Re: [asterisk-users] Fax .pdf from Asterisk
James,
That is amazing details. I can use all of this. Thank you for sharing.
I am assuming you installed res_fax from repository?
*yum install asterisk18-res_fax_digium.i386*
And how did you install SpanDSP? Is there a guide you used?
I am aiming for multi-channels fax so the digium one won't do for me as
it's one channel limit like you mentioned. I probably don't need T.38 but
hey it won't hurt to have it.
Thanks again,
On Fri, May 4, 2012 at 12:37 AM, James Sharp wrote:
> On 5/3/12 9:16 PM, Bruce B wrote:
>
>> Lee,
>>
>> Much appreciated for the input.
>>
>> I am running all VoIP. SIP and IAX2 to our ITSPs. Please elaborate on
>> HylaFax and IAXmodems. Is there a guide posted on to get it running, or
>> is it part of the repository? Once installed how would one send .pdf as
>> fax?
>>
>
> You can either use the "Free Fax for Asterisk" single channel at a time
> fax system or "app_fax" that uses the SpanDSP library. I'm working on a
> similar system myself that uses the SpanDSP system because I could never
> get FFFA to talk T.38 right to my provider (Gafachi).
>
> I use spoolfiles to create a call that lands in my dialplan and from
> there, I can pick up the fax results in the dialplan.
>
>
> Here's my callfile
>
> Channel: SIP/1771655@gafachi1a
> CallerID: 18005551212
> MaxRetries: 0
> RetryTime: 60
> WaitTime: 30
> Context: faxout
> Extension: 5
> Priority: 1
>
>
> And my extensions file
>
> [faxout]
> exten => 5,1,SendFAX(/tmp/test.tiff,a)
> exten => 5,n,Noop(${LOCALSTATIONID});
> exten => 5,n,Noop(${LOCALHEADERINFO});
> exten => 5,n,Noop(${FAXSTATUS});
> exten => 5,n,Noop(${FAXERROR});
> exten => 5,n,Noop(${REMOTESTATIONID});
> exten => 5,n,Noop(${FAXPAGES});
> exten => 5,n,Noop(${FAXBITRATE});
> exten => 5,n,Noop(${FAXRESOLUTION});
>
> Note that I use the "a" option in SendFAX. That makes it change behaviors
> on negotiating things like T.38. It was needed for my provider, but it may
> not be needed for yours. You may or may not need/even have access to T.38
> faxing.
>
>
>
> You will have to convert the PDF to a TIFF file before you can send it
> with either Fax subsystem.
>
> gs -q -dNOPAUSE -dBATCH -sDEVICE=tiffg4 -sPAPERSIZE=letter
> -sOutputFile=/tmp/test.tiff test.pdf
>
> --
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>
> http://lists.digium.com/**mailman/listinfo/asterisk-**users
>
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Re: [asterisk-users] Fax .pdf from Asterisk
On 5/3/12 9:16 PM, Bruce B wrote:
Lee,
Much appreciated for the input.
I am running all VoIP. SIP and IAX2 to our ITSPs. Please elaborate on
HylaFax and IAXmodems. Is there a guide posted on to get it running, or
is it part of the repository? Once installed how would one send .pdf as fax?
You can either use the "Free Fax for Asterisk" single channel at a time
fax system or "app_fax" that uses the SpanDSP library. I'm working on a
similar system myself that uses the SpanDSP system because I could never
get FFFA to talk T.38 right to my provider (Gafachi).
I use spoolfiles to create a call that lands in my dialplan and from
there, I can pick up the fax results in the dialplan.
