similar to: Transfer calls from cellphone

Displaying 20 results from an estimated 600 matches similar to: "Transfer calls from cellphone"

2006 May 04
1
Switchboard solutions, interactions with handset
Hi there, I'm looking into developing an in-house switchboard application. Does anyone here know of a way to control a hard-phone from such an application. For example, the attendant forwards a call with another one in queue. Once the first call has been forwarded (by keyboard shortcuts or dragging-n-dropping) - she presses a button (on the computer) to answer the waiting call. Now, if the
2006 Jan 05
3
TE110p and pri_cpe signalling not recognized
Hi guys, I've been installing and configuring a TE110p card. The compile and install went very well. I'm using this on FC4 and I compile with linux26 as well checked I on the udev configs. zttool and ztcfg both indicate that the card is ready. But when I try to "load chan_zap.so" then I get the following Unable to load module chan_zap.so Jan 5 21:43:33 ERROR[6808]:
2002 May 29
3
inverse gaussian random numbers
Dear R-people Does someone have a routine to ngenerate inverse-gaussian random numbers. I am thinking of something similar to rinvgauss, pinvgauss etc. in S-plus. best regards Helgi -- Helgi Tomasson FAX: 354-552-6806 University of Iceland PHONE:354-525-4571 Faculty of Economics and Business Administration
1998 Aug 31
3
Case sensitivity
I have had a bit of a problem getting consistency in filenames between NTFS and HPUX-Samba shares. Every time files are moved or copied to the samba-shares they convert all cases to lower. I have put the following in to smb.conf case sensitivity = yes preserve case = yes short preserve case = yes which I put in the global section and in to several of the shares sections also, but the
2006 Apr 13
13
Digium cards, so disappointing !
I am so fed up with Digium cards. My company first owned a TE410P, I installed it in a Dell server and "enjoyed" its instability (we bought it months before Digium warned about the incompatibility issues). Then we switched to a TE411P for the hardware echo cancellation. Now we want to receive fax (< 20/day) on it and guess what ? Since April 2006 (again a few months after we bought
2001 Apr 16
2
Dump utility?
Is there any dump utility that exists for vorbis streams? What I am intersted in is something that will do a break down like: How many bits are used for encoding _each_ codebooks, how many bits are used for the residue, how much is used for the lpc coefficints. A selective dump of the codebooks themselves would be nice too of course. I'm wondering if anyone has such a little dump utility.
2000 Apr 17
1
source-parse difference in 0.9 and 1.0
In version 0.9 I ran a command file of 489 lines with source(" ") In version 1.0 I do the same and get Error in parse(file, n, text, prompt) : unable to open file for parsing This has something to do with the length of the command file. If I paste the individual lines into R it is OK, if I only use few lines of the code is is OK as well. The problem is affected by
2004 Mar 31
5
3-4 port FXO card recommendations
*This message was transferred with a trial version of CommuniGate(tm) Pro* In setting up Asterisk, I'm looking to dump my current phone system (Nortel Venture). I presently have three POTS lines. I would use a VOIP provider, but now are presently available in the Toronto, ON, CANADA area that support user owned hardware/software. I need a 416/647 area code number. In looking at FXO cards
2004 Feb 03
2
Pictures of new multiport FXO/FXS from digum
Does the voicetronix card work with Asterisk? -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of woody+asterisk@solutionsfirst.com.au Sent: Monday, February 02, 2004 11:06 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Pictures of new multiport FXO/FXS from digum > -----Original Message-----
2008 Apr 05
2
IAX IP Phone
Hi All; Till now I am not able to find a good IAX IP Phone or Gateway that can be used with good quality. Anyone can advise for good one? Regards Bilal ____________________________________________________________________________________ You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost.
2012 Apr 11
4
Dahdi-2.4.0+2.4.0 means ??
Hi, can anyone tell me what does that 2.4.0+2.4.0 means in dahdi release numbering ??? 2.4.0????? regards upendra. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120411/9b160392/attachment.htm>
2009 Jan 15
2
[patch] libc Berkeley DB information leak
Hi, FreeBSD libc Berkeley DB can leak sensitive information to database files. The problem is that it writes uninitialized memory obtained from malloc(3) to database files. You can use this simple test program to reproduce the behavior: http://www.saunalahti.fi/~jh3/dbtest.c Run the program and see the resulting test.db file which will contain a sequence of 0xa5 bytes directly from malloc(3).
2005 Jul 25
3
Zap channel configuration problem
Hi, I would like to use a digum card to call an external number through my PSTN. I think that I have a problem in the configuration. Asterisk returns me app_dial.c:764 dial_exec: Unable to create channel of type 'Zap' I use Fedora core 3. I installed libpri, zaptel and asterisk. I plugged my line on the FXS module (green part). I make modprobe zaptel && modprobe wctdm without
2011 Mar 24
4
Issues with Digum Repos / AsteriskNOW & Bad Packages
I wish to use AsteriskNOW (the Digium repository + CentOS) with imap voicemail storage and Asterisk 1.4. After having installed AsteriskNOW with Asterisk 1.4 and Asterisk GUI I run the yum package manager and replace voicemail with imap voicemail and attempt to start Asterisk, however the voicemail module is not loaded: [Mar 23 23:30:03] WARNING[12592]: loader.c:382 load_dynamic_module: Error
2005 Feb 24
1
Which Codec(s) to use..?
Hey Everyone, I am playing around with my * box, and I have a few different phones hanging off it it right now. I have a Cisco 7960 capable of g729, ulaw and alaw, I have a Cisco ATA186 with a Panasonic cordless phone attached to it, I have a Digum IAXy with a dumb analog phone attached to it, and I have a Linksys PAP2-NA with an AT&T 959 analog phone attached to it. I also have several
2009 Mar 06
2
question about MeetMe performance.
hello, I will do a server to do a lots of conferences (MeetMe). I want to know that if I dont use a digum card, the limit of simultaneous calls is harder without a card than with a card ?if, yes, how harder is the limit? thank you Cordialement, BERGANZ Fran?ois P Pensez ? l'Environnement, n'imprimez ce mail que si n?cessaire. -------------- next part --------------
2003 Feb 05
2
big ps-files
To R-users I have been using R for many years and I am very happy with it. One thing puzzles me. Graphic postscript files tend to become quite big, much bigger than corresponding splus postscript files. Does anybody have a hint to avoid this? best regards Helgi -- Helgi Tomasson FAX: 354-552-6806 University of Iceland
2004 Jun 22
1
Copying and printing lattice graphics (PR#7004)
Full_Name: Olafur Arnar Ingolfsson Version: 1.9.1 OS: Win XP Submission from: (NULL) (213.236.225.194) Copying lattice graphics only works with copy as bitmap. Printing the graphic window doesn't work either. I have the same problem with 1.9.0 and 1.9.1, just updatet packages If I have done som (obvious) mistake, I do apologize. x.val <- rnorm(100,50,3) library(grid);library(lattice)
2005 Feb 02
4
new install
hi, i got an error while running the asterisk -v error message: error while writing audio data ===== R.B.Roa Traffic Management Engineer PhilCom Corporation __________________________________ Do you Yahoo!? Yahoo! Mail - You care about security. So do we. http://promotions.yahoo.com/new_mail
2009 Apr 24
3
timing source problem
hi all, we do have some troubles with zaptel timing source - we have a setup with 3 telco PRI lines, connected to asterisk digium card 0 - asterisk does some handling - calls are leaving on digium card 1 - going to a siemens hipath - there is some call handling - some of the calls then are going from the hipath over a qsig line to a bosch integral PBX - handling the rest of the calls. To be able