Use an existing dialer like ViciDialer?
l.
2009/10/14 kaustuva...@bbsr.syscomes.com
Hello,
I am using Asterisk 1.4 version.
How to dial multiple numbers per second through asterisk manager
Thanks and regards
--
Loway - home of QueueMetrics - http://queuemetrics.com
On 15/10/09 4:42 AM, Faheem wrote:
Through Asterisk AMI, you can not dial multiple number at the same time.
If you are going to implement a concurrent call scenario, then AMI would
not be a valid choice. Multiple calls can be implemented with callfile.
Totally incorrect.
We do hundreds of
Hello.
I have a problem with Asterisk, sometimes it hangs up an external call after 20
seconds, apparently without any reason.
The call comes from a SIP server hosted from EuteliaVoIP, many peers rangs and
one of them answer, the call ends itself after 20 seconds from the answer.
I've tried
Hi
We've had this a few times and never got to the bottom of exactly why it
happens but stopped it by upgrading the phone and the router it was
going through to the latest firmware versions.
Hope that helps
Ish
Gianni Fioretta wrote:
Hello.
I have a problem with Asterisk, sometimes it
Elliot Otchet elliot.otc...@callingcircles.com writes:
Have you tried autopause=yes in your queue configuration? You can then
unpause the member by either the dialplan (e.g. having the cell phone
user log back in) or using an AMI based program to change the
paused state.
You can read more
Lenz Emilitri lenz.lo...@gmail.com writes:
You could configure them as agents and have them log off automatically
after a while they're not responding.
Agents have to log in and wait for calls though, don't they? There used
to be AgentCallbackLogin, but that has been replaced by dialplan code
--- On Thu, 10/15/09, Gianni Fioretta gianni.fiore...@yetopen.it wrote:
I have a problem with Asterisk, sometimes it hangs up an
external call after 20 seconds, apparently without any
reason.
The call comes from a SIP server
Hi,
This may or may not apply to your case:
Hi
i test a new equipment on my backbone: a Cisco AS5300 with voice dsp
ressource
connected at a E1 Voice Link.
I want that all call incoming on the cisco 5300 are sent to Asterisk and
all Asterisk outgoing
call are sent to Cisco AS5300.
Actually, i configure the AS5300:
isdn switch-type
plz do not send for me e-mail
thanks
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Gianni Fioretta wrote:
Hello.
I have a problem with Asterisk, sometimes it hangs up an external call after
20 seconds, apparently without any reason.
The call comes from a SIP server hosted from EuteliaVoIP, many peers rangs
and one of them answer, the call ends itself after 20 seconds
Hello. To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
On 10/15/09, as asd sa11...@yahoo.com wrote:
plz do not send for me e-mail
thanks
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Hi,
Where can I find doc related to IMAP storage.
Usually, config options can be found either in voicemail.conf or
voip-info.org but almost none relates to IMAP configuration.
At the moment, I'm looking for data related to imapflags possible values.
Regards
That shouldn't be too hard to accomplish. If you've got the addons (and mysql)
installed you could store them in a MySQL table (timestamp, device) and have a
cron job set to run at X frequency that un-pauses the queue members via AMI.
Don't want to go to MySQL? Use system() to 'touch' files
Elliot Otchet elliot.otc...@callingcircles.com writes:
That shouldn't be too hard to accomplish. If you've got the addons
(and mysql) installed you could store them in a MySQL table
(timestamp, device) and have a cron job set to run at X frequency that
un-pauses the queue members via AMI.
I don't have any experience with E1, but here are some comments from
the T1 perspective (on a 2800 series Cisco). Here is also a link to
my collection of Cisco voice debugging commands:
http://thurmantech.com/node/5
On Thu, Oct 15, 2009 at 3:27 AM, Phibee Network Operation Center
n...@phibee.net
Where can I find doc related to IMAP storage.
Usually, config options can be found either in voicemail.conf or
voip-info.org but almost none relates to IMAP configuration.
At the moment, I'm looking for data related to imapflags possible values.
More or less everything I know I found on this
Hi,
Here and there, I can mentions to Asterisk Desktop Assistant versions 1.1 or
Pro but I can't find any place to download or buy it.
