On Thu, Jul 06, 2006 at 03:32:04PM -0400, C F wrote:
I have recently build 2 machines, one with an Intel Pentium Dual Core
CPU, and one SATA HDD, and the other with a single AMD 64 bit CPU, and
a RAID 1 w 2 SATA HDD. Both costed the same even though the AMD had 2
HDDs. Here are the show
bram kortleven wrote:
We are currently looking for a way to easily configure a 'auto attendant'
system on our asterisk pbx.
More in detail, I'm looking for a webbased (or something similar)
configuration generator, that has a feature like asking me how many 'menu
levels' I want, what text to
The server name in the caller ID is a little
annoying
Cisco have removed that in the latest firmware
On 7/6/06, Ryan Amos [EMAIL PROTECTED] wrote:
The 8.2 firmware works just fine (asterisk 1.2.6). I have had exactly
zero problems with it, nothing even weird about it. Pretty trouble-free
Thanx alot for the tips.i'll try then out and let u know about the result
On 7/6/06, Alexander Lopez [EMAIL PROTECTED] wrote:
Yep, forgot 'bout that.
Or you could use web-meetme, it has this feature.
On 7/6/06, Alexander Lopez
[EMAIL PROTECTED] wrote:
Snip, snip. Chop Chop.
Is this the answer to all our prayers?
SIP protocol support (EXPERIMENTAL) (IP_NF_SIP)
SIP is an application-layer control protocol that can establish,
modify, and terminate multimedia sessions (conferences) such as
Internet telephony calls. With the ip_conntrack_sip and
the ip_nat_sip modules
Im in the process of installing Asterisks for clients and the question i
need answered is whats best practice for Asterisk. Should i use trixbox
or freebpx or should i install manually. Im getting very conversant with
Asterisk dailplan and concepts. What i am doing currently is looking at
On Fri, Jul 07, 2006 at 08:11:07AM +0100, Pele Zico wrote:
Im in the process of installing Asterisks for clients and the question i
need answered is whats best practice for Asterisk. Should i use trixbox
or freebpx or should i install manually. Im getting very conversant with
Asterisk
Hello Andrea,
Thursday, July 6, 2006, 3:19:13 PM, you wrote:
AS Hello everyone,
AS I'm trying to set up an Asterisk machine with a quad-port BRI
AS Junghanns card, and I want to use the mISDN drivers.
Hi andrea,
best way to do it is to download the install-misdn mqueue from beronet
web site
BC == Brian Capouch [EMAIL PROTECTED] writes:
BC Everyone's mileage varies, and IMO it doesn't do any of us any
BC good for negative opinions to be presented to the public as fact.
BC You disclaimed, indeed, but you would have been better off to say
BC something like, Grandstreams have been
i cant find any help about installing the web meetme tool. on
www.voip-info.org a link is given for installation instructions about
web meetme but i thinks its dead.
http://asteriskpr.blogspot.com/2005/09/guide-to-install-web-meetm_112614171575673316.html
i cant find anyother source of info about
2006/7/6, Maxim Vexler [EMAIL PROTECTED]:
NVFaxDetect does just that ;)Do you think NVFaxDetect is reliable ?Could you use it along a voicemail (I mean : someone having a single extension for voice and fax call, forward all incoming calls to its voicemail when leaving the office)
Cheers
Hello,
Do you know where i can download some rings for a PA1688 based Phone?
