On Sun, Jan 08, 2006 at 06:16:16PM -0800, Mike Fedyk wrote:
Tzafrir Cohen wrote:
Experimental: Asterisk 1.2:
At the moment they are not that experimental anymore and should be ready
for use, but are not well-tested yet.
To use it, define both sources:
deb http://rapid.dotsrc.org/
Weuse Juniper/Netscreen 5GT's with the latest 5.3 firmware.It is fully sip aware and in a NAT environment it modifies the addresses in the SIP frames according the NAT table.The netscreen also checks the sip frame for the udp ports to be opened for the audiochannels and openn them for the session
Hi,
we are trying a beronet ISDN card with asterisk 1.2 on debian sarge distro.
Everything seems fine except for outbound calls: it seems we cannot send
outbound digits so we cannot use phone digits to use ivr menus.
I followed beronet dinstallation document.
Is there some parameter missing to
On 1/8/06, Chris Bagnall [EMAIL PROTECTED] wrote:
Has anyone had experience using Asterisk-IM/Jive Messenger with any IMclients apart from Trillian and Spark? (Trillian costs money and I'm not
that keen on Spark's lack of configurability)
I've been looking as well. Unfortunately there's really
Hi,
The hostname that you used in your register directive ('provider.ie')
must have a corresponding section in sip.conf. In your example, you used
'[provider-in]'. If that is what you actually used, then this might
explain why your incoming goes to the default context because it
couldn't
In article [EMAIL PROTECTED], [EMAIL PROTECTED]
says...
If you're wanting to scroll through output from a CLI command, use:
asterisk -rx command | less
Thank to bouth of you.
--
Tomislav Parcina
[EMAIL PROTECTED]
___
--Bandwidth and Colocation
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says...
An example SIP friend is defined as [112], so we could now type, from
the CLI:
sip notify polycom-check-cfg 112
sip notify cisco-check-cfg 214
doesn't seam to do anything. I have sip_notify.conf in my /etc/asterisk/
directory. Cisco
Sorry in advance if this is a FAQ...
I've got a working Asterisk setup based on [EMAIL PROTECTED] 2.2. I have a
TDM400
card with 2 FXS and 2 FXO ports; PSTN connections come in via the TDM
card.
I haven't been able to get inbound fax with spandsp and rxfax to work.
Occasionally an
I am probably thinking that [EMAIL PROTECTED] might be a better way to start
your journey
PaulH
- Original Message -
From:
luke
devon
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Monday, January 09, 2006 4:50
PM
Subject: Re:
Hi , I wanted to connect GradStream Budge Tone - 100 phone with a Asterisk box for acc them as extentions on the LAN .1. After configure Asterisk in a Linux box with different ip network can i , use the other IP phones over the LAN ??2. Asterisk installed machine can wein the LAN as
Hi,
Thanks to both Iqbal and Kokmeng for the replies.
Kokmeng I tried what you suggested however no luck...
What I have done which is currently working(kind of) is that in my
sip.conf in the [general] section I have set context=incomingpstn. My
register line looks like:
register =
On Mon, 9 Jan 2006, James Harper wrote:
I would suggest extend the libcapi20. I already did such an extension
to
libcapi20 to support the bintec remote-capi. This means with that
libcapi20,
each program (including chan_capi) can do remote-capi without any
change...
The more I
Would anyone recommend a medium size company choosing Asterisk over 3COM
- Original Message -
From: Kerry Garrison [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Saturday, January 07, 2006 10:23 PM
Subject: RE:
Tzafrir Cohen wrote:
On Sun, Jan 08, 2006 at 06:16:16PM -0800, Mike Fedyk wrote:
Tzafrir Cohen wrote:
Experimental: Asterisk 1.2:
At the moment they are not that experimental anymore and should be ready
for use, but are not well-tested yet.
