I ran into this same problem compiling ndiswrapper on my MacBook with FC6. I
uninstalled the old-time GCC and used yum to install gcc++3.3, I think it
was and then re-compiled. FC6 has some wierdness with compiling. Ask me how
fun it was getting everything working in my MacBook.
-Original
When I have HylaFAX answer a call redirected to the fax extension in
Asterisk when it detects CNG, Asterisk hangs up:
Jan 30 14:32:59 VERBOSE[1098]: -- IAX2/ttyIAX0552/12 is ringing
Jan 30 14:32:59 VERBOSE[1098]: -- IAX2/ttyIAX0552/12 answered Zap/23-1
Jan 30 14:32:59 DEBUG[1098]: Ooh,
Sorry my mistake it is in Snom 360 firmware 3.60b and higher, not 320.
-Original Message-
From: Mike Hammett [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 24, 2007 7:57 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] RE: Snom 320 echo
Where do I find more out in
On * 1.0.9
User logs into voicemail, dials Option Zero, then Option Three. Records
name, accepts the recording. greet.wav is generated in the user's mailbox.
It plays back fine. The Directory app still spells out his name!
What am I missing?
___
Plantronics 510SL Bluetooth
We use this on Snom 360's in CAP positions and they work well although the
lifter is a bit of a Rube Goldberg contraption. However, the receptionists
are extremely happy with it, she can walk around almost the entire building
with it and answer calls. One issue we
Later firmware versions have an echo-cancelling component in it, upgrade to
latest version and also turn down the gains on the mic, the default setting
is way too high. A setting of 3 or 4 max is all that is nessisary.
hth
-Original Message-
From: Mike Hammett [mailto:[EMAIL PROTECTED]
Used the Directory application instead of the Directory AGI.
-Original Message-
From: Colin Anderson [mailto:[EMAIL PROTECTED]
Sent: Tuesday, January 23, 2007 11:29 AM
To: 'asterisk-users@lists.digium.com'
Subject: [asterisk-users] * 1.0.9 Voicemail record name does not
playback
They do. The ExternalIVR application was developed in cooperation with
them.
lol so the $2.6b they spent on Skype was well worth it, then. It'd be nice,
though to have an OSS chan_skype.
___
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IMO, the 480i, by a LONG shot.
Yeah I have a rollout of 36 480i's right now and they are cat's ass, but I
have found that the cordless will cause interference with Bluetooth headsets
- static.
Otherwise, yeah, my favorite phone. Good implementation all around.
A little php and SCP would make this work. You could do a web interface like
vmail.cgi.
-Original Message-
From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED]
Sent: Friday, January 19, 2007 9:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
If you use a channel bank like the Adtran Atlas 550, you can specify a
primary sync to the telco, and every subsequent connection to the Atlas uses
that sync as a timing source. Expensive, but I expect you can pick one up or
something like it on Ebay. Nothin beats an Atlas, though.
-Original
On the voip-info.org wiki there are good tips to get snoms to play nice on
lans. I personally have experienced wierdness using particular switches
(cheap ones).
also note that snom now has an auto-update subscription URL in their support
wiki, if you use the URL it makes updating a 4.X to 6.X
I use that one myself. Does the Snom attempt to dial asterisk when
you hit Retrieve? What error do you get? Sure it's in the right
context (I screw that up ALL the time)...
Sometimes it's asterisk, sometimes it's unknown sometimes, it's
Unknown so:
exten =
folder.
- Original Message -
From: Colin Anderson mailto:[EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - mailto:asterisk-users@lists.digium.com
Non-Commercial Discussion'
Sent: Thursday, January 11, 2007 12:10 AM
Subject: RE: [asterisk-users] Directory too difficult?
I got
Why not use DBGet / DBPut? I use it for Caller ID and I have over 50K
entries in the DB, and there is no appreciable load hitting the DB in the
dialplan. And my one install (admittedly modest) hits the DB a few thousand
times a day, with up to 46 concurrent calls.
-Original Message-
I am not sure that the security guy for this network will allow me to put
up the asterisk box dual homed to the public IP and the LAN.
Your security guy needs to go back to school. If eth0 is on the LAN and eth1
is on the WAN, and the WAN connection is properly secured with only the
ports you
In the current setup, asterisk is behind a different nat/firewall than
the LAN phones. The phones are using sccp. If the asterisk box is
compromised, it is not on the local LAN. This is what I think he
doesn't want to give up.