All rings on this link are not very nice...:
http://www.aredfox.com/edownloadsring.htm
Best regards,
--
Olivier Saulnier
STEGANUX
1er étage DIAMECANS
BEL AIR
03410 St-Victor
T: 04.70.02.27.62
F: 04.70.09.97.41
Instaling web-meetme is pretty easy ... did you try to install and use
it? Or checking for help even before trying? If you had already tried
.. let us know where you got stuck. You could find the download and
installation instructions here http://areski.net/Web-MeetMe/about.php
~VamsiOn 7/7/06,
Hi all !
someone deployed Junghanns's 0.3.0-PRE-1q (* 1.2.9.1) with a quadBri
card on a production system ?
drawbacks ?
thanks for your time
.mike
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Sent RTP packet to 293.67.65.3:43294 (type 18, seq 59050, ts
697456, len 2)
Got RTP packet from 21.98.11.200:58654 (type 18, seq 6246,
ts 3559220, len 20)
ANY ONE KNOWS WHAT THIS rtp DEBUD MEANS
THANKS
*
No employee
If you Answer() before, yes.
On 7/7/06, Olivier [EMAIL PROTECTED] wrote:
2006/7/6, Maxim Vexler [EMAIL PROTECTED]:
NVFaxDetect does just that ;)
Do you think NVFaxDetect is reliable ?
Could you use it along a voicemail (I mean : someone having a single
extension for voice and fax call,
since i'm going to try bristuffed * 0.3.0pre-1 right now,
could you please tell me where is that patch to libpri ?
thank you very much for your time
.mike
On Thu, 2006-07-06 at 22:30 +0200, stoffell wrote:
On 7/6/06, Kevin Savoy [EMAIL PROTECTED] wrote:
I'm having an issue where Asterisk
If you just save the page (from the browser) it will have the entire
configuration asa continuous html file. NewSipuraUtil is good too (more as a
backup and restore sort of facility). Bothways no passwords are passed.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
no i havent tried to install. I dont know how to do that in the first
place. i already have checkedout this link, there is no information
about installation over there. i cant find any make file, or configure
file in the source. i think its installatiojn is not like other
asterisk packages.On
It would be hard to bill all this calls, if you are using dialout call
files instead of Asterisk Manager API no ?
How would you colect the call duraction of both call legs?
Thks,
Marco Mouta
On 7/6/06, Marnus van Niekerk [EMAIL PROTECTED] wrote:
Also have a look at .call files.
You web
On Fri, Jul 07, 2006 at 11:03:41AM -0400, mike wrote:
since i'm going to try bristuffed * 0.3.0pre-1 right now,
could you please tell me where is that patch to libpri ?
0.3.0PRE-1*q* . Otherwise you'll end up with a patch for Asterisk 1.2.0
or something similar.
The patch to libpri is under
Hi all,
I've been planing to implement a webcall portal to dial SIP
extensions from my pbx, I've implemented this with dialout call files.
Could you advice me on the best way to collect call duration of this
calls, only this way i can allow my users to place external outgoing
calls.
I've need
yes i understand
you are talking about the libpri.patch in the patches folder
i was able to install
sucks that doing a rmmod qozap caused a kernel panic
On Fri, 2006-07-07 at 12:16 +0300, Tzafrir Cohen wrote:
On Fri, Jul 07, 2006 at 11:03:41AM -0400, mike wrote:
since i'm going to try
Ciao James,
Hello everyone,
I'm trying to set up an Asterisk machine with a quad-port BRI
Junghanns card, and I want to use the mISDN drivers.
I'm having some trouble configuring it: do I need to use CAPI
drivers? I haven't found good links, could you please provide some
info?
Hi,
I'm not an expert with gsm gateways, but as far as i know at least if
you have NT device for to connect ISDN line to Asterisk, you should
have your asterisk as TE (terminal equipment) and your NT as Network
Terminal.
Did you try to set the your BRI port where you connect your GSM
gateway as
Hi to all,
I am just trying for Asterisk on my fedora core3 machine. here's my sip extensions config files
sip.conf
[general]
bindport=5060
bindaddr=0.0.0.0
allow=all
context=ECPT
localnet=192.168.0.1
localmask=255.255.255.0
[phone1]
type=friend
host=192.168.0.53
context=Embedded
Brian Capouch wrote:
Thomas Kenyon wrote:
For some reason when I do this, It only works if I have callerID
switched off, otherwise I get authentication errors.