Any help ?
Regards
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Thanks very much, it worked as I needed :)
On Wed, 14 Oct 2009 17:14:53 +0530, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
hey In 1.6 version actually not wrote any code for option 'o'
you need to add following line into file
Index: apps/app_chanspy.c
On Thu, Oct 15, 2009 at 6:27 AM, Phibee Network Operation Center
n...@phibee.net wrote:
dial-peer voice 10 voip
destination-pattern .T
session protocol sipv2
session target ipv4:IP_OF_ASTERISK:5060
session transport udp
dtmf-relay rtp-nte
codec g711alaw
no vad
!
dial-peer voice 42
Hi,
I am planning to deploy an Asterisk PBX for 100-200 users. I am not
sure about PSTN incoming/outgoing line ratio for SIP users. I mean if
you recall dial up internet the common line ratio is 1:10 (one line
for 10 users on access server or an E1 for 300 users). Can somebody
tell me
We have several offices running Asterisk version 1.4.20.1, and OSLEC
with Rhino R4FXO-EC and one running a Digium TDM800P card for interface
to analog lines. All offices are running Snom 300 phones. Phones all
have static addresses and are on the same physical network as the server.
The
I need for asterisk to call me at a predetermined number once a day at a
predetermined time and once connected to me make 5-10 simultanious calls to
a DID filling all available channels. What is the best way to do this?
Eric
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Shahnawaz Mir wrote:
I am planning to deploy an Asterisk PBX for 100-200 users. I am not
sure about PSTN incoming/outgoing line ratio for SIP users. I mean if
you recall dial up internet the common line ratio is 1:10 (one line
for 10 users on access server or an E1 for 300 users). Can
On Thu, 15 Oct 2009, Shahnawaz Mir wrote:
I am planning to deploy an Asterisk PBX for 100-200 users. I am not sure
about PSTN incoming/outgoing line ratio for SIP users. I mean if you
recall dial up internet the common line ratio is 1:10 (one line for 10
users on access server or an E1 for
On Thu, 15 Oct 2009, Eric Fort wrote:
I need for asterisk to call me at a predetermined number once a day at a
predetermined time and once connected to me make 5-10 simultanious calls
to a DID filling all available channels. What is the best way to do
this? Eric
What's best for you may
- Steve Edwards asterisk@sedwards.com wrote:
On Thu, 15 Oct 2009, Shahnawaz Mir wrote:
I am planning to deploy an Asterisk PBX for 100-200 users. I am not
sure
about PSTN incoming/outgoing line ratio for SIP users. I mean if you
recall dial up internet the common line ratio is
Shahnawaz Mir wrote:
Hi,
I am planning to deploy an Asterisk PBX for 100-200 users. I am not
sure about PSTN incoming/outgoing line ratio for SIP users. I mean if
you recall dial up internet the common line ratio is 1:10 (one line
for 10 users on access server or an E1 for 300 users).
2009/10/15 John A. Sullivan III jsulli...@opensourcedevel.com
Hello, all. I have a user who needs to monitor their voice mail box and
the general delivery voice mail box. I defined them in sip.conf as
follows:
[tkeeley](a10f)
mailbox=...@a10, 6...@a10
From memory, I could successfully
2009/10/15 Matthew Harrell lists-sender-6a8...@bittwiddlers.com
Where can I find doc related to IMAP storage.
Usually, config options can be found either in voicemail.conf or
voip-info.org but almost none relates to IMAP configuration.
At the moment, I'm looking for data related to
On Thu, 2009-10-15 at 19:24 +0200, Olivier wrote:
2009/10/15 John A. Sullivan III jsulli...@opensourcedevel.com
Hello, all. I have a user who needs to monitor their voice
mail box and
the general delivery voice mail box. I defined them in
sip.conf as
Just a thought... If the SNOM has multiple lines, tying one to 612 and the
other to 610 should make the MWI active for both lines. Asterisk AFAIK only
actives the first entry in the list, so you would need two entries for
tkeeley with mailbox=612 in the first instance and mailbox=610 in the
Thanks Tim,
Your response is really helpful. Its not going to be very busy. I was
expecting 10:1 but I will start some where between 4-10. Thank you
very much.