Here's my callfile
Channel: SIP/1771655@gafachi1a
CallerID: 18005551212
MaxRetries: 0
RetryTime: 60
WaitTime: 30
Context: faxout
Extension: 5
Priority: 1
And my extensions file
[faxout]
exten => 5,1,SendFAX(/tmp/test.tiff,a)
exten => 5,n,Noop(${LOCALSTATIONID});
exten => 5,n,Noop(${LOCALHEADERINFO});
exten => 5,n,Noop(${FAXSTATUS});
exten => 5,n,Noop(${FAXERROR});
exten => 5,n,Noop(${REMOTESTATIONID});
exten => 5,n,Noop(${FAXPAGES});
exten => 5,n,Noop(${FAXBITRATE});
exten => 5,n,Noop(${FAXRESOLUTION});
Note that I use the "a" option in SendFAX. That makes it change
behaviors on negotiating things like T.38. It was needed for my
provider, but it may not be needed for yours. You may or may not
need/even have access to T.38 faxing.
You will have to convert the PDF to a TIFF file before you can send it
with either Fax subsystem.
gs -q -dNOPAUSE -dBATCH -sDEVICE=tiffg4 -sPAPERSIZE=letter
-sOutputFile=/tmp/test.tiff test.pdf
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Re: [asterisk-users] Fax .pdf from Asterisk
Lee, Much appreciated for the input. I am running all VoIP. SIP and IAX2 to our ITSPs. Please elaborate on HylaFax and IAXmodems. Is there a guide posted on to get it running, or is it part of the repository? Once installed how would one send .pdf as fax? Thanks, Bruce On Thu, May 3, 2012 at 7:42 PM, Lee Howard wrote: > On 05/03/2012 01:28 PM, Bruce B wrote: > >> I want to send out 1000 faxes. I have an excel sheet of numbers and I >> have Asterisk 1.8 installed from repository. I don't want to use a fax >> machine or any ATAs or analogue equipment. How would Asterisk help me with >> faxing these? and what add-ons do I need to make this possible? >> > > Not interested in HylaFAX with IAXmodems? (I presume that you are using > PSTN circuits and not VoIP.) > > Thanks, > > Lee. > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax .pdf from Asterisk
On 05/03/2012 01:28 PM, Bruce B wrote: I want to send out 1000 faxes. I have an excel sheet of numbers and I have Asterisk 1.8 installed from repository. I don't want to use a fax machine or any ATAs or analogue equipment. How would Asterisk help me with faxing these? and what add-ons do I need to make this possible? Not interested in HylaFAX with IAXmodems? (I presume that you are using PSTN circuits and not VoIP.) Thanks, Lee. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI disconnects
On Thursday 03 May 2012 1:47:09 pm Paul Belanger wrote: > On 12-05-03 01:45 PM, Mike Diehl wrote: > > Hi all. > > > > I've got a perl script that connects to Asterisk's management interface > > using Asterisk::AMI. So far, its proven to be very useful. > > > > I'm hoping to use this to detect and respond to asterisk restarts and sip > > reloads. > > > > However, my script gets disconnected quite frequently, causing false > > alarms in my monitoring. > > When this scrip disconnects, the asterisk console simply says that the > > manager logged off from 127.0.0.1 and got logged back on. No reall > > error indication given. > > > > Any suggestions on how to make this script keep it's connection? > > This belongs on the asterisk-users list. I posted this to the asterisk-users list... Did I miss something? > However, something like StarPY > (Python) has this functionality by using the Twisted. It simply > monitors events on the AMI, and if the connect break, can be setup to > re-connect. My code doesn't have any problems getting re-connected in a timely fashion. The issue is that it's getting disconnected for no apparent reason! -- Take care and have fun, Mike Diehl. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fax .pdf from Asterisk
Hello, I want to send out 1000 faxes. I have an excel sheet of numbers and I have Asterisk 1.8 installed from repository. I don't want to use a fax machine or any ATAs or analogue equipment. How would Asterisk help me with faxing these? and what add-ons do I need to make this possible? I can work my way around doing bash script and do Asterisk spool files, but I am unclear as to what happens from that point on to getting the result of fax sent or not. Some guidance is much appreciated. Thanks, Bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI disconnects
On 12-05-03 03:47 PM, Paul Belanger wrote:
On 12-05-03 01:45 PM, Mike Diehl wrote:
Hi all.
I've got a perl script that connects to Asterisk's management
interface using Asterisk::AMI. So far, its proven to be very useful.