Do you know of anyway to bulk-save the contents of all the config
screens on that unit?
If so, I could scrub the passwords and
Ciao Andrea,
Hello everyone,
I'm trying to set up an Asterisk machine with a quad-port BRI
Junghanns card, and I want to use the mISDN drivers.
I'm having some trouble configuring it: do I need to use CAPI drivers?
I haven't found good links, could you please provide some info?
Well, CAPI
Douglas Garstang wrote:
Somewhat off topic...
I have a Sipura SPA-3000 ATA. Calls are coming in from the FXO port. I'm trying
to get all calls forwarded to Asterisk. However, (and this is hard to believe),
the docs say that 1-stage calling (I presume that means no PIN is required) is
not
Hello
Just use NoCDR() in the non bridged local context.
Jon
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Marco Mouta
Sendt: 7. juli 2006 11:22
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: [asterisk-users] How to collect Call
Sorry i didn't get your idea.
could you explain me what you mean? Are you saying to make CDR in only
one of the legs?
Best regards,
Marco Mouta
On 7/7/06, Jon Schøpzinsky [EMAIL PROTECTED] wrote:
Hello
Just use NoCDR() in the non bridged local context.
Jon
-Oprindelig meddelelse-
When I used the .call files, I made so that the call went through a Local
extension, where I didn't record the call, so that it would only be logged on
the outgoing channel
Jon
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Marco Mouta
Sendt: 7.
did u try asterisk manager api?
On 7/7/06, Jon Schøpzinsky [EMAIL PROTECTED] wrote:
When I used the .call files, I made so that the call went through a Local
extension, where I didn't record the call, so that it would only be logged on
the outgoing channel
Jon
-Oprindelig
its a .php file. Don't know about it but should be as simple as
editing some sort of config.inc and open it on a web browser ;)
On 7/7/06, Rizwan Hisham [EMAIL PROTECTED] wrote:
no i havent tried to install. I dont know how to do that in the first place.
i already have checkedout this link,
Dear Group,
I have a requirement for the agents in the call queue to be able to
transfer calls to other people within the organization and/or outside.
Unfortunately when I add tT to the Dial Command
i.e. exten = 0423,1,Dial(SIP/phone51,20,tT)
When the agent presses # to acknowledge the call it
Hello guys,
Not sure if it's me or what, but I'm starting to learn Asterisk. And I'm
currently reading the Oreilly book and another one. And I was at the
point to test the s extension. But when I try to use it it doesn't work
and the call gets rejected in Asterisk.
Here's a part of my
I switched to using the Manager API, as I thought it was easier to use.
I use it from a PHP script, which uses the flaAPI.php class.
Just use the originate action, set the channel to a Local channel and connect
to a context.
Are both calls going out over PSTN, or are one of them going via SIP?
On 7/7/06, Shad Mortazavi [EMAIL PROTECTED] wrote:
Dear Group,
I have a requirement for the agents in the call queue to be able to
transfer calls to other people within the organization and/or outside.
Unfortunately when I add tT to the Dial Command
i.e. exten = 0423,1,Dial(SIP/phone51,20,tT)
both cases.
And i would like to control call duration when the calls are bridged:
call A number pstn
then call B number sip or pstn
and CDR the call duration of real call between A and B, is there
something that does this?
Best regards,
On 7/7/06, Jon Schøpzinsky [EMAIL PROTECTED] wrote:
Just change the key for the transfer in features.conf ;)
Example:
[featuremap]
blindxfer =*1; Blind transfer
disconnect = #1897; Disconnect
;automon = *1; One Touch Record
atxfer = *; Attended transfer
had Mortazavi a écrit :
Dear Group,
I have a
Hello,
I am new to Asterisk, looking for a PBX solution doing and automatic
response based on a database query. I spent some hours googling, reading
manuals without too much luck. (Maybe I am just blind ...)
