search for: mantaing

Is there a function that will allow me to run all model iterations if I specify a full model? I am using information criteria to choose between possible candidate models. I have been writing out all possible model combinations by hand, and I am always worried that I am missing models or have made a mistake somewhere. It is also difficult to alter models if I want to change a term. For example,

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

Hello, thank You, this solution worked too. But had to do same thing for those files: test-http-client-errors.c:388 test-http-client-errors.c:484 test-http-client-errors.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?

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

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' is negative My configuration is like this: Install prefix . : /usr/local File offsets ... : 64bit I/O polling .... : poll I/O notifys .... : none SSL ............ : yes (OpenSSL) GSSAPI ......... : no

Dear List, I am having problems running stepAIC with a negative binomial regression model.  I am working with data on manta ray abundance, using 20 predictor variables.  Predictors include variables for location (site), time (year, cos and sin of calendar day, length of day, percent lunar illumination), oceanography (sea surface temp mean and std, sea surface height mean and std), weather (cos

Dear List, I am trying to do model averaging for a negative binomial 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)

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

Dear List, I am new to lattice plots, and am having problems with getting my plot to do what I want. Specifically: 1. I would like the legend to have the same symbols as the plot. I tried simpleKey but can't seem to get it to work with autoKey. Right now my plot has dots (pch=19) and my legend shows circles. 2. I have nine groups but xyplot seems to only be using seven colors, so two

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).

Dear List, I am analyzing the home range area of fish and seem to have lost the individuals ID names during my manipulations, and can't find out how to rename them. I calculated the MCP of the fish using mcp() in Adehabitat. MCP's were converted to spatial points data frame and exported to qGIS for manipulations. At this point the ID names were lost. I brought the manipulated

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