I bought a couple Polycom Soundpoint 300's, and have them working nicely
with SIP... but I'd like to be able to do automatic config via FTP, but it
requires some XML config files. The docs discuss them in detail, but I
can't seem to d/l them from Polycom. [No, it doesn't appear to be on the
CD
I can't figure out how one upgrades the boot ROMs for the Polycoms.
I've read the wiki, I've checked the docs, I've looked here, there,
everywhere; I hoped it might be through the web interface, through
DHCP... but I can't see how to do it.
Suggestions?
Thanks!
-Ken
Put the new bootrom.ld and bootrom.ver files on the config server (FTP
or TFTP) that your phones load from, and they will upgrade automatically.
Easy. Too easy. ;-) Seriously, though: I'd've never thought of that.
Thanks much!
-Ken
___
I'm RTFM'ing, but I can't figure out how the dhcpd.conf file specifies the
boot server, and how it differentiates between whether it's FTP or
TFTP. I've
tried option 66/next-server, and option 150, to no avail. And the docs
just don't -- leastwise, in the way I'm reading them -- make sense.
Interesting... I'll stick with FTP anyway, since I can partially secure
it, and it works across NAT :-)
Kevin: I don't think Matt made himself quite clear enough -- going with
his dhcpd.conf setup works for FTP, regardless of what the option name
is; it's working great! And I don't even have
I'm very, very confused. Dialing out, through VoicePulse, with both gsm
and ulaw CODECs, most of my calls are great. However, calling my
(non-Asterisk) voicemail at my job, and calling my cell phone both
produce horrendous (~ 1/3-second delay) echo. I've tried with different
phones (Polycom
I've got a 400P, with a couple FXO's and and FXS. I've got an analog
plugged into the FXS, and it gets dialtone fine. However, whenever I
press any digits, I get the doo-dah, doo-dah unhappy sound. I've got a
functioning FXS system at home, but I was trying to plug this into an
AMP-created
Argh. I can't figure out what I'm doing wrong. I can dial with my SIP
phones just fine, but I want to set up an analog phone plugged into my FXS
port... and, while it gets dialtone, no matter what digit I press, I get
stuff like:
VERBOSE[21963]: -- Starting simple switch on 'Zap/1-1'
I know the reasons I like Asterisk -- and that's all well and good. But
if I'm bidding against (say) Cisco, money aside, what should my argue
points be? Is there a compare-and-contrast Asterisk-vs-the-other-guys
page out there somewhere?
Just curious,
-Ken
Olson, Dana wrote:
I have this Aastra 480i phone, and you can set the TFTP server IP
address manually in the phone, but there should be a way to have it
find the TFTP server information via DHCP. Does anyone know if this is
possible, and if so, what is the attribute I have to set on my DHCP
I'd dearly love to be able to give an Asterisk demo by just toting my
notebook, a PC/PCMCIA card, and a couple SIP phones. Is there any way
to do this? Or should I look for a small-profile box with PCI slots,
instead?
___
Asterisk-Users mailing list
Okay, I'm stumped. When I call the PSTN (through POTS lines), my analog
phone phone works fine. My SIP phones -- a Grandstream and a Polycom --
have major echo; roughly a .25 second delay. Eventually, it goes away,
which I guess is echo cancellation in action. But, dammit, why does my
I've got all three CODECs the 300 supports -- G.711u, G.177A, and
G.729AB -- enabled, I've changed the order, I've got them all in allow
lines in my sip.conf, as follows:
disallow=all
allow=ulaw
allow=alaw
allow=G729
From sip debug I get the following snippets:
Howdy! I'm VERY interested in your HOWTO... but the link you have,
below, times out. Any chance you could mail me the HOWTO, or point me
to a new link?
Thanks much!
-Ken
[EMAIL PROTECTED] wrote:
I've created a pretty complete HOWTO on creating a Linux Bridge (using
Fedora) to shape LAN --
files for both; I've tried
enabling/disabling ULAW, ALAW and G279 on the Soundpoint, to no avail.
Plug 1.0.0 back in -- works like a champ.
To say that I'm confused would be understating things rather severely.
Thanks much,
Ken D'Ambrosio
___
Asterisk
To say that I'm confused would be understating things rather severely.
Kevin P. Fleming wrote:
To say that we can't help you without seeing your config files would
also be an understatement. Unfortunately, we are not all-knowing nor
telepathic, so just saying it doesn't work won't generate
I could probably play with, but I'd like to be sure
it'll... well, you know: work.]
