*All,
To meet the target for the month, we are running a special promotion.
$5 activation fee waived for all new DID purchases.*
Buy DIDs from DIDForSale http://www.didforsale.com/ today and *your $5
activation fees will be WAIVED* for all the DIDs purchased before July 20
2009.
There is no
Sincere Apologies--
Send the mail to wrong list, Meant to send to asterisk-biz list.
-J
On Wed, Jul 8, 2009 at 11:35 PM, Jai Rangi jpra...@gmail.com wrote:
*All,
To meet the target for the month, we are running a special promotion.
$5 activation fee waived for all new DID purchases.*
Buy
If you create a peer definition and put the host address in it and the
context you want it to go to you should be fine
Cheers Duncan
David Klaverstyn wrote:
Hi All,
I never saw a reply to this question. Is anyone able to assist?
Regards
David.
*From:*
Hi List
I've a CID lookup hooked onto an inbound route (i m using trixbox) ...it runs
well but it returns the value as CIDNAMECIDNUMBER ... if i just want to
display the CIDNAME [leaving the quotes and CIDNUMBER] .. how can i do it ?
do i have to edit some macro in extensions.conf ?
rgds
Hi,
As my extensions.ael is becoming quite long (3000 lines), I'm wondering if
existing indentation tools
such as vim, indent, ... could improve its formatting (split long lines into
several ones, align {}, ..)
Has anyone tried ?
Regards
___
--
On Thu, Jul 9, 2009 at 12:21 AM, Miguel Molinammol...@millenium.com.co wrote:
Christian Gansberger escribió:
Hi all!
I want to autopause my queue member when they are not answering within
20 seconds, and the autopause
should affect all queues they are member of, not only the queue where
the
Have you tried Fring?
It's a softphone software for mobile phones
http://www.fring.com/
Ish
Nick Hill wrote:
Thank you for the info
Does anyone know if the cdc-modem interface which is available on mobile
phones
can actually potentially be used to initiate, or register for receiving a
Hello Ishfaq
I have used Fring, but I don't believe it is capable of initiating GSM calls
from VOIP.
As I understand it, Fring does
VOIP---Data bearer ---Fring-Microphone/speaker
(wifi, 3G data)
I am proposing
\|/
|
3G/GSM---Asterisk
the bit can be achieved now using
Hi all,
I´m a beginner with asterisk and I want to know if with asterisk I can send
sms to a mobile, I´m on Spain, and I don´t know this can be a problem (with
the operators...)
I have Elastix 1.3.2 and I have seen this url:
http://mirror.su.lt/voip-info/wiki/view/Asterisk+cmd+Sms.html
I have
Hi,
I've got a problem with rtp keepalives. I'm using basically the same
config on 2 hosts, but one of them sends rtp comfort noise when it's
on hold, the other doesn't. The only difference I can think of now is
that one of the machines is multihomed, but that might be unrelated.
rtpkeepalive is
On Thu, Jul 9, 2009 at 3:01 AM, Sriramd_r_sri...@hotmail.com wrote:
Hi List
I've a CID lookup hooked onto an inbound route (i m using trixbox) it
runs well but it returns the value as CIDNAMECIDNUMBER ... if i just
want to display the CIDNAME [leaving the quotes and CIDNUMBER] .. how can
Hi
use www.kannel.org
On Thu, Jul 9, 2009 at 3:26 PM, ESGLinux esggru...@gmail.com wrote:
Hi all,
I´m a beginner with asterisk and I want to know if with asterisk I can send
sms to a mobile, I´m on Spain, and I don´t know this can be a problem (with
the operators...)
I have Elastix 1.3.2
Same question here : How about in Belgium ??
Because core show application like sms gives information about the UK.
Jonas.
On Thu, 2009-07-09 at 11:26 +0200, ESGLinux wrote:
Hi all,
I´m a beginner with asterisk and I want to know if with asterisk I can
send sms to a mobile, I´m on
Hi Sir,
I just want to confirm that it is located in zaptel.com or zapata.conf?
because i have find resetinterval in zapta.conf but not in zaptel.conf..