To use it, define both sources:
deb
Small, medium and large are relative. What do you want it to do, and
why do you want to change your phone system? With the right talent,
(consultant or in-house) Asterisk can be used in most situations. With
that no more details, then a simple answer will have to suffice.
Most likely yes.
On Fri, Dec 30, 2005 at 10:23:26PM +0100, Armin Schindler wrote:
On Fri, 30 Dec 2005, Louis-David Mitterrand wrote:
Hello,
I just received a couple SX440isdn phones and was wondering if they can
be plugged into a Diva 4BRI port in NT mode and work with
asterisk+chan_capi?
Yes, I
I have setup wake up call in * following those instructions
http://www.voip-info.org/wiki/view/Asterisk+tips+Wake-Up+Call+PHP
and it works fine. Now I have few questions.
- When I arrange wake up call, does it call me only that day or I can
set it up for whoole week?
- Can I set it up for
Hi I have been trying to get SNOM (320,360) and hotdesking working with
asterisk.
I can get it working fine with SER but it fails with asterisk unless I
have no SIP password/secret in sip.conf
This is how it works with SER,
1. reset phone (removes accounts)
2. phone prompts for username and sip
In article 6A1C243A7E2E824293FABC3042045790930851
@dtw_localmail.strtrade.com, [EMAIL PROTECTED] says...
Hello all, is anyone aware of any open source call accounting software for
Asterisk? Something that can parse out Asterisk's call detail records and
generate on-demand reports?
Check out
Hi,
I raised this with SNOM and they say it is purely an asterisk problem
and it needs to be fixed (asterisk that is).
If asterisk sent a 401 instead of a 403 the phone would work fine and we
would all be happy.
Here you can find a patch that will fix it:
Hello,
I have a couple of questions:
1) Before heading off for a bit of vacation, I was having a wierd
problem where I was getting more than one call per callfile placed in
the outgoing/ spool. I describe it here:
http://forums.digium.com/viewtopic.php?t=3455
so far, so good - it's not doing
Ah cool, thanks ill look at it.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Maik Schmitt
Sent: 09 January 2006 11:24
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SNOM
On Mon, 2006-01-09 at 12:07 +0100, Tomislav Parcina wrote:
I have setup wake up call in * following those instructions
http://www.voip-info.org/wiki/view/Asterisk+tips+Wake-Up+Call+PHP
and it works fine. Now I have few questions.
- When I arrange wake up call, does it call me only that day
Hi all,
I would like to know if there is a solution to this question.
Scenario:
Two asterisk servers connected across the Intenet ( in SIP or IAX mode, no
matter) with both of them having static ip addresses
Then I add a second link (with another provider), with another NIC at both
side, and
Hi,
Two asterisk servers connected across the Intenet ( in SIP or IAX mode, no
matter) with both of them having static ip addresses
Then I add a second link (with another provider), with another NIC at both
side, and again both of them having static ip addresses.
Is there a way to tell
Hello!
After many troubles, I have received my Wildcard 406. There is a label
on antistatic bag stating that this is 406. The card itself is marked
as 405. Kernel modules shows in dmesg that card is 405.
Is 406 the same as 405 with additional board installed?
Hi Armin,
You can also use one Linux Server running CAPI cards with rcapid and have
your Asterisk/OpenPBX with chan_capi on another maschine...
Did you ever try something like that? What kind of implication had the
remote CAPI with regards to sound quality?
--
Best regards
Peer Oliver
On Mon, 9 Jan 2006, Peer Oliver Schmidt wrote:
Hi Armin,
You can also use one Linux Server running CAPI cards with rcapid and have
your Asterisk/OpenPBX with chan_capi on another maschine...
Did you ever try something like that?
I just tried it. But I never really used it longer.
What
That's great... I didn't know about the persistentagents features!
I'll test it asap!
Thank you
Dov
- Original Message -
From:
Alexander
Lopez
To: Dov Bigio ; Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Saturday, January 07, 2006 5:16
PM
Are you sure you have the FTP server's IP address set correctly in the phone's
configuration?
