Jesus Mogollon wrote:
Hi all
Does anyone know of any motherboards with PCI slots that can take the
TE412P card? Is there such a MB for Athlon 64 or P4 procs?
I have no experience of it, but you could look at the Asus M2N32 WS
which has 2 x PCI-X (3.3V) slots. It is a socket AM2 (Athlon64
Carlos Rojas wrote:
Anyone know a good carrier of voip for international calls?
Please use asterisk-biz list
http://lists.digium.com/mailman/listinfo/asterisk-biz
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Jason:
The issue is indeed VoicePulse. Their equipment is not correctly setup
and/or capable to recieve DTMF from many sources, one of those is Sprint
CDMA mobile phones, they claim the issue is Sprint however Sprint is
correctly sending DTMF and every other carrier is able to recieve them.
Best
VoicePulse is the absolute worst. You can get additional channels for
$25/month but that includes no usage whatsoever. That's almost double what
the same capacity WITH MINUTES on a PRI port costs!
Any decent provider will be able to give you an unlimited number of channels
because you are paying
Google is your friend!!
http://www.eweek.com/article2/0,1895,1773983,00.asp
http://www.eweek.com/article2/0,1895,1773832,00.asp
http://www.eweek.com/article2/0,1895,1772661,00.asp
Let us hope SS isn't a communications lawyer
The FCC DOES have jurisdiction
John Novack
Steve Sobol wrote:
On Fri
The Asterisk Development Team is pleased to announce the release of
Asterisk 1.4.0-beta4.
This release contains a number of updates:
- a bug fix for the ExternalIVR application and addition of 'silence'
sound files to support it
- various SIP interoperability improvements
- memory and dialog leak
The Asterisk Development Team is pleased to announce the release of
Zaptel 1.4.0-beta3.
This release contains a number of updates:
- compatibility with Linux kernel 2.6.19
- bug fixes to the Xorcom Astribank driver (XPP)
- support for Digium's TE110P Rev C, VPMOCT064 and new revisions of the
S110
The Asterisk Development Team is pleased to announce the release of
Asterisk 1.2.14.
This release contains a number of updates:
- a bug fix for the ExternalIVR application and addition of 'silence'
sound files to support it
- various SIP interoperability improvements
- memory and dialog leaks in
The Asterisk Development Team is pleased to announce the release of
Zaptel 1.2.12.
This release contains a number of updates:
- compatibility with Linux kernel 2.6.19
- bug fixes to the Xorcom Astribank driver (XPP)
- various other bug fixes
Thanks for supporting Asterisk and Zaptel!
___
But who in there right state if mind would use ulaw?
Just take them away to the funny farm ha ha ho ho!! :-P
I do. Exclusively. I personally don't like the g729 compression (audio
quality and license issues) any my customers definitely notice the
difference right away and wonder why the quality
So, my "peak" would need 4.5 mega-bits per second of bandwidth.
Am I in the ballpark?
Sounds about right. Or the other way around (if you need to know the
peak bandwidth usage):
For audio:
1,000,000 minutes/month = 33,000 minutes/day
10% daily usage in 1 hour = 3,300 minutes used
3,300 minutes
But who in there right state if mind would use ulaw?
Just take them away to the funny farm ha ha ho ho!! :-P
gsm, ilbc, g729 etc are a lot better choice.
Best regards,
Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email
(VoIP PBX) 1-866-638-1254
For Information on PBX Systems
This may expose my ignorance, but here goes :)
I've been asked to figure out how much bandwidth would be needed to handle
1,000,000 minutes a month.
Here's the environment:
) All calls are received via SIP.
) All calls use the ulaw codec.
) Calls average 10 minutes in duration.
) The "busi
Hardware is an SM56 card (X100P clone). When the line hangs up, ztmonitor
displays full bar (or whatever maximum allowed by rxgain) in RX. It only
drops zero when the line picks up (and remote was silent). Is this
something of concern? The zap channel seems to work despite echo.
Additional
>
> I see that the digium card doesn't share the IRQ however
> Digium has recommended diabled USB still... additionally the
> Digium card is on 169 which isn't a valid IRQ.. how can I
> find out what it is sharing with?