Thanks
_
cricket and news. Logon to MSN Video for the latest clips
Hi thanks for your answer,
I don´t want to install code in the machine with the asterisk, (I have tried
and I have dependencies that I can´t solve) so,
the real question is with the software that comes with my elastix release
can I send sms?
thanks again,
ESG
2009/7/9 Shahid Tel
Hi, all , hope u all are good and fine .
me getting new error which i am pasting below.. This will came when i am
reloading the asterisk. i also tried [reload chan_zap.so ] on asterisk cli,
then out is same as
i mention below, i there is any misconfiguration in zapata.conf?? i am posting
resetinterval=never in zapata.conf.
you may want to reset them though, just not as frequently. The
resetinterval can take an integer as well.
Thanks,
Steve Totaro
On Wed, Jul 8, 2009 at 9:35 AM, Aman Dhallyaman.dha...@live.com wrote:
Hi All,
Hope you all are fine and good, Today i have
Am Donnerstag, den 09.07.2009, 11:26 +0200 schrieb ESGLinux:
Hi all,
I´m a beginner with asterisk and I want to know if with asterisk I can
send sms to a mobile, I´m on Spain, and I don´t know this can be a
problem (with the operators...)
Hi,
the SMS code in Asterisk is - afaik - only
Bah, my mistake, as Steve said the entry goes in zapata.conf.
On 09/07/2009, Steve Totaro stot...@asteriskhelpdesk.com wrote:
resetinterval=never in zapata.conf.
you may want to reset them though, just not as frequently. The
resetinterval can take an integer as well.
Thanks,
Steve Totaro
2009/7/2 Administrator TOOTAI ad...@tootai.net
Carlos Ruiz Diaz a écrit :
Check chan_mobile.
[...]
Or use GSM gateway
Using a GSM gateway is possible but it's quite different as you need to
insert a SIM card inside to let it work.
___
--
2009/7/2 Carlos Ruiz Diaz carlos.ruizd...@gmail.com
Check chan_mobile. Now is mature enough to be used in a server with low
CPS.
The USB connectivity will be introduced in the close future (I think) but
by now it can be connected via bluetooth device.
Where did you get this info (USB
2009/7/9 Anselm Martin Hoffmeister ans...@hoffmeister-online.de
Am Donnerstag, den 09.07.2009, 11:26 +0200 schrieb ESGLinux:
Hi all,
I´m a beginner with asterisk and I want to know if with asterisk I can
send sms to a mobile, I´m on Spain, and I don´t know this can be a
problem (with
I read it in this list. I buit an application on top of chan_mobile
and i needed usb connectivity to improve the bandwidth so i googled
for the answer and one of the hits was from here.
On 7/9/09, Olivier oza-4...@myamail.com wrote:
2009/7/2 Carlos Ruiz Diaz carlos.ruizd...@gmail.com
Check
Hi All,
I've just upgraded our CRM and it has an Asterisk Integration Module
that I would like to test out.
The CRM is running on one of our hosted servers in the cloud. The
Asterisk server is running in my office.
I am running Asterisk 1.4.21.2~dfsg-1ubuntu3.
Reading the page
On 10/7/09 12:05 AM, Alan Lord (News) wrote:
Hi All,
I've just upgraded our CRM and it has an Asterisk Integration Module
that I would like to test out.
The CRM is running on one of our hosted servers in the cloud. The
Asterisk server is running in my office.
I am running Asterisk
While I agree with Steve on a philosophical level, there are a lot of
merits to command lines and direct editing of configuration, there also
comes a time when just getting the job done is benefited by a nice
point-n-click.
I have found in my career that I may spend a month neck deep in a
hi,
making may way through all this...internal sip registration works,(cant call
yet but anyhow)...
the asterisk box is obvisoulsy behind a router. im not 100% sure if i should
go with port forwarding or NAT and if a or b, what additional setup is
actually correct?
sip_nat.conf # this is when i
On 9 Jul 2009, at 13:05, Alan Lord (News) wrote:
Reading the page
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20manager.conf
got me a little concerned regarding having an open channel between the
two machines and there is scant information about setting up a more
secure
Just google/bing it. http://voip-info.org/wiki/view/chan_mobile
On Thu, Jul 9, 2009 at 12:56 PM, Olivier oza-4...@myamail.com wrote:
2009/7/2 Carlos Ruiz Diaz carlos.ruizd...@gmail.com
Check chan_mobile. Now is mature enough to be used in a server with low
CPS.