Thanks,
Ken D'Ambrosio
Sr. SysAdmin,
Xanoptix, Inc.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
, that's readily findable).
Thanks much,
Ken D'Ambrosio
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo
, maybe I'm just dumb, but I'm
thinking that there's an admin manual I didn't get. Anyone have any
pointers? Either web-wise, or simple stuff like the password for the
unlock config option?
Thanks much...
Ken D'Ambrosio
___
Asterisk-Users mailing list
Hi, all. I've got my Uniden UIP200 configured via TFTP (had to get DHCP
3.0.1 -- Debian's latest is 2.0.x!), and all seems well... except for the
minor detail that it doesn't work. It registers fine with Asterisk, but
when I copied my Grandstream's sip.conf info and plugged in the Uniden
Ryan Courtnage wrote:
If it registers fine with Asterisk, then what exactly is the problem?
Does the Uniden phone display an error? Asterisk? Can you make/receive
calls?
The firmware version and unidenmac.txt might also be relevant to the
problem.
Jeepers. You want a description of the
R A wrote:
Then
what do you think i have to do?
i install a sniffer and the phone make an ARP request
to 67.153.142.69.
the phone is 192.168.0.160 and i set my pc to
192.168.0.161 and i can't ping the phone.
Get a new phone. :(
What he's saying -- and I agree with him -- is that, if you
Hi! I've got a Comdial PBX that I would dearly love to replace with an
Asterisk box. However, for various reasons, it appears not to be in the
cards. Regardless of what management does, or does not, want, our
current VM solution -- some Dialogic card with a KeyVoice application
-- is dying.
Perhaps 90% of my calls -- over a Uniden and a Grandstream -- are fine.
The other 10% get some nasty echoes. Is there some magic something I
should be tweaking? I kind of thought that echo cancellation was
static, rather than dynamic, and that it's difficult to be able to cope
with something
I -think- that the Sipura 841 was PoE... and I'd been anxious to find out.
However, according to Atacomm.com, it's been delayed until mid-January.
*sigh* So: does anyone know of a (decent) phone that meets the following
criteria, and isn't too expensive?
- SIP
- two (or more) lines
- some form
(and maybe even inbound
;-) calls, so I can get something working, and build from there. Any
suggestions? If it means I need to go over docs I've already read,
that's fine, but I'm pretty confused right now...
Thanks,
- Ken D'Ambrosio
___
Asterisk-Users
!
-Ken D'Ambrosio
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
When I dial out, from both my analog and my SIP phones, it fails roughly
80% of the time with various telco messages (eg., The number you have
dialed...). The messages take a good 30 seconds before I hear them,
which makes me think the telco isn't seeing enough digits. 20% of the
time, my calls
I got dragged away from Asterisk (somebody made me an offer I couldn't
refuse for system administration), but I'm thinking about seeing if I
can't get it deployed at my new employer. Regardless, there are two
things about older voicemail that used to annoy me:
- Dial by name. Has anyone made it
On Mon, September 17, 2007 7:21 pm, Matt Riddell wrote:
In the past, you could help someone sort a problem, only for the config
files to be overwritten the next time the user did something in the GUI.
Are there any Asterisk GUIs out there that actually parse the data files,
themselves, instead
Hi, all. My company is setting up a branch office in Germany, and I'm
very interested in a VoIP provider over thataway. However, I'd need a few
things:
- Reliability. Can't have my branch office's DID's just going down. A
company with a proven track record would be very, very good.
- English.
Hey, all. I've got a non-PRI T1 that doesn't do DID correctly: I can't
get the DID from the proper variables, and, instead, I direct it based on
the four least valuable DTMF digits dialed by the T1 for in-bound calls.
Which really works pretty well; Asterisk plugs them quite nicely into
Andrew Furey wrote:
Huh? My 7905 takes well under 10 seconds, including Asterisk
registration and NTP update. Granted, if it were DHCP it might take
marginally longer, but 5 _minutes_?
Yeah, the Polycoms *do* take a while to boot -- but not five minutes.
I've timed mine (Polycom 501's) and
Hi, all -- some (but not all) of the faxes I receive on my Sangoma card
have an offset, where the right 3/4 of the fax are shifted left, and the
remaining portion is pasted on to the right edge; see
jots.org/~ken/fax.jpg for an example.