>
lspci -vb will give you the irq as seen by the cards on the PCI bus
--
When booting Ubuntu 6.06.1 (Linux 2.6.15-27-386), wcfxo would load but not
configure. I have three ways to manually force wcfxo to configure: 1)
ztcfg, 2) modprobe -f wcfxo, or of course 3) unload and reload wcfxo. Each
works equally well.
As a workaround, now I have to put ztcfg in rc.local
Larry Alkoff wrote:
I have a Sipura 3k connected to Asterisk 1.2.
All I want to do here is have incoming PSTN calls ring POTS phones
connected to the Sipura.
The web interface for the Sipura, on the PSTN line tab lists
VoIP User 1 Auth ID: asterisk
and
Dial Plan 8: ()
How do I put the
On 12/15/06, Paul Connolly <[EMAIL PROTECTED]> wrote:
We currently have an Asterisk system with a PRI and 6 POTs lines for
backup. We are looking to add service such as Voicepulse Connect as an
extra level of redundancy and a cost saving alternative to PRI calls. VP
Connect only allows 4 simu
I have a Sipura 3k connected to Asterisk 1.2.
The web interface for the Sipura, on the PSTN line tab lists
VoIP User 1 Auth ID: asterisk
and
Dial Plan 8: ()
How do I put the Dial Plan 8 information in sip.conf or extensions.conf?
Is 66610 a sip extension in sip.conf or a context in extension
Hello everybody
Anyone know a good carrier of voip for international calls?
Regards
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Is there any good iax2 softphone capable of attended transfer ( like sjphone
for sip ) . ? I tried iaxcomm and idefisk both seems unable to handle
attended transfers.
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Actually port block is on softphone side and not on asterisk server's
internet connection .I put this in iptables of asterisk server
iptables -t nat -A PREROUTING -i eth0 -p udp --dport 5091 -j REDIRECT
--to-port 127.0.0.1:5060
server is listening on port 5060
Now strange part is everything s work
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Rob Schall
> Sent: Friday, December 15, 2006 11:14 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Fast Busy Followup
>
> So I might have a b
Actually port block is on softphone side and not on asterisk server's
internet connection .I put this in iptables of asterisk server
iptables -t nat -A PREROUTING -i eth0 -p udp --dport 5091 -j REDIRECT
--to-port 127.0.0.1:5060
server is listening on port 5060
Now strange part is everything s wor
Hi all
Does anyone know of any motherboards with PCI slots that can take the
TE412P card? Is there such a MB for Athlon 64 or P4 procs?
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To UNSUBSCRIBE or update op
On Fri, 15 Dec 2006, John Novack wrote:
> Are you in the US?
> If so, such blocking is not legal
I'd like to see a citation for that. ISPs aren't common carriers and
aren't required to carry specific types of traffic.
> and you should file a complaint with
> the FCC
The FCC regulates common c
Does anyone on-list have experience doing this? I'm curious about setting it
up. I own a domain and might like to try making sip:[EMAIL PROTECTED] a
workable idea.
Is this just an experimental thing, or might it be really usefull...say for
video calling?
Michael
Scenario:
A call is sent from one Asterisk system to another with IAX. The remote
Asterisk system runs the Queue application, which then starts to play a
different music on hold class then the standard 'default'. The console on this
system displays:
-- Executing Queue("IAX2/xxx.yyy.142.203
We currently have an Asterisk system with a PRI and 6 POTs lines for backup.
We are looking to add service such as Voicepulse Connect as an extra level
of redundancy and a cost saving alternative to PRI calls. VP Connect only
allows 4 simultaneous calls; we are looking for 4 to 5 times that to sup
Am Samstag, den 16.12.2006, 02:32 +0530 schrieb Vicky:
> I have shifted asterisk port to 5091 . Now i am able to register
> properly using sjphone but still when dialing number it keep on
> showing calling .. and do not go ahead . I change asterisk's rtp ports
> too but still i am unable to make c
I have shifted asterisk port to 5091 . Now i am able to register properly
using sjphone but still when dialing number it keep on showing calling ..
and do not go ahead . I change asterisk's rtp ports too but still i am
unable to make call . My other softphone on different internet isp is
working
I've set it up as...
span=2,1,0,esf,b8zs
bchan=6-27
dchan=28
It is a paetec full pri t1.
