Re: [asterisk-users] faxdetect on/off on the fly?
Hi Danny, Can you please elaborate on how in the dialplan we can set faxdetect on and off? We currently have it set on in sip.conf. Thanks. On 3 January 2013 17:21, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Cunningham *Sent:* Thursday, January 03, 2013 3:13 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] faxdetect on/off on the fly? ** ** Hello, We want the ability to choose from an AGI script whether or not to enable faxdetect for calls over SIP or DAHDI. Is this possible, or can anyone suggest a workaround? Thanks for any advice. You should be able to call the AGI and set a dialplan variable and use Gotoif to do/not do faxdetect. Reading the .sample files for 11.0 it seems that normally these are “configured until restart/reload” but with a little testing, the default should be overrideable. -- _ -- 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 -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] faxdetect on/off on the fly?
Hello, We want the ability to choose from an AGI script whether or not to enable faxdetect for calls over SIP or DAHDI. Is this possible, or can anyone suggest a workaround? Thanks for any advice. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] faxdetect on/off on the fly?
Hi Steve, We have all calls going to an AGI, which decides where the number will get routed to, and if fax detection should be enabled for this call. The choice should only apply to the current call. Thanks very much. On 3 January 2013 17:46, Steve Edwards asterisk@sedwards.com wrote: On Thu, 3 Jan 2013, David Cunningham wrote: We want the ability to choose from an AGI script whether or not to enable faxdetect for calls over SIP or DAHDI. What's the 'use case?' You're going to call in and execute an AGI that will enable faxdetect for future calls to this channel or other channels? Should the 'change' survive an Asterisk restart or an OS reboot? -- Thanks in advance, --**--** - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- __**__**_ -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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 as a translating proxy only?
Hello, We want to use Asterisk as a proxy to translate between Skinny/SCCP and SIP, with as little as possible work required in between. Does Asterisk have a way for custom programs to read and write raw packets? If we can get the input data in a readable format and output it in the required format, that may do the trick. Thank you for any advice. -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] SendFAX timestamp
Steve, Thanks for the reply. Would anyone else know if Asterisk allows use of SpanDSP's time zone conversion? On 27 June 2012 00:24, Steve Underwood ste...@coppice.org wrote: On 06/26/2012 10:24 AM, David Cunningham wrote: Hello, Does SendFAX have the ability to put the caller ID and timestamp on the fax? If so, is there a way to adjust the timezone used for the timestamp? Thanks for any assistance. SpanDSP has that ability, including per instance time zones, but I don't know if the Asterisk module exposes that facility. 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] SendFAX timestamp
Hello, Does SendFAX have the ability to put the caller ID and timestamp on the fax? If so, is there a way to adjust the timezone used for the timestamp? Thanks for any assistance. -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] FastAGI script and DIAL execution
is ringing
-- SIP/193.17.66.71-004d is making progress passing it to
SIP/139255423-004c
-- SIP/193.17.66.71-004d answered SIP/139255423-004c
-- Executing [h@customer:1] Set(SIP/139255423-004c,
CDR(q931)=16) in new stack
-- Executing [h@customer:2] Set(SIP/139255423-004c,
CDR(userfield)={agi:,a-leg-id:2118d872-305e-4bb4-8c47-30e1514cb934,b-leg-id:36b232e73ac326bd0407b1594627c589@y.y.y.y:5060})
in new stack
SIP/139255423-004cAGI Tx 200 result=-1
SIP/139255423-004cAGI Tx HANGUP
SIP/139255423-004cAGI Rx GET VARIABLE HANGUPCAUSE
SIP/139255423-004cAGI Tx 200 result=1 (16)
SIP/139255423-004cAGI Rx GET VARIABLE Q16
SIP/139255423-004cAGI Tx 200 result=1 (0)
SIP/139255423-004cAGI Rx SET VARIABLE AJ_AGISTATUS SUCCESS
SIP/139255423-004cAGI Tx 200 result=1
-- SIP/139255423-004cAGI Script agi://localhost/auth completed,
returning 4
SIP/139255423-004cAGI Tx HANGUP
== Spawn extension (customer, 003462999, 8) exited non-zero on
'SIP/139255423-004c'
--
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Re: [asterisk-users] Continue AGI after Dial() following caller hang up?
Kingsley,
We have the same - the daemon forks child processes to handle individual
calls.
We need the fastAGI to continue so it can take some further action
recording details of the call. This could be done using the 'h' extension,
but it would be nice to avoid this method for simplicity sake. It does
appear that some people can continue after the Dial and we can't for some
reason.
On 22 November 2011 21:21, Kingsley Tart kings...@skymarket.co.uk wrote:
When something makes a socket connection to your fastAGI daemon, does
your daemon fork a child process to deal with that connection, or handle
it in the main process?
I've set ours up to fork a child process and detach itself from the
parent socket. When it ends, the child exits (which is what we want) and
the parent stays running (which is also what we want).
Is there any particular reason you want your fastAGI instance to persist
for the duration of the call?
Cheers,
Kingsley.
On Tue, 2011-11-22 at 14:27 +1100, David Cunningham wrote:
The strange thing is that we are using fast AGI, and for some reason
the AGI always exits when the caller hangs up - even when I set HUP to
IGNORE. If I set HUP to a subroutine that just logs a message, that
message is never logged.
Thanks for all the help.
On 22 November 2011 05:23, Kingsley Tart kings...@skymarket.co.uk
wrote:
Yeah fastAGI is great, I've been using it for a while for
performance
reasons but yes I guess it would solve problems like this too.
Cheers,
Kingsley.
On Mon, 2011-11-21 at 08:34 -0600, Danny Nicholas wrote:
Just offhand, I think you should utilize the FastAGI
protocol, since it
doesn't seem to live or die based on when the call hangs up.
Otherwise,
the
$SIG{'HUP'} = 'IGNORE';
Statement will separate the process so it doesn't die on a
hangup.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf
Of Kingsley Tart
Sent: Monday, November 21, 2011 7:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Continue AGI after Dial()
following caller
hang up?
Yeah I think I slightly misread your original question,
which I realised
when I saw Thorsten's reply. I initially thought you just
wanted to avoid
going into the h extension.
I'm not doing any AGI stuff here that hangs around while the
call does stuff
- the AGI process just runs quickly then quits, returning
control back to
the dialplan. I had incorrectly assumed you were doing the
same.
Cheers,
Kingsley.
