17 feb 2010 kl. 19.12 skrev Joseph:
Does the sort order matter in sip.conf file?
I know sort order might effect:
allow=ulaw
allow=alaw
but does it matter where I place: insecure=invite ?
The reason I'm asking is that I've loaded almost two identical (sip.conf and
extension.conf)
17 feb 2010 kl. 23.15 skrev Michelle Dupuis:
Is it possible to just send an event from one Asterisk server to another?
(Perhaps some custom event that I could define?) Or would that break the SIP
protocol/handling in asterisk?
I think this discussion would be easier if you told us what you
Hi
Daniel Bareiro a écrit :
[...]
Hours ago the IP changed and the domain was updated satisfactorily, but
in spite of this I was obtaining the registering failures that I
mentioned above. After to restart Asterisk (1.4.24.1), I no longer had
this problem of registering. But there would be
On Tue, 16 Feb 2010, Armin Schindler wrote:
On Tue, 16 Feb 2010, Marcus Hunger wrote:
Hi,
did you see this one: https://issues.asterisk.org/view.php?id=16774 ? It
looks related to your issue.
Oh thanks, I missed that one.
It really looks related. I have added a note.
Now I know how to
Hi,
Am Donnerstag, den 18.02.2010, 10:49 +0100 schrieb Armin Schindler:
On Tue, 16 Feb 2010, Armin Schindler wrote:
On Tue, 16 Feb 2010, Marcus Hunger wrote:
Hi,
did you see this one: https://issues.asterisk.org/view.php?id=16774 ? It
looks related to your issue.
Oh thanks, I
Hi!
Is it possible to decouple the internal Fax communication from ISDN
using t38modem or as an alternative Fax For Asterisk, perhaps in
conjunction with HylaFAX?
Look at HylaFax first, and I guess that you will be happier with leaving
T.38 out of the picture.
Do note that hardware can
Yes that's cool! :)
l.
2010/2/17 Miguel Molina mmol...@millenium.com.co
Ok, if I get it the simplest workaround would be changing this:
exten = _X.,1,Dial(SIP/${EXTEN})
To this:
exten = _X.,1,Dial(SIP/${FILTER(0123456789,${EXTEN})})
If you're intended to receive only numbers from the
Hi All
I am using a Asterisk 1.6.1.6 and I have Digium cards TE122B for the PRI
line and TDM800P cards for connecting the telephone lines.The voice
calls are working fine.Now I need to connect FAX machines to this
TDM800P cards.Kindly let me know what all changes I need to make to make
the
Thanks for the answer. My actual solution is setting the name before
entering the queque. I promise I'll think about using h extension, but I
think there should be a way out with doing so many processing before and
after entering a queue.
Thanks again
ML
2010/2/17 Warren Selby
On 02/18/2010 03:40 PM, dle...@lstelcom.com wrote:
Hi,
I am trying to fix a Asterisk setup with buggy (POTS) Fax machines. The
setup consists of the following components:
- A Digium TE121 for connectiong to E1 ISDN
- Debian box with Asterisk 1.4
- Grandstream GXW-4008 SIP ATA to which the
Hi,
I've set up an Asterisk as voip gatway:
VOIP - Asterisk - hfc-s card - NTBA - Siemens Gigaset Dect ISDN pbx.
Outgoing calls from dect handset to the world are working. Incoming calls don't
even ring the handset.
I'm using the dahdi driver with the zaphfc kernel module. The hfc-s card is
I'm trying to pass additional call information (eg: customer ID) to a call
center along with the call itself.
At this point I would be happy just seeing everything that I can get from
the SIP header from within the dialplan...is there an example of how to
access all header content from within the
Michelle Dupuis wrote:
I'm trying to pass additional call information (eg: customer ID) to a call
center along with the call itself.
At this point I would be happy just seeing everything that I can get from
the SIP header from within the dialplan...is there an example of how to
access all
*Our Product offerings:
*We sell DIDs all over US in 2600+ rate center. For the list of all the
available ratecenters please visit us at
http://www.didforsale.com/moreinfo.php?help=ratecenter.
We have inbound DIDs in 2 different configurations.
1) DID with unmetered inbound and 20 channels
I don't think this mailing list is intended for posts advertising
peoples services.