Does this help with the diagnosis, or do you need more info?
Rob
Steven wrote:
> What kinf of line do your DIDs come in on?
> How many spans do you have configured and where do they go? Telco/legacy PBX?
>
Yes i read that on voip-info wiki but i have bindport = under device
(extension) which should make that extension work on other port but its not
working . :(
On 16/12/06, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote:
port= specifies the REMOTE port.
You can't have multiple bindport= and
What kinf of line do your DIDs come in on?
How many spans do you have configured and where do they go? Telco/legacy PBX?
Does span 2 have a context defined?
--
--
Steven
http://www.glimasoutheast.org
"Rob Schall" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]
> So I might hav
I am sure rtp ports arent blocked ..
On 16/12/06, Derek Whitten <[EMAIL PROTECTED]> wrote:
Mail list wrote:
> Hello my isp has blocked outgoing and incoming connection for port 5060
> . I
> have ssh access to server so i want to send all traffic from port 5091
to
> port 5060 of asterisk .so
well that should map incoming packets to 5091 to 5060, but may not rewrite
[new] outbound packets from 5060 to 5091, which your isp may be blocking.
an iptables SNAT or MASQUERADE might help you there. i'm not positive on
if this would be needed or not.
more importantly, however, if your is
forking CDR could help Ricardo.
On 12/15/06, Ricardo Martins <[EMAIL PROTECTED]> wrote:
Hi John, I´m very interested into this call forwarding capabilities and
I solved this problem filtering on the web-script (in my case, php) the
number the user can intert on the database. (I know it´s not an
On Fri, Dec 15, 2006 at 12:10:40PM -0600, John French wrote:
> The setvar command
It is not a dialplan command. It is a configuration key.
> below works fine in iax.conf and in sip.conf
> but not here in zaptel.conf. I need it to retrieve info from the
> AstDB. Advice is apreciated, can't seem
Are you in the US?
If so, such blocking is not legal, and you should file a complaint with
the FCC
John Novack
Mail list wrote:
Hello my isp has blocked outgoing and incoming connection for port
5060 . I have ssh access to server so i want to send all traffic
from port 5091 to port 5060
port= specifies the REMOTE port.
You can't have multiple bindport= and it must be in [general]
Mail list wrote:
I am using a month old svn version of asterisk 1.2 . I have set
bindport=5091 for a sip peer ( type = friend) and nat=yes .. in sip show
peer it shows port 5091 for peer but aster
I have
1.2.12.1
Voicepulse using IAX
I get about 30-40% issues with not having the DTMF tones work.
I have 3 questions
#1. Voicepulse says they are sending them, Is there some setting I can
adjust to make sure my end is working?
#2. I have set the Dialplan to play a sound "Operator" then go to
Am Freitag, den 15.12.2006, 13:08 -0600 schrieb Alvin Austin:
> Hello,
>
> In Asterisk 1.4 beta 3, the UPGRADE.txt file says:
>
> Variables:
> * The builtin variables ${CALLERID}, ${CALLERIDNAME}, ${CALLERIDNUM},
> ${CALLERANI}, ${DNID}, ${RDNIS}, ${DATETIME}, ${TIMESTAMP},
> ${ACCOUNTCODE},
>
Mail list wrote:
> Hello my isp has blocked outgoing and incoming connection for port 5060
> . I
> have ssh access to server so i want to send all traffic from port 5091 to
> port 5060 of asterisk .so i tried
>
> iptables -t nat -A PREROUTING -i eth0 -p udp --dport 5091 -j DNAT --to
> 127.0.0.1
Hello my isp has blocked outgoing and incoming connection for port 5060 . I
have ssh access to server so i want to send all traffic from port 5091 to
port 5060 of asterisk .so i tried
iptables -t nat -A PREROUTING -i eth0 -p udp --dport 5091 -j DNAT --to
127.0.0.1:5060
Now my softphone is abl
So I might have a bit of a more narrow question from my earlier one.
Previous, I had been wondering what would cause a phone dialing into a
DID that connects to the asterisk box to get a fast busy.