Anyone have any ideas on how to get this working? Since I'm
Hi! For reasons that I won't bore people with, I'd like to disable echo
cancellation on-the-fly, depending on which DID a call came in on. I've
seen things like spandsp disable EC for faxes, so I know it's possible.
Any idea where to start looking? (I assume I'll have to make a helper
I've some DID's that I'm using for in-bound faxing, but I'm having some
trouble with getting that working perfectly on my T1. So I'm thinking of
pointing them to an analog line. Will the DID's simply come in over the
analog, presumably sending the DID digits via DTMF? Or is that not
something
If I do a supervised forward on a call (Polycom 501, Asterisk 1.2.1), all
goes well until the second time I hit forward (to join the caller with the
extension); then, the caller's MoH goes away (making them think they've
been hung up on), and the server spits out:
asterisk-cw*CLI
-- SIP read from
I had the /exact/ same problem. Turns out it's the FTP server; in the
docs, there are several FTP servers specified as being compatible;
proftp is the one I went with, and it fixed it right up. (Note that I
was using the default Debian FTP server when it was rebooting, so it's
not just a 'doze
-- as the .sample file suggests -- appears not to
work for Polycoms.]
-Ken
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ken D'Ambrosio
Sent: Tuesday, January 31, 2006 11:20 AM
To: asterisk-users@lists.digium.com
Cc: [EMAIL PROTECTED]
Subject
From your description, it sounds as if the SIP phones are local to the
Asterisk box. If this is so, having nat=yes might be a problem.
-Ken
sdgesa gaeharth wrote:
please help!!!
I am dialing into our asterisk server(TDM400p) from the psnt. I hear
our voicemail message and I press the
Hi! I've got MWI working just fine for my 501, but it's on if I have
-any- VM messages. I only want it on if there are *new* messages. Any
ideas as to what I should be changing?
Thanks!
-Ken
___
--Bandwidth and Colocation provided by Easynews.com --
Anthony Rodgers wrote:
Hi Ken,
When you say -any-, what do you mean? Messages in the Old folder, or
what?
Precisely. If there are messages in the Old folder, the MWI still
blinks. (I suppose I should've been more explicit; apologies...)
-Ken
___
For the record, I've done a couple of Asterisk installs, and HATED echo
-- or feebly attempting to get Asterisk's flakey software algorithms to
do anything about it. Finally got sick 'n tired, and threw money at it
-- got the Sangoma quad-span T1 card.
And echo freaking VANISHED. (Note that,
Rich Adamson wrote:
More than that, in their fine print some only claim to pass maybe two or
three of the tests. There is nothing that defines what you must achieve
before you can claim G.168-2002 compliance.
Well, isn't that just wonderful :-) Standards are amazing things, from a
I just tried to go from CAS to PRI on my T1 (Sangoma), and failed pretty
badly. Seemingly everything worked -- Asterisk would see the incoming
call (including CID and DID info), try to route it, and fail -- giving
me a telco (not Asterisk) call failure message. My zapata.conf and
zaptel.conf
Michael Graves wrote:
Does anyone on-list have direct experience with the new analog cards
from Sangoma? I'm thinking about FXOs with echo cans. Need 2-4 ports
but don't want to go through another TDM400 style experience.
First impressions (of which one should always be wary):
1) I really,
]
[mailto:[EMAIL PROTECTED] On Behalf Of Ken
D'Ambrosio
Sent: Thursday, February 16, 2006 1:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: [EMAIL PROTECTED]
Subject: [Asterisk-Users] No D-channels available!
I just tried to go from CAS to PRI on my T1 (Sangoma), and failed
Hi, all. I've got a T1 through Choice One Communications
(www.choiceonecom.com), a provider in the US northeast. I recently tried
to switch to ISDN on it -- and failed miserably. I've posted my config
files, and nobody's seen anything obviously wrong. Has anyone else used
their ISDN T1's? If
Hi, all. I've just had my T1 re-provisioned to ISDN. Everything comes up
and seems to work fine, with the minor detail that there is no audio
whatsoever.
So: voice prompts are played, caller ID and DID information is seen and
acted on, etc., etc., etc., but at no point is any audio heard on
I'd like to set up a sort-of follow-me: on a call to a given extension,
I'd like to simultaneously call several different numbers, play them all a
prompt upon answering, and monitor for DTMF digit 1. I know how to get
Dial() to dial multiple numbers, and I know how to play prompts and
monitor for
On Mon, March 6, 2006 4:19 pm, [EMAIL PROTECTED] wrote:
I have installed several hundred polycom's, and I have never seen a
500/501
with a power jack. All with the inline cable, as you mention.
