Displaying 20 results from an estimated 2000 matches similar to: "linksys SPA-941"
2006 Mar 24
2
3Com Phones
Greetings,
We are looking at installing a VoIP system with Asterisk and are
currently looking at the line of 3Com phones. Has anybody had
success with using the following phones? We need to buy a lot and we
don't want to end up with phones that don't work properly with asterisk.
3Com 3101 (model with speakerphone)
3Com 3102 Business Phone
3Com 3103 Manager Phone
3Com 3105
2006 Dec 21
2
asterisk crashed
our * crashed twice in a month with segmentation fault &
a core dump. here's the stack trace:
#0 0xb7e11965 in mallopt () from /lib/tls/libc.so.6
#1 0xb7e10c43 in malloc () from /lib/tls/libc.so.6
#2 0xb7e17090 in strdup () from /lib/tls/libc.so.6
#3 0x08057ada in ast_verbose (fmt=0x0) at logger.c:879
#4 0xb7b29009 in moh_files_release (chan=0x9455ca0, data=0xb657be48) at
2005 Jun 07
1
connecting Asterisk to NEC NEAX system
hi. i connected Asterisk to an NEC NEAX system with a crossover T1 cable
and the Digium TE405P using E&M wink signaling. the connection's ok. however
when dialing from the NEC to the Asterisk. most of the time the Asterisk only
sees the first digit of the dialed number(which is 4 digits). some time if i
dialed the 4 digits very fast it might get through. seems like there's a timming
2004 Dec 09
5
Sipura SPA-841
Froogle found me one supplier for the SPA-841, not sure I trust them
though. Does this phone even exist yet? Does anyone have any
experience with it? Does anyone know a vendor other than
Atacomm/voipsupply?
2010 Jun 08
1
early media issue from phone co.
hi folks. i have the following puzzle:
when i call certain cell phone# using a regular phone & POTS.
the called cell phone co. usually return a message such as
phone travel out of range or phone is busy etc. if the phone is
unreachable. now when i have the following setup:
sip phone -> asterisk -> PRI -> phone co.
i call the same cell# and if it's unavailable. the PRI return
2005 Sep 28
3
cisco phones problems
hi folks.
we recently deployed 10 Cisco 7960G w/ SIP firmware 7.3 on our network and
we start having problems of dropping calls (actually the calls wasn't dropped
it just the sound was muted for about 5-10 seconds, but most users will think
the call dropped and hangup/redial). i've check the console output.
there was a lot of messages like the following:
Sep 28 15:00:49 NOTICE[8182]:
2006 Oct 23
2
Polycom SP4000 ftp problem
i recently bought an SP4000 conference phone but having problem
provisioning it using ftp, every time it just hangs at
"Updating initial configuration..." screen. when i switch it
to tftp, it'll work fine. i though it was bootrom/firmware issue
so i upgrade it to bootrom 3.2.2/sip 2.0.1 but it makes no
difference. any thoughts?
p.s. i'm using debian sarge proftpd 1.2.10 and the
2009 Nov 20
1
server unresponsive
hi folks.
we've experienced some weird problems lately. we have about 600
SIP phone on a single system running *1.4.26.2 for about a month.
recently there was massive UNREACHABLE messages like this one
showed up:
chan_sip.c: Peer '2699' is now UNREACHABLE! Last qualify: 1252
then they all became reachable again in a few seconds. sometimes
it last for couple minutes. but sometimes
2008 Apr 05
3
iaxmodem + hylafax w/ DID routing
hi folks.
i'm experimenting with iaxmodem + hylafax using DID to determine
where to send the fax to it's final destination. however i have
difficulties passing the DID information from iaxmodem to
hylafax.
in extensions.conf:
exten => _XXXX,1,Dial(IAX2/iaxmodem0/${EXTEN}|20|r)
exten => _XXXX,n,Dial(IAX2/iaxmodem1/${EXTEN}|20|r)
exten => _XXXX,n,Busy
exten => _XXXX,n,Hangup
2005 Sep 24
2
CDR problem
Hi to All,
I've an Asterisk CVS Head working with Mysql.
My problem is that instead of ANSWERED or something like, into the CDR
database records, I find only numbers.
