions.conf down to the bare minimum: Here is my extensions.conf file [globals] [general] autofallthrough=no [default] [fromprovider] exten => YYYYYYYYYY,1,Dial(SIP/151,20) [phones] exten =>

Dear Sirs: I am not yet a user of R. My background includes the use of (Turbo) Pascal for scientific analysis of underwater acoustics problems (e.g. sound ray tracing and array gain in directional noise fields.) I have become interested in the following type of problem: (1) select , say, 1000 random locations within the continental United States; (2) characterize (statistically) the

Hi All, Have completely moved off the old ESI system, and things have been going pretty good with the new server. I have one issue, which has been reported by several of our customers. I've tested it, and it does indeed seem to be a problem. When the customer is asked to dial in the first three letters of the person they are trying to reach, they will be routed to the wrong extension.

All, The users in our organization are well, quite frankly, sick of phone service that is being provided. The choppy phone calls, and drop outs are detrimental to our sales force. I've tried about everything I can think of. Moved the asterisk server from VM machine to dedicated machine More than enough bandwidth Setting 802.1p = 7 Set Dedicated voice traffic 35% of bandwidth. Not sure

All: My company has an existing ESI IVX E-class system with 45 phones. I can add one more card, to expand it another 6 phones, but it's $8000, and then the system will have to be replaced. I have the Asterisk server up and running, with 2 sip lines from the local phone service. (Thanks to you guys, it is working great!). I'm pretty sure this is the way the company will move, and

Hi everyone! I'm quite a newbie at this Asterisk stuff so please bear with me. We've recently decided to start training in Asterisk via AsteriskNow! Asterisk version is 1.4.18.1 through AsteriskNow! 1.02 The box we have is paired with a Digium TE110P and we've managed to get it to the point where incoming calls via a single DID (from NTT Japan) can be received and answered

All: RKG needs an asterisk consultant to help us track down issues we are having with our system. Mainly dropouts and dropped calls. If you have experience in troubleshooting these issues, please contact me at email attached to this messages. Regards, Eddie -- Eddie H. Mikell Senior Systems Engineer RKG Office: 434.970.1010 x 124 Email: emikell at rimmkaufman.com --

"sip show channels" shows some info about active sip channels, the current codec included. What does it say? jg" jg, sip show channels reports the Format as being ulaw for 17 active calls. Holds - no Peer User/ANR Call ID Format Hold Last Message Expiry Peer xxxxxxxxxx kbrown xxxxxxxx (ulaw) No Rx:

All: Starting switching over my phone lines. I got phone line 1 switched. Everyone working. I switched the second phone line, and it worked about an hour, then I started getting errors from the cli saying the server could not register with the providing. I restarted the system, and it worked ok for about 30 minutes, and then started giving he same errors. The error is [Jul 13 11:21:14]

resending to provide a more informative subject line.... On Fri, Jan 2, 2009 at 3:21 PM, Kingsford Jones <kingsfordjones at gmail.com> wrote: > Hi David, > > A general answer to your question is: yes, R would be useful for such > analyses - particularly when interfaced with a GIS. For an > introduction to the types of spatial tools available in R see the Task > View for

All: Getting ready to put the system in production. Any suggestions on "stress testing" the system? I'd like to initiate say 10 sip phone calls to make sure the provider has the bandwidth. Can you do that in CLI? I've called 4 numbers simultaneously with the hard phones I currently have and am thinking of adding 6 or so soft-phones to various pc's to make a total of

Jim, and all: Thanks for the response. If I can repeat what you are saying: you don't have to define the multiple lines in sip.conf? For comparison, with my current esi setup, we have 10 outgoing lines. If one line is busy, then the service just rolls to the next number. This is set up with the phone service. That doesn't have to done with outgoing sip lines? Does the dialstatus

I have added the three lines to the sip.conf file based on the latest changes from broadvoice. I can receive incoming calls but cannot place any outgoing calls. The error I get is: *CLI> -- Registered to '69.73.19.178', who sees us as IPADDRESS:4569 -- Attempting call on SIP/Broadvoice/5068012 for application Playback(demo-congrats) (Retry 1) Mar 8 08:35:21 NOTICE[29290]:

All: Still experimenting with the asterisk server for the company. My local phone company has given me two sip numbers to experiment with, say 444-456-1234 & 444-456-5678 Calling in and out works, and I've spread a couple of the phones out with some co-workers. My question is this: Do I have to define multiple sip lines in either the sip.conf or the extensions.conf? Now when I

All: Some of the people who dial into to our system will press the pound key when entering an extension for the directory key. When waitexten gets that, I get an error messages as, for example 123# doesn't match any extension. I was going to use ${EXTEN} to just use the first three numbers, but I'm not sure how to use this with WaitExten. so I have exten =>

All: I am looking to install another asterisk server in an office located in a different part of the country. I think I can configure the sip and extension conf files, so that the internal phones at the two locations can call each other. My question is this, how do I properly configure the sip file for a different provider at the new location? Can I use a different register statement for

All: I am using the standard voicemail in asterisk. Everything works well, except, if a users wants to record their own personal greeting, it doesn't playback. I can see the soundfile being created. I suspect it is a setting in the voicemail.conf, or an option I am over-looking on the grandstream, but if anyone can point me in the write direction, I would certainly appreciate the help.

Hello. I'm trying to upgrade my asterisk installation to most current CVS version. Currently I am running CVS-03/24/04-07:26:16 and dialing out works fine. When I install the latest CVS, outbound dialing fails, but inbound and internal calls work just fine. == Spawn extension (it, 9651246****, 2) exited non-zero on 'SIP/8202-d359' -- Executing

Hello-- I'd like to do a little processing on external phone numbers from within the asterisk pbx. Fairly simple stuff, but... devilishly hard to make it work so far! 1. I'd like to dial 9 to get an outside line. 2. If the number dialed after the 9 is 754XXXX, I'd like it to go thru unmodified. It's the only local number available here. 3. I'd like all 1 XXX XXX XXXX numbers

I am trying to supplement and ultimately provide a patch for ''foreman'' which is an adjunct to puppet. Essentially, there is a Hosts class which belongs_to Hostgroup and Hostgroup class has a column called ''ancestry'' which is actually a Hostgroup (probably what is referred to as STI but I am not sure) and thus within Foreman, nesting Hostgroups is not uncommon.