The USB connectivity will be
On 09/07/09 14:40, Steve Howes wrote:
On 9 Jul 2009, at 13:05, Alan Lord (News) wrote:
Reading the page
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20manager.conf
got me a little concerned regarding having an open channel between the
two machines and there is scant
Christian Gansberger escribió:
On Thu, Jul 9, 2009 at 12:21 AM, Miguel Molinammol...@millenium.com.co wrote:
Christian Gansberger escribió:
Hi all!
I want to autopause my queue member when they are not answering within
20 seconds, and the autopause
should affect all queues they are
Use CALLERID(name).
http://www.voip-info.org/wiki/view/Asterisk+func+callerid
Steve Totaro wrote:
On Thu, Jul 9, 2009 at 3:01 AM, Sriramd_r_sri...@hotmail.com wrote:
Hi List
I've a CID lookup hooked onto an inbound route (i m using trixbox) it
runs well but it returns the value as
ESGLinux wrote:
2009/7/9 Anselm Martin Hoffmeister ans...@hoffmeister-online.de
mailto:ans...@hoffmeister-online.de
Am Donnerstag, den 09.07.2009, 11:26 +0200 schrieb ESGLinux:
Hi all,
I´m a beginner with asterisk and I want to know if with asterisk
I can
Hi,
Some users have been reporting a peculiar problem.
The are having an issue when they dial out to some multi-level IVRs
where you make 2 or 3 touchtone choices and then are connected to a
live operator.
When the live operator connects the operator cannot hear them or
sometimes it results in
Hello Sasa
The page you point to doesn't talk about USB connectivity for chan_mobile. It
does talk about bluetooth connectivity, which can be achieved by way of a USB
bluetooth dongle, but that is not the same thing.
I am talking about using standard interfaces exposed by mobile devices
On Thu, Jul 9, 2009 at 1:13 PM, James Lamannajlama...@gmail.com wrote:
Hi,
Some users have been reporting a peculiar problem.
The are having an issue when they dial out to some multi-level IVRs
where you make 2 or 3 touchtone choices and then are connected to a
live operator.
When the live
Nick Hill escribio':
Hello Sasa
The page you point to doesn't talk about USB connectivity for chan_mobile. It
does talk about bluetooth connectivity, which can be achieved by way of a USB
bluetooth dongle, but that is not the same thing.
I am talking about using standard interfaces
Can someone tell me how to setup a Aastra 75i phone? I have been trying to set
it up and have pointed it to our asterisk server and selected http for
download. What is the path? I have created two extension in asterisk for
testing. I can't even get the phones to call each other.
I understand that standalone macros have been deprecated in 1.6 for
gosub routines. I've been working on converting them all but was
wondering about dial macros - it doesn't look like there's a replacement
yet to call a gosub routine from the dial command. Or am I looking at
this wrong?
hose
Hi all,
I need to test the following scenario:
+---+ +---+
| asterisk 1| | asterisk 2|
+---+ +---+
| |
| |
___|__|___
| |
|
It should be pretty simple. Follow the instructions on this page
http://www.voiptalk.org/products/aastra-setup.html
put the username from sip.conf into the first 4 fields, the secret into the
password field and your asterisk ip into the fields that say voiptalk.org
users.conf
[207]
Hello,
I've found a little documentation on voip-info and on the asterisk-
users list, although I was hoping for an example of a tried-and-true
failover setup between PRI and SIP.
We are an outgoing call center that uses asterisk 1.4 connected to 2
PRIs from the local telephone company in
I followed it the best I could. the phones say no service. I haven't got to
setting up the SIP trunk yet I was told I could get the extensions to work so I
could test between the two phones i have. I have to nics in my server. one is
connect to the phone router the other to a network switch.