On Mon, 2011-11-21 at 23:01 +1100, David Cunningham wrote:
Kingsley,
Thanks for the reply, but I am looking to continue within
the same AGI
process and I believe that method would require starting a
new AGI.
On 21 November 2011 22:22, Kingsley Tart
kings...@skymarket.co.uk
wrote:
We do that with the F option in Dial().
From http://www.voip-info.org/wiki/view/Asterisk
+cmd+Dial :
F(context^exten^pri): When the caller hangs up,
transfer the
called
party to the specified context and extension and
continue
execution.
Cheers,
Kingsley.
On Mon, 2011-11-21 at 17:38 +1100, David
Cunningham wrote:
Hello,
We would like to continue a Perl AGI after a
Dial() it has
done
completes following caller hangup. We would like
to do this
in the
same AGI, and not using a new AGI from the 'h'
extension. It
works
fine when the called party hangs up and the 'g'
option is
used, but
not for caller hangup.
Is this possible?
If not a confirmation that this is the case
would be very
helpful.
Thanks for any advice!
--
David Cunningham, Voisonics
http://voisonics.com
Re: [asterisk-users] Continue AGI after Dial() following caller hang up?
Thorsten, We have SIGHUP set to 'IGNORE', but it still does not continue the AGI after the Dial(). Do you have any idea why that might happen? Thanks for your advice. On 21 November 2011 22:19, Thorsten Göllner t...@ovm-group.com wrote: If the caller hangs up Asterisk sends a SIGHUP. You can catch the signal and do whatever you want to do. Am 21.11.2011 07:38, schrieb David Cunningham: Hello, We would like to continue a Perl AGI after a Dial() it has done completes following caller hangup. We would like to do this in the same AGI, and not using a new AGI from the 'h' extension. It works fine when the called party hangs up and the 'g' option is used, but not for caller hangup. Is this possible? If not a confirmation that this is the case would be very helpful. Thanks for any advice! -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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 -- Thorsten Göllner OVM Office Voice Media GmbH Herderstrasse 68 40237 Düsseldorf Tel.: +49(0)211 / 618 57 53 Fax: +49(0)211 / 618 57 54 -- _ -- 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 -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Continue AGI after Dial() following caller hang up?
Kingsley, Thanks for the reply, but I am looking to continue within the same AGI process and I believe that method would require starting a new AGI. On 21 November 2011 22:22, Kingsley Tart kings...@skymarket.co.uk wrote: We do that with the F option in Dial(). From http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial : F(context^exten^pri): When the caller hangs up, transfer the called party to the specified context and extension and continue execution. Cheers, Kingsley. On Mon, 2011-11-21 at 17:38 +1100, David Cunningham wrote: Hello, We would like to continue a Perl AGI after a Dial() it has done completes following caller hangup. We would like to do this in the same AGI, and not using a new AGI from the 'h' extension. It works fine when the called party hangs up and the 'g' option is used, but not for caller hangup. Is this possible? If not a confirmation that this is the case would be very helpful. Thanks for any advice! -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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 -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Continue AGI after Dial() following caller hang up?
The strange thing is that we are using fast AGI, and for some reason the
AGI always exits when the caller hangs up - even when I set HUP to IGNORE.
If I set HUP to a subroutine that just logs a message, that message is
never logged.
Thanks for all the help.
On 22 November 2011 05:23, Kingsley Tart kings...@skymarket.co.uk wrote:
Yeah fastAGI is great, I've been using it for a while for performance
reasons but yes I guess it would solve problems like this too.
Cheers,
Kingsley.
On Mon, 2011-11-21 at 08:34 -0600, Danny Nicholas wrote:
Just offhand, I think you should utilize the FastAGI protocol, since it
doesn't seem to live or die based on when the call hangs up. Otherwise,
the
$SIG{'HUP'} = 'IGNORE';
Statement will separate the process so it doesn't die on a hangup.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kingsley
Tart
Sent: Monday, November 21, 2011 7:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Continue AGI after Dial() following caller
hang up?
Yeah I think I slightly misread your original question, which I realised
when I saw Thorsten's reply. I initially thought you just wanted to avoid
going into the h extension.
I'm not doing any AGI stuff here that hangs around while the call does
stuff
- the AGI process just runs quickly then quits, returning control back to
the dialplan. I had incorrectly assumed you were doing the same.
Cheers,
Kingsley.
On Mon, 2011-11-21 at 23:01 +1100, David Cunningham wrote:
Kingsley,
Thanks for the reply, but I am looking to continue within the same AGI
process and I believe that method would require starting a new AGI.
On 21 November 2011 22:22, Kingsley Tart kings...@skymarket.co.uk
wrote:
We do that with the F option in Dial().
From http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial :
F(context^exten^pri): When the caller hangs up, transfer the
called
party to the specified context and extension and continue
execution.
Cheers,
Kingsley.
On Mon, 2011-11-21 at 17:38 +1100, David Cunningham wrote:
Hello,
We would like to continue a Perl AGI after a Dial() it has
done
completes following caller hangup. We would like to do this
in the
same AGI, and not using a new AGI from the 'h' extension. It
works
fine when the called party hangs up and the 'g' option is
used, but
not for caller hangup.
Is this possible?
If not a confirmation that this is the case would be very
helpful.
Thanks for any advice!
--
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019
--
_
-- Bandwidth and Colocation Provided by
http://www.api-digital.com --
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every Thurs:
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asterisk-users mailing list
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Thurs:
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--
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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--
Cheers,
Kingsley.
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New to Asterisk? Join us for a live
[asterisk-users] Continue AGI after Dial() following caller hang up?
Hello, We would like to continue a Perl AGI after a Dial() it has done completes following caller hangup. We would like to do this in the same AGI, and not using a new AGI from the 'h' extension. It works fine when the called party hangs up and the 'g' option is used, but not for caller hangup. Is this possible? If not a confirmation that this is the case would be very helpful. Thanks for any advice! -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Permanent sip and agi debug on?
Hi all, I can't find the answer to this via google - is there some way to permanently enable sip set debug on and agi set debug on in Asterisk? I want this to be automatically enabled even after restarts. Thanks for any advice. -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Permanent sip and agi debug on?