Oho, now I see. Well, there's the philisophical endless debate
Yes let's use a very limited vocabulary engine to make it simple:
A good thought, but I'd be raked over the coals the first time a suit with a
super thick accent (we have a few) would drive the recognition engine crazy.
Here in Alberta, we talk about things that piss us off. One of the things a
If you say: Agent you are transferred to a person. The IVR clearly states
that when you call in. I got a demo of Mitel's speech platform last year and
it has algorithms that measure apparent stress in a voice. If the voice
sounds to stressed, it transfers to an operator.
-Original
I have a group of users whos complaint about Asterisk is that the directory
application is too hard too use. (yeah, yeah, I know. For the record,
they're Calgarians) Now I'm in a pickle: I don't want to have to create a
custom directory for these guys. Anyone have any tips for making the
directory
://www.sheltonjohns.com
On Jan 10, 2007, at 11:10 AM, Colin Anderson wrote:
I have a group of users whos complaint about Asterisk is that the directory
application is too hard too use. (yeah, yeah, I know. For the record,
they're Calgarians) Now I'm in a pickle: I don't want to have to create a
custom
Followup on this issue, it appears that using a single PRI's clock as the
master clock avoids clock drift between the PRI's and we get no more
artifacts. So, :
wanpipe1.conf:
TE_CLOCK= NORMAL
TE_REF_CLOCK= 0
wanpipe2.conf:
TE_CLOCK= MASTER
TE_REF_CLOCK= 1
zaptel.conf:
I've replaced 2XTE110 with an A102 with echo cancellation specifically to
deal with echo problems. However, user feedback has indicated to me that on
some calls (not a lot, but some) the call is unusable, with audio
artifiacts, described by one user, as: very bad phasing reverb feedback
(from my
Aha, it just happened to me, so now I can characterize the audio: It
basically sounds like it's missing every other sample - fuzzy and distorted.
Timing?
___
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To
I think you are absolutely right. The audio I heard earlier sounds exactly
like a timing issue. So:
wanpipe1.conf:
TE_CLOCK= NORMAL
TE_REF_CLOCK= 0
wanpipe2.conf:
TE_CLOCK= MASTER
TE_REF_CLOCK= 1
zaptel.conf:
span=1,1,0,esf,b8zs
span=2,0,0,esf,b8zs
I'm going to make
ASUS motherboards, in particular, have worked for me perfectly, everytime
with both Digium and Sangoma cards. They are also easy to work with and well
documented.
-Original Message-
From: Doug [mailto:[EMAIL PROTECTED]
Sent: Tuesday, January 02, 2007 1:04 PM
To: Asterisk Users Mailing
LD [M] /root/wanpipe/patches/kdrivers/wanec/wanec.o
Building modules, stage 2.
MODPOST
*** Warning: register_wanec_iface
[/root/wanpipe/patches/kdrivers/wanec/wanec.ko] undefined!
*** Warning: unregister_wanec_iface
[/root/wanpipe/patches/kdrivers/wanec/wanec.ko] undefined!
CC
not have Zaptel source installed, or WanPipe can't
find the Zaptel source.
Colin Anderson wrote:
LD [M] /root/wanpipe/patches/kdrivers/wanec/wanec.o
Building modules, stage 2.
MODPOST
*** Warning: register_wanec_iface
[/root/wanpipe/patches/kdrivers/wanec/wanec.ko] undefined
Sangoma support did something and the driver is there. Rebooting now with
the new card. Hold me, I'm scared.
-Original Message-
From: Colin Anderson [mailto:[EMAIL PROTECTED]
Sent: Friday, December 22, 2006 10:09 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject
Does IAXmodem allows you to receive faxes with any extensions
(auto-detecting incoming faxes).
You just let Asterisk do the fax detection for you, and when it hears CNG,
send it to the fax extension, and your fax extension would just Dial() one
of the IAXmodems (using IAX)
[EMAIL PROTECTED]
I don't see why rxfax would be less reliable than iaxmodem/hylafax as
it's using the same spandsp to receive fax.
I will defer to Lee Howard on this but IIRC the big factor is ECM which is
not supported in SpanDSP. And another difference is that it is *HylaFAX*
that is recieving the fax itself
If you are using Windows to generate the .call files, make sure they are in
Unix format (LF only at EOL, not CR+LF) - Notepad makes bad Unix files. Use
Crimson Editor www.crimsoneditor.com to make the file, and click Document
File Format Unix Format.