I've noticed the following message:
chan_zap.c: Ring requested on unconfigured channel 0/1 span 2
ecuting NoOp("", "DATETIME() : 0") in new stack
-- Executing NoOp("", "DATETIME : 20061215-12:56:26") in
new stack
Dec 15 12:56:26 ERROR[26373]: pbx.c:1383 ast_func_read: Function
TIMESTAMP not registered
-- Executing NoOp("", "TIMESTAMP(
nik600 wrote:
Hi
i am trying to do the same thing:
receive a call from a cisco callmanager and forward it to a SIP user.
Asterisk is compiled with h323 support, and is configured as a gateway
in the cisco callmanager.
The incoming call is in the g.729 format, you should be able to fix this
i
On 12/15/06, Pavel Jezek <[EMAIL PROTECTED]> wrote:
1) why you Answer() before Dial()
sorry, it is a my error
2) try Dial(SIP/user) instead of Dial(SIP/[EMAIL PROTECTED]) asterisk
knows, what IP has peer (sip show peers)
no, because the user isn't registered on asterisk server.
asterisk is
some idea, how to make BLF working on ci$co 7961 (sip)?
Steve Langstaff wrote:
I've not used the Cisco kit for this, but you might try adding 'hints'
to your agent extensions, and then defining a BLF button to subscribe to
this.
e.g. If you have an agent with ID 1001, add this to extensions.c
Hi John, I´m very interested into this call forwarding capabilities and
I solved this problem filtering on the web-script (in my case, php) the
number the user can intert on the database. (I know it´s not an asterisk
solution).
There is an issue that I couldn´t handle. When I forward the call,
The setvar command below works fine in iax.conf and in sip.conf but not here in
zaptel.conf. I need it to retrieve info from the AstDB. Advice is apreciated,
can't seem to find an answer.
; define channels
group=1
context=longdistance_users
signalling=fxo_ks ;FXO Sig for Phone
callerid="John Fr
Yuan LIU wrote:
I just didn't want to accept fxotune.c's claim about working only with
TDM. Several other users indicated that they were not able to tune
X100P. There's also a README.debian note that specifically indicated
exclusion of X100P.
fxotune is written to change register values
Hi John,
I would try to use on sip.conf and iax.conf and zapata.conf:
on every "user (friend or whatever)" defined add this:
[useraccount]
setvar=mycontext=yourcontext
--
This variable will become available for every user, so you ju
I am using a month old svn version of asterisk 1.2 . I have set
bindport=5091 for a sip peer ( type = friend) and nat=yes .. in sip show
peer it shows port 5091 for peer but asterisk isnt listening on port 5091 at
all . I tried both port=5091 as well as binport=5091 but asterisk does not
listen
I'm simply trying to forward calls to users who have the call forwarding
feature enabled (FWD Status and FWD Ph Number kept in the astDB). The
problem is that I want users to be able to forward calls to numbers that
they would normally be allowed to dial within their own context. (I
don't want
Asterisk seems to be ignoring DTMF for features in Asterisk 1.4.0b3
My SNOM sends the dtmf-relay and Asterisk gets it because I can
see it in the sip debug.
However, once seen, Asterisk doesn't actually do anything about it. I've
checked features and that seems fine. Is this a bug or something t
Pavel wrote:
> I think, callmanager needs media termination point (mtp) for
> sip trunk, so rtp stream will always go through callmanager...
That is true for CCM 4.X, so SIP works with CCM 4.X, but is
far from ideal. As of CCM 5.X added RFC 2833 support to the
SCCP endpoints, so a MTP is not req
1) why you Answer() before Dial()
2) try Dial(SIP/user) instead of Dial(SIP/[EMAIL PROTECTED]) asterisk
knows, what IP has peer (sip show peers)
3) try call echo() test aplication from callmanager phone
nik600 wrote:
On 12/15/06, Pavel Jezek <[EMAIL PROTECTED]> wrote:
in h323.conf you have
> adding g in your dial application and the call will go on the extension
> when the callee hangup
Yes, i could also use "h" extension, but how to know which one hangup first
? ${HANGUPCAUSE} always say "16" (Normal clearing), and ${CHANNEL} is set to
the current channel in the dial plan...
Greg
I will be out the office on vacation.