Kevin, Thank you very much! On 10 November 2011 00:15, Kevin P. Fleming kpflem...@digium.com wrote: On 11/09/2011 04:22 AM, David Cunningham wrote: Hi all, I can't find the answer to this via google - is there some way to permanently enable sip set debug on and agi set debug on in Asterisk? I want this to be automatically enabled even after restarts. In recent versions of Asterisk, you can put CLI commands into cli.conf and they will be run automatically when Asterisk starts. There are even examples of doing this for 'sip set debug' in cli.conf.sample :-) -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Voice recognition recommendations?
Hi all, We have a project involving voice recognition, and will need a vocabulary of 10,000 words (actually names). Can anyone recommend a product that works with Asterisk? Thanks, -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Change to pickups in Asterisk 1.8 - not working on local channels?
Hello all,
We have a customer who upgraded from Asterisk 1.6 to 1.8, and pickup groups
which previously worked fine have stopped working.
Can anyone advise if there has been a change in how pickups work?
Here is an example where 1000101 is trying to pick up a call to 1000103:
SIP/product-local-0005AGI Rx EXEC Dial
Local/1000103@product-pickup
/n,60,M(product-answered^0^1306286740.11)orL(360:6)
-- AGI Script Executing Application: (Dial) Options:
(Local/1000103@product-pickup
/n,60,M(product-answered^0^1306286740.11)orL(360:6))
Limit Data for this call:
timelimit = 360 ms (3600.000 s)
play_warning = 6 ms (60.000 s)
play_to_caller = yes
play_to_callee = no
warning_freq = 0 ms (0.000 s)
start_sound=
warning_sound = timeleft
end_sound =
-- Called 1000103@product-pickup/n
-- Executing [1000103@product-pickup:1]
Pickup(Local/1000103@product-pickup-db70;2, 1000103@product-phone) in
new stack
[May 25 11:25:40] NOTICE[1020]: app_directed_pickup.c:313 pickup_exec: No
target channel found for 1000103.
-- Auto fallthrough, channel 'Local/1000103@product-pickup-db70;2'
status is 'UNKNOWN'
The context doing the pickup looks like:
[product-pickup]
exten = _[0-9*#]!, 1, Pickup(${EXTEN}@product-phone)
Thanks for any advice,
--
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019
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Re: [asterisk-users] Higher CPU usage on 1.6.1 than 1.4?
Jared, Thank you for that information! Has anyone else had an experience like this? On 12 May 2011 20:25, Jared Geiger compuw...@gmail.com wrote: Hi David, When I was testing 1.6.1 for high volume channels, I couldn't get over 1000 channels / 40 CPS without the load average spiking up due to io wait. I switched back to 1.4 and I can go to 3000 channels / 75 CPS with no io wait and a load average in the 1s. It seemed like it was caused by the new timing system in 1.6.1 even though I wasn't proxying media using only SIP. I haven't tried 1.8 yet to see if it handles large call volumes any better. ~Jared On Wed, May 11, 2011 at 8:29 PM, David Cunningham dcunning...@voisonics.com wrote: Hello, We have a customer who upgraded from Asterisk 1.4 to 1.6.1.22 and is now experiencing higher CPU utilization on their server. I can't see anything wrong, so is this just expected with 1.6? Can anyone help explain it? Thanks for any advice. -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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 -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Higher CPU usage on 1.6.1 than 1.4?
Hello, We have a customer who upgraded from Asterisk 1.4 to 1.6.1.22 and is now experiencing higher CPU utilization on their server. I can't see anything wrong, so is this just expected with 1.6? Can anyone help explain it? Thanks for any advice. -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Error compiling Dahdi: invalid use of undefined type struct module
/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:5945: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:5951: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:5966: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:5976: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:6011: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:6018: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c: In function ‘dahdi_unregister’: /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:6049: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:6058: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:6062: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:6092: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c: In function ‘process_echocan_events’: /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:7092: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:7102: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c: In function ‘__putbuf_chunk’: /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:7594: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:7668: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:7810: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c: In function ‘dahdi_poll’: /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:8082: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c: In function ‘coretimer_func’: /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:8448: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c: In function ‘dahdi_receive’: /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:8559: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c: In function ‘dahdi_init’: /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:8712: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:8722: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:8723: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c: In function ‘dahdi_cleanup’: /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:8752: error: invalid use of undefined type ‘struct module’ make[2]: *** [/usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.o] Erreur 1 make[1]: *** [_module_/usr/src/dahdi-linux-2.4.0/drivers/dahdi] Erreur 2 make[1]: quittant le répertoire « /usr/src/linux-headers-2.6.34.6 » make: *** [modules] Erreur 2 -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Error compiling Dahdi: invalid use of undefined type struct module
Shaun, CONFIG_MODULES wasn't enabled - thanks for the advice! On Mon, Jan 31, 2011 at 4:02 PM, Shaun Ruffell sruff...@digium.com wrote: On 1/30/11 8:45 PM, David Cunningham wrote: I'm installing Asterisk with Dahdi on a server with a custom kernel compile. I've got the kernel source in /lib/modules/2.6.34.6--grs-ipv6-64/build which points to /usr/src/linux-headers-2.6.34.6 and I think that's fine, but am getting all these struct module errors. Can anyone advise? Thanks! # make make -C drivers/dahdi/firmware firmware-loaders make[1]: entrant dans le répertoire « /usr/src/dahdi-linux-2.4.0/drivers/dahdi/firmware » make[1]: quittant le répertoire « /usr/src/dahdi-linux-2.4.0/drivers/dahdi/firmware » make -C /lib/modules/2.6.34.6--grs-ipv6-64/build SUBDIRS=/usr/src/dahdi-linux-2.4.0/drivers/dahdi DAHDI_INCLUDE=/usr/src/dahdi-linux-2.4.0/include DAHDI_MODULES_EXTRA= HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m make[1]: entrant dans le répertoire « /usr/src/linux-headers-2.6.34.6 » CC [M] /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.o /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c: In function ‘dahdi_register_tone_zone’: /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:1440: error: invalid use of undefined type ‘struct module’ Normally this is the result of not having CONFIG_MODULES set in your kernel config. This is set when you check Enable loadable module support on the top level menu in menuconfig. Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 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 -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Introducing easySysAdmin - automated security and telecom fraud protection
Hello all, Voisonics is pleased to introduce easySysAdmin, an automated support/security platform, designed to save your engineer's time and prevent hacking attempts and telecom fraud. It comprises of an online service run by us, and a lightweight and easy-to-install client on your side. Specifically of interest to Asterisk users is the monitoring of SIP registrations, and automatic blocking of repeated failed attempts. This is done though a local application on your server so your call data is totally secure. In addition, blocked IP addresses are shared via the service so you can pre-emptively block addresses that others experienced attacks from. We offer a FREE one week no obligation trial, and during the month of January we also offer a DISCOUNT for Asterisk users. Just enter voucher asterisk01 at the checkout to receive 30% off. Protection is available from just US$84 per server with the discount. You can read more and give the free trial a go at: http://easysysadmin.com/ Our standard security package includes: - Monitor VoIP traffic (SIP registrations) and block attackers. - Watch remote server access (SSH logins) and block attackers. - Detect spam relay attempts (SMTP) and block attackers. - Scan of network ports to find vulnerabilities. - Check of software for vendor (distribution) security updates. - Custom monitoring of any TCP port or log file you want. - Flexible configuration to set warn and blocking levels. The background to this service is that prior to founding Voisonics I worked with IBM for 10 years, and became responsible for the security planning and audit compliance of many of IBM's voicemail and IVR platforms across the world. Using this experience and knowledge of their standards we have created easySysAdmin. If you have any questions please don't hesitate to contact me directly. Regards, -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] progressinband, how much extra CPU load?