On Thu, Jan 05, 2006 at 05:17:41PM -0500, Ken D'Ambrosio wrote:
Anthony Rodgers wrote:
Is the mac-address.cfg file name in lower case?
Yeah, it is. Hell -- I've cut-and-pasted the filename from
Hi,
I want to pick up a call with the snom's programmable buttons(snom190
-SIP 3.60x, snom360-SIP 4.1) with asterisk server (v 1.2.0), I tried
with the option 'Destination' and when the incoming call arrive to
another snom phone the button blinking.
In this way I can only pick down it
On Monday 09 January 2006 07:32, Dmitry Ivanov wrote:
After many troubles, I have received my Wildcard 406. There is a label
on antistatic bag stating that this is 406. The card itself is marked
as 405. Kernel modules shows in dmesg that card is 405.
Is 406 the same as 405 with additional
The more I look, the more I think that the bintec protocol might be
the
one required to talk to the Cisco anyway. Do you have those patches
somewhere?
I have placed the patched libcapi20 sources (libcapi20.tgz) on the
public
ftp server ftp://isdn4linux.org/pub/capi4linux
Thanks!
It
On 1/9/06, Rich Adamson [EMAIL PROTECTED] wrote:
Sorry in advance if this is a FAQ...
I've got a working Asterisk setup based on [EMAIL PROTECTED] 2.2. I have a
TDM400
card with 2 FXS and 2 FXO ports; PSTN connections come in via the TDM
card.
I haven't been able to get inbound
Hi can anyone help.
I just updated my CentOS and ran the command rebuild_zaptel and
genzaptelconf with a Reboot in between each step.
Now I have no Zaptel devices (I used to have 3 FXO X100P Cards)
Summary of what happens below:
(Zaptel.conf contains no card info after running this command.)
On Saturday 07 January 2006 02:30, Philipp von Klitzing wrote:
Hi!
Now, one user, not the receptionist, has gone in and set his personal
numbers to these function keys thinking that DESTINATION meant setting a
number to dial out. So now I have a ton of SIP SUBSCRIBE messages for his
Tomislav Parcina schrieb:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I am aware of the possibility to add the option t or T to dial, so #33
transfers the call to extension 33.
It needs to be deined in feautres.conf file. So when you dial #1 you'll
hear transfer and than
On Tue, 10 Jan 2006, James Harper wrote:
The more I look, the more I think that the bintec protocol might be
the
one required to talk to the Cisco anyway. Do you have those patches
somewhere?
I have placed the patched libcapi20 sources (libcapi20.tgz) on the
public
ftp server
Anyone got the screen xml function to work yet? i've setup an URL in my line 1
(the only line I use) but i don't even see a GET request to my webserver.
Kind regards,
Erik
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users
Have you checked
http://www.voip-info.org/wiki/view/Asterisk+High+Availability+Solutions
Regards,
Evert
[EMAIL PROTECTED] wrote:
Hi all,
I would like to know if there is a solution to this question.
Scenario:
Two asterisk servers connected across the Intenet ( in SIP or IAX mode, no
Hi all,
problem solved!
The parameter /s at the end of Dial string command was necessary.
Giorgio Incantalupo
gincantalupo wrote:
Hi,
we are trying a beronet ISDN card with asterisk 1.2 on debian sarge
distro.
Everything seems fine except for outbound calls: it seems we cannot
send
Richard,
This also happened to me over the weekend. What happened to me was yum
updatd two files found in /etc/udev/permissions.d/ and the other in
/etc/udev/rules.d/
Yum makes backup copies of each of these files. All you need to do is
copy the missing lines from both files and paste them
Starting and stopping the recording is based off of the message
taking software which knows when I call is going on. They do make
recording devices that go in between the headset and phone, but they
take batteries. I can't really have a recording device running off
batteries in a call center.