What do you get from sip show peers in CLI? Do you have your ip address
in sip.conf?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose
Sent: Thursday, July 09, 2009 4:12 PM
To: asterisk-users@lists.digium.com
On Thursday 09 July 2009 14:13:28 Hose wrote:
I understand that standalone macros have been deprecated in 1.6 for
gosub routines. I've been working on converting them all but was
wondering about dial macros - it doesn't look like there's a replacement
yet to call a gosub routine from the dial
On Thu, Jul 9, 2009 at 4:37 PM, Jason Martin jmar...@metrixmatrix.comwrote:
Hello,
I've found a little documentation on voip-info and on the asterisk-
users list, although I was hoping for an example of a tried-and-true
failover setup between PRI and SIP.
We are an outgoing call center
On Thu, Jul 9, 2009 at 5:31 PM, Steve Totaro stot...@first-notification.com
wrote:
On Thu, Jul 9, 2009 at 4:37 PM, Jason Martin jmar...@metrixmatrix.comwrote:
Hello,
I've found a little documentation on voip-info and on the asterisk-
users list, although I was hoping for an example of a
On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose sixfourimp...@hotmail.com wrote:
I followed it the best I could. the phones say no service. I haven't got
to setting up the SIP trunk yet I was told I could get the extensions to
work so I could test between the two phones i have. I have to nics in my
Hi all,
I've just built a new installation of CentOS release 5.3 (Final) and
have installed both
http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.1.1.tar.gzAsterisk
1.6.1.1 and subsequently Asterisk 1.6.0.10 (thinking that I was maybe
trying to be too cutting edge)
I was wondering if (2) quad T1 cards
will work nicely in 1 server with a quad core AMD 3.0 gig cpu?
Basically used to dial out and deliver messages. play wav files for the
message.
Any thoughts.
Jerry
___
-- Bandwidth and Colocation Provided by
What you say...Tilghman Lesher (tilgh...@mail.jeffandtilghman.com):
On Thursday 09 July 2009 14:13:28 Hose wrote:
I understand that standalone macros have been deprecated in 1.6 for
gosub routines. I've been working on converting them all but was
wondering about dial macros - it doesn't
Jerry Geis wrote:
oh you mean a telemarketing pest server ?
I was wondering if (2) quad T1 cards
will work nicely in 1 server with a quad core AMD 3.0 gig cpu?
Basically used to dial out and deliver messages. play wav files for the
message.
Any thoughts.
Jerry
On Thu, 9 Jul 2009, Jerry Geis wrote:
I was wondering if (2) quad T1 cards will work nicely in 1 server with a
quad core AMD 3.0 gig cpu?
Basically used to dial out and deliver messages. play wav files for the
message.
Within the environment you described, yes.
But why would you want to
On Thu, Jul 9, 2009 at 6:48 PM, Steve Edwards asterisk@sedwards.comwrote:
On Thu, 9 Jul 2009, Jerry Geis wrote:
I was wondering if (2) quad T1 cards will work nicely in 1 server with a
quad core AMD 3.0 gig cpu?
Basically used to dial out and deliver messages. play wav files for the
This is not a telemarkeing machine.
Customer wants to be able to contact their own people with this
(I dont ask why)
I thought about a second machine and using SIP to connection back to the
server. Is that a better solution? have 1 card in the server and another
card in
another machine
On Thu, Jul 9, 2009 at 6:55 PM, Jerry Geis ge...@pagestation.com wrote:
This is not a telemarkeing machine.
Customer wants to be able to contact their own people with this
(I dont ask why)
I thought about a second machine and using SIP to connection back to the
server. Is that a better
On Thu, Jul 9, 2009 at 6:34 PM, Jerry Geisge...@pagestation.com wrote:
I was wondering if (2) quad T1 cards
will work nicely in 1 server with a quad core AMD 3.0 gig cpu?
Yes. Buy a server that has the corresponding ports to accommodate the
cards. A modern server is probably going to have PCI-E
David Backeberg wrote:
On Thu, Jul 9, 2009 at 6:34 PM, Jerry Geisge...@pagestation.com wrote:
I was wondering if (2) quad T1 cards
will work nicely in 1 server with a quad core AMD 3.0 gig cpu?
Yes. Buy a server that has the corresponding ports to accommodate the
cards. A modern
Hello!
I've been asked to get a show of hands for some analysts for users
in Higher Ed - Universities, Colleges, or any other 2 or 4 year degree-
granting institutions. If this fits you, please let me know your
contact data and briefly how you're using Asterisk, and if you don't
mind I
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