Hi everyone, We have an Asterisk 1.4.17 user who has problems with sometimes not getting a ring tone on the calling phone. We're considering setting progressinband = yes, but would like to know how much extra CPU load this will require? If anyone can give something even roughly specific (eg 30% increase) that would be great, rather than just lots. Also, are there any ATAs which are known to not work with progressinband = yes? We have Polycom, Linksys and Audiocode. Thanks for any advice, -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] 'Bookmarking' a place in a sound file
Hi all, Is it possible to somehow 'bookmark' a place in a sound file? That is, the user presses a key while a sound file is playing and that point is saved, and some time in the future we can play the same sound file and tell it to start playing from that point. This would be done within a perl AGI program. Thanks for any advice! -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 -- _ -- 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] 'Bookmarking' a place in a sound file
Steve, that looks just the job, thank you very much. On Wed, Dec 8, 2010 at 2:32 AM, Steve Edwards asterisk@sedwards.comwrote: On Tue, 7 Dec 2010, David Cunningham wrote: Is it possible to somehow 'bookmark' a place in a sound file? That is, the user presses a key while a sound file is playing and that point is saved, and some time in the future we can play the same sound file and tell it to start playing from that point. This would be done within a perl AGI program. The AGI command 'stream file' will return 'endpos' when interrupted with a keypress. You could then save that in a channel variable or a database. A subsequent call to 'stream file' would include 'endpos' as the 'sample offset.' -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 -- _ -- 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] Recording maximum time and stop on silence
Danny, thank you! On Wed, Sep 22, 2010 at 10:31 PM, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Cunningham *Sent:* Wednesday, September 22, 2010 4:28 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Recording maximum time and stop on silence All, Two questions: 1. Is there a limit on how long a call can be recorded for? For example is 4 hours a problem? 2. Can recording be stopped after a configured period of silence? Thanks in advance, -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 AFAIK, #1 is limited only by available disk space, #2 is yes, but you may have to tweak some settings to “get it right” -- _ -- 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 -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 -- _ -- 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] Recording maximum time and stop on silence
All, Two questions: 1. Is there a limit on how long a call can be recorded for? For example is 4 hours a problem? 2. Can recording be stopped after a configured period of silence? Thanks in advance, -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 -- _ -- 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] Cause and cure for Exceptionally long voice queue length queuing to Local?
Leif - thank you! Will try that. On Fri, May 21, 2010 at 12:19 AM, Leif Madsen leif.mad...@asteriskdocs.org wrote: David Cunningham wrote: Hello, We're seeing lots of warnings like the following, running Asterisk 1.6.1.12. Does anyone know the cause or cure? One explanation I've come across is that the peer is congested and sending RTCP messages asking us to slow the RTP down. Is there any way we can verify this? [May 17 13:42:45] WARNING[27482] channel.c: Exceptionally long voice queue length queuing to Local/12126412...@asterisk-phone-7e3d;1 Thanks in advance! Try upgrading to 1.6.1.13. You're using a version of Asterisk from early December 2009. Doing a search for closed issues on the Asterisk issue tracker at https://issues.asterisk.org caused me to find bug 15609 (https://issues.asterisk.org/view.php?id=15609) which was committed on December 30, 2009. On January 11, 2010, Asterisk version 1.6.1.13-rc1 was created which contains the commit from December 30, 2009, and was subsequently released as 1.6.1.13 on January 15, 2010. The ChangeLog showing the commit is here: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.1.13 The release is available here: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ Leif. -- _ -- 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 -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 -- _ -- 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] Cause and cure for Exceptionally long voice queuelength queuing to Local?
I don't see anything in the SIP trace related to the warning messages. Would anyone have any further tips? Thanks for any help! On Wed, May 19, 2010 at 9:12 PM, David Cunningham dcunning...@voisonics.com wrote: What should I expect see if it is the peer asking us to slow down RTP? Thanks again. On Wed, May 19, 2010 at 9:05 PM, Danny Nicholas da...@debsinc.com wrote: Sip debug peer? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Cunningham Sent: Wednesday, May 19, 2010 3:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Cause and cure for Exceptionally long voice queuelength queuing to Local? Hello, We're seeing lots of warnings like the following, running Asterisk 1.6.1.12. Does anyone know the cause or cure? One explanation I've come across is that the peer is congested and sending RTCP messages asking us to slow the RTP down. Is there any way we can verify this? [May 17 13:42:45] WARNING[27482] channel.c: Exceptionally long voice queue length queuing to Local/12126412...@asterisk-phone-7e3d;1 Thanks in advance! -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 -- _ -- 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 -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 -- _ -- 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] Cause and cure for Exceptionally long voice queue length queuing to Local?
Hello, We're seeing lots of warnings like the following, running Asterisk 1.6.1.12. Does anyone know the cause or cure? One explanation I've come across is that the peer is congested and sending RTCP messages asking us to slow the RTP down. Is there any way we can verify this? [May 17 13:42:45] WARNING[27482] channel.c: Exceptionally long voice queue length queuing to Local/12126412...@asterisk-phone-7e3d;1 Thanks in advance! -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 -- _ -- 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] Cause and cure for Exceptionally long voice queuelength queuing to Local?