Of course, if someone can provide photo evidence I will stand corrected.
I think the confusion
HTTP's nice, but FTP does the job. Check the docs for supported FTP
servers -- many of the stock Linux FTP servers will give the exact problem
you discussed, below. I should know -- took me almost a week before
trying proftpd, and WHAMMO, worked like a champ.
-Ken
On Tue, March 7, 2006 12:37
Hi, all. I see that the Polycom SoundPoint IP 330 supports VLAN... but I
don't quite see how that works. Do you point a non-VLAN'd segment at it
(akin to when you uplink a VLAN_enabled switch), and have the phone
implement the VLAN? Or...? *puzzled*
Thanks much,
-Ken
Hi, all. I'm trying to do some rudimentary testing of an Asterisk system,
but, for various reasons, I have to do this covertly, which means I'm
paying out-of-pocket. So I'm looking for somewhere that will do *cheap*
SIP and/or IAX termination, preferably with at least two simultaneous
calls, and
I used to do lots of Asterisk, but got an offer I couldn't refuse, and
went SysAdmin. Well, now I'm trying to bring Asterisk in-house, and want
to set up a test system. One thing I'd really like to get my hands on is
recent firmware, etc., for SoundPoint IP 430's. Freedomphones.net, my old
Hi, all. I'm doing an upgrade from an [EMAIL PROTECTED] (Asterisk 1.x) system
to stock Asterisk 1.4. Everything's working great, except that all the
prompts (both stock system prompts on the new system and people's old
recorded VM prompts) sound HORRIBLE. Call quality is great, both internal
I badly want to roll out Asterisk at my job. Unfortunately, my boss is
dazzled by shiny objects. We had a vendor in today who showed us their
system which, honestly, didn't suck -- but boy, is it going to be
expensive! One major component of the eye candy was an end-user interface
that allowed
http://www.youtube.com/user/voiceroute
Ming
On 8/7/08, Ken D'Ambrosio [EMAIL PROTECTED] wrote:
I badly want to roll out Asterisk at my job. Unfortunately, my boss is
dazzled by shiny objects. We had a vendor in today who showed us their
system which, honestly, didn't suck -- but boy
Hi, all. I have a Sangoma A104D (on-board, DSP-based echo can); I'm
currently passing through some of my in-bound calls to a legacy PBX (which
I hope to eventually replace). That being said, until I do, I'd like to
kill echo cancellation for the passed-through calls -- I don't want to
mess with
Hey, all. I haven't really gotten deep into Asterisk since 1.0.x, and I'm
afraid I've forgotten a fair bit. One big thing that I've forgotten is
the syntax, etc., for extensions.conf. Where do I find that? I'm looking
for stuff about commands, syntax for commands, variables, etc. Is there a
Hi, all. We've got a PoS legacy PBX at my company that doesn't have call
accounting. I figured, Hey, why not stick a dual-span T1 Asterisk-based
system in the middle? Then, I just passively pass in-bound calls to the
PBX, and outbound calls to the PSTN. I can then have Asterisk do all the
call
You have handsets connected to your proprietary PBX. Most domestic
things you dial on your proprietary PBX handsets get passed directly
through to your asterisk box without getting mangled by your
proprietary PBX. International calls that are prefixed by 011 are
getting mangled by your
the provider may be tagging it on. have you checked pridialplan, or
prilocaldialplan settings and playing around with that in zapata.conf ?
Oooh. That makes sense. I've poked around, but don't really see much
documentation on this. 'Cause going outbound is easy, but how do I check
to see if
Hey, all. We're rolling out VoIP, and I'm wondering about PoE
recommendations, as we're going to have to replace our current network
equipment. My first inclination would be to just plunk down the cash and
do a Cisco system, but I'm relatively certain that would get shot down by
finance. Any
Hi, all. This e-mail is a follow-up to an exchange I had several weeks
ago. I've got an Asterisk box with a dual-span T1 card. I want to place
it between the PSTN and my company's legacy PBX. I actually did do that,
but international calls from the legacy PBX were having the 011 stripped
off
Hi, all. I've got a customer who's complaining of low volume, especially
for conference calls. If this were a Zap system, I'd just bump up txgain
in their zaptel.conf file... but it isn't. Should I crank the volume of
the phones (they're Polycoms), or is there some other, more graceful,
Hi, all. I just upgraded my sip.cfg for my Polycoms, and that damn warble
on registration(? -- maybe it's on acquiring an IP?) has started again.