David N. Welton [EMAIL PROTECTED] wrote:
2) app_txfax
I need to know if a fax has gone through or not. My reading of txfax
seems to indicate that it basically just fails, rather than giving me
anything I can work with to try and fail gracefully (letting the user
know that things didn't go
Hi, I've used Agents + Queues before with success, but I can't figure
out why this trivial setup is not functioning...
stage*CLI show agents
1306 (gdh) available at '[EMAIL PROTECTED]' (musiconhold is 'default')
1 agents configured [1 online , 0 offline]
and the internal context is
I would like to know if anyone out there has a known and working solution
in Asterisk 1.2.1 for dialtone detection. We currently use the
Chanisavail command on Zap channels and then need dialtone detection after
that. Please respond on or off list.
v o i p 3
a t t a
n i b b l e d o t n e t
On Mon, 9 Jan 2006, Louis-David Mitterrand wrote:
On Fri, Dec 30, 2005 at 10:23:26PM +0100, Armin Schindler wrote:
On Fri, 30 Dec 2005, Louis-David Mitterrand wrote:
Hello,
I just received a couple SX440isdn phones and was wondering if they can
be plugged into a Diva 4BRI port in
Alexander Lopez wrote:
I would incoparate dundi, After using it I have fallen in love with it
for distributed applications such as this. It makes configuration at
first a bit steeper but it picks up monentum as your deploy. With Dundi
you basicaly broadcast what extensions or numbers are served
Sorry in advance if this is a FAQ...
I've got a working Asterisk setup based on [EMAIL PROTECTED] 2.2. I have
a TDM400
card with 2 FXS and 2 FXO ports; PSTN connections come in via the TDM
card.
I haven't been able to get inbound fax with spandsp and rxfax to work.
The syntax for the options in chanspy are not well documented. How do
I use multiple options?
I am using the Asterisk Manager API and am using
ChanSpy(|q)
but would like to include volume
ChanSpy(|q,v3) ?
Any insight would be appreciated.
Dan Littlejohn
www.littlejohnconsulting.com
Hi can anyone help.
I just updated my CentOS and ran the command rebuild_zaptel and
genzaptelconf with a Reboot in between each step.
Now I have no Zaptel devices (I used to have 3 FXO X100P Cards)
Summary of what happens below:
(Zaptel.conf contains no card info after running this
Fonality just received an influx of capital, you can read about it here.
http://gigaom.com/2006/01/09/fonality/
Cory Andrews
Purchasing Manager
++
VOIPSupply.com
A Division of b2 Technologies
454 Sonwil Drive
Buffalo, NY 14225
direct - 716.250.3402
mobile -
I would if the tech that sets it up knows exactly
what he or she is doing.
Regards,
Dovid
: "Dakota" [EMAIL PROTECTED]Subject: Re:
[Asterisk-Users] Asterisk vs 3COMTo: "Asterisk Users Mailing List -
Non-Commercial Discussion"asterisk-users@lists.digium.comMessage-ID:
[EMAIL
Thanks Cory. Awesome... and their in LA too. They'll be hearing from me. :)
-Original Message-
From: Cory Andrews [mailto:[EMAIL PROTECTED]
Sent: Monday, January 09, 2006 8:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Jobs
I'm looking for some input from someone with real experience of
telephony. I am having problems with the sound quality on our PSTN line
calls. Our channel banks are Adtran 600 and 750 and I spent a lot of
time on the phone with Adtran trying to work out the problem.
We are getting hum and noise
Darren Nickerson wrote:
3) I'm working on a small, simple email-fax system. Just out of
curiosity, what else is out there for Asterisk? I found AsterFax, but
it looks a little bit hairy to set up...
You really should consider HylaFAX - www.hylafax.org. It has what you're
missing - a fully
Does anyone know if it would cause problems to have the same Zap
channel in multiple goups? So, for example, if I have two PRIs
would the following work or would it cause problems:
channel = 1-23
group = 1
channel = 25-47
group = 2
channel = 1-23,25-47
group = 3
I am just curious if anyone has
Hi all,
Please see the dialplan snippet below. Any hint why it does not pass the
correctly entered 10 digit number as calleridnum on to the SIP phone?