--
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To UNSUBSCRIBE or update options visit:
Neha Khandelwal escribió:
/_Our Product offerings:
_/
//Use the asterisk-biz list instead to advertise your asterisk-related
products.
Regards.
//
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
--
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Hello list !
Can realtime dialplan be combined with 'hardcoded' dialplan in
extensions.conf ??
Does a context need completely be written or in extensions.conf or in
the mysql-table 'extensions_table' ? Or can I combine the two with the
'switch'-statement ??
Kind regards,
Jonas.
--
On Thu, 2010-02-18 at 19:46 +0100, jonas kellens wrote:
Does a context need completely be written or in extensions.conf or in
the mysql-table 'extensions_table' ? Or can I combine the two with the
'switch'-statement ??
You can certainly combine the two with a switch statement. Asterisk
will
Hello mike,
this feature is only available with an higher firmware for the spa941 (
5.x.x)
You can set this up on the Phone itself or over the Web IF (the USER part).
in the SPA941 its called Call Waiting Service. This would also do what
you want.
Best regards
Steve Smith
Mike A. Leonetti
How about something like :
[mycontext]
exten = 100,1,NoOp(calling 100)
exten = 100,n,NoOp(going realtime)
switch = Realtime/mycont...@realtime_extensions ; from here on we use
realtime
And then my MySQL-DB contains :
`extensions_table` VALUES (1, 'mycontext', '100', n, 'Wait', '2');
On Thu, 2010-02-18 at 19:43 +0800, Gopalakrishnaiyer Venugopal-Q16770
wrote:
Hi All
I am using a Asterisk 1.6.1.6 and I have Digium cards TE122B for the
PRI line and TDM800P cards for connecting the telephone lines.The
voice calls are working fine.Now I need to connect FAX machines to
this
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Hi, Warren.
On Thursday, Feb 18, 2010 at 00:01:23 -0300, Warren Selby wrote:
; DGB - 20100211
externip = sysadminhaiku.com.ar
localnet = 10.1.0.0/24
If you're using dynamic dns, shouldn't you be using externhost instead
of externip?
It can
Hey, all. Got an SNOM 820 in the other day to kick the tires. As with
many phones, provisioning it was a bit of a PITA. The biggest problem, as
far as I could tell, was that their firmware just doesn't seem that
stable, and is sometimes hard to get to.
- I managed to corrupt the firmware twice;
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Hi, Warren.
On Thursday, Feb 18, 2010 at 16:30:40 -0300, Daniel Bareiro wrote:
; DGB - 20100211
externip = sysadminhaiku.com.ar
localnet = 10.1.0.0/24
If you're using dynamic dns, shouldn't you be using externhost
instead of externip?
It can
Hey, all. I love having a PRI to play with -- lets me do all sorts of
things with DIDs, fax-to-e-mail, etc. But for a small shop, a T1 is
pretty pricey. Is there any reason that a BRI can't do exactly the same
stuff, but on 2B+D instead of 23B+D?
Enquiring minds, etc.
-Ken
--
This message
On Thu, Feb 18, 2010 at 03:05:14PM -0500, Ken D'Ambrosio wrote:
Hey, all. Got an SNOM 820 in the other day to kick the tires. As with
many phones, provisioning it was a bit of a PITA. The biggest problem, as
Thanks for the review, I was wondering if snom's mass deployment tools
they have
i want to use Transfer Function (T) i.e when ever caller calls and if his call
is not answered he should hear
the voice, enter the extention of the person.n if he presses the wrong
extention he should hear u had entered wrong extention.if he dosent
dial in a given period of time than it should
On Thu, 18 Feb 2010, Karsten Wemheuer wrote:
Am Donnerstag, den 18.02.2010, 10:49 +0100 schrieb Armin Schindler:
On Tue, 16 Feb 2010, Armin Schindler wrote:
On Tue, 16 Feb 2010, Marcus Hunger wrote:
Hi,
did you see this one: https://issues.asterisk.org/view.php?id=16774 ? It
looks related
On Thu, February 18, 2010 3:56 pm, Alex Samad wrote:
On Thu, Feb 18, 2010 at 03:05:14PM -0500, Ken D'Ambrosio wrote:
Hey, all. Got an SNOM 820 in the other day to kick the tires. As with
many phones, provisioning it was a bit of a PITA. The biggest
problem, as
Thanks for the review, I
Ken D'Ambrosio schrieb:
...
pretty pricey. Is there any reason that a BRI can't do exactly the same
stuff, but on 2B+D instead of 23B+D?