I still have the old sip.cfg, but can't figure out which option it is.
Any help?
Thanks!
-Ken D'Ambrosio
--
This message has been scanned
Hey, all. When I last was heavily into Asterisk (1.0.x), setting up to
receive faxes was, well, a PITA, what with having to patch the Asterisk
install with various driver patches and this, that, and the other.
Is that still true? Is there a fax HOWTO out there that reflects Asterisk
1.4.x?
Hi, all. I just tried to fire up app_txfax and app_rxfax, only to find
that I can't seem to compile them. The problem appears to be that my
libtiff library is wrong. Only problem is that, according to the README,
I need libtiff =3.8 and 4.0, which is all well and good... except that
there is no
What version of spandsp do you use?
Based on the fact that you asked that question, I suddenly got suspicious
that, despite his warnings, it might have worked for you with libtiff-4.
So I went and re-tried (using spandsp 0.0.4-pre16), and it failed
*differently*. So then I got suspicious that
For no reason other than it would be cool, I'd like to be able to dial an
extension and have it play a random MP3. Since, however, MP3s are
kinda-sorta weird due to patents, I'm not sure what the right approach for
this is. Any pointers on how to go about this?
Thanks!
-Ken
Hi, all. I've got app_rxfax going and nicely receiving a fax, which I
then throw to a script, and have it convert it to a PDF and mail it.
Works great... a lot of the time. But a fair bit of the time, rxfax
throws errors, and extensions.conf seems never to invoke my script. Here
are the
Hi, all. I want to execute a script, and return the value of said
(Python) script to the dialplan. I thought something like
exten = 1,1,Set(MyWorkingDir=System(/bin/pwd))
might work, but apparently not. I also looked into AGI stuff, but that
doesn't quite seem to be the right approach.
Hi, all. I'm getting a lot of
[Feb 3 13:56:36] WARNING[3721]
/usr/src/asterisk/spandsp/agx-ast-addons/app_rxfax.c: Channel T30 DONE
0.
in my log file, and incomplete fax reception. Any idea what might be
going on? I've googled a fair bit, but haven't seen anything leap out at
me.
Thanks,
Hi, all. My subject line says it all: is there a WiFi SIP phone with VPN
abilities? Failing that, a WiFi phone that runs Linux? I already know
one phone that does meet my requirements -- the iPhone. The new software
comes with a Cisco VPN client, and a SIP client can be had from
third-party
Hey, all. I was going through a make configure on my Asterisk 1.4.23
Ubuntu box, and noticed something I'd forgotten: Asterisk now supports
IMAP_STORAGE. However, when I highlight it, it tells me that there's an
unmet dependency, presumably for imap_tk. I've apt-get installed
everything I can
Asterisk segfaulted on me the other day; how do I tell it to generate a
core file so -- if it happens again -- I can attempt to debug? I looked
in the obvious places in make menuconfig and didn't see anything
appropriate.
Thanks,
-Ken
--
This message has been scanned for viruses and
Hey, all. I was just wondering if there were any
tools/utilities/what-have-you out there that would allow a user to click
on a contact in Outlook, and have their phone dial it? (Or, I guess, have
Asterisk dial both their phone and the destination number, and put the two
into a conference.)
Idle curiosity: I like the look and feel of the Grandstreams, but it's
been my experience that the speakerphones suck (esp. when compared to the
pretty damn flawless Polycoms). I've used the BT-100/101 and GS-2000;
have any of their newer models changed that?
Thanks!
-Ken
--
This message has
Hey, all. I'm all over MWI, but I gotta say that I think the Polycoms go
a bit over the top. The blinking LED is enough for me; how do I disable
the stuttered dialtone and the audible warble? I've looked through the
config files, but there are a HELL of a lot of options, and I haven't been
able
Hi, all. My autoattendant looks like this:
exten = s,1,Answer()
exten = s,n,Background(corporate-greeting)
exten = s,n,Set(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds
exten = s,n,Set(TIMEOUT(response)=10) ; Set Response Timeout to 10 seconds
exten = s,n,WaitExten(30)
When the call
Hi, all. I've got an old Telrad PBX with an Emagen(?) voicemail box. The
VM box, itself, is beginning to show its age. Big-time. We're thinking it
might be time to look for a replacement. I'd love to install Asterisk
with an FXO card or something, but I don't think it supports whatever
Make a call to VM (has to go out on the port you have the handset plugged
into), answer it and listen.