The SIP phone always shows Unknown.
exten = s,1,PrivacyManager(1,10)
exten = s,n,GotoIf($[${PRIVACYMGRSTATUS} =
SUCCESS]?privok:privfailed)
exten
just as with any asterisk application, options are separated each by a
pipe option1|option2|option3
regards
On 1/9/06, Dan Littlejohn [EMAIL PROTECTED] wrote:
The syntax for the options in chanspy are not well documented. How do
I use multiple options?
I am using the Asterisk Manager API
Patrick Conroy wrote:
Does anyone know if it would cause problems to have the same Zap channel in
multiple goups? So, for example, if I have two PRIs would the following
work or would it cause problems:
The internal structures in chan_zap can only store one group association
for each
why dont use ChanSpy or Monitor? An AGI or MAGI script would let you
monitor all the incoming and/or outgoing calls of anyone, taking the
info from a database will make it flexible so you can add more
monitored people, and then download the audio via web, or even email
it to who it may concern.
On Mon, 2006-01-09 at 16:40 +0100, David N. Welton wrote:
Hi,
I thought about using Hylafax, but after looking around a bit, I got the
impression that it's not exactly trivial to integrate it with Asterisk,
and that it will require a dedicated incoming line. Perhaps I'm mistaken?
ok, look for the file /etc/asterisk/modules.conf . There disable
autoload. Then try loading as less modules as you can. This is a
list of my modules. Im attaching you a copy of my modules.conf so you
can use it as a start. From there start to disable modules, I dont
think is a core problem. What
andrutto wrote:
Yeah, but to traditional PBX central you can plug fax machine hassle free.
Well, in theory you should be able to do the same with Asterisk: plug
fax machines into FXS ports on the box.
I say in theory because I've not done that myself, and I've heard
rumors of past
On Mon, Jan 09, 2006 at 10:44:58AM -0500, Patrick Conroy wrote:
Does anyone know if it would cause problems to have the same Zap channel in
multiple goups? So, for example, if I have two PRIs would the following
work or would it cause problems:
channel = 1-23
group = 1
channel = 25-47
Hello,
I have installed a brand new asterisk 1.2.1 server.
OS is centos (RH enterprise kernel) 4.1.
Asterisk suddenly stops working.
It does not generate a core dump what so ever.
I looks like a clean stop of asterisk, as if you
where to enter stop now in asterisk CLI.
Hi Everyone,
Does anyone know of any ATAs that can do proxy failover without using
SRV. I don't want to rely on dns if at all possible.
Basically, I have Asterisk boxes in two different data centers and I
need ATAs to be able to uses the server at DC2 if DC1 goes down. The
servers are already in
I think cisco ATA can handle 2 proxies,
This option is called altproxy in the web based management
joash
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
Thomas
Sent: Monday, January 09, 2006 5:32 PM
To: asterisk-users@lists.digium.com
Subject:
On Mon, January 9, 2006 16:44, Patrick Conroy said:
Does anyone know if it would cause problems to have the same Zap channel
in
multiple goups? So, for example, if I have two PRIs would the following
work or would it cause problems:
channel = 1-23
group = 1
channel = 25-47
group = 2
We currently have most of our voicemail forwarded to user's email
addresses, but the message is coming in at a way low volume. It sounds
great when you listen on the phone, but it's very hard to hear when you
listen on the computer. Does anyone know of a way to increase the gain
on the file
Rich Adamson wrote:
I'm certainly not the expert on this topic, but I believe the issue has
to do with the pci bus and probably relates to the TigerJet chip used on
the card. Until that's addressed, any analog modem use through the card
will be marginal at best. (Same issue as with the older
I'm looking for some input from someone with real experience of
telephony. I am having problems with the sound quality on our PSTN line
calls. Our channel banks are Adtran 600 and 750 and I spent a lot of
time on the phone with Adtran trying to work out the problem.