What we want to reach:
Caller dials in, gets prompted to dial an extension depending on
r u using queues. if u are, then open the agents.conf file and fine the following line
;ackcall=yes
this option when enabled and set to 'yes' forces the agents to press # to acknowledge the call. if you set
ackcall=no
then agents dont have to press # and call will be acknowledged without
I am thinking of using this machine to run asterisk. Has anyone had any
experience with this machine?
Thanks for any info.
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You should look for dialplan (extensions.conf) commands. They include
everything you mansion. Examples include mysql queries. Maybe you spend
sometime for Sybase connections.
-Original Message-
From: Dirk Enrique Seiffert [mailto:[EMAIL PROTECTED]
Sent: Friday, July 07, 2006 4:16 PM
To:
AHH HA! I did not realize that.
On 7/6/06, Kyle Hagan [EMAIL PROTECTED] wrote:
Shayne wrote:
I'm currently having the same issue. If anyone has a solution to this
it would be greatly appreciated.
I have a room with 70+ agents logged and and am not seeing this.. One
thing to keep in mind is
Dirk Enrique Seiffert ha scritto:
Hello,
I couldn't find any examples on the auto attendent reading responses from
a database, though it looks like a common task to me. Can anybody provide
some hints, links, directions or experiences for this kind of
configuration?
You should look at the AGI
Hi guys,I need to make a configuration to test a E1 channel, so, in the same context I created two extensions:exten = 555666,1,Dial(Zap/1/5556662)exten = 5556662,1,Dial(SIP/test)
On the E1 card I linked with a cross cable the ports 1 and 2. The leds are signaling that the connection is ok.But when
Could you explain a little more - what are the best tools for a bare bone
asterisk install. would you use a gui maybe for clients to add extensions,
gui to monitor calls and asterisk usage and managent concerns. HELP really
needed
Tzafrir Cohen wrote:
On Fri, Jul 07, 2006 at 08:11:07AM
I actually just deployed this server for a customer with only 4 users.
It worked great. The bios control over the IRQ's isn't the best. I
would definitely recommend against an integrated NIC. Other than that is
works great for them...
-Original Message-
From: [EMAIL PROTECTED]
We use NV's fax detection and it works very well.
(However this can still congest your system with junk
faxes).
We combined this with cid_rewrite (from www.generationd.com) which allows blocking
of calls based on CID name/num (and also rewrites the CID name based on your own
SQL database).
i have set dtmf=rfc2833 both in h323 and sip configuration files as my quintum uses h323 channel.
here is the dialplan snippet
[default]
exten= s,1,Answer
exten= s,2,Goto(1234,1)
exten= 1234,1,MeetMe(1234|AMpPw|)
exten= 1234,2,Hangup()
it always uses the s extension path to go to the MeetMe
Dirk Enrique Seiffert wrote:
Hello,
I am new to Asterisk, looking for a PBX solution doing and automatic
response based on a database query. I spent some hours googling, reading
manuals without too much luck. (Maybe I am just blind ...)
What we want to reach:
Caller dials in, gets prompted
On 7/7/06, mike [EMAIL PROTECTED] wrote:
someone deployed Junghanns's 0.3.0-PRE-1q (* 1.2.9.1) with a quadBri
card on a production system ?
drawbacks ?
Not really... But be sure to test if you don't have the hangup bug.
(call your cell phone, don't pickup, just hangup as soon as its
ringing,
Hi,
i try to use asterisk together with a webstar mgcp cable modem. if i
configure the cable modem to act as a nuera v5.2 call agent it connects
to asterisk. but if i try to make a call the line will disabled.
in order to get a deeper look here comes the logs:
--snip--
voip*CLI
MGCP read:
Hi,Do you need to buy an SIP/MGCP spare licence (GPL-SW-SM-UL-7960=) along a Cisco 7960 hardphone (GPL-CP-7960G=) when you simply want to connect it to a SIP enabled Asterisk server ?Regards
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What is the best method to detect the state of a sip trunk from an external
monitoring application?