If you hear a bunch of DTMF then you are golden.
Sounds like good stuff, but my most substantial concerns involved things
like MWI: is asterisk able to push that back to the PBX?
--
Wow. Thanks for all the replies! Something just occurred to me, though:
which side would be FXO, and which side would be FXS? The PBX? Or the
Asterisk/VM side?
Thanks again for all the info!
-Ken
On Wed, July 1, 2009 3:36 pm, Jared Smith wrote:
On Wed, 2009-07-01 at 13:05 -0400, Ken
Hi, all -- since the new 2.x firmware seems to support NAT -- and, since
I'm not an authorized dealer -- I'm kind of wondering if anyone knows
where I can get it. freedomphones.net/polycom/files/ only goes up to
1.6.7. If anyone can either mail it to me, or mail me a link, I'd
certainly be
I've set up a bunch of plain-jane Asterisk systems, but had heard good
things about the more recent incarnations of [EMAIL PROTECTED] errr, Trixbox.
So I installed it, and fired it up, and it works fine.
Until I try to do an asterisk -r. I get the does /var/run/asterisk.ctl
exist? question,
Just wondering if there are any IAX phones worthy of the name phone out
there -- looking for hard phones, but I suppose a Linux-based softphone
wouldn't, you know, hurt. ;-)
Thanks!
-Ken
___
--Bandwidth and Colocation provided by Easynews.com --
where to start with something like
this, but I have to imagine it's been done before.
Thanks!
Ken D'Ambrosio
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update
wed
that the FTP transactions were being executed properly, but the phone
wasn't responding correctly. It was only when I went with ProFTPd that
things got better -- for me, at least. ;-) YMMV, etc.
Doug.
-Original Message-
From: Ken D'Ambrosio [mailto:[EMAIL PROTECTED]]
Sent: Tuesday, Ma
Hey, all. I'm trying to set my CID such that, internally, I see a
four-digit extension (which is also handy when checking VM), but
externally, I see the full 10-digit number. So I plugged these lines into
my extensions.conf:
exten = _XXX,1,GotoIf($[ ${CALLERIDNUM} != 1625]?4:2)
exten =
Hi, all. I want to have a button on my receptionist's 601 that, when
pressed, will forward her current call to a given extension. Is there any
way to do that? I've tried to RTFM, but I'm coming up empty.
Thanks,
-Ken D'Ambrosio
___
--Bandwidth
Hi, all. Every now and then, some of my users get Error on their
phones. A reboot fixes it, but it's quite annoying/inconvenient. I'm
running Asterisk 1.2.4, and have the following firmware, etc.:
Bootrom: 2.6.2.0032
BootBlock: 2.5.0(11500_030)
SIP application: 1.6.2.0041
Any ideas as to why
screws stuff up).
Any ideas?
Thanks,
-Ken D'Ambrosio
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Hey, all. I've got a client who's interested in possibly using a
softphone for his receptionists. While I've certainly used some
softphones for single extensions, I'm not sure which one I'd suggest for a
receptionist.
Any favorites?
Thanks,
-Ken
with timestamp, it mentioned either a line number, or -- more likely -- a
context/extension/priority triplet.
Is there anything like that?
Thanks,
Ken D'Ambrosio
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
Hi! I've got two POTS lines coming in to my * box, but I only want the
primary of the two lines available for outbound dialing. I can't quite
figure out how to make that happen. Suggestions?
Thanks,
-Ken D'Ambrosio
___
--Bandwidth and Colocation
to.
Suggestions?
Thanks!
-Ken D'Ambrosio
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
I've got an account that's looking at doing some cable/VoIP
integration. They were wondering if it were possible to set up
something like this:
PSTN (T1) - Asterisk - (some VoIP protocol, probably SIP) - Siemens
soft switch - their product
It sure sounds nice in theory, but I've never
!
-Ken D'Ambrosio
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
decent, but there are a lot
of things it doesn't do, too.
So: which GUI do -you- like?
Thanks!
-Ken D'Ambrosio
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http
1 - 100 of 160 matches
Mail list logo
