Great tip Duncan :)
On Thu, Jul 12, 2012 at 10:29 AM, Duncan Turnbull dun...@e-simple.co.nzwrote:
You can also specify routes with an callerid qualifier as 09XX/20X
This would only have it apply to extensions in the 200-209 range
That route can then point to a trunk going nowhere if
Thanks Samy
I am figuring you may know but with freepbx if you want to make it a bit more
tailored then send it to a custom trunk
In freepbx add a custom trunk with the custom dial string
Local/$OUTNUM$@blocked-number-custom
/etc/asterisk/extensions_custom.conf
[blocked-number-custom]
exten =
Hi Akhilesh,
Probably this link would give you some idea on ASR. With the help of it,
add some logic in dialplan to develop an application of your choice.
(Courtesy Lefteris Zafiris)
Goto https://github.com/zaf/asterisk-speech-recog/ and read README
--Satish Barot
On Thu, Jul 5, 2012 at 12:46
Hello everyone,
I am trying to apply
thishttp://www.voip-info.org/storage/users/496/27496/images/499/rbt.patch.diffpatch
on chan_ss7-2.1.0 for RingBack tone but its not accepting and
throwing errors:
Hunk #1 FAILED at 704.
Hunk #2 FAILED at 715.
I have done the patch modifications manually in
Hi Chandrakant,
for what I can see the patch you're trying to use (
http://sourceforge.net/projects/aterisk-amr/files/) is going to work with
1.8.13 as well, however the issue behind the crash is not in Asterisk's
code but in 3GPP's original AMR implementation when running on 64-bit hosts.
I've
Is there a tool integrated with asterisk which can give us the pitch of the
utterance?
On Thu, Jul 12, 2012 at 3:09 PM, Satish Barot satish4aster...@gmail.comwrote:
Hi Akhilesh,
Probably this link would give you some idea on ASR. With the help of it,
add some logic in dialplan to develop an
I have this very specific problem with two dect sets. Problem that I
have is one-way audio, in this very rare situation.
I am calling with a Siemens N510 with C610 handset to Panasonic
KX-TGP500 with KX-TPA50 handset. This gives me problems when I am
calling to a SIP account that is
Hello mailinglist,
I want to connect Asterisk with OpenBTS and make a call with a mobile
phone.
I use:
Ubuntu 11.10 + Kernel 3.0.22
GnuRadio 3.3.0
Asterisk 1.8.13
OpenBTS 2.8
Nokia Mobile Phone
OpenBTS works and I can send sms from the OpenBTS server to the
mobile phone. What I also need is a
Hello
Is your server behind nat? This problems sounds me nat problems.
Regards
On Thu, Jul 12, 2012 at 7:53 AM, Roland o/d Akker aster...@rolandow.com wrote:
I have this very specific problem with two dect sets. Problem that I have is
one-way audio, in this very rare situation.
I am
Kevin P. Fleming kpflem...@digium.com writes:
That's quite interesting; can you describe a scenario where this occurs?
Imagine you have a server with two interfaces, eth0 with 192.168.1.1/24
and eth1 with 10.0.2.1/24. Further imagine that you wish to be able to
move phones between the networks
On 07/12/2012 09:19 AM, Benny Amorsen wrote:
Kevin P. Fleming kpflem...@digium.com writes:
That's quite interesting; can you describe a scenario where this occurs?
Imagine you have a server with two interfaces, eth0 with 192.168.1.1/24
and eth1 with 10.0.2.1/24. Further imagine that you wish
On 07/11/2012 11:36 PM, Jeff LaCoursiere wrote:
This does exhibit the problem though. Your OS stack assumes one of
those addresses - the first identified interface? - is the one that all
replies will appear to come from. So phones on the 192.168.2.0/24
network that try to register get replies
I must be missing something. If a phone sends a UDP packet to
192.168.1.1, how does that get routed to (arrive at) the 10.0.2.1
interface on the Asterisk server? The only way I can imagine that
happening is if a router in between the phone and the server has been
told that 192.168.1.0/24
On 07/12/2012 12:38 PM, Freddi Hansen wrote:
We have since Asterisk 1.2 been using a configuration with 6 NIC's
bonding to 3 networks, one public internet and 2 private networks.
Routing calls between networks and having phones on all 3 networks is no
problem.
There is one case though where we
We realize there is an issue with a ticket system subscribed to the list
and responding directly to member's posts. We are working on resolving
the issue, please bear with us!
Thanks,
--
Rusty Newton
Digium, Inc | Open Source Community Support Manager
Check us out at: www.digium.com
Kevin P. Fleming kpflem...@digium.com writes:
I must be missing something. If a phone sends a UDP packet to
192.168.1.1, how does that get routed to (arrive at) the 10.0.2.1
interface on the Asterisk server?
The easiest way is that the Asterisk server itself is the router. Phones
on
On 07/12/2012 03:53 PM, Benny Amorsen wrote:
chan_sip does have the ability to use connect()-ed sockets for dialogs
now, since that is required for TCP, TLS and WebSocket support. It
wouldn't be a huge leap to use them for UDP as well, if that was
beneficial.
It would be greatly appreciated
On Thursday 12 Jul 2012, Kevin P. Fleming wrote:
On 07/11/2012 11:36 PM, Jeff LaCoursiere wrote:
This does exhibit the problem though. Your OS stack assumes one of
those addresses - the first identified interface? - is the one that
all replies will appear to come from. So phones on the
Hi,
i wanted to make dial plan in such a way that the any incoming call to the
sip phone should auto answer.(auto pickup) .
Help.
regards
Upendra
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
Different phones use different methods. What kind of sip phones do you have?
On Jul 13, 2012, at 12:17 AM, upendra uppi...@gmail.com wrote:
Hi,
i wanted to make dial plan in such a way that the any incoming call to the
sip phone should auto answer.(auto pickup) .
Help.
20 matches
Mail list logo