I am currently testing a series of sip providers, and I wish to monitor
their state from an external application so that I may collect statistics on
their uptime/availability.
If I use
ok understood, thank you very much !
i'm testing it heavily from all the day and it seems to resist
i'll try the hangup bug, thanks for the tip and the patch !
thank you very much for your time !
.mike
On Fri, 2006-07-07 at 15:53 +0200, stoffell wrote:
On 7/7/06, mike [EMAIL PROTECTED] wrote:
When you dial directly you are bypassing
the zap and just dialing an internal extension. So that is probably why dialing
directly works. As far as the cross over cable between ports 1 and 2 I have
never attempted something like that before.
James Hawks
-Original Message-
I think you are really confused. I dont see a reason why dialing
555666 the call should go to client SIP/test. What you are doing is
dialing to Zap channel 1 (whatever it is) the number 5556662, so, what
do you have connected at the other end of the Zap/1 channel?
On 7/7/06, Ralph Liebessohn
-Original Message-
From: Brian Capouch [mailto:[EMAIL PROTECTED]
Sent: Thursday, July 06, 2006 4:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] OT: Sipura SPA-3000 ATA Directing Calls
toAsterisk
Douglas Garstang wrote:
Somewhat
I have been having no success with my zap interface, and am trying to
use ztmonitor to fix it.
I am running Trixbox 1.1 and so far have one x100p card (shows up as
'Wildcard).
I tried many manual attempts at setting rx and tx gain with no results.
Then I read:
On Fri, Jul 07, 2006 at 10:57:02AM -0400, Robert Moskowitz wrote:
I have been having no success with my zap interface, and am trying to
use ztmonitor to fix it.
I am running Trixbox 1.1 and so far have one x100p card (shows up as
'Wildcard).
I tried many manual attempts at setting rx
Hi
this is the problem i am experiencing:
I have 3 sip account (A,B,C) in a queue with ringall strategy.
suppose this scenario:
A is hold
B anc C are free
when a call is queued:
A is still hold
B is ringing
C is ringing
if B and C doesn't answer and A hangs up his call, A doent' ring!
On Fri, 2006-07-07 at 08:33 -0600, Douglas Garstang wrote:
-Original Message-
From: Brian Capouch [mailto:[EMAIL PROTECTED]
Sent: Thursday, July 06, 2006 4:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] OT: Sipura SPA-3000 ATA
Is there something out there that does something similar? Or does anyone
know how to make such a script? If possible, we prefer mysql-driven
menu's... as all other stuff is in mysql already...
Try looking at the FreePBX front end or the Trixbox Distrobution.
Regards,
Raymond McKay
President
Hi,
I am considering using * to run the phones in the small(12 unit) apartment building and I am looking for your input. Does it make sense? What setup would you use? What works and what does not.
robert
Read this topic online here:
Sure it makes sense but why are you doing
this in the first place?
Do you have a budget? Is it high end
features you are looking for or hoping to make money on call costs.
What are they using at the moment?
Cheers,
Dean
From:
[EMAIL PROTECTED]
[mailto:[EMAIL
Hi,
I've got a serious problems.
I have an * box set up at a custommer office.
* seems to work well until this message appears when i try to call from
the outside to any number managed by *
Jul 7 17:12:08 WARNING[8792]: chan_zap.c:9256 pri_dchannel: Ring
requested on channel 0/1 already
Hello Bill
Zapata.conf
switchtype=national
secallerid=yes
echocancel = yes
echocancelwhenbridged = yes
rxgain = 0.0
txgain = 0.0
signalling=fxo_ks
immediate=no
#include zapata_additional.conf
context=from-internal
group=1
channel=110
*CLI
-- Starting simple switch on 'Zap/110-1'
--
Hello
When I lift handset at phone hear silence
This is my config
*CLI zap show channel 110
Channel: 110
File Descriptor: 42
Span: 4
Extension:
Dialing: no
Context: from-internal
Caller ID: 2812
Calling TON: 0
Caller ID name: 2812
Destroy: 0
InAlarm: 0
Signalling Type: FXO Kewlstart
Radio: 0
This happens when your * box gets out of sync with your telco provider.