We are getting hum and
I thought about using Hylafax, but after looking around a bit, I got the
impression that it's not exactly trivial to integrate it with Asterisk,
and that it will require a dedicated incoming line. Perhaps I'm mistaken?
It isn't that bad basically download compile and install the trick is to
find
Colin Anderson [EMAIL PROTECTED] wrote:
The big weakness in Hylafax is the client. 90% of the time the client will
be under Windows, and your choices are Cypheus, which is pretty and user
friendly but slow and crash-y or WHFC which is ugly and nasty but works
100%
and has slick features like
I'm certainly not the expert on this topic, but I believe the issue has
to do with the pci bus and probably relates to the TigerJet chip used on
the card. Until that's addressed, any analog modem use through the card
will be marginal at best. (Same issue as with the older x100p card.)
On Mon, 2006-01-09 at 11:15 -0600, Rich Adamson wrote:
It would be very interesting to know the real numbers that have it working.
The archives (and about two/three years of attempting to help others with
the exact same problem) suggests no better then maybe one in ten or twenty
will ever get
Rich Adamson wrote:
I'm certainly not the expert on this topic, but I believe the issue has
to do with the pci bus and probably relates to the TigerJet chip used on
the card. Until that's addressed, any analog modem use through the card
will be marginal at best. (Same issue as with the older
It would be very interesting to know the real numbers that have it working.
The archives (and about two/three years of attempting to help others with
the exact same problem) suggests no better then maybe one in ten or twenty
will ever get spandsp to work with the digium x100p or TDM card.
Hello.
A while back, I noticed an odd problem with our SPA-841 phones connected
to Asterisk. Now we are having a different odd problem, and I'm not sure
if they're related. I wonder if anyone else has experienced anything
else like this, and/or if there is any reasonable explanation?
Hi, I have now managed to get it working with asterisk 1.0.10 I had to
modify the patch http://bugs.digium.com/bug_view_page.php?bug_id=6035 as
its for the latest version of asterisk but it works very well now.
Thanks for the pointer.
-Original Message-
From: [EMAIL PROTECTED]
Anyone have any information regarding the ectoolkit on svn?
~ron
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To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Unfortunately I asked the same question a day or two with no response...
It appears the only way is to use a very beta patch, look on
bugs.digium.com and search for snom pickup, you should find it. But I
wouldn't recommend using it in a production environment just yet.. It's
funny cause asterisk
Hello, hoping someone out there has some ideas -
I have a VOIP line that has call waiting. It is terminated at a Sipura
3000 and the POTS side of that device connects to an FXO port in my *
box. I also have a POTS/PSTN line that terminates in another FXO port
on my * box.
There are two FXS ports
I knew someone will not be able to resist :)
On 1/8/06, Steve Totaro [EMAIL PROTECTED] wrote:
I am not sure why you are looking for jobs doing Asterisk work when less
than two weeks ago you were publicly bashing on the list.
Steve
Consulting is fine, as long as I'm working for someone
I've recently started PIC'ing some calls into a asterisk box across a
feature group D trunk from Verizon. Everything seems to work ok, except
for some reason Asterisk doesn't grab the full caller ID from Verizon. I
can see that they do send it, but Asterisk drops the first 2 numbers.
Looking at
We've been having lost dialtone problems on one of our analog station
ports. Just before rebooting this time I noticed these in our dmesg
outputonce the PBX comes back I'll get the times, but I can't help but
think this must have something to do with it. Anyone? Do we need to have
faxguy, maybe you can tell me why
As I've noted in previous posts I'm evaluating HylaFax with IAXModem. When I
run iaxmodem and faxgetty through a console the modem works 100% I have yet
to find a fax that it won't tie up with. When I run IAXmodem and faxgetty in
initttab, the modem is extremely
[EMAIL PROTECTED] zaptel-1.2.1]# make
gcc -I/lib/modules/2.4.21-4.ELsmp/build/include -O6 -DMODULE
-D__KERNEL__ -DEXPORT_SYMTAB
-I/lib/modules/2.4.21-4.ELsmp/build/drivers/net -Wall -I.
