similar to: SipTone II

Displaying 20 results from an estimated 900 matches similar to: "SipTone II"

2004 Nov 23
1
Paul Mahlers Book
Anybody know of a UK supplier of "Voip Telephony with Asterisk" " by Paul Mahler ? I've searched on the web, and the only suppliers I can find are US based, and the postal charge is as much as the book. cheers -- Clive Email : clive.carter@sbcs.co.uk Alt : clivecarter@orange.net Tel : 0845 0043366 Alt : 01952 402032 SIP : 84416002@voiptalk.org Mobile : 07970 856261
2004 Nov 27
2
rtp compile error
Hi Just uploaded source from cvs (CVS-HEAD-11/27/04-12:56:51) Zaptel and libpri make install works ok, but I get the following error when running make install in asterisk directory rtp.c : in function 'ast_rtp_bridge': rtp.c : 1552 internal compiler error : Illegal instruction Please submit a full debug report ........... make *** [rpt.o] : Error 1 What have I done wrong ? (Its got to
2004 Nov 21
4
UK available SIP phone?
Hi, Anybody here from the UK using Asterisk at home? I'm looking for a SIP phone which will work with Asterisk and not leave me broke! I got one of the Tecom ones from Solwise but it refuses to login to Asterisk server for some reason. May have to send it back. What are the other options please? Thanks Mike
2003 Jun 27
10
Voicemail issue
Hi,. How can I make that Voicemail app to play only my own recorded message without the default one? Thanks, Dan
2004 Nov 23
2
-lssl
Hi Having my first go at compiling Asterisk from cvs source. Compiled and installed zaptel ok Running make asterisk returns the following error message /usr/bin/ld cannot find -lssl collect2: ld returned 1 exit status The last part of the compile messages on screen are- editline/libedit.a db1.ast/libdb1.a stdtime/libtime.a -ldl -lncurses -lm -lresolv -lssl There is obviously something I have
2005 Jan 10
6
UK * group
Is there a UK Asterisk users group? Would be interested in contacting others in the UK who use asterisk for either home or business applications. If there is, could someone provide me with some contact details, else anyone who's also interested, contact me off list. Cheers, Ben Merrills -------------- next part -------------- An HTML attachment was scrubbed... URL:
2004 Jul 20
2
No Ringing.
Dear Asterisk Group. I have two Asterisk servers serving two data/help desk centers, both centers have a near identical setup. However, when connected to one of my data centers, I call a user, I can see on the CLI that the phone is ringing, but I hear no ringing on my SIP soft phone? Has anyone had a similar scenario? How as it resolved. Warm Regards Shad Mortazavi
2005 Feb 10
12
asterisk@home scary log
Hi everybody, I'm testing asterisk@home 0.4, looks great so far I was working when I have been alerted by a bip comming from the * pc... I connected a screen to it and saw that there was a message which looked like : Message from syslogd@asterisk1 at Thu Feb 10 09:01:00 2005 ... asterisk1 so I stopped asterisk, type mail and got a strange mail saying that user xxxx@yahoo.com could
2004 Jul 28
0
SipTone 4 Sale...
Hey Folks, I'm selling my SipTone on eBay... starting at $100, 17 hours left. It's been modified (the firmware) so that you are able to telnet into it and possibly (thanks to cross compiling) run your own software on it. http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&rd=1&item=5711945656&ssPageName=STRK:MESE:IT Just so this post doesn't seem all to be about selling it
2004 May 25
1
SipTone II and Choppy/Stuttering Audio
Hi All, * is running a dream now, however we have an odd problem that I am sure some guru will be able to sort out for me in no time!! When receiving or making a call about 60 seconds or so into the call we develop choppy/stutter audio problems. It then seems to clear itself only to return again, and so the pattern carries on! This has got me stumped! Our equipment is SipTone II handsets, AVM
2003 Apr 02
1
FW: ipDialog Ethernet SIP Phone $199
Here is a SIP phone I haven't seen before. Does anyone have any experience with this one? -----Original Message----- From: George Richardson [mailto:georger@netxusa.com] Sent: Wednesday, April 02, 2003 4:56 PM To: clay@ctitec.com Subject: ipDialog Ethernet SIP Phone $199 pad <http://us.st1.yimg.com/store1.yimg.com/Img/trans_1x1.gif>
2004 Jun 16
4
UIP200
Hi, We've recently deployed 6 Uniden UIP200 phones (running firmware 4.54). We've been having some serious problems: 1) All the phones randomly reboot themselves. Typically when trying to answer or initiate a call. 2) All the phones will disconnect from a calls with the PSTN after 2-3 minutes. 3) The phones are unable to interact with a remote IVR (digit presses are not received at
2003 Oct 15
4
SIP Telephone Quality/Price
Hi! I am doing a research about the prices of SIP telephones. If someone can tell me which one are the cheapest and have an acceptable quality... it will be very kind. Best Regards, Mireia
2003 Aug 12
12
IP phone recommendation
Hello, I would like to buy a SIP IP phone, but I don't know wich one to choose... Can you tell me wich IP Phone is known to work with Asterisk please. I've seen the Cisco 7940, but I don't know if it works, and how expensive is it ? I'm french, so if you know some french resellers, tell me. Thanks a lot, ---------------------- Fabrice Tereszkiewicz Sawadka.org
2005 Jul 10
2
chan_capi ASTCC trouble
Hi all I am wondering if anyone has had a similar trouble to this: The timeout arguments in the dial command does not work. The caller does not get disconnected when the timeout reaches zero. I am not sure if this is a chan_capi issue, or a asterisk issue. I am using CVS-head and chan_capi CVS head also. Any suggestions or help will be appreciated. Thanks Clive
2012 Jul 24
1
Patchy 'front-end' package installation problems using -R- 2.15.1
I think this is the fourth attempt to send this blessed message, so let's hope this gets through without any 'unprocessed' or 'ignored' in-lines on auto-reply. I wish to report to you some strange problems I'm experiencing with installing packages directly into my -R- 2.15.1 (there is an indirect solution, which I note below). First, here's some essential information:
2005 Sep 20
6
iax2 trunking wackyness
Hi I was doing some bandwidth testing, and my incomming usage is 36% more than my outgoing bandwidth. The setup is IAX2 trunking using GSM codec. Is there any obvious reason I am overlooking to figure out why there is such a big difference between the two.? I am using CVS-head September 3rd, maybe there is a version skew? Any suggestions will be appreciated. Thanks Clive
1998 Mar 17
0
R-beta: locfit -> CRAN
The locfit library is now available through CRAN, in the Contributed R Code directory. Locfit fits local regression, likelihood and density estimation models, in the spirit of loess but with many additional features. To install, unpack the locfit_19980309.tar.gz file, and R INSTALL locfit Most of the functionality and examples on my home page http://cm.bell-labs.com/stat/project/locfit/ should
1998 Mar 17
0
R-beta: locfit -> CRAN
The locfit library is now available through CRAN, in the Contributed R Code directory. Locfit fits local regression, likelihood and density estimation models, in the spirit of loess but with many additional features. To install, unpack the locfit_19980309.tar.gz file, and R INSTALL locfit Most of the functionality and examples on my home page http://cm.bell-labs.com/stat/project/locfit/ should
2005 Jan 11
2
ASTCC - error on call end
Hi I get an error popping up when the call ends as follows: DBD::mysql::db do failed: Unknown column 'callstart' in 'field list' at /var/lib/asterisk/agi-bin/astcc.agi line 90, <STDIN> line 32. Does anyone else get this same thing? Looks as if my database table is wrong, or something else is up...not sure Thanks Clive