similar to: Reg. EMT-22 IP Phone

I would like the text plotted with 'mtext' to be alighned like it is for printing on the console. Here is what I have: > print(emt) ME RMSE MAE MPE MAPE MASE original -1.034568e+07 1.097695e+08 2.433160e+07 -31.30554 37.47713 1.5100050 xreg 1.561235e+01 2.008599e+03 9.089473e+02 267.05490 280.66734

Hello, everyone I have a piece of code that looks like this: mrets <- merge(mrets, BMM.SR=apply(mrets, 1, MyFunc, ret="BMM.AV120", stdev="BMM.SD120")) mrets <- merge(mrets, GM1.SR=apply(mrets, 1, MyFunc, ret="GM1.AV120", stdev="GM1.SD120")) mrets <- merge(mrets, IYC.SR=apply(mrets, 1, MyFunc, ret="IYC.AV120",

Hello, I have tried to install this phone for hours now and I can't get it working. Maybe someone can help me :) I have searched for more info from everywhere but there isn't much about 7910 :( >From the CLI I get this: NAME ADDRESS MAC Reg. State ================ =============== ================ ========== telefon --

In the process of migrating to Dovecot 1.0beta3, any tips or step-by-step instructions that might relate to Solaris 9. Thanks Bertrand Leboeuf leboeuf at emt.inrs.ca

Hi moises, Hi i need to done some modify/changes in one of the asterisk c source code ,,eg: app_meetme.c How can i compile and debug it without compile the whole module of asterisk... and also let me know which editor suitable it ,I'm using suse linux.. Plz help me reg. this .. -nsthi, -------------- next part -------------- An HTML attachment was scrubbed... URL:

Has anyone written an app that monitors SIP/IAX registration attempts? A couple of clients are being flooded with SIP registrations (but the source IP changes every few hours so IPtables won't do).. I would think that any attempt to reg 5 times with a bad password should cause a 5 minute timeout until reg is considered again. Has anyone written such an app? The name app_hackblock is my

Hi, ? sip.conf is missing in /etc/asterisk after installing the package AsteriskNow 1.5 Beta release? Is there any guide for FreePbx Administration? Thanks ? Explore and discover exciting holidays and getaways with Yahoo! India Travel http://in.travel.yahoo.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi, Is asterisk a registrar server. thanks, satish Confidentiality Notice The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or Mailadmin@wipro.com immediately and

Three months ago, I was experiencing all sorts of issues with my Asterisk box maintaining a connection to multiple trunks, etc. I also experienced various timing issues as well. In addition, Asterisk would sometimes take almost a minute to fully load and register its SIP and IAX trunks. Puzzled, I recompiled several times. No result. I checked my hardware. Didn't find anything. However, I did

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031208/0128c586/attachment.htm -------------- next part -------------- Thanks alot. I found the problem, with SIP firmware image was 2.* large size SIPDefault.cnf leads to Buffer overflow, I kept firmware_version field and deleted rest. That helped in upgrading the firmware to higher version.

Hi All; Asterisk is IP PBX, so there should be special IP Phones to be used with this IP PBX and this IP Phone should has special buttons working with Asterisk IP PBX, like: Line1, Line2, Line3, ... Line8 (these to be able to capture direct PSTN lines), also a button for Hold and transferring for another extension ... So can some one advise me which is the recommended IP Phone to be used with

How should I configure my asterisk server so that I can receive calls from an unregistered peer from whom I also receive registrations of sip phones? I'm asking you this, because with my actual configuration, when I register a contact from that peer's IP, no more inbound calls are accepted from that peer, as my asterisk rejects those INVITEs with "407 Proxy Authentication

Hi, ? Did anyone tried static build of asterisk 1.6 version? Installation fails when tried with static build. warning: Using 'initgroups' in statically linked applications requires at runtime the shared libraries from the glibc version used for linking asterisk.o: In function `cli_prompt': warning: Using 'getgrgid' in statically linked applications requires at runtime the

Dear all I have configure Asterisk 1.4 with sccp and configure Cisco phone 7975G model with Asterisk and it is working fine but i have one problem when i going to confance like when i call to someone and put it on hold and want to take in another New call to new person and want to take both live call in confrance how to do it in asterisk is there any configuration XML

Any recommendations on good wireless SIP phones? Thanks, Pedram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081105/6248fa14/attachment.htm

I have as large customer that would like to repalce all their Avaya PBXs with a OpenSer/Asterisk solution. Now the plan is to replace their 12,000 Avaya desk sets with Polycoms. My question is has anyone successfully used with OpenSer and or Asterisk, if so I would like to talk with you about hiring you to build this in our lab envirnment. Bob G. bobg at techie.com- --

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040106/eae682d3/attachment.htm -------------- next part -------------- Hello Whenever I receive voicemail on CISCO or SNOM or Analog Phones (Scitec), I would like to have some kind of indication in terms of beep sound or blinking voicemail indicator....Could you please tell me the way to enable

I would like some assistance getting a Polycom IP 500 working with Asterisk. I have dug through the mail-list archives and looked through wiki. I would like some more specific tips/tweaks to get the phone working with all possible features supported. Anyone willing to share their configuration files and advice? I will gladly add to the wiki pages what I have learned so that all can share in

Hello, all. I have some problem when testing my IP phone with Windows Messenger. My IP phone supports such codecs as g729, g711. And Windows Messenger supports red, g711, g723 as you know. The problem comes up when testing with this sip.conf file. ([general] context displayed only) =================================================================================== [general] port=5060

Hi all, Does anyone know if it's possible to continue talking to one member of an initiated conference call on Cisco 79xx ? In other words- disconnect one of the parties. Thanks. Assaf -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040821/525e26b0/attachment.htm