I ran into this same problem, and it turns
Short answer: a single group should be fine. Long answer: It depends.
Your Dial() command determines the order in which Asterisk plucks channels
from your PRI. Most north american system call inbound channel 1 first, then
2, etc. It makes sense, then for you to take channels from the topmost
Message-
From: Lee [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 19, 2006 2:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: SPAM-LOW: Re: [asterisk-users] .Call files do not seem to
wo rk
Colin Anderson wrote:
If you are using Windows to generate the .call files
')
-- Playing 'transfer' (language 'en')
-- Executing NoOp(IAX2/[EMAIL PROTECTED]/2, INBOUND Regular call
exiting for user Hansen Li from Colin Anderson 7028247) in new stack
-- Hungup 'IAX2/[EMAIL PROTECTED]/2'
Any pointers would be welcome
In 2.6.15 kernels and higher, you can use taskset to pin a task to a
certain CPU. Here's a way to set httpd to the 2nd processor in a 4 way
system:
HTTPDPID=`ps -A | grep -a -A0 httpd`
taskset 0x0002 -p ${HTTPDPID:0:5}
-Original Message-
From: Don [mailto:[EMAIL PROTECTED]
Sent:
IIRC, a call from the SIP perspective is any transaction or interaction
with a SIP device. So things that qualify as a call are things like
registration and qualification. Nothing to sweat about. You can suppress it
with sip no debug from the command prompt.
hth
-Original Message-
From:
http://dev.mmgsecurity.com/projects/lat/
http://dev.mmgsecurity.com/projects/lat/
I run Open Xchange in a couple of sites and administrating LDAP thru the
command line is akin to enjoying a case of anal warts.
-Original Message-
From: Steven Baker [mailto:[EMAIL PROTECTED]
Sent:
Unfortunately a lot of people don't bother to set TSI and have blocked
Caller ID on their fax line so you would get false positives if you filtered
out those faxes. I just did a HylaFAX install last week where the enduser
was extremely pleased about the fax-to-email - when a junk fax came in
My * at home is a P-3 400 256 meg with a TDM 400. 2 cordless phones, 3 snom
200's. Termination is through an IAX provider. All of the standard stuff
works, transfer to cell, web voicemail,etc but the interesting thing that I
do is a script that polls the Canadian weather service every 10 min
I use alphanumeric names as extensions in my Asterisk architecture,
which are the username part of the e-mail of each person at my site.
Because Asterisk was primarily built to use numeric extensions, I'm
having some problems with people that have usernames with dots between
letters, like
what happens when you drop your gains? use /etc/asterisk/zaptel.conf and
fiddle with tx and rx values. Works, most of the time.
-Original Message-
From: John McEntee [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 11, 2006 12:25 PM
To: Asterisk Users Mailing List - Non-Commercial
I've used the prompt pls-wait-connect-call to give my users a cue to cool
their heels for a second or two in circumstances like this, and no one has
complained. That's probably the most useful prompt in Asterisk!
-Original Message-
From: James Harper [mailto:[EMAIL PROTECTED]
Sent:
Had
the same problem in fc2. Solution was to chkconfig zaptel off chkconfig
asterisk off then in rc.local modprobewct1xxp (i think) then
ztcfgthen start safe_asterisk. Dunno why.
Hey,
is OnStar using Asterisk? Details, please.
-Original Message-From: Shea, Matt
You could uses System() and the Logger command. Wouldn't be hard.
-Original Message-
From: Douglas Garstang [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 04, 2006 12:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Dialplan Syslog
Just a
When you Dial() the cell, are you passing the 't' parameter? Also: When the
call hits the cell, is Asterisk still in the media stream? canreinvite=no
should be explicitly specified in the SIP accounts of your providers in
sip.conf. One more thing: Do you know for a fact that inband DTMF is being
??? I do it with a Zap channel no problem. In my case,
1. Call comes in from PSTN (Zap channel)
2. Call is routed back out a Zap channel using the Dial() command with the
't' option
3. Asterisk is still in the media stream, so it listens for inband DTMF
4. User presses Hash, Asterisk says
I, for one, welcome our new Republican overlords.
lol you are just full of pop culture references, aren't you?
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-Original Message-
From: Colin Anderson [mailto:[EMAIL PROTECTED]
Sent: Saturday, September 30, 2006 3:56 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] 480i phone: Is there a trick to
registering with *??
oj that's awesome info thanks i
is to clean up the web provisioning interface it's super sparse and maybe a
little better codec selection.
good job, Aastra! -Users: check this phone out it is a good value.