-Wstrict-prototypes -fomit-frame-pointer
-I/lib/modules/2.4.21-4.ELsmp/build/drivers/net/wan
If you've ever compiled and installed an older version of * on this box,
specifically from the 1.0 era, it's possible you need to try removing
/usr/include/asterisk and see if that helps.
Moj
Leandro Rzezak wrote:
[EMAIL PROTECTED] zaptel-1.2.1]# make
gcc
On Mon, 9 Jan 2006, Michael Baird wrote:
I've recently started PIC'ing some calls into a asterisk box across a
feature group D trunk from Verizon. Everything seems to work ok, except
for some reason Asterisk doesn't grab the full caller ID from Verizon. I
can see that they do send it, but
Aaron Daniel wrote:
We currently have most of our voicemail forwarded to user's email
addresses, but the message is coming in at a way low volume. It sounds
great when you listen on the phone, but it's very hard to hear when you
listen on the computer. Does anyone know of a way to increase
Removed /usr/include/asterisk, same thing.. Any clue?On 1/9/06, Mojo with Horan Company, LLC [EMAIL PROTECTED]
wrote:If you've ever compiled and installed an older version of * on this box,
specifically from the 1.0 era, it's possible you need to try removing/usr/include/asterisk and see if that
I just upgraded to Asterisk 1.2.1 and Asterisk fails
to start with the error below.
Jan 9 21:25:38 NOTICE[1339]: cdr.c:1171 do_reload:
CDR simple logging enabled.
Jan 9 21:25:38 WARNING[1339]: loader.c:326
__load_resource:
/usr/lib/asterisk/modules/chan_zap.so: undefined
symbol: pri_restart
Jan
Hey, all. I quoted a customer about $100 for some cheap SIP phones. I
was planning on using the BT-102's, but he called said they look like
Princess phones, and I have to admit that he has a point. Some of the
other inexpensive phones look decent, but (for example) the SPA-841's
wiki entry says
It's any voicemail from any line on the system, whether it's SIP, IAX,
or ZAP... The voicemail message is basically so low in volume that my
boss has almost blown his speakers switching between listening to
voicemail and listening to whatever music he listens to lol... I've got
the rxgain's
Grandstream GXP-2000 is decent.
On 1/9/06, Ken D'Ambrosio [EMAIL PROTECTED] wrote:
Hey, all. I quoted a customer about $100 for some cheap SIP phones. I
was planning on using the BT-102's, but he called said they look like
Princess phones, and I have to admit that he has a point. Some of
Hello All
Everything was working OK, and decided to install AMP 1.10.010... and
problem started.
AMP took control of Asterisk... For some odd reasons I can not
connect to Asterisk CLI any more. I get the following error: -
[EMAIL PROTECTED] ~]$ sudo /usr/sbin/asterisk -r
Unable to
I know this isn't a specifically asterisk question, but does anyone know
how to make the phone NOT use it's old config? I'm trying to get rid of
the second line registration crap and it's not working.
Aaron
___
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On Mon, 2006-01-09 at 15:28 -0500, Ken D'Ambrosio wrote:
Hey, all. I quoted a customer about $100 for some cheap SIP phones. I
was planning on using the BT-102's, but he called said they look like
Princess phones, and I have to admit that he has a point. Some of the
other inexpensive phones
Check manager.conf and manager.custom.conf (installed by amp) for access
lists which may be preventing you from reaching it.
-Original Message-
From: Nitesh Divecha [mailto:[EMAIL PROTECTED]
Sent: Monday, January 09, 2006 2:45 PM
To: Asterisk Users Mailing List - Non-Commercial
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