Regards
Shahnawaz Mir
On 15-Oct-09, at 11:11 AM, Tim Nelson wrote:
- Steve Edwards asterisk@sedwards.com wrote:
On Thu,
No, Extenspy was introduced in 1.4 as far as I know.
Chanspy is simple :) Helpful as I am, I'm gonna paste here the output of show
application chanspy
callcenter*CLI show application ChanSpy
callcenter*CLI
-= Info about application 'ChanSpy' =-
[Synopsis]
Listen to a channel, and optionally
On Thu, 2009-10-15 at 10:52 -0400, Matthew Harrell wrote:
Where can I find doc related to IMAP storage.
Usually, config options can be found either in voicemail.conf or
voip-info.org but almost none relates to IMAP configuration.
At the moment, I'm looking for data related to imapflags
Ah, interesting. I wasn't aware that it only used the first value.
What's the purpose of the secondary values then? If I understand you
correctly, you are saying I should have one entry for tkeeley with two
entries for mailbox=? Thanks - John
On Thu, 2009-10-15 at 12:54 -0500, Danny Nicholas
The secondary value is used, just not by the MWI functionality.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A.
Sullivan III
Sent: Thursday, October 15, 2009 1:13 PM
To: Asterisk Users Mailing List -
No, I'm saying you need two tkeeley entries with one mailbox each. The
multiple entry is fine for other mailbox functionality, just not MWI.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A.
Sullivan III
At 3:37 PM on 15 Oct 2009, Benny Amorsen wrote:
Perhaps the problem could be restated in a different way: After a
queue member rejects a call (instead of just not answering), the
queue should wait X amount of time before sending the next call.
Queues.conf has a million settings, but I can't
Hello,
Does anyone know how to test the timing device?
I've tried the following but with no luck.
Zaptel is installed.
I'm trying to use ztdummy as a timer.
[r...@templateasteriskserver ~]# dahdi_test
Unable to open dahdi interface: No such file or directory
You have to check and verify the SIP trunk details, as ext to ext works once
the pbx is up, but to call out, it should go through your provider.so
just recheck your provider's details.
Regards
Sandesh
On Wed, Oct 14, 2009 at 8:24 AM, Ott Rose sixfourimp...@hotmail.com wrote:
here is the
At 11:32 AM on 15 Oct 2009, C. Chad Wallace wrote:
At 3:37 PM on 15 Oct 2009, Benny Amorsen wrote:
Perhaps the problem could be restated in a different way: After a
queue member rejects a call (instead of just not answering), the
queue should wait X amount of time before sending the
You need to do /etc/init.d/dahdi start or /etc/init.d/zaptel start to load
the devices or dummy devices
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Thursday, October 15, 2009 1:58 PM
To: Asterisk
Hello,
I´ve found information about NVFax, app_fax, NVBackgroundDetect, rxfax, etc
But which is the best way for *detecting fax in Asterisk 1.6*???
I will use it in an automatic dialer.
Thank you very much,
Pablo Bernasconi
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Hi All,
We are trying to implement a DS3 capacity calls (672 concurrent calls) using
asterisk server. I wanted to ask are there any compatible DS3 cards with
asterisk? I tried searching a lot but could find DS3000P from digium but
unable to get this product. Does anybody have any idea of having
Hi,
I've downloaded for a demo, a P0S3-08-12.zip file which is suppose to work
with 7960.
Is it supposed to be the same file that the one needed to 7942 model ?
At the moment, my 7942 is blocked when trying to download a
P0S3-8-12-00.loads file.
Regards
das sandesh sandesh...@gmail.com wrote:
Hi All,
We are trying to implement a DS3 capacity calls (672 concurrent calls) using
asterisk server. I wanted to ask are there any compatible DS3 cards with
asterisk? I tried searching a lot but could find DS3000P from digium but unable
to get
Hi Danny,
I've tried that but I get the following errors:-
[r...@templateasteriskserver ~]# /etc/init.d/dahdi start
Loading DAHDI hardware modules:
FATAL: Module dahdi not found.
wct4xxp: FATAL: Module wct4xxp not found.
