similar to: Cisco ATA 186 and Asterisk

I'm running: Asterisk CVS-HEAD-07/06/04-17:49:49 built by root@gf-002-pbx-001 on a i686 running Linux I've tried placing files (both ending in .call and not) in the correct format in /var/spool/asterisk/outgoing.... I get _nothing_. No log messages, nothing on the console, zip. Permissions seem to be correct on both the files and the directory, as well (* is running as root, for right

I see that there's now a 2.16.1 upgrade path for Cisco ATA-186 devices, dated (variously) July 11 or July 14 2003. Here are some interesting bugs that claim to be fixed. Most notable is CSCeb17953, at least from my perspective, as I've hit this bug before. CSCea42480 The Cisco ATA ignores the Require:100rel header and processes call. CSCea69889 The Cisco ATA builds a 302 Moved

Hi all: I'm testing a new installation of *, bringing up some ATA186. In * environment, all stuff works greats. The only thing that don't work is a Call Transfer, but the 3Party works ok. Some time ago I read that somebody had proven this functionality successfully. If somebody knows what I missing, please let me know. Thanks in advance, Gus -------------- next part -------------- An

Hello, I have a couple of questions with regards to fitting a coxph model to a data set in R: I have a very large dataset and wanted to get the baseline hazard using the basehaz() function in the package : 'survival'. If I use all the covariates then the output from basehaz(fit), where fit is a model fit using coxph(), gives 507 unique values for the time and the corresponding cumulative

I've never seen this, and have no idea how to reproduce it. For resloution you are going to have to give me a working example of the failure. Also, per the posting guide, what is your sessionInfo()? Terry Therneau On 08/09/2012 04:11 AM, r-help-request at r-project.org wrote: > I have a couple of questions with regards to fitting a coxph model to a data > set in R: > > I have a

I'm running a Linksys WRT54GL router from my CentOS-7 home server. Every now and then (maybe once every 2 days) the router's WiFi cuts out, and I've found no way to solve this except to disconnect the power from the router, wait 10 seconds and then re-connect. This always works. The router is running under dd-wrt. My question is - which makes it a tiny bit CentOS-related - does anyone

I'm looking for somehting like the standard house hold linksys/dlink router. Basically it needs to have at least 1x100mbit port, wireless G capabilitys and at least 1 x anolog voip/sip connection. I've found linksys's WRT54GP2 which appears to do what i want. Anybody use this? Does it require vontage's service? I'm looking for any recommendations. Thanks -- ~Shaun

Hi, anyone can confirm if the Linksys's ATA and Router (PAP2-NA and RT31P2-NA) have the same limitation of just one G.729 call like the Cisco ATA 186 ? I'm testing both appliances here and found this issue but could not confirm this anywhere (nothing on the manual, no document or post from any user about this). In my tests they use G.729 only on the first call and G.711 on the

I just bought two new Linksys WRT54G routers. Sveasoft has loaded Linux on this router and included a bunch of Linux tools, one of which is Bandwidth Management. The QoS aspect of this is supposed to be much more granular than the previous solution (Wonder Shaper). I have not been able to find any suggestions for how to impliment QoS (Bandwidth Management) using the web interface of Alchemy.

I've been attempting to calculate the predictions from a poisson glm object, along these lines: predict(foo.glm, type = "response") and predict(foo.glm, type = "response", se.fit = TRUE) foo.glm is arrived at this way: foo.glm <- glm(Insects ~ Dad * Mum + Location, offset = log(MM), family = "poisson", data = model.df) There are two

I am getting (with older image): RFC3389 support incomplete. Turn off on client if possible How do I turn that off for the DG104s? Or if I can't how do I tweak asterisk? I see posts about ATA-186's having an audiomode, but the closet I came to was inbanddtmf. I tried =0 and =1, no effect. Thanks! -- Zot O'Connor <zot@zotconsulting.com> White Knight Hackers, Inc.

Hi Guys/Gals - I don't post here often but I read with interest all the postings. - I've been on a lot of mailing lists, but this one is by far the most interesting. I've been doing a lot of work with 'tftp' loading Cisco 79xx phones with firmware, configs. for asterisk, etc. And I see where a lot of folks have trouble with 'tftp', use alternate port numbers

Hi All, I'm looking at setting up a Asterisk system, and hope to use ATA 186's with it. Im in Australia, and am getting mixed answers to if its the I1 or I2 i need, does anyone have any experience with using ATA 186's in Australia Also, can anyone recommend a good place to obtain these locally? Cheers, Steven

I was doing a daily make update for asterisk. On the 19th the new version compiled fine. I installed it. All of my ata 186's can call out to pstn etc. All inbound calls, the phones ring but when you pickup, just silence both local and remote with no complaints in the cli. I backed down to the r 1.0 1.0.3 on the 20th CVS-v1-0-12/20/04-18:24:52 it worked ok. Yesterday I did a cvs update on the

I'm running a Linksys WRT54GL router from my CentOS-7 home server. Every now and then (maybe once every 2 days) the router's WiFi cuts out, and I've found no way to solve this except to disconnect the power from the router, wait 10 seconds and then re-connect. This always works. The router is running under dd-wrt. My question is - which makes it a tiny bit CentOS-related - does anyone

Hi, I'm having trouble getting caller*id to appear on my phone connected to an ATA186, and being called from Asterisk. Does anyone out there successfully see callerid on their ata186-connected phone? The "From:" header in the INVITE to the ATA seems to have the "right stuff" - eg From: "Study phone" <sip:6002@195.217.255.45:5062>;tag=as412db061 But

I have seen the list of codecs for the ATA 186's but not sure if it was 100% or not. I want to know really is it possible to run GSM or ilbc on them or is a G729 lic the only way to get a low bandwidth codec? This is the list of codecs that I have seen. RxCodec and TxCodec?Configure the codec ID. * G.723.1?Codec ID 0 * G.711a?Codec ID 1 * G.711u?codec ID 2 * G.729a?codec

Hello All, We are running Asterisk on a linux server as a SIP proxy with Cisco ATA 186's at the subscriber end. For long distance we have iax2 connectivity with a ip carrier. For local calls we are routing out through a commercial VEGA voicestream pots unit to an adtran channel bank and then from there to our class 5 soft switch. The sip to sip calls and the long distance calls work great.

Ahh Mr Carbuyer ... you should have _specified_ you wanted tires with that new car We can still help you though, it will just be an extra $$ above the price we quoted you I understand the concept. I see it in many industries until a company comes along that cares about it's customers I still think that digium is the best buy (for the small scale stuff that I'm interested in anyway) ...

Is the Sipura 3000 the only way to interface a remote pstn line and connect incoming calls to Asterisk? I have a location connected by network that has a phone line, when the room is occupied I want the line ti ring there as normal, but when the employee is travelling I want the line to be conencted to a ATA that then feeds it as an incoming pstn line to the pbx located at my office so it can