>>> "asterisk-users@lists.digium.com" 12/15/06 11:25 >>>
On Friday 15 December 2006 10:52 am, Eric "ManxPower" Wieling wrote:
> DACS is not done in Asterisk. DACS is done in the Zaptel drivers. In
> fact, you can do DACS without Asterisk even being install
On Friday 15 December 2006 10:52 am, Eric "ManxPower" Wieling wrote:
> DACS is not done in Asterisk. DACS is done in the Zaptel drivers. In
> fact, you can do DACS without Asterisk even being installed on the
> system. Also the channels that are DACS'd are not even accessible to
> Asterisk.
You
Andrew Kohlsmith wrote:
On Friday 15 December 2006 10:21 am, Eric "ManxPower" Wieling wrote:
Hardware cross connect is called DACS and is not done by Asterisk.
Asterisk does support DACS with Zaptel TDM boards. I know that this is done
on-card with the multiport TExxx boards, but I'm not sur
Gregory Duchatelet a écrit :
Hi,
Using AMI or dial plan, how can i know which leg (channel ?) of a
bridged call, hangup ?
AMI send 2 hangup events, which have both cause 16 (normal clearing),
and the first hangup event is the called leg hangup event, not the one
who hangup…
Greg
---
On Friday 15 December 2006 10:21 am, Eric "ManxPower" Wieling wrote:
> Hardware cross connect is called DACS and is not done by Asterisk.
Asterisk does support DACS with Zaptel TDM boards. I know that this is done
on-card with the multiport TExxx boards, but I'm not sure if the TDM4xx/24xx
boar
Jonathan k. Creasy wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Eric "ManxPower" Wieling
Sent: Thursday, December 14, 2006 4:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hardwa
[EMAIL PROTECTED] wrote:
Is this only possible in a hard configuration way? or is it possible
that asterisk handles the call and tell the zap channel, now you have to
connect this 2 zap channels cross connect to each other?
You can tell Asterisk to connect any channel to any channel. This i
On 12/15/06, Pavel Jezek <[EMAIL PROTECTED]> wrote:
in h323.conf you have still:
disallow=all
allow=all
try change to:
disallow=all
allow=alaw
i've tried but it gives me the same error...
-- Got SIP response 606 "Not Acceptable" back from 193.x.x.x
__
On Fri, 2006-12-15 at 11:09 -0800, Khaled wrote:
> Dear
>
> Please how can I make a local dns naptr on my system ,ro resolve local
> calls using enum
http://www.oreilly.com/catalog/dns4/
Regards,
Patrick
___
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in h323.conf you have still:
disallow=all
allow=all
try change to:
disallow=all
allow=alaw
nik600 wrote:
On 12/15/06, Pavel Jezek <[EMAIL PROTECTED]> wrote:
probably you haven't g729 installed in asterisk, use g711 instead, put
this in h323.conf and in callmanager place asterisdk g
On 12/15/06, Pavel Jezek <[EMAIL PROTECTED]> wrote:
probably you haven't g729 installed in asterisk, use g711 instead, put
this in h323.conf and in callmanager place asterisdk gateway in region
that will use g711...
disallow=all
allow=alaw
alternatively you can find g729 codecs binaries here:
ht
Hi all,
I'm using 1.2.9.1, with the "metermaid" patches to show parking spot
status on Snom BLF lights.
I see from http://www.asterisk.org/node/97 that the metermaid code has
changed substantially since 1.2.9.1.
Is anyone successfully using the "new" metermaid functionality in 1.4.x?
I'd l
use contex
On 12/15/06, Nigel Kendrick <[EMAIL PROTECTED]> wrote:
Hi Folks,
Can you point me towards some info on how to specify that certain extensions
use specific outbound trunks - we can only set outbound caller ID against
the SIP accounts managed by our service provider (we cannot pass CID
I think, callmanager needs media termination point (mtp) for sip trunk,
so rtp stream will always go through callmanager...
JR Richardson wrote:
Hi All,
I haven't started sip traces or debug yet, but was wondering what the deal
is with the CCM and reinvite, why it doesn't work with Asteri
Hi all,
I am using Asterisk 1.2.10 on Debian Sarge and currently I am rewriting my
extensions.conf with ael.