I'm hoping to use this to detect and respond to asterisk restarts and
sip reloads.
However, my script gets disconnected quite frequently, causing false
alarms in my monitoring.
Here's what the code looks like:
=
while ($more) {
$change = 0;
while (!$ami->connected(5)) {
Event::Shutdown(Event=>"Shutdown");
send_to_subscribers( {"Event" => "Shutdown", "customer_id" => 0},
"Connection to Asterisk lost on $server");
$change++;
sleep(1);
ami_connect();
next;
}
$watchdog = 10;
while ($ami->get_event(1)&& $watchdog--) {
if ($watchdog< 2) { print "Watchdog timer too short.\n"; }
$change++;
}
Do other stuff from here.
if ($change == 0) {
sleep(2);
}
}
exit 1;
=
When this scrip disconnects, the asterisk console simply says that the
manager logged off from 127.0.0.1 and got logged back on. No reall
error indication given.
Any suggestions on how to make this script keep it's connection?
This belongs on the asterisk-users list. However, something like StarPY
(Python) has this functionality by using the Twisted. It simply monitors
events on the AMI, and if the connect break, can be setup to re-connect.
Heh, apparently this is the asterisk-user list. Sorry, dealing with a
new mail client and misread my folder labels.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) |
Github: https://github.com/pabelanger | Twitter:
https://twitter.com/pabelanger
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Re: [asterisk-users] AMI disconnects
On 12-05-03 01:45 PM, Mike Diehl wrote:
Hi all.
I've got a perl script that connects to Asterisk's management interface using
Asterisk::AMI. So far, its proven to be very useful.
I'm hoping to use this to detect and respond to asterisk restarts and sip
reloads.
However, my script gets disconnected quite frequently, causing false alarms in
my monitoring.
Here's what the code looks like:
=
while ($more) {
$change = 0;
while (!$ami->connected(5)) {
Event::Shutdown(Event=>"Shutdown");
send_to_subscribers( {"Event" => "Shutdown", "customer_id" => 0},
"Connection to Asterisk lost on $server");
$change++;
sleep(1);
ami_connect();
next;
}
$watchdog = 10;
while ($ami->get_event(1)&& $watchdog--) {
if ($watchdog< 2) { print "Watchdog timer too short.\n"; }
$change++;
}
Do other stuff from here.
if ($change == 0) {
sleep(2);
}
}
exit 1;
=
When this scrip disconnects, the asterisk console simply says that the manager
logged off from 127.0.0.1 and got logged back on. No reall
error indication given.
Any suggestions on how to make this script keep it's connection?
This belongs on the asterisk-users list. However, something like StarPY
(Python) has this functionality by using the Twisted. It simply
monitors events on the AMI, and if the connect break, can be setup to
re-connect.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) |
Github: https://github.com/pabelanger | Twitter:
https://twitter.com/pabelanger
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Re: [asterisk-users] Asterisk Vs FreeSWITCH for Fax
We handle tousands of faxes a day with asterisk 1.8 and spandsp but we dictate what ATA the customers can use. Most of the faxes come in and are processed as e-fax from asterisk.. It really comes down to how good are your black art skills when It comes to fax and source providers channels clean for fax. Thanks Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AMI disconnects
Hi all.
I've got a perl script that connects to Asterisk's management interface using
Asterisk::AMI. So far, its proven to be very useful.
I'm hoping to use this to detect and respond to asterisk restarts and sip
reloads.
However, my script gets disconnected quite frequently, causing false alarms in
my monitoring.
Here's what the code looks like:
=
while ($more) {
$change = 0;
while (!$ami->connected(5)) {
Event::Shutdown(Event=>"Shutdown");
send_to_subscribers( {"Event" => "Shutdown", "customer_id" =>
0}, "Connection to Asterisk lost on $server");
$change++;
sleep(1);
ami_connect();
next;
}
$watchdog = 10;
while ($ami->get_event(1) && $watchdog--) {
if ($watchdog < 2) { print "Watchdog timer too short.\n"; }
$change++;
}
Do other stuff from here.
if ($change == 0) {
sleep(2);
}
}
exit 1;
=
When this scrip disconnects, the asterisk console simply says that the manager
logged off from 127.0.0.1 and got logged back on. No reall
error indication given.