The only way to currently fix it is to restart *.
Do you log your CDR's to a database? If so asterisk will wait until the call
is logged before hanging up the channel which might be too long for your
telco provider.
James
no i havent tried to install. I
dont know how to do that in the first place. i already have
checkedout this link, there is no
information about installation over there. i cant find
any make file, or configure file
in the source. i think its installatiojn is not like other asterisk
Dean,
I am looking for additional stream of income from the property.
I have no hard budget yet.
They have straight lines from the phone company.
How would you do it?
Thanks in advance,
robert
Read this topic online here:
Greetings.
I sincerely apologize if this is an inappropriate use of this mailing
list.
I would like to announce a new software package intended to extend the
functionality of Asterisk and ease daily maintenance and administration.
The package is called Asterisk Manager Suite (AMS) and contains
I have this in sip.conf:
[ata1]
username = ata1
accountcode = ata1
qualify = yes
secret = foo
type = friend
host = dynamic
fromdomain = ipt.gumby.com
context = fromata_start
qualify = yes
When a call comes in from this device, if I have type=peer, Asterisk doesn't
match it, but it does if
I do know that the P4 core has a much longer pipeline (20 instructions?)
than the AMD. That is not good for a real time application and that is one
of the reasons AMD cpu's are able to do more with less clock cycles. Nobody
has mentioned heat and power consumption either. AMD's A64's generate
Depending on how much you can get one for, you may opt for a PRI and then get a channel bank to wire all the apartments with phones. If you are looking at straight VoIP you will have to be concerned about 911 service, etc..
Thanks,KyleOn 7/7/06, augustynr [EMAIL PROTECTED] wrote:
Dean,
I
I have been experimenting with the new metermaid application that allows
phones to monitor the status of a parked call using BLF. Does anyone know
what BLF feature the phone needs to support to make this work. Is it
basically the same as the Bristuff Devstate()? Anyone know which phones do
I have to say that from the screenshots it looks very impressive. It's also interesting to hear that you created a proxy for the AMI, I'm sure other projects may be able to make use of that as well. I'm not going to install it on my production server (yet), but look forward to trying it at home!
Why would they want to use you? What additional
functionality do you think you are going to offer?
How can you really improve over the
existing service that they have from the carrier?
Is it really worth the risk (billing fraud
etc) for the money you are going to make.
I agree with the channel bank approach,
and then just put inexpensive analog phones in the apartments, or an analog
base station with wireless handsets. Thats what most folks are used to
seeing in a residential setting. They dont need Cisco phones in the
kitchen or livingroom where they
The big player show us, to limit the free phone calls per week to a
certain amount.
How can we do that with ASTCC?
bye
Ronald
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I have patched astcc.agi with the HUP patch, but it still hangs from
time to time.
Asterisk SVN-branch-1.2-r25165M built by root @ vpbx on a x86_64 running
Linux on 2006-05-07 00:31:09 UTC
bye
Ronald
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Olivier wrote:
Do you need to buy an SIP/MGCP spare licence (GPL-SW-SM-UL-7960=)
along a Cisco 7960 hardphone (GPL-CP-7960G=) when you simply want to
connect it to a SIP enabled Asterisk server ?
Yes, as far as our sales rep can tell us.
Florian
Thanks!
It's sort of my first 'go' with GTK.
I tried to make the software as useful as possible even separately, so
even if you don't use AMA, you can still use the proxy, and developers
might find libAMI useful as well.
I look forward to hearing your feedback on it!
j
On Fri, 2006-07-07
Jason Adams wrote:
I actually just deployed this server for a customer with only 4 users.