-Original Message-
From: Colin Anderson [mailto:[EMAIL PROTECTED]
Sent: Monday, October 02, 2006 9:12 AM
To: 'Asterisk
If
reliability is the issue, then use the PRI *first* then failover to
VoIP.
If
cost savings are the issue, use VoIP then have a 2nd VoIP provider to fail over
to, and no PRI.
In
either scenario, inbound call routing is thorny, some guys that provide both PRI
and VoIP can route calls
Of Colin Anderson
Sent: 29 September, 2006 6:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] 480i phone: Is there a trick to registering
with
*??
Running * 1.0.9, sip.conf allow=all is set.
Based on the advice of -users earlier this week I've
at 00:47 -0600, Colin Anderson wrote:
yeah, weird that, i did set the proxy after I posted so it's now
sip:[EMAIL PROTECTED] but still no dice. There is a port number as
well
that I left at 0, should I change it to 5060?
btw, the hardware is a sweet little package if I get this working I
I got a bad batch of 360's where the hookswitch was damaged in shipping.
Snom fixed this by sticking a piece of packing foam between the switch and
the hook socket, wedging it into place. While this worked fine, I found I
had to be careful unpacking the phone - if you just yanked on the foam, a
Also with PRI:
-Fax works
-No 911 issues
-SIP provider may or may not honor your arbitrarily set caller ID - PRI
always will if your telco isn't a dick
-Easier to break out an analog channel if needed (give me a channel bank
over an ata any day)
-Faster to troubleshoot - if you get red alarm
Running * 1.0.9, sip.conf allow=all is set.
Based on the advice of -users earlier this week I've ordered an Asstra 480i
CT for evaluation. Phone is up, sees Asterisk, tries to register, Asterisk
refuses. I though it might be codec mismatch so I specified allow=all. Valid
account, password OK,
[8247]
username=8247
type=peer
secret=
quaify=no
port=5060
pickupgroup=
nat=never
[EMAIL PROTECTED]
host=dynamic
dtmfmode=rfc2833
allow=ulaw
allow=alaw
context=from-internal
callerid=Colin Anderson 702-8247
canreinvite=no
Tried peer , friend allow=all etc no go
I concur with your approach, but Tier 1 means as little here as it
does when evaluating Internet backbone carriers. could you expand on
what evaluation criteria you use? I'm going to be pre-speccing some
stuff myself this month...
Sorry I should have been more clear. A good Asterisk install
Erm, I think what the OP was referring to was something like this:
____
_
A. SIP service--B. His Asterisk install-C. His
customer's install--- Enduser handsets
__
Since I'm told that you can only have, say, one Digium card
per chassis, this can be an issue.
???
lspci | grep Jens
01:01.0 Network controller: Individual Computers - Jens Schoenfeld Intel 537
01:04.0 Network controller: Individual Computers - Jens Schoenfeld Intel 537
asterisk -rx zap show
Were those people -- who, unlike me, had done it and had problems -- wrong?
There are more variables than the Digium card itself. Things like bus
design, chipset etc all come into play. I've noticed that there is a
concerted effort with Asterisk implmentors to often roll out Asterisk in a
white
There
seems to be three tiers in my experience:
1.
Only your DID's
2.
Arbitrary, but the pilot number of the PRIwill appear if you suppress your
Caller ID
3.
Completely arbitrary, including null --this is the fa
shizzle
So you
want 2) or 3) but definitely it is a telco thing. You need
Try a
different, (larger orsmaller)room with different acoustical
characteristics. You may be talking, the audio is transmitted from a primary
source - you - but then itmay pick up the reflections of your
voicebouncing off of the walls in the room, and the phone may be picking
that up as
Looks good, great price:
http://www.aastratelecom.com/ipphones/pro_243.asp
Anybody using these? How's the cordless? Does it play nice with * ?
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It's excellent home phone. I wouldn't use it in a business environment.
No
hold, no one-touch voicemail. However, it works great!
aw crap, that's a biggie but I think I can work around it, teach the user to
dial *98 for voicemail, *700 for park and hash to transfer, currently the
users dial
:[EMAIL PROTECTED]
Sent: Monday, September 25, 2006 10:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] OT: Opinions on Aastra 480i CT?