Any suggestions on how to make this script keep it's connection?
TIA.
--
Take care and have fun,
Mike Diehl.
--
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Re: [asterisk-users] Asterisk Capacity
Or you could use a System call in the hangup dialplan and trigger a new call as soon as an old one just finished. Maybe a silly idea but it shpuld just work. Ioan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Capacity
Ashish Agarwal писал 03.05.2012 18:53: > So what is a better approach to achieve this I've switched to AMI originate, call files did not satisfy me for some reasons. Besides, originating calls from script gives you full control on when does each call start ant stop, thus allowing you to manage your capacity instead of guessing if those 30 call files were all executed successfully. -- With Best Regards Mikhail Lischuk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Capacity
"easiest" way is "service asterisk status" or asterisk -rx "core show channels verbose" From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ashish Agarwal Sent: Thursday, May 03, 2012 11:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Capacity How can I check how many lines are currently being used? On May 3, 2012 9:23 PM, "Duncan Turnbull" wrote: Hi Ashish On 4/05/2012, at 3:41 AM, Ashish Agarwal wrote: Hello, We are currently working on a project where using .call file on asterisk spool, outbound calls will be made from a pri line and a voice clip will be played. We know that pri has a capacity of handling only 30 channels at a time. Therefore, my worry is what happens if we write 100 files at a time on the spool. Will asterisk manage the queue or how exactly will it behave. Only write 30 or less files at a time. Asterisk will try and send them all and they will fail. Just count how many files are in the spool dir and how many calls are active and stay below your limit Cheers Duncan Regards, Ashish -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Capacity
How can I check how many lines are currently being used? On May 3, 2012 9:23 PM, "Duncan Turnbull" wrote: > Hi Ashish > > On 4/05/2012, at 3:41 AM, Ashish Agarwal wrote: > > Hello, > > We are currently working on a project where using .call file on asterisk > spool, outbound calls will be made from a pri line and a voice clip will be > played. > > We know that pri has a capacity of handling only 30 channels at a time. > Therefore, my worry is what happens if we write 100 files at a time on the > spool. Will asterisk manage the queue or how exactly will it behave. > > Only write 30 or less files at a time. Asterisk will try and send them all > and they will fail. Just count how many files are in the spool dir and how > many calls are active and stay below your limit > > Cheers Duncan > > Regards, > > Ashish > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Capacity
If you set the ctime (or maybe mtime) of your spool file to a date in the future, then asterisk won't process the .call file until that future time. I recommend creating your call files with a random ctime/mtime for 0 - 240 seconds in the future and make sure you have a random retry time in your .call file. Also NEVER create your .call files in the asterisk outgoing directory -- asterisk might try processing the file before your script finishes writing the data. Create the file in a different directory on the same filesystem and move it to the outgoing directory. This is not a perfect solution, but it is "good enough" and worked well at the time. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, May 03, 2012 11:56 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk Capacity The "best" approach to this would be to have a "sender" that uses AMI to monitor channels and release .call files as channels become available. About 100 lines in PERL. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ashish Agarwal Sent: Thursday, May 03, 2012 10:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Capacity So what is a better approach to achieve this On May 3, 2012 9:20 PM, "Mitul Limbani" wrote: The other 70 will result into failure with .call file approach. Regards, Mitul Limbani, Chief Architech & Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-61447605 Cell: +91-9820332422 On Thu, May 3, 2012 at 9:11 PM, Ashish Agarwal wrote: Hello, We are currently working on a project where using .call file on asterisk spool, outbound calls will be made from a pri line and a voice clip will be played. We know that pri has a capacity of handling only 30 channels at a time. Therefore, my worry is what happens if we write 100 files at a time on the spool. Will asterisk manage the queue or how exactly will it behave. Regards, Ashish -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Capacity