It worked great. The bios control over the IRQ's isn't the best. I
would definitely recommend against an integrated NIC. Other than that is
works great for them...
-Original Message-
From: [EMAIL
Tzafrir Cohen wrote:
On Fri, Jul 07, 2006 at 10:57:02AM -0400, Robert Moskowitz wrote:
I have been having no success with my zap interface, and am trying to
use ztmonitor to fix it.
I am running Trixbox 1.1 and so far have one x100p card (shows up as
'Wildcard).
I tried many manual
You have a little confusion:
friend = can GENERATE and RECEIVE calls
peer = can only GENERATE calls
user = can only RECEIVE callsAlyed
Return-Path: [EMAIL PROTECTED] Fri Jul 07 09:27:13 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by
If its a highend apartment complex
and you are doing it for additional functionality then polycoms are appropriate.
Cheers,
Dean
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Cory Andrews
Sent: Friday, 7 July 2006 12:42 PM
To: 'Asterisk
Users Mailing
Oops, i missed the crossover cable part. I have used crossover cable,
so it should work, but the DNID must be complete. Wich signaling are
you using?
Regards
On 7/7/06, James Hawks [EMAIL PROTECTED] wrote:
When you dial directly you are bypassing the zap and just dialing an
internal
Hi there!
I'm setting up an E1 with a new Telco and they are asking me to add the
extension number (CallerID)into an "Additional calling party number". Guess it
refeers to a part of the E1 trace they are getting. I've been
playing around with the callerid and in zapata.conf and sip.conf
Interesting... will this patch (metermaid) work with 1.2.7 asterisk?
On 7/7/06, shadowym [EMAIL PROTECTED] wrote:
I have been experimenting with the new metermaid application that allows
phones to monitor the status of a parked call using BLF. Does anyone know
what BLF feature the phone needs
Hi,
Yes we have that installed. Music on hold works fine when called in extensions.conf
e.g.
exten = 5000,2,MusicOnHold()
However when I put someone on hold the music does not play
I am using the Polycom Soundstation IP301 and X-lite phones.
Thanks
Julian
Date: Fri, 7 Jul 2006 09:35:56
Benny Amorsen wrote:
BC == Brian Capouch [EMAIL PROTECTED] writes:
BC Everyone's mileage varies, and IMO it doesn't do any of us any
BC good for negative opinions to be presented to the public as fact.
BC You disclaimed, indeed, but you would have been better off to say
BC something like,
The building is in Canada and pri cost $700 so probably no.
Thanks though.
Read this topic online here:
http://forum.globalvoicenet.com/viewtopic.php?p=1557#1557
Cory,
Channel banks on the fxs side makes sense what about fxo?
The same? I cannot bring pri as it is too expensive.
Thanks,
Read this topic online here:
http://forum.globalvoicenet.com/viewtopic.php?p=1561#1561
Dean,
No it is not higher end apartment complex.
Thanks,
robert
Read this topic online here:
http://forum.globalvoicenet.com/viewtopic.php?p=1562#1562
___
On 15:53, Fri 07 Jul 06, stoffell wrote:
Replace in libpri/q931.c the first block of text with the second
The fix is in libpri so maybe this is a funny question, but
does this apply to E1/T1 linecards as well ?
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key:
It is working for my and my Snom phones.
On 7/7/06, Matt [EMAIL PROTECTED] wrote:
Interesting... will this patch (metermaid) work with 1.2.7 asterisk?
On 7/7/06, shadowym [EMAIL PROTECTED] wrote:
I have been experimenting with the new metermaid application that allows
phones to monitor the
On 7/7/06, James Hawks [EMAIL PROTECTED] wrote:
When you dial directly you are bypassing
the zap and just dialing an internal extension. So that is probably why dialing
directly works. As far as the cross over cable between ports 1 and 2 I have
never attempted something like that
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