What should I expect see if it is the peer asking us to slow down RTP? Thanks again. On Wed, May 19, 2010 at 9:05 PM, Danny Nicholas da...@debsinc.com wrote: Sip debug peer? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Cunningham Sent: Wednesday, May 19, 2010 3:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Cause and cure for Exceptionally long voice queuelength queuing to Local? Hello, We're seeing lots of warnings like the following, running Asterisk 1.6.1.12. Does anyone know the cause or cure? One explanation I've come across is that the peer is congested and sending RTCP messages asking us to slow the RTP down. Is there any way we can verify this? [May 17 13:42:45] WARNING[27482] channel.c: Exceptionally long voice queue length queuing to Local/12126412...@asterisk-phone-7e3d;1 Thanks in advance! -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 -- _ -- 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 -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 -- _ -- 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] What does Asterisk give to reject a re-invite?
Hi Kevin, We don't have mohinterpret set at all, so I think it uses default. Is there anything else you can suggest? Any other places to go for help? Thanks for your assistance! On Thu, May 13, 2010 at 11:32 PM, Kevin P. Fleming kpflem...@digium.com wrote: On 05/13/2010 05:16 PM, David Cunningham wrote: We're having an issue where a peer's response to an INVITE includes a=sendonly. Later it sends a re-invite with a=sendrecv, however Asterisk responds to that with an OK that includes a=recvonly. The end result is the called party can't hear the caller. Do you have any idea why this is, or where I could go for more information? That would seem to indicate that the peer is placing Asterisk 'on hold', and then taking it back 'off hold' later. I do not know why Asterisk would respond with 'recvonly', it should only do that when it thinks the channel is still on hold. Are you using 'mohinterpret=passthrough', where Asterisk would send the hold indication to the bridged channel instead of reacting to it locally? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com 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 -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 -- _ -- 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] What does Asterisk give to reject a re-invite?
Hello, If you have canreinvite=no and a peer sends you a re-invite, what will Asterisk reply with? Thanks, -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 -- _ -- 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] What does Asterisk give to reject a re-invite?
Kevin, Thank you for that reply! We're having an issue where a peer's response to an INVITE includes a=sendonly. Later it sends a re-invite with a=sendrecv, however Asterisk responds to that with an OK that includes a=recvonly. The end result is the called party can't hear the caller. Do you have any idea why this is, or where I could go for more information? Thanks for the help. On Thu, May 13, 2010 at 11:06 PM, Kevin P. Fleming kpflem...@digium.com wrote: On 05/13/2010 01:41 PM, David Cunningham wrote: If you have canreinvite=no and a peer sends you a re-invite, what will Asterisk reply with? It will accept it. 'canreinvite' is mis-named, and that's why in more modern versions of Asterisk it has been renamed to 'directmedia'. Asterisk will *always* accept properly formed re-INVITEs that don't require capabilities that are not available, and it will also generate them for non-directmedia purposes (like switching to and from T.38) when necessary, regardless of whether 'canreinvite' is set to yes or no. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com 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 -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 -- _ -- 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] One way audio problem, a=sendonly and a re-invite
Hello all, I have a problem where problem with one way audio, and I think it's related to a=sendonly and a re-invite. Can anyone please assist? The scenario is as follows - We send an INVITE to a peer, and it replies with a 100 Trying, and then a 183 Session Progress message containing a=sendonly. - Asterisk plays the caller music on hold, which I believe is correct if we have an a=sendonly. - Then the peer sends a 200 OK which also has a=sendonly, and then sends a re-invite which I've copied and pasted below. - We have canreinvite=no set in sip.conf, but I'm not sure if we should be rejecting this re-invite or not because it does contain a=sendrecv. If it should be rejected what error should Asterisk return, and how can we establish two way audio? - After this re-invite Asterisk replies with a 100 Trying and then a 200 OK which contains a=recvonly. - Call is established but called party cannot hear caller. Here's the re-invite message - note that Asterisk is on port 5070: U 2010/05/05 12:47:38.139701 (peer):5060 - (asterisk):5070 INVITE sip:(called number)@(asterisk):5070 SIP/2.0. Via: SIP/2.0/UDP (peer):5060;branch=z9hG4bK2sansay7330954rdb6594. To: User sip:(called number)@(asterisk):5070;tag=as3ddcc528. From: sip:(called number)@(peer):5060;tag=sansay7330954rdb6594. Call-ID: 58eb52aa414c5e465c3c1a15603093fb@(asterisk). CSeq: 2 INVITE. Contact: sip:(called number)@(peer):5060. Max-Forwards: 69. Content-Type: application/sdp. Content-Length: 297. . v=0. o=Sansay-VSXi 188 1 IN IP4 (peer). s=Session Controller. c=IN IP4 (other unknown IP, maybe of called number?). t=0 0. m=audio 6932 RTP/AVP 18 0 8 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=ptime:20. Any help would be much appreciated! -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 -- _ -- 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] How to use AGI php script function $agi - exec_dial
You might find this helpful: http://www.voip-info.org/wiki/view/Asterisk+AGI+php Regards, On Mon, Jan 11, 2010 at 2:19 AM, Zhang Shukun bit...@gmail.com wrote: hi, i want to use $agi - exec_dial() to dial . this is in extention.conf [tutorial] exten = 1234,1,Dial(SIP/ivan) is that i use $agi - exec_dial(SIP,tutorial|1234|1) can dial ? BTW, i want to know some turorial on how to use PHPAGI funtions? can you tell me some? Thanks! -- Best regards, Sucan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham Voisonics Limited IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Really Silly Question From Total Newbie
___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham Voisonics Limited IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't load cdr_radius.so module?