The "best" approach to this would be to have a "sender" that uses AMI to monitor channels and release .call files as channels become available. About 100 lines in PERL. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ashish Agarwal Sent: Thursday, May 03, 2012 10:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Capacity So what is a better approach to achieve this On May 3, 2012 9:20 PM, "Mitul Limbani" wrote: The other 70 will result into failure with .call file approach. Regards, Mitul Limbani, Chief Architech & Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-61447605 Cell: +91-9820332422 On Thu, May 3, 2012 at 9:11 PM, Ashish Agarwal wrote: Hello, We are currently working on a project where using .call file on asterisk spool, outbound calls will be made from a pri line and a voice clip will be played. We know that pri has a capacity of handling only 30 channels at a time. Therefore, my worry is what happens if we write 100 files at a time on the spool. Will asterisk manage the queue or how exactly will it behave. Regards, Ashish -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Capacity
So what is a better approach to achieve this On May 3, 2012 9:20 PM, "Mitul Limbani" wrote: > The other 70 will result into failure with .call file approach. > > Regards, > Mitul Limbani, > Chief Architech & Founder, > Enterux Solutions Pvt. Ltd. > 110 Reena Complex, Opp. Nathani Steel, > Vidyavihar (W), Mumbai - 400 086. India > http://www.enterux.com/ > http://www.entvoice.com/ > email: mi...@enterux.in > DID: +91-22-61447605 > Cell: +91-9820332422 > > > > > On Thu, May 3, 2012 at 9:11 PM, Ashish Agarwal wrote: > >> Hello, >> >> We are currently working on a project where using .call file on asterisk >> spool, outbound calls will be made from a pri line and a voice clip will be >> played. >> >> We know that pri has a capacity of handling only 30 channels at a time. >> Therefore, my worry is what happens if we write 100 files at a time on the >> spool. Will asterisk manage the queue or how exactly will it behave. >> >> Regards, >> >> Ashish >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Capacity
In my experience the first 30 calls will go simulatenously, then the remainder will go as lines become available. You should use /g1 or /r1 to allow the call file to pick an open channel. Mitul is somewhat correct; all 100 calls will try to process at once, so the 70 “laggards” will have to re-process. The better solution would be to send the calls in chunks of 20 or 30, especially if you don’t specify async=yes and multiple tries. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mitul Limbani Sent: Thursday, May 03, 2012 10:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Capacity The other 70 will result into failure with .call file approach. Regards, Mitul Limbani, Chief Architech & Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-61447605 Cell: +91-9820332422 On Thu, May 3, 2012 at 9:11 PM, Ashish Agarwal wrote: Hello, We are currently working on a project where using .call file on asterisk spool, outbound calls will be made from a pri line and a voice clip will be played. We know that pri has a capacity of handling only 30 channels at a time. Therefore, my worry is what happens if we write 100 files at a time on the spool. Will asterisk manage the queue or how exactly will it behave. Regards, Ashish -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Vs FreeSWITCH for Fax
IAX Modem with Hylafax is a perfect combo as such, it just works !! Regards, Mitul Limbani, Chief Architech & Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-61447605 Cell: +91-9820332422 On Thu, May 3, 2012 at 9:20 PM, Duncan Turnbull wrote: > Hi Anita > > On 4/05/2012, at 12:27 AM, Anita Hall wrote: > > > Hi > > > > We are using Spandsp + FreeSWITCH for receiving Fax over T.30 E1/PRI and > the results make us sad :( > > > I am presuming you do mean T.30 (standard fax protocol but people don't > mention it much) not T.38 as I am not familiar with that so much. Whats > making you sad? Cost? performance, end customer compatibility? > > > I suppose Asterisk also has the option of using spandsp or a commercial > version from Commetrex. What are your experiences with receiving Fax on > spandsp or commetrex on Asterisk ? > > When I was doing fax I used Hylafax+ & asterisk & iaxmodem for sending / > receiving thousands of faxes using analogue lines, I would have thought E1 > would be almost perfect > > > > > Does it really matter whether I use Asterisk or FreeSWITCH ? > Cant see why it would but if you use iaxmodem then Asterisk is needed for > the IAX interface. > > Cheers Duncan > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Capacity