The replacement of the following part makes me mad:
[set-language]
exten => _X./_0031.,1,Set(incoming_call=1|lang=nl)
exten => _X./_0031.,2,Goto(incoming,${EXTEN},1)
exten => _X./_0049.,1
probably you haven't g729 installed in asterisk, use g711 instead, put
this in h323.conf and in callmanager place asterisdk gateway in region
that will use g711...
disallow=all
allow=alaw
alternatively you can find g729 codecs binaries here:
http://kvin.lv/pub/Linux/Asterisk/
nik600 wrote:
H
Dear all,
I'm trying to receive a call from a VoIP provider to my Asterisk which
is behind a router, with a port forwarding (5060).
This configuration has already been validated with another VoIP
provider, but in the present case, not.
I suppose (thanks to the sip trace) my asterisk is not able
Hi All,
I haven't started sip traces or debug yet, but was wondering what the deal
is with the CCM and reinvite, why it doesn't work with Asterisk (using
1.2.9.1). I can make calls back and forth all day with canreinvite=no, when
I try to reinvite an inbound sip call from the CCM with Asterisk
On Friday 15 December 2006 4:18 am, Apesys wrote:
> exten =>
> s,1,ParkAndAnnounce(call-parked-at:PARKED|30|PAGE(LOCAL/[EMAIL PROTECTED]
>o page&LOCAL/[EMAIL PROTECTED]|)
why not Local/[EMAIL PROTECTED], and then have something like this:
[group_page]
exten => ,1,Dial(SIP/555)
exten => ,1
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Eric "ManxPower" Wieling
> Sent: Thursday, December 14, 2006 4:30 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Hardware TDM Switching
>
I'm using asterisk blind/attended transfer feature on a queue (also tried
with sip phones feature), and both type of transfers work fine. The problem
is that attended trasfers doesn't get logged to queue_log, but blind
transfers are logged just fine. Anyone knows if this is the correct
behavior?
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Eric "ManxPower" Wieling
> Sent: Thursday, December 14, 2006 4:30 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Hardware TDM Switching
Hello,
we are using asterisk in combination with the voicemail system. I´m just
wondering if it is possible to switch the voicemail to an "I am on
holiday" mode.
This means that the unavailable message is played to the caller but no
possability to record a message.
So far I did not find an optio
Hi
i am trying to do the same thing:
receive a call from a cisco callmanager and forward it to a SIP user.
Asterisk is compiled with h323 support, and is configured as a gateway
in the cisco callmanager.
h323.conf:
[general]
port = 1720
bindaddr = 193.x.x.x ; this SHALL contain a single,
They provided DIDs too. It was not that straight forward, but I've figured
it out how to use them.
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Hi,
Using AMI or dial plan, how can i know which leg (channel ?) of a bridged
call, hangup ?
AMI send 2 hangup events, which have both cause 16 (normal clearing), and
the first hangup event is the called leg hangup event, not the one who
hangup.
Greg
use dundi
Best regards,
Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email
(VoIP PBX) 1-866-638-1254
For Information on PBX Systems for SOHO
http://www.bochterservices.com/?j=PBX&t=email
Need A Toll Free Number?
http://www.bochterservices.com/?t=TFdid&t=email
For new and use
Is this only possible in a hard configuration way? or is it possible that
asterisk handles the call and tell the zap channel, now you have to
connect this 2 zap channels cross connect to each other?
Thanks
Nico
PS: Sorry, but there is NO info about this on voip-info.org
On Thu, 14 Dec 200
Hi Folks,
Can you point me towards some info on how to specify that certain extensions
use specific outbound trunks - we can only set outbound caller ID against
the SIP accounts managed by our service provider (we cannot pass CID info to
them at the moment although they are promising this facility
I've not used the Cisco kit for this, but you might try adding 'hints'
to your agent extensions, and then defining a BLF button to subscribe to
this.
e.g. If you have an agent with ID 1001, add this to extensions.conf (or
equivalent)...
exten => 1001,hint,Agent/1001
[Sorry it's the third time I send this message as I couldn't see it in the
list. I hope it will not come three times].
Hi everybody.
It is possible to announce the parking position through a paging to a group
of extensions?
I would like that when someone parks a call, some phones will anno
Dear
Please how can I make a local dns naptr on my system ,ro resolve local calls
using enum
Regards
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