What does /etc/dahdi/modules look like? I suspect that it has each of the
wc* entries in it. If so, remove those lines and put in dummy (just
once).
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent:
Hello All,
I have a need for a wireless solution and have been looking at the
Aastra 57i CT phone that have the wireless handset with them. Aastra
says they will cover up to 300,000 square feet.
I am finding this hard to accept. I was also wondering about the
secure WDCT cordless technology
Ok, its a little better now.
But I still get a fatal message:-
[r...@templateasteriskserver dahdi]# /etc/init.d/dahdi start
Loading DAHDI hardware modules:
FATAL: Module dahdi not found.
dummy: [ OK ]
Any ideas?
Thanks
Dan
On Thu, 15 Oct 2009, Shahnawaz Mir wrote:
I am planning to deploy an Asterisk PBX for 100-200 users. I am not
sure about PSTN incoming/outgoing line ratio for SIP users. I mean if
you recall dial up internet the common line ratio is 1:10 (one line
for 10 users on access server or an E1
Hello.
I've been setting up an Asterisk server, and I am now supposed to move
it to a different network than the one it was set on.
I'd like to give the server 2 IP address:
-1- The first IP address is the IP it will have on the LAN, meaning that
softphones will register to the Asterisk
Try this thread
http://forums.digium.com/viewtopic.php?p=132042sid=297f2470a0a3d87e91efc1a5
9defcab9
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Thursday, October 15, 2009 3:02 PM
To: Asterisk Users
At 12:50 PM 10/15/2009, you wrote:
I have a need for a wireless solution and have been looking at the
Aastra 57i CT phone that have the wireless handset with them. Aastra
says they will cover up to 300,000 square feet.
I am finding this hard to accept. I was also wondering about the
secure WDCT
Yes it is possible, the only thing that you need to do is to configure
correctly your network routes, if your ip devices are on the same net of
your elastix you wont need to do any route configuration.
Just leave the default gateway for your wan provider, it should work
without any trouble
On
The 57i and 480i are good wireless phones but after 100ft you are out of
range (assuming business interiors). Of you still have to deal with buggy
firmware(and hit and miss tech support).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
- Steve Edwards asterisk@sedwards.com wrote:
- Steve Edwards asterisk@sedwards.com wrote:
42[:1]
(The fact that you ask such a generic question implies you have a
high
probability of failure. You should hire somebody with a bit more
clue
and learn from them.)
On Thu, 15 Oct 2009, das sandesh wrote:
We are trying to implement a DS3 capacity calls (672 concurrent calls)
using asterisk server. I wanted to ask are there any compatible DS3
cards with asterisk? I tried searching a lot but could find DS3000P from
digium but unable to get this product.
Jorge Gutiérrez a écrit :
Yes it is possible, the only thing that you need to do is to configure
correctly your network routes, if your ip devices are on the same net of
your elastix you wont need to do any route configuration.
Just leave the default gateway for your wan provider, it should
Hi Men,
I believe that .T is anything + a Time out of (probably) 3 sec. before to
dial the complete called number.
Best Regards,
Francois
destination-pattern .T
What does destination-pattern .T mean? I'm not familiar with what
.T would match. I would suggest using a more specific pattern
No usable DS3 cards for Asterisk. There is a standing consensus, as far
as I've been able to tell (and I could be wrong), that this would be
rather difficult - if not impossible - to do given the liberal timing
tolerance of PCI buses and PC architecture once you're talking about
that much
On Thu, Oct 15, 2009 at 4:58 PM, David Backeberg dbackeb...@gmail.com wrote:
There's no one-step solution I'm aware of. Cisco sells something
called an AS5300 that supposedly can terminate a DS3 and convert it
all to SIP. Otherwise, you need a channel bank like the Adtran MX2800
I was close,
We're also working fine with it but I also do not know what the
available imapflags are and what they mean. I have seen notls and
novalidatecert. Out of curiosity, I spent the last 20 minutes googling
for information on c-client imapflags and didn't find any definitions or
even a simple
On Thu, Oct 15, 2009 at 3:20 PM, das sandesh sandesh...@gmail.com wrote:
Hi All,
We are trying to implement a DS3 capacity calls (672 concurrent calls) using
asterisk server. I wanted to ask are there any compatible DS3 cards with
asterisk? I tried searching a lot but could find DS3000P from
David Backeberg wrote:
On Thu, Oct 15, 2009 at 4:58 PM, David Backeberg dbackeb...@gmail.com wrote:
There's no one-step solution I'm aware of. Cisco sells something
called an AS5300 that supposedly can terminate a DS3 and convert it
all to SIP. Otherwise, you need a channel bank like the
Customer has 2 call manager systems and I am using asterisk to place
calls through the CCM.