Hi Ashish On 4/05/2012, at 3:41 AM, Ashish Agarwal wrote: > Hello, > > We are currently working on a project where using .call file on asterisk > spool, outbound calls will be made from a pri line and a voice clip will be > played. > > We know that pri has a capacity of handling only 30 channels at a time. > Therefore, my worry is what happens if we write 100 files at a time on the > spool. Will asterisk manage the queue or how exactly will it behave. > Only write 30 or less files at a time. Asterisk will try and send them all and they will fail. Just count how many files are in the spool dir and how many calls are active and stay below your limit Cheers Duncan > Regards, > > Ashish > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Vs FreeSWITCH for Fax
Hi Anita On 4/05/2012, at 12:27 AM, Anita Hall wrote: > Hi > > We are using Spandsp + FreeSWITCH for receiving Fax over T.30 E1/PRI and the > results make us sad :( > I am presuming you do mean T.30 (standard fax protocol but people don't mention it much) not T.38 as I am not familiar with that so much. Whats making you sad? Cost? performance, end customer compatibility? > I suppose Asterisk also has the option of using spandsp or a commercial > version from Commetrex. What are your experiences with receiving Fax on > spandsp or commetrex on Asterisk ? When I was doing fax I used Hylafax+ & asterisk & iaxmodem for sending / receiving thousands of faxes using analogue lines, I would have thought E1 would be almost perfect > > Does it really matter whether I use Asterisk or FreeSWITCH ? Cant see why it would but if you use iaxmodem then Asterisk is needed for the IAX interface. Cheers Duncan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Capacity
The other 70 will result into failure with .call file approach. Regards, Mitul Limbani, Chief Architech & Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-61447605 Cell: +91-9820332422 On Thu, May 3, 2012 at 9:11 PM, Ashish Agarwal wrote: > Hello, > > We are currently working on a project where using .call file on asterisk > spool, outbound calls will be made from a pri line and a voice clip will be > played. > > We know that pri has a capacity of handling only 30 channels at a time. > Therefore, my worry is what happens if we write 100 files at a time on the > spool. Will asterisk manage the queue or how exactly will it behave. > > Regards, > > Ashish > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Capacity
Hello, We are currently working on a project where using .call file on asterisk spool, outbound calls will be made from a pri line and a voice clip will be played. We know that pri has a capacity of handling only 30 channels at a time. Therefore, my worry is what happens if we write 100 files at a time on the spool. Will asterisk manage the queue or how exactly will it behave. Regards, Ashish -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Vs FreeSWITCH for Fax
On 05/03/2012 10:35 PM, cjwstudios wrote: If you're going full time hosted fax you will ultimately end up buying a t.38/sip gateway like an Audiocodes Mediant. Many people handling hundreds of thousands of FAXes per day would disagree with that assessment. On Thu, May 3, 2012 at 5:27 AM, Anita Hall wrote: Hi We are using Spandsp + FreeSWITCH for receiving Fax over T.30 E1/PRI and the results make us sad :( I suppose Asterisk also has the option of using spandsp or a commercial version from Commetrex. What are your experiences with receiving Fax on spandsp or commetrex on Asterisk ? Does it really matter whether I use Asterisk or FreeSWITCH ? regards, Anita Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Vs FreeSWITCH for Fax
If you're going full time hosted fax you will ultimately end up buying a t.38/sip gateway like an Audiocodes Mediant. On Thu, May 3, 2012 at 5:27 AM, Anita Hall wrote: > Hi > > We are using Spandsp + FreeSWITCH for receiving Fax over T.30 E1/PRI and the > results make us sad :( > > I suppose Asterisk also has the option of using spandsp or a commercial > version from Commetrex. What are your experiences with receiving Fax on > spandsp or commetrex on Asterisk ? > > Does it really matter whether I use Asterisk or FreeSWITCH ? > > regards, > Anita > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hangup problem on T1 span
Tzafrir,
Thanks for your response. I'll check into those items.