Hello,
this depends on your operator and the telcom regulation in
your country.
In Germany, the main difference (besides the number of channels)
is the
Ken D'Ambrosio wrote:
Hey, all. Got an SNOM 820 in the other day to kick the tires. As with
many phones, provisioning it was a bit of a PITA. The biggest problem, as
Thanks for the review! I need to get my hands on one.
--
Stelios Koroneos wrote:
On Thu, 2010-02-18 at 19:43 +0800, Gopalakrishnaiyer Venugopal-Q16770
wrote:
Hi All
I am using a Asterisk 1.6.1.6 and I have Digium cards TE122B for the
PRI line and TDM800P cards for connecting the telephone lines.The
voice calls are working fine.Now I need to
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Thursday, Feb 18, 2010 at 17:29:44 -0300, Daniel Bareiro wrote:
; DGB - 20100211
externip = sysadminhaiku.com.ar
localnet = 10.1.0.0/24
If you're using dynamic dns, shouldn't you be using externhost
instead of externip?
It can be. I was
Asterisk Project Security Advisory - AST-2010-002
++
| Product| Asterisk|
The Asterisk Development Team has announced security releases for the following
versions of Asterisk:
* 1.2.40
* 1.4.29.1
* 1.6.0.24
* 1.6.1.16
* 1.6.2.4
These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The releases of Asterisk 1.2.40,
It should use the context of the device
On Wed, Feb 17, 2010 at 8:40 PM, Joseph syscon...@gmail.com wrote:
Is there any asterisk guru who can explain me how how asterisk knows which
context forward the call to?
--
Joseph
--
I've got several SIP clients with dynamic IP addresses
Asterisk has one public and one private IP address
SIP clients might connect to Asterisk from either the internet or the
private network (192.168.1.255) - they're portable
By default, directmedia/canreinvite is enabled and Asterisk sets up
I think that's what I need! In what version of * do these functions appear?
(Docs just say trunk)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P.
Fleming
Sent: Thursday, February 18, 2010 12:33 PM
Yes, it should but it doesn't.
And the gurus at Audiocodes support can not explain why?
--
Joseph
On 02/18/10 19:27, C F wrote:
It should use the context of the device
On Wed, Feb 17, 2010 at 8:40 PM, Joseph syscon...@gmail.com wrote:
Is there any asterisk guru who can explain me how how
On 02/18/10 09:00, Olle E. Johansson wrote:
17 feb 2010 kl. 19.12 skrev Joseph:
Does the sort order matter in sip.conf file?
I know sort order might effect:
allow=ulaw
allow=alaw
but does it matter where I place: insecure=invite ?
The reason I'm asking is that I've loaded almost two
Hi experts,
The extensions.conf has the dial plan set as
exten == _988XXX,1,Dial(DAHDI/g1/${EXTEN},20)
I want to modify this so that i can dial numbers with more than 10
digits for example like accessing an IVR menu.
Warm Regards
Venugopal G
On Fri, Feb 19, 2010 at 12:21 PM, Gopalakrishnaiyer Venugopal-Q16770
venui...@motorola.com wrote:
Hi experts,
The extensions.conf has the dial plan set as
exten == _988XXX,1,Dial(DAHDI/g1/${EXTEN},20)
I want to modify this so that i can dial numbers with more than 10 digits
for
I hope I'm not wrong but I think the problem is related to the fact
that on incoming calls Asterisk find the peers based on their IP and
not on their IP+PORT. Thus, if you have several extensions on the same
devices (= one single IP with different SIP ports), the last entry
into your sip.conf file
Hi
I tried the below expression.However it didn't work.
I got the below error message on my CLI
app_dial.c:871 wait for answer:Unable to forward voice or dtmf
pbx.c:3897 _ast_pbx_run: Timeout, but no ruke 't' in context 'Internal'
Warm Regards
Venugopal G
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