I'm not familiar with cdr_radius, but is there a debugging option? Anything in /var/log/messages? On Thu, Dec 24, 2009 at 1:35 AM, Zhang Shukun bit...@gmail.com wrote: Thank you ! i have load cdr_radius.so successfully! but another error occur. -- Executing [4...@tutorial:1] Dial(SIP/ivan-0a07dc80, SIP/test) in new stack -- Called test -- SIP/test-0a08b0f0 is ringing -- SIP/test-0a08b0f0 answered SIP/ivan-0a07dc80 -- Packet2Packet bridging SIP/ivan-0a07dc80 and SIP/test-0a08b0f0 [Dec 24 09:30:32] ERROR[10747]: cdr_radius.c:227 radius_log: Failed to record Radius CDR record! == Spawn extension (tutorial, 4321, 1) exited non-zero on 'SIP/ivan-0a07dc80' it says Failed to record Radius CDR record. Could you tell me , what's wrong with it? 2009/12/23 Olle E. Johansson o...@edvina.net: 23 dec 2009 kl. 11.25 skrev David Cunningham: Shukun, It tells you No such file or directory. Is the file in your modules directory? Actually, to be more specific. The module cdr_radius.so exists, but can't bind to the radius library libradiusclient-ng.so.2. Check LD_LIBRARY_PATH /O On Wed, Dec 23, 2009 at 10:09 AM, Zhang Shukun bit...@gmail.com wrote: hi , all when i do the command module load cdr_radius.so ,error happens. i have installed radiusclient-ng , what's wrong with it? thanks! error message as follow: ZHANGSHUKUN*CLI module load cdr_radius.so Unable to load module cdr_radius.so Command 'module load cdr_radius.so' failed. [Dec 23 17:55:41] WARNING[31072]: loader.c:380 load_dynamic_module: Error loading module 'cdr_radius.so': libradiusclient-ng.so.2: cannot open shared object file: No such file or directory [Dec 23 17:55:41] WARNING[31072]: loader.c:730 load_resource: Module 'cdr_radius.so' could not be loaded. -- Regards, Sucan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- o...@edvina.net - http://edvina.net Open Unified Communication - building platforms with SIP and XMPP From PBX to large scale implementations for carriers. Contact us today! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Sucan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Connect my Asterisk to external sip?
It looks to me like calls from your Dial will route back to the sip-outgoing context and Dial again... it's loop. You'd really need to provide more logging information to advise further. On Thu, Dec 24, 2009 at 5:16 AM, hadi motamedi motamed...@gmail.com wrote: On Wed, Dec 23, 2009 at 1:55 PM, David Cunningham dcunning...@voisonics.com wrote: AsteriskWin32 does have SIP server functionality, same as the linux version. I can't think of any reason why having your CentOS Asterisk be both client and server and register with itself wouldn't work. Although I am wondering how much help all this will be in debugging a connection problem to another SIP provider... On Wed, Dec 23, 2009 at 12:55 PM, hadi motamedi motamed...@gmail.comwrote: On Wed, Dec 23, 2009 at 10:22 AM, David Cunningham dcunning...@voisonics.com wrote: Hadi, You could use Asterisk as a sip server, it's installable on Windows. Using sip set debug on might help you with the Host '192.168.0.139' does not implement 'REGISTER' problem. On Wed, Dec 23, 2009 at 9:12 AM, hadi motamedi motamed...@gmail.comwrote: On Sat, Dec 19, 2009 at 1:33 PM, Fred Posner f...@teamforrest.comwrote: On Dec 19, 2009, at 7:57 AM, hadi motamedi wrote: On Sat, Dec 19, 2009 at 12:07 PM, Fred Posner f...@teamforrest.com wrote: On Dec 19, 2009, at 6:51 AM, hadi motamedi wrote: Dear All I have an application that calls for my Asterisk sip to be connected to an external sip server for voip routing . Please be informed that my Asterisk sip is at @192.168.0.2 and the external sip is at @ 192.168.0.139 . To this end , I modified my sip.conf extensions.conf as the followings : Under sip.conf : - [general] register = toronto:welc...@192.168.0.139/osaka [osaka] type=friend secret=welcome context=osaka_incoming host=dynamic disallow=all allow=alaw [6672019] type=friend host=dynamic context=phones Try this: [general] register = toronto:welc...@osaka [osaka] type=friend username=toronto authname=toronto secret=welcome context=osaka_incoming host=192.168.0.139 disallow=all allow=alaw Although your error shows the other server does not allow register. What is the other server? ---fred http://qxork.com Thank you for your reply . The other server is not an Asterisk sip server . It is a sip server inside a softswitch from a third party vendor . As the external sip server man is asking me to disable for the authentication at the first stage , can you please let me know how can I disable for the authentication at this stage (when the calls get through I will enable it again) ? Thank you in advance [general] ;register = toronto:welc...@osaka [osaka] type=friend ;username=toronto ;authname=toronto ;secret=welcome context=osaka_incoming host=192.168.0.139 disallow=all allow=alaw ---fred http://qxork.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thank you for your reply . Please be informed that I want to simulate this case in the Laboratory , i.e. connecting my Asterisk sip to external sip server with the guidelines you sent me . Can you please propose for an Voip application sw that I can install on my MS Windows client and plays the external sip server side role ? It seems that Skype is not suitable for this case as it cannot be configured to play the role of external sip server . Thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I downloaded installed the AsteriskWin32 PBX but it doesn't have sip server functionality . Can you please propose for an alternative to be used on the MS Windows client as external sip server for my Asterisk on CentOS ? Thank you ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll
Re: [asterisk-users] 1.6 Troubleshooting help
It looks like whatever is being transmitted, or the response, isn't getting through. Possibly due to NAT or a firewall? It would help if you described the scenario where this is occurring. On Thu, Dec 24, 2009 at 7:18 AM, listu...@spamomania.co.uk listu...@spamomania.co.uk wrote: Hi, How would I go about troubleshooting this: [Dec 24 07:15:11] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission a50346a4-bfdc3...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:12] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaae...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:13] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaae...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:14] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaae...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:15] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission a50346a4-bfdc3...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:16] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaae...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:19] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission a50346a4-bfdc3...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:20] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaae...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:22] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission a50346a4-bfdc3...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:24] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaae...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:26] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission a50346a4-bfdc3...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:28] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaae...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:30] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission a50346a4-bfdc3...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:32] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaae...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:35] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission a50346a4-bfdc3...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Connect my Asterisk to external sip?