Regards,
Stephen J Alexander
MPBX, LLC
http://mpbx.com
832-713-6729
On Thu, May 3, 2012 at 4:39 AM, Tzafrir Cohen wrote:
> On Wed, May 02, 2012 at 11:18:54AM -0500, Stephen J Alexander wrote:
> > Hello all,
> >
> > I'm trying to solve a problem on a T1 span setup wherein calls are
> > apparently not hanging up properly.
>
> CAS or PRI?
>
> >
> > The system in question is using a Xorcom Astribank with 1 full and 1
> > partial T1 span, and running Asterisk 1.4.36.
> >
> > The symptom is that when a call hangs up on a DAHDI channel (according to
> > Asterisk), and another outgoing call tries to open a new channel on the
> > same line as the hung-up call within approximately a minute of the
> hangup,
> > the new call gets a congestion notice ("all circuits busy") from
> > asterisk. After about a minute passes after the hangup, the line becomes
> > available again. So it seems like the channels are not hanging up when
> > Asterisk tells them to, and Asterisk doesn't know it.
> >
> > I suspected a signaling issue, and this appeared confirmed when I
> > discovered that the signalling was set in chan_dahdi.conf as "fxs_ks"
> (this
> > installation had been converted from analog lines by another company; I
> > guess that was an oversight?).
>
> The signalling and such is probably set in
> /etc/asterisk/dahdi-channels.conf so that setting does not matter.
>
> >
> > So I changed it to pri_cpe, as my reading of the docs indicated was
> proper.
> > After this change and restarting everything, though, the symptoms
> persist.
> > So I figure that either my reading of the docs is wrong (and therefore
> > pri_cpe is not the right signaling) OR something totally unrelated is
> going
> > on.
> >
> > Can someone please clue me in here? I am a bit at a loss. Let me know if
> > you need further information about the system/environment.
>
> What is the output of 'dahdi show channel N' for one such a "bad"
> channel when not in a call? Are you sure it's not in a call? See the
> output of 'core show channels'.
>
>
> --
> Tzafrir Cohen
> icq#16849755 jabber:tzafrir.co...@xorcom.com
> +972-50-7952406 mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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[asterisk-users] Asterisk Vs FreeSWITCH for Fax
Hi We are using Spandsp + FreeSWITCH for receiving Fax over T.30 E1/PRI and the results make us sad :( I suppose Asterisk also has the option of using spandsp or a commercial version from Commetrex. What are your experiences with receiving Fax on spandsp or commetrex on Asterisk ? Does it really matter whether I use Asterisk or FreeSWITCH ? regards, Anita -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] End-To-End Secured Communications
On 05/03/2012 07:17 AM, Fernando Berretta wrote: Hi, I'm analyzing how to make Asterisk communications secured End-To-End, and not sure which is the best approach, SRTP + TLS seems to be secured but.. at least by default, doesn't appear to be End-To-End allowing Asterisk administrators to wiretap communications.. some sites I've hear that with SRTP is also possible End Points exchange keys between them directly avoiding Man in the Middle, is it possible with asterisk ? how On the other hand I've found ZRTP seems to be secured end-to-end, but we couldn't find any IP phones with support for it.. just SoftPhones Could someone please point me to the right direction ? This is a fundamental architectural issue with all back-to-back User Agents used in SIP networks. They are pretty much by definition a 'man in the middle'. If they are used, the administrators will have access to call signaling and media for all calls passing through them. It is also important to realize that if you want end-to-end media security, then you would not be able to use any of Asterisk's features that involve media handling (transcoding, recording, whispering/spying, music-on-hold, conferencing, etc.) Given that, what you really want is a pure SIP proxy like Kamailio or OpenSIPs. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] End-To-End Secured Communications
Hi, I'm analyzing how to make Asterisk communications secured End-To-End, and not sure which is the best approach, SRTP + TLS seems to be secured but.. at least by default, doesn't appear to be End-To-End allowing Asterisk administrators to wiretap communications.. some sites I've hear that with SRTP is also possible End Points exchange keys between them directly avoiding Man in the Middle, is it possible with asterisk ? how On the other hand I've found ZRTP seems to be secured end-to-end, but we couldn't find any IP phones with support for it.. just SoftPhones Could someone please point me to the right direction ? Thanks, Fernando -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call-back php/agi script
Hello, Can anyone please suggest a simple call back script for asterisk? Thanks, Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hangup problem on T1 span
On Wed, May 02, 2012 at 11:18:54AM -0500, Stephen J Alexander wrote:
> Hello all,
>
> I'm trying to solve a problem on a T1 span setup wherein calls are
> apparently not hanging up properly.