This is also a problem to let ASTPP works, infact I receive an error:
ERROR - ERROR - ERROR - ERROR - ERROR
DISPOSITION NOT MATCHED
and the call has no cost.
Any suggestions?
Thanks
--
.:FaberK:.
2006 Nov 15
2
Question about TFTPD server
Hi all,
I have installed this package onto my Debian and placed the files i want the Cisco 7960 phone to get from the tftpdboot directory. But it doesn't seem to work. Are ther any special settings I should do to this server?
Many thanks for all your help,
Christian
2008 Mar 10
2
Global Variables on Reload
I'm running Asterisk 1.4.18 and having a problem with the
clearglobalvars option.
I have a NIGHT_SERVICE variable which I initially set equal to off. I
then have an extension they can dial which will toggle that variable. My
problem is when you enter the CLI and type "reload", it resets to "off"
again. I've tried setting the clearglobalvars=no as well as just
2010 Sep 09
2
DAHDI fxstest?
Greetings all-
During some recent testing and debugging, I wanted to use the 'fxstest' application. However, I found it hasn't been built when doing the standard 'make, make install' shtick with dahdi-linux-complete-2.3.0.1+2.3.0...
Can anyone tell me how to build fxstest?
Thanks!
--Tim
2011 Apr 24
1
Realtime and priority labels
In the following example
exten => _1NXXNXXXXXX,1,Set(GROUP(outbound)=myprovider)
exten => _1NXXNXXXXXX,n,Set(COUNT=${GROUP_COUNT(myprovider at outbound)})
exten => _1NXXNXXXXXX,n,NoOp(There are ${COUNT} calls for myprovider)
exten => _1NXXNXXXXXX,n,GotoIf($[ ${COUNT} > 2 ]?denied : continue)
exten => _1NXXNXXXXXX,n(denied),NoOp(There are too many calls up)
exten =>
2007 Jun 13
3
Using Modems with Asterisk
Has anyone had any experience using a modem through the Asterisk system?
I have some technical support personnel that need to use a computer
modem to connect to a remote system for troubleshooting. Is there a SIP
compliant gateway that will support a modem connection at decent speeds
(minimum of 28.8) that anyone knows of? If not, has anyone used a
Digium FXS card for this?
Thanks
2012 Nov 03
3
PRI got event HDLC Abort
hi folks.
recently some of our customers complained about bad voice
quality on the phone system. i looked at the logs and found
a lot of these:
[2012-11-03 08:26:38] NOTICE[11305] chan_dahdi.c: PRI got event: HDLC Abort (6) on
D-channel of span 1
[2012-11-03 08:26:45] NOTICE[11305] chan_dahdi.c: PRI got event: HDLC Abort (6) on
D-channel of span 1
[2012-11-03 08:26:54] NOTICE[11305]
2010 Oct 04
3
take input and store in variable
I am using a context to change values in a DB. Currently in my context, I
am passing it to
exten => s,1,WaitExten(7) ; 7 seconds to input
exten => s,n,Set(NEW_VAR=${EXTEN}) ;Here is my problem. This is the only
way I know how to 'grab' user input, which was normally from ${EXTEN} but
I realize this won't work for extension 's'......
The short google search I did
2009 Nov 06
2
Routing incoming call based on caller id
I am not that good at regex and it's use in Asterisk. I am running
Asterisk 1.4.13
Currently I have this in my extensions.conf for incoming calls on our
house phone line:
[housemenu]
exten => s,1,GotoIF($["${CALLERID(num)}" = "815xxxxxxx"]?s|12);
815xxxxxxx is our home phone number, when caller id fails or is missing
that is what is recorded.
I want to expand this
2010 Oct 23
5
a2billing muting "enter the phone number"
How can I mute the message "please enter the number you wish to call and
press the # key" in a2billing???
I tried
use_dnid = YES
but still I keep getting the message prompt...
thanks
2008 Mar 12
2
TXFax/RXFax/AGX-Addons/SpanDSP Crashing
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi all,
anyone else seen RX/TXFax crashing Asterisk on latest Asterisk SVN?
I've now seen it on two machines I tried to set up - one it seems
because the tiff file was malformed, but the other is doing:
tiff -> tx fax -> zaptel -> pstn -> ddi -> zaptel -> rx fax -> tiff
The above crashes every time.
If no one else has