Colin Anderson wrote:
Looks good, great price:
http://www.aastratelecom.com/ipphones/pro_243.asp
Anybody using
Just spitballing:
1. Execute a macro in the dial command to spawn an AGI that would return
it's PID to Asterisk and accept the IP address or SIP address of the phone
as an argument. Call the variable, say, ${INCREMENTCOSTPID}
2. The AGI would store call cost variable plus the increment. It would
hello asterisk community,
most of us have participated in this project since the beginning of
Asterisk. I remember entering the Asterisk community nearly 3 years ago at
version 0.3.0. since than the system has undergone many changes, things
improoved dramatically. the system has matured to a
You
need to use AGI to do this. You would put a shell script yourscript.agi in
/var/lib/asterisk/agi-bin
If you
want an HTTP response dumped into your dialplan as a variable, you would use
wget:
myagi.agi:
#!/bin/bash
TMPFILE=/tmp/$$.$RANDOM
wget
-q -t3 --output-document=$TMPFILE
Why not use the application or function Curl?
lol some of us cavemen are running 1.0.9.
instead of that:
meh, you say Tomato, I say Tomato. (funny, that doesn't look right when you
read it)
___
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I found this, which looked interesting:
http://wiki.onmac.net/index.php/Triple_Boot_via_BootCamp
Also, Apple released a new version of BootCamp that supports the Xserve on
Aug 16. If it'd work, and you could shoehorn a PRI card into it, man
wouldn't that make a nice Asterisk box? And at $2999,
I
don't see anything in there thatI'm not doing already (and have been for
over a year, with 200 users)with Asterisk 1.0.9, HylaFAX, and Exchange
5.5, with the exception of the text-to-speech stuff which is do-able with
Cepstral / Festival and some scripts that hook MAPI on the Exchange
I run
HylaFAX on a separate box from my dual PRI Asterisk box, and Asterisk relays the
call to HylaFAX when it detects the fax. It relays the call on a private subnet
with a crossover Ethernet cable with the slin codec. I have over 200 IAXmodems
running on the HylaFAX box, which is an
it,
interesting idea though.
-Original Message-
From: Lee Howard [mailto:[EMAIL PROTECTED]
Sent: Friday, August 11, 2006 11:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk IAXmodem HylaFax?
Colin Anderson wrote:
The only thing that keeps
Anyone having issues with the message waiting indicator and retrieve
button on SNOM 320's and 360's.
When the MWI light is not lit on a 360, and the user hits the voicemail
button, the Snom phone dials the extension 'unknown', 'default' or
'asterisk'. If you don't have an unknown etc extension
I was wondering if we could uplink small switches to the wall data ports
to
the switch, and connect the additional SIP phones to them to get them
connectivity to Asterisk?
Yes, we do it and it works fine, as long as you don't cascade more than 3
switches between two devices your latency
The
3300 uses the MiNet protocol so you wouldn't be able to interface it over a LAN,
but hooking it up the "old school" way (crossover PRI cable) should work fine,
you would have to set up DID's on your 3300 that correspond to the extension
numbers you would want to dial the 3300 from
SIP is
on the "low-end" product (3200?) which is meant as an Asterisk-style mini-PBX
that provided an entry point to get Mitel into orgs with 50 users, but for
some reason it seems discontinued I can't find it on their website anymore.
There is a SIP stack for the 5220 and higher phones
We are using an Ateus VoiceBlue to GSM gateway calls on our * 1.0.9
server.
It works perfectly fine, except at peak periods, say, 10 AM and 3 PM. At
that point, calls get dropped (not gateway'd) and Asterisk jumps to the
next
priority in the dialplan. Our interpretation of this is that the
We are using an Ateus VoiceBlue to GSM gateway calls on our * 1.0.9 server.
It works perfectly fine, except at peak periods, say, 10 AM and 3 PM. At
that point, calls get dropped (not gateway'd) and Asterisk jumps to the next
priority in the dialplan. Our interpretation of this is that the local
So you need a divide and conquer strategy here:
1. Is it Asterisk or the WAN? This should be easy enough to test for. Do
call dropouts happen in your datacentre? If not, your Asterisk install is
good. My money's on the 10mbit WAN pipe, and that's what I would be
focussing on.
2. If it's the WAN,
Also, when I connect to the server locally (the server is in the room
next to me, in other words, and i have 1 Gbit of bandwidth all the way
to the back of the server, I still get call dropouts.