One for the main use - CCMMAIN and another for disaster CCMSLAVE.
Can asterisk be setup in such a way that calls first try to use CCMMAIN
and if thats not available use CCMSLAVE.
Example if I place a
Here are two ways to address this
1. Dial(SIP/CCMMAINSIP/CCMSLAVE) - this tries both at once
2. exten = s,1,Dial(SIP/CCMMAIN,10,KkTt)
Exten = s,n,Dial(SIP/CCMSLAVE,10,KkTt)
CCMSLAVE only gets called if no one answers CCMMAIN in 10 seconds (2-3
rings)
-Original Message-
From:
On Oct 14, 2009, at 1:04 AM, Dan Journo wrote:
Thanks Eric,
I'd love to be able to make it to an Astricon one day. At the
moment, its a bit out of my price range.
Do you happen to know whether RackspaceCloud.com offers a Kernel
with a timing device enabled?
Many thanks and good
Based on interest expressed at AstriCon, we've published Asterisk and
FreePBX Amazon EC2 instances in Europe (previously they were only
available in the U.S. region).
More information is available at:
http://voxilla.com/2009/10/15/asterisk-on-the-cloud-with-a-click-1405
Does anybody know here I can find old Sipura firmware 2.0.13 for SPA-3000
I have Cisco 3.1.20 but it is not working as it suppose to.
--
Joseph
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On Wed, 2009-10-14 at 22:41 -0400, John A. Sullivan III wrote:
Hello, all. I have a user who needs to monitor their voice mail box
and
the general delivery voice mail box. I defined them in sip.conf as
follows:
[tkeeley](a10f)
mailbox=...@a10, 6...@a10
I think you've got the syntax
On Thu, Oct 15, 2009 at 08:42:22PM +0100, Dan Journo wrote:
Hi Danny,
I've tried that but I get the following errors:-
[r...@templateasteriskserver ~]# /etc/init.d/dahdi start
Loading DAHDI hardware modules:
FATAL: Module dahdi not found.
The dahdi kernel modules are not
Here are two ways to address this
1. Dial(SIP/CCMMAINSIP/CCMSLAVE) - this tries both at once
2. exten = s,1,Dial(SIP/CCMMAIN,10,KkTt)
Exten = s,n,Dial(SIP/CCMSLAVE,10,KkTt)
CCMSLAVE only gets called if no one answers CCMMAIN in 10 seconds (2-3
rings)
Danny thats good to know for
On Thu, 2009-10-15 at 15:29 -0700, Jared Smith wrote:
On Wed, 2009-10-14 at 22:41 -0400, John A. Sullivan III wrote:
Hello, all. I have a user who needs to monitor their voice mail box
and
the general delivery voice mail box. I defined them in sip.conf as
follows:
[tkeeley](a10f)
On Wed, 2009-10-14 at 22:56 -0400, John A. Sullivan III wrote:
Hello, all. I've got a problem where we set up call pickup for a
customer. If the Bob's extension rings and Bob is in Jim's office, Bob
can press the button on his Snom 320 that says Bob and pick up his
line. It works great for
I sent a query about this before, but have some further information and am
hoping somebody has a suggestion as to what to try next to debug this.
I'm using an Asterisk box primarily for MeetMe conferencing. There are
two sources: TDM via two Q.SIG T1's and SIP phones. Conferencing works
fine
Hi,
As you may know by now, yesterday on the Astricon the City of
Amsterdam presented their large scale asterisk deployment of
2 phones. Because they do not allow brand names to be used
within the city, they call it 'IP Business Manager', but the
software they use is in fact the Astium PBX,
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