Hadi, You could use Asterisk as a sip server, it's installable on Windows. Using sip set debug on might help you with the Host '192.168.0.139' does not implement 'REGISTER' problem. On Wed, Dec 23, 2009 at 9:12 AM, hadi motamedi motamed...@gmail.com wrote: On Sat, Dec 19, 2009 at 1:33 PM, Fred Posner f...@teamforrest.com wrote: On Dec 19, 2009, at 7:57 AM, hadi motamedi wrote: On Sat, Dec 19, 2009 at 12:07 PM, Fred Posner f...@teamforrest.com wrote: On Dec 19, 2009, at 6:51 AM, hadi motamedi wrote: Dear All I have an application that calls for my Asterisk sip to be connected to an external sip server for voip routing . Please be informed that my Asterisk sip is at @192.168.0.2 and the external sip is at @192.168.0.139. To this end , I modified my sip.conf extensions.conf as the followings : Under sip.conf : - [general] register = toronto:welc...@192.168.0.139/osaka [osaka] type=friend secret=welcome context=osaka_incoming host=dynamic disallow=all allow=alaw [6672019] type=friend host=dynamic context=phones Try this: [general] register = toronto:welc...@osaka [osaka] type=friend username=toronto authname=toronto secret=welcome context=osaka_incoming host=192.168.0.139 disallow=all allow=alaw Although your error shows the other server does not allow register. What is the other server? ---fred http://qxork.com Thank you for your reply . The other server is not an Asterisk sip server . It is a sip server inside a softswitch from a third party vendor . As the external sip server man is asking me to disable for the authentication at the first stage , can you please let me know how can I disable for the authentication at this stage (when the calls get through I will enable it again) ? Thank you in advance [general] ;register = toronto:welc...@osaka [osaka] type=friend ;username=toronto ;authname=toronto ;secret=welcome context=osaka_incoming host=192.168.0.139 disallow=all allow=alaw ---fred http://qxork.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thank you for your reply . Please be informed that I want to simulate this case in the Laboratory , i.e. connecting my Asterisk sip to external sip server with the guidelines you sent me . Can you please propose for an Voip application sw that I can install on my MS Windows client and plays the external sip server side role ? It seems that Skype is not suitable for this case as it cannot be configured to play the role of external sip server . Thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with chan_sip
Jonas, Some possible causes: - File permission problem - Firewall blocking - Other network problem like no route On Wed, Dec 23, 2009 at 10:20 AM, jonas kellens jonas.kell...@telenet.bewrote: Calling my home numbers has always worked. Till now. The Asterisk CLI show the following : [Dec 23 10:53:22] NOTICE[25159]: chan_sip.c:12640 handle_response_invite: Failed to authenticate on INVITE to 'sip:092xx9...@85.xx.xx.xx;tag=as5b139383' And after restarting Asterisk, the CLI is flooded by : [Dec 23 11:11:06] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b6e710 (len 549) to my_ip:5063 returned -1: Operation not permitted[Dec 23 11:11:06] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b704e0 (len 543) to my_ip:5062 returned -1: Operation not permitted[Dec 23 11:11:06] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b722b0 (len 543) to my_ip:5061 returned -1: Operation not permitted[Dec 23 11:11:07] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b6e710 (len 549) to my_ip:5063 returned -1: Operation not permitted[Dec 23 11:11:07] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b704e0 (len 543) to my_ip:5062 returned -1: Operation not permitted[Dec 23 11:11:07] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b722b0 (len 543) to my_ip:5061 returned -1: Operation not permitted[Dec 23 11:11:08] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b6e710 (len 549) to my_ip:5063 returned -1: Operation not permitted[Dec 23 11:11:08] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b704e0 (len 543) to my_ip:5062 returned -1: Operation not permitted[Dec 23 11:11:08] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b722b0 (len 543) to my_ip:5061 returned -1: Operation not permitted[Dec 23 11:11:09] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b6e710 (len 549) to my_ip:5063 returned -1: Operation not permitted[Dec 23 11:11:09] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b704e0 (len 543) to my_ip:5062 returned -1: Operation not permitted[Dec 23 11:11:09] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b722b0 (len 543) to my_ip:5061 returned -1: Operation not permitted[Dec 23 11:11:10] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b6e710 (len 549) to my_ip:5063 returned -1: Operation not permitted[Dec 23 11:11:10] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b704e0 (len 543) to my_ip:5062 returned -1: Operation not permitted[Dec 23 11:11:10] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b722b0 (len 543) to my_ip:5061 returned -1: Operation not permitted[Dec 23 11:11:20] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b6e710 (len 549) to my_ip:5063 returned -1: Operation not permitted[Dec 23 11:11:20] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b704e0 (len 543) to my_ip:5062 returned -1: Operation not permitted[Dec 23 11:11:20] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b722b0 (len 543) to my_ip:5061 returned -1: Operation not permitted What is going on here ?? Jonas. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't load cdr_radius.so module?
Shukun, It tells you No such file or directory. Is the file in your modules directory? On Wed, Dec 23, 2009 at 10:09 AM, Zhang Shukun bit...@gmail.com wrote: hi , all when i do the command module load cdr_radius.so ,error happens. i have installed radiusclient-ng , what's wrong with it? thanks! error message as follow: ZHANGSHUKUN*CLI module load cdr_radius.so Unable to load module cdr_radius.so Command 'module load cdr_radius.so' failed. [Dec 23 17:55:41] WARNING[31072]: loader.c:380 load_dynamic_module: Error loading module 'cdr_radius.so': libradiusclient-ng.so.2: cannot open shared object file: No such file or directory [Dec 23 17:55:41] WARNING[31072]: loader.c:730 load_resource: Module 'cdr_radius.so' could not be loaded. -- Regards, Sucan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Connect my Asterisk to external sip?
