similar to: Bridging old system (ESI IVX E) with new Asterisk server

All, Thanks for the suggestions, but the system is a plan non-sip, non-ip, non pri setup. It's pretty much a closed box setup. And the prices for the card and support are robbery - which is why we aren't going to go with another setup like that. While it has been reliable - I don't think there has ever been an issue with it, expansion is expensive. The local company was

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: 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 =>

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

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

All: Yesterday I discovered something interesting. I dialed 1800ANCESTRY from the asterisk system I am testing and got the number doesn't exist message. I then dialed the same number from our old system and it went through. I realized that the "Y" in ancestry made the number too long, and went back to my dialplan. How do I ignore numbers that are too long? Obviously,

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: 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, 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: 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: 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 --

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.

I've had success on "older" model Lenovos.....(T-410 / T-420 / T-430) but anything beyond those seems to have some issue or another. I was even able to swap the standard drive to an SD (250GB) on a T-430 and it's running g like a champ. A lot of the newer stuff is OK as long as you don't have any boutique drivers for video network or sound. YMMV. On Sep 29, 2016 9:18 PM, John

hi all, I have a strange problem using the libsmbclient api.. retrieving or stating files named with percents in is for some of them impossible. Example: a file located at "smb://test/test/%7C2.txt" cannot be retrieved with my code, smbc_open fails with the code NT_STATUS_OBJECT_NAME_INVALID. I am not sure but, url encoding/decoding that would escape strings with "percent"

I have 3 4TB WD drives I want to put in a RAID1 array. Two WD4000FYYZ and One WD4000F9YZ All enterprise class but two are WD Re and one is WD Se. I ordered the first two thinking 2 drives in the raid array would be sufficient but later decided its a long drive to the server so I would rather have 3 drives and ordered a third in accidentally did not get EXACT same thing. Would there be ANY

Hi all, I just wanted to share a project I''ve been working on now for a few months. It''s still far from complete, but has already been very useful for me and my coworkers. I''m calling it mongrel-esi. ESI stands for Edge Side Include. The specs live here, http://www.w3.org/TR/esi-lang. I currently only have support for <esi:include, <esi:try,

Hi, I’d like to run clang with some optimization passes (-reassociate, -simplifycfg, -dce ) But in the end I need to dump the llvm IR (no link to assembly generation). I didn’t find any way to do that. I found out that passes are run via opt, but opt generates bc code, and I want IR. Any help will be welcome Thanks, Régis -------------- next part

Dear all, I'm new R's user and I'm looking for package dealling with Path analysis. Does it exist ? Where ? Best, Regis Martin PhD Student Laboratory of Altitutdinal Population Biology UMR CNRS 5553 Universté de Savoie Bât. Belledonnes 00 33 (0)4 79 75 86 44 regis.martin at univ-savoie.fr

Hello, I am developing an agent-based model to simulate the spread of infectious diseases in heterogeneous landscapes composed of habitat polygons (or clumps of connected cells). To simplify the model, I consider a habitat grid (or raster) containing the polygon ID of each cell. In addition, I have epidemiological parameters associated with each polygon ID. At each time step, the parameter values

Hi, I'm trying to use a similarity matrix (triangular) as input for pam() or fanny() clustering algorithms. The problem is that this algorithms can only accept a dissimilarity matrix, normally generated by daisy(). However, daisy only accept 'data matrix or dataframe. Dissimilarities will be computed between the rows of x'. Is there any way to say to that your data are already a