CAS or PRI?
>
> The system in question is using a Xorcom Astribank with 1 full and 1
> partial T1 span, and running Asterisk 1.4.36.
>
> The symptom is that when a call hangs up on a DAHDI channel (according to
> Asterisk), and another outgoing call tries to open a new channel on the
> same line as the hung-up call within approximately a minute of the hangup,
> the new call gets a congestion notice ("all circuits busy") from
> asterisk. After about a minute passes after the hangup, the line becomes
> available again. So it seems like the channels are not hanging up when
> Asterisk tells them to, and Asterisk doesn't know it.
>
> I suspected a signaling issue, and this appeared confirmed when I
> discovered that the signalling was set in chan_dahdi.conf as "fxs_ks" (this
> installation had been converted from analog lines by another company; I
> guess that was an oversight?).
The signalling and such is probably set in
/etc/asterisk/dahdi-channels.conf so that setting does not matter.
>
> So I changed it to pri_cpe, as my reading of the docs indicated was proper.
> After this change and restarting everything, though, the symptoms persist.
> So I figure that either my reading of the docs is wrong (and therefore
> pri_cpe is not the right signaling) OR something totally unrelated is going
> on.
>
> Can someone please clue me in here? I am a bit at a loss. Let me know if
> you need further information about the system/environment.
What is the output of 'dahdi show channel N' for one such a "bad"
channel when not in a call? Are you sure it's not in a call? See the
output of 'core show channels'.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
--
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Re: [asterisk-users] realtime config for general settings in sip.conf
You need 2 but they can point to the same table sipusers => sippeers => You can get table definitions by downloading the source and then looking in the contrib/realtime/ directory Ish On Thu, 2012-05-03 at 04:56 +, Kamlesh Kumar wrote: > Hello, > > For realtime configuration, in /etc/asterisk/extconfig.conf file, what > should be the family name to configure general sip.conf parameters. > > => ,, > > thanks, > Kamlesh > > > > > From: i...@pack-net.co.uk > > To: asterisk-users@lists.digium.com > > Date: Wed, 2 May 2012 13:59:58 +0100 > > Subject: Re: [asterisk-users] realtime config for general settings > in sip.conf > > > > On Wed, 2012-05-02 at 12:04 +, Kamlesh Kumar wrote: > > > Hi, > > > > > > I need to configure global parameters in sip.conf like rtptimeout, > > > rtpholdtimeout, rtpkeepalive, domain, session-timers etc... in > real > > > time architecture. Please suggest the way to do it. > > > > > > thanks, > > > Kamlesh > > > > > > > Hi > > > > You can set defaults in the column definitions and you can still set > > globals in the sip.conf > > > > Ish > > > > -- > > Ishfaq Malik > > Department: VOIP Support > > Company: Packnet Limited > > t: +44 (0)845 004 4994 > > f: +44 (0)161 660 9825 > > e: i...@pack-net.co.uk > > w: http://www.pack-net.co.uk > > > > Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD > STREET > > NORTH, MANCHESTER > > SCIENCE PARK, MANCHESTER, M156SE > > COMPANY REG NO. 04920552 > > > > > > -- > > > _ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com > -- > > New to Asterisk? Join us for a live introductory webinar every > Thurs: > > http://www.asterisk.org/hello > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