AsteriskWin32 does have SIP server functionality, same as the linux version. I can't think of any reason why having your CentOS Asterisk be both client and server and register with itself wouldn't work. Although I am wondering how much help all this will be in debugging a connection problem to another SIP provider... On Wed, Dec 23, 2009 at 12:55 PM, hadi motamedi motamed...@gmail.comwrote: On Wed, Dec 23, 2009 at 10:22 AM, David Cunningham dcunning...@voisonics.com wrote: Hadi, You could use Asterisk as a sip server, it's installable on Windows. Using sip set debug on might help you with the Host '192.168.0.139' does not implement 'REGISTER' problem. On Wed, Dec 23, 2009 at 9:12 AM, hadi motamedi motamed...@gmail.comwrote: On Sat, Dec 19, 2009 at 1:33 PM, Fred Posner f...@teamforrest.comwrote: On Dec 19, 2009, at 7:57 AM, hadi motamedi wrote: On Sat, Dec 19, 2009 at 12:07 PM, Fred Posner f...@teamforrest.com wrote: On Dec 19, 2009, at 6:51 AM, hadi motamedi wrote: Dear All I have an application that calls for my Asterisk sip to be connected to an external sip server for voip routing . Please be informed that my Asterisk sip is at @192.168.0.2 and the external sip is at @ 192.168.0.139 . To this end , I modified my sip.conf extensions.conf as the followings : Under sip.conf : - [general] register = toronto:welc...@192.168.0.139/osaka [osaka] type=friend secret=welcome context=osaka_incoming host=dynamic disallow=all allow=alaw [6672019] type=friend host=dynamic context=phones Try this: [general] register = toronto:welc...@osaka [osaka] type=friend username=toronto authname=toronto secret=welcome context=osaka_incoming host=192.168.0.139 disallow=all allow=alaw Although your error shows the other server does not allow register. What is the other server? ---fred http://qxork.com Thank you for your reply . The other server is not an Asterisk sip server . It is a sip server inside a softswitch from a third party vendor . As the external sip server man is asking me to disable for the authentication at the first stage , can you please let me know how can I disable for the authentication at this stage (when the calls get through I will enable it again) ? Thank you in advance [general] ;register = toronto:welc...@osaka [osaka] type=friend ;username=toronto ;authname=toronto ;secret=welcome context=osaka_incoming host=192.168.0.139 disallow=all allow=alaw ---fred http://qxork.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thank you for your reply . Please be informed that I want to simulate this case in the Laboratory , i.e. connecting my Asterisk sip to external sip server with the guidelines you sent me . Can you please propose for an Voip application sw that I can install on my MS Windows client and plays the external sip server side role ? It seems that Skype is not suitable for this case as it cannot be configured to play the role of external sip server . Thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I downloaded installed the AsteriskWin32 PBX but it doesn't have sip server functionality . Can you please propose for an alternative to be used on the MS Windows client as external sip server for my Asterisk on CentOS ? Thank you ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New 1.4 system: registered, but not responding to invite?
Doug, It doesn't respond to the INVITE - the trace says No response to the INVITE?. If the phone doesn't even ring it's probably not getting anything, which points to a problem with the router it's behind. How is the router set up to deliver SIP and RTP to the phone? On Tue, Dec 22, 2009 at 5:33 AM, Doug d...@natel.net wrote: At 00:46 12/21/2009, Alex Balashov wrote: A packet capture would be needed to illuminate the source of the problem. Thanks, Alex for your suggestion. Here is a link for the packet capture: http://www.A7H.com/~stuph/TCPdump-2009-Dec-21-2304.txthttp://www.A7H.com/%7Estuph/TCPdump-2009-Dec-21-2304.txt I just don't see where the extension responds to the INVITE. What would prevent that? By the way, I have a bunch of phones behind this same router that work just fine on our old v1.2 system. On 12/21/2009 01:39 AM, Doug wrote: I've turned on NAT everywhere I can think, but even though I hear ringing on the calling phone (different system) the called phone does not ring. Has anyone bumped into this lately? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP to Analog Devices
Brian, From http://www.voiplink.com/Linksys_Analog_Telephone_Adapters_s/51.htm they have adaptors compatible with Asterisk, but explicitly say in the product titles that they're unlocked, which I think is the key. On Thu, Dec 17, 2009 at 4:16 AM, Brian Cline br...@nw.brian.fm wrote: Hello, I'm running Asterisk v1.6.1.11 internally with a few Linksys SIP phones and will be receiving a machine containing a Dialogic card for a development project (in a nutshell, the card receives analog calls while the accompanying software handles automated prompts, etc). The Dialogic card is not SIP-based but will work with an analog line, so I'm looking into adapters that act themselves as SIP devices but provide an analog port on the other end so that I can internally dial that device's extension, and thus interact with the Dialogic card as though I had a basic POTS line attached. Mostly looking for input and recommendations on these kinds of adapters. Here are a few I've found so far. * Cisco/Linksys PAP2T - appears to be locked into specific mainstream VoIP vendors. Seems to be confirmed with * Cisco/Linksys SPA2101 and SPA3102 - looks promising, but not clear from the product literature whether either is just a straight SIP device and not VoIP vendor specific. Can anyone provide input on these, or other recommendations for this kind of device -- or in the event I am totally wrong about whether this would actually work, suggest alternatives? Many thanks. Brian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham Voisonics Limited IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best way ro run 2 or more asterisk servers?
Since you're not registering either server with the other, where is the call
going to pick up the required username and password to connect to the other
server?
On Wed, Dec 16, 2009 at 12:09 AM, Landy Landy landysacco...@yahoo.comwrote:
I'm trying to get two server communicate with each other and call from one
to the other but, I'm having a lot of problems.
I tried to create a iax trunk between the two:
At the server:
[client]
type=friend
username=asterisk2
authuser=asterisk2
fromuser=asterisk2
secret=sss
auth=md5
context=from_client
;peercontext=from_asterisk
host=172.16.0.11
trunk=yes
qualify=yes
iax2 show peers
Name/UsernameHost Mask Port Status
client/asterisk 172.16.0.11 (S) 255.255.255.255 4569 (T) (E) OK (3
ms)
1 iax2 peers [1 online, 0 offline, 0 unmonitored]
extensions.conf
[to_client]
exten = _3XX,1,Verbose(. To Asterisk2 Server .)
exten = _3XX,n,Dial(IAX2/${ext...@client)
exten = _3XX,n,Hangup()
[from_client]
include = local-dial
At the client:
[server]
type=friend
host=172.16.0.3
username=asterisk
authuser=asterisk
fromuser=asterisk
secret=xxx
context=from_server
trunk=yes
auth=md5
qualify=yes
iax2 show peers
Name/UsernameHost Mask Port Status
server/asterisk 172.16.0.3 (S) 255.255.255.255 4569 (T) (E) OK (3
ms)
1 iax2 peers [1 online, 0 offline, 0 unmonitored]
extensions.conf
[from_server]
include = local-dial
[to_server]
exten = _5XXX,1,Verbose(. Trying to contact ${EXTEN:1} @ asterisk
.)
exten = _5XXX,n,Dial(IAX2/${ext...@server)
exten = _5XXX,n,Hangup
According to some reading, I do NOT need to register neither one.
When I try to call from one end to the other I get:
[Dec 15 03:06:04] NOTICE[4265]: chan_iax2.c:10338 socket_process: Host
172.16.0.3 failed to authenticate as 300
Please help.
Thanks.
___
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--
David Cunningham
Voisonics
IVR development, VOIP consultancy
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3411 5024
Australia: +61 (0) 2 9037 2180
___
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