search for: mantas

...missing models or have made a mistake somewhere. It is also difficult to alter models if I want to change a term. For example, below are the set of models I would like to run. Is there a way to specify the full model and have R generate the rest? I.e. specify m1234567<-glm.convert(glm.nb(mantas~site*year+cosmonth+sinmonth+coslunar+sinlunar+plankton, data=mydata)) and have R run all the other models. library(MASS) #Intercept only m0<-glm.convert(glm.nb(mantas~1,data=mydata)) #One term - 7 models #Manta abundance is greater at one of the two sites m1<-glm.convert(glm.nb(ma...

Hello, I have a 4 port FXO digium card with 3 PSTNs attached to it and AsteriskAtHome setup. Everything is working fine except outbound calls. When I dial a outside number, it works fine, but when another employee trys to dial out while I am on a line, it will not go. I have a outgoing route setup in the AMP interface. Dial Pattern: 1NXXNXXXXXX NXXNXXXXXX NXXXXXX Trunk

...s.c:556 test-http-client-errors.c:636 test-http-server-errors.c:594 main.c:63 director.c:1445 director.c:1448 imap-client.c:253 director.c:1445 director.c:1448 mail-stats.c:56 Is this an old compiler issue or something else? Timo Sirainen <tss at iki.fi> ra??: > On 2 Feb 2017, at 19.38, Mantas Gegu?is <mantas.geguzis at ittc.vu.lt> wrote: >> >> Hello, >> >> I am tying to compile Dovecot 2.2.27 on Solaris 10, and I get this error: >> test-ioloop.c: In function `test_ioloop_pending_io': >> test-ioloop.c:188: error: size of array `type name'...

I'm getting this messages trying to call with one sip trunk: Feb 3 16:43:09 DEBUG[3389] channel.c: Avoiding initial deadlock for 'SIP/usa-e2ea' Feb 3 16:43:09 VERBOSE[3491] logger.c: -- SIP/usa-e2ea answered Zap/1-1 Feb 3 16:43:09 WARNING[3491] channel.c: No path to translate from Zap/1-1(68) to SIP/usa-e2ea(256) Feb 3 16:43:09 WARNING[3491] app_dial.c: Had to drop call

...cassandra Full text search : squat : -lucene -solr Last version that I have compiled was 2.2.24, version 2.2.25 failed with error: In file included from guid.c:6: sha1.h:80: error: static or type qualifiers in abstract declarator Is there anyone who can help me? -- Pagarbiai Mantas Gegu?is VU Informacini? technologij? taikymo centras tel. 8 5 236 6208

...irection, wind speed, temperature, barometric pressure), and tides (dummy variables for high, falling, low, and rising).  So predictors are binary, categorical, and continuous variables.  I ran glm.nb to fit my full model containing all first order terms.  The model I ran was: glm.nb.full<glm.nb(mantas~site+year+cosday+sinday+daylength+lunarpercent+sstmean+sststd+ sshmean+sshstd+cosdir+sindir+spd+temp+alt+tideht+high+falling+low+plankton)   However, when I use stepAIC on the model I get the message:   > stepAIC(glm.nb.full) Start:  AIC=19240.46 mantas ~ site + year + cosday + sinday + daylengt...

...ial model using the package AICcmodavg. I need to use glm() since the package does not accept glm.nb() models. I can get glm() to work if I first run glm.nb and take theta from that model, but is there a simpler way to estimate theta for the glm model? The two models are: mod.nb<-glm.nb(mantas~site,data=mydata) mod.glm<-glm(mantas~site,data=mydata, family=negative.binomial(mod.nb$theta)) How else can I get theta for the family=negative.binomial(theta=???) Thanks! Tim Tim Clark Department of Zoology University of Hawaii

Hello, Can you recommend a good billing software for asterisk that supports reseller accounts? Will be better if it haves opensource licence. Best regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : gsalas@manta.telconet.net www : http://www.manta.telconet.net