However, this IS the only server (of 8 total, all in the same rack and
connected to the telco via
I do have another hand-rolled install of Asterisk 1.2.9.1 with identical
hardware (Digium T1 and FXS boards) on a PRI with the same telco that is
working fine with the account codes.
I'm stumped. Anyone have any ideas or pointers?
shot in the dark:
pridialplan=unknown
dumps digits exactly
It seems only the Far-end (called party), is hearing this and not the
calling party.
So, looks like the A in ADSL is showing here. What is your upstream
bandwidth? If it is 1mbit this is the most likely cause. If your VoIP
provider allows it, change your codec to GSM, which IMO has the best
A GSM
gateway will allow you to specify a ruleset so a channel on the gateway is
always locked to a particular mobile number, then you just send the call from
Asterisk to the gateway and it will do the hunt for you.
-Original Message-From: Lito Lampitoc
[mailto:[EMAIL
, 2006 9:03
AMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: Re: [Asterisk-Users] asterisk to mobile
phonewhat brand of gsm gateway do you think works well
with asterisk?
On 6/27/06, Colin
Anderson [EMAIL PROTECTED]
wrote:
A GSM gateway
if you go with 1.0.X you can't go wrong, and there is plenty of stuff that
it can do that will keep you busy. Problem is it is sooo tempting to use the
new candy in 1.2.X or head. But for me, 1.0.X is the way to go as long as
you can deal with echo cancellation problems (think Sangoma or go to
yes. Wind whistling in a car, female voices at a particular pitch and
volume, fax machine running in the background of a voice call with the
speaker on. It happens. Whether this is a problem or not depends on your
pain threshold. I get a couple reports a week, which means that it actually
happens
Yes, which is why I disable faxdetect entirely. My sister-in-law was
constantly being detected as a fax machine several minutes into
conversations with my wife. As funny as that may seem at first ...
those two eventually make it a not-so-funny situation for me.
lol, Spousal Acceptance
Surely once the call has been bridged the fax detection should turn off ?
I'd like to find out a way it can be done, can anyone else comment?
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Should be part of the FAQ for the list, as well as the setting for Exchange
5.5 which a *lot* of orgs still run (we do too)
I wonder if the list SW can be modded to automatically plonk any mail with
the subject string: Out of Office
-Original Message-
From: Steven [mailto:[EMAIL
thing is i don't have these issues using the
same phone on my test box at home which is a old 256M Sun Ultra5 connected
to a old Netgear Cable/DSL wireless router (can you say 0 (zero) QOS).
Terrelle
From: Colin Anderson [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
it to determine your $SENDTO setting.
Lee.
Colin Anderson wrote:
example config, please?
-Original Message-
From: Lee Howard [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 21, 2006 5:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: faxdetect
He's probably using Exchange which has a global setting to either send OOO
replies to SMTP addresses or not. It's a dumbass Exchange administrator who
enables this option (it is actually on by default)
Same thing happened to the mac-asterisk list last week, except the OOO
message would reply to
Seems to me that the problem here is one of automatic distribution. Why not
create a virtual faxmodem for each user, or group of users, and assign a DID
to each of them, then have HylaFAX email the PDF direct to the user via
custom settings in FaxDispatch? THe user can then print it, or file it or
In the Snom web management page under Advanced make sure Challenge response
on phone is turned to OFF. This is a stupid feature to have on by default
from the factory.
-Original Message-
From: Edward de Zeeuw [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 21, 2006 11:54 AM
To:
yes. works fine.
-Original Message-
From: Lee Howard [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 21, 2006 2:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: faxdetect questions - Please HELP!
Colin Anderson wrote:
I have 200
If I understand this correctly, this is a user outside your firewall dialing
in to your office over the Internet. Always, inbound calls work, but
sometimes, outbound calls do not work. So if you have replaced the hardware
totally, and you still have the same problem, it could be a routing issue
. Most people would have run enough
iaxmodems to handle the load and then use the DID for routing the faxes,
rather than the device.
Lee.
Colin Anderson wrote:
yes. works fine.
-Original Message-
From: Lee Howard [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 21, 2006 2:48 PM
From
your Asterisk console:
tcpdump -i eth0 -e | grep -A1 your target phone's IP
address
Then:
Make a
call on your target phone.
Disclaimer: not tested
-Original Message-From: mojowrkn
[mailto:[EMAIL PROTECTED]Sent: Monday, June 19, 2006 11:21
AMTo:
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