...rea$Manta, main = "Cummulative area of 100% MCP", xlab = "Data Point", ylab = "MCP Area", ylim = c(0,150), scales = list(tck = c(1, 0)), #Removes tics on top and r-axis pch=19,cex=.4, auto.key = list(title = "Mantas", x = .05, y=.95, corner = c(0,1),border = TRUE)) #Legend Thanks, Tim Tim Clark Department of Zoology University of Hawaii

Hi: Iam looking for a sip softphone that supports g729 codec Any one have an idea ? Reagrds; jonnyhashem --------------------------------- Don't get soaked. Take a quick peak at the forecast with theYahoo! Search weather shortcut. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I've one astribank with 8 FXO unit and 8 pstn lines connected to the astribank. When I receive calls on my ipphone I get always Unknown callerid. It's is possible to receive the callerid from the lines on the astribank unit? This is my config: [channels] language=es context=from-zaptel signalling=fxs_ks ;rxwink=300 usecallerid=yes callerid=asreceived ;cidsignalling=bell

Hi, Anyone is using LG ip phone LIP-100 with Asterisk. I've two of this phones but seems to work only with net2phone, in the product page http://isupport.lge.co.kr/html/ibu_lgic_modelView.jsp?jgrcode=D2_IPTP&modelid=M_IP100C the features are showing SIP and H.323 support. Can be used with my asterisk box? Best regards, -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq.

Hello All, I got one quick question on A2Billing. Specs: - - A2Billing v1.3 - OS CentOS 4.5 - Asterisk 1.2 - Zaptel 1.2 Did the installation and everything is working as it suppose to... Using the A2Billing documentation, I created the RateCard, SIP Trunks, and SIP Customers. I was also able to login using XLite Dialer and was able to call out to my SIP Trunk also. Now how can I remove the

Dear Sirs, we ... 1) buy minutes from other providers 2) sell minutes to out clients some calls terminate to our equipment, others - to h323 proxies. we want calls to be routed according to costs (a route is chosen from many by lowest cost). at the end of it, we'd like to bill our clients and see how much have we earned (money we receive from client on one side, money we pay to proxies on

Hi, Already posted once but I need more feedback. What kind of servers is everyone using for asterisk and what problems have you ran in to ? Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050630/dd52bf35/attachment.htm

It took a little longer then expected, but here it finally is, a field test for the idefisk for linux iax2 softphone. Freely downloadable from http://www.asteriskguru.com/tools/ You will probably need to copy the iaxclient lib into your library directory and run ldconfig before starting the phone. Please note that this is the first copy in the wild of the linux version and is not as tested

Hi, I've asterisk-oh323 0.7.3 and after 2 days of test finally I can make calls from one h.323 device to the world using sip trunks :) I can call to sip devices from the h.323 one. Now I want to make calls from sip to h.323 but it does not work. Maybe one of us have a configuration example to do this? I'm using the latest svn version (compiled yesterday).

...shapefile my.mcp<-mcp(xy, id=id, percent=100) spol<-area2spol(my.mcp) spdf <- SpatialPolygonsDataFrame(spol, data=data.frame +(getSpPPolygonsLabptSlots(spol), +row.names=getSpPPolygonsIDSlots(spol)), match.ID = TRUE) writeOGR(spdf,dsn=mcp.dir,layer="All Mantas MCP", driver="ESRI +Shapefile") #Read shapefile manipulated in qGIS mymcp<-readOGR(dsn=mcp.dir,layer="All mantas MCP land differenc") My spatial points data frame has a number of "Slot"s, including one that contained the original names called Slo...

Hi, I appreciate it if someone knows what is available for SIP web phones out there. I am interested in putting a soft phone on a website that registers with Asterisk using SIP. Then, when someone uses it, it directly calls into an asterisk call queue.. Any ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello group, I'm running asterisk @ home 1.5 - I would like to change these messages(call attend) to Spanish, how I can do that. Thanks, Nelson