On Thursday 07 March 2013, Luis H. Forchesatto wrote:
Greetings.
I got an extension on my Elastix who cannot pick calls on the other
extensions, but It can transfer his calls to the other extensions. When
this extension tries to pickup a call pressing *8 it simply does not pick
it up.
-Original Message-
From: Carlos Alvarez car...@televolve.com
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] asterisk
I think I found the problem. Better looking the sip_additional.conf file I
noticed that a few extensions didnt had a callgroup and pickgroup
configured, even with the interface appointing otherwise.
I manually configured this options and reloader asterisk and now I'm gonna
test the extensions and
2013/3/8 nik600 nik...@gmail.com
Dear all
i'm planning a migration to asterisk for a high volume IVR service
(from 1000 to 1500 concurrent call)
The IVR service is based only on DTMF tones so the features required is
- play feature
- dtmf detection
Asterisk will receive calls via VOIP
If you accept calls on.g711 and static ivr dialplan you should be able to
do around 300-400 concurrent on the box config that you provided.
And If you pay some expert consultant, he may be to fine tune it to be able
to handle 500 concurrent as well.
Which version of asterisk are you planning to
Yes, it worked :D
Thankyou guys for the help.
2013/3/8 Luis H. Forchesatto luisforchesa...@gmail.com
I think I found the problem. Better looking the sip_additional.conf file I
noticed that a few extensions didnt had a callgroup and pickgroup
configured, even with the interface appointing
Le 08/03/2013 13:17, Leandro Dardini a écrit :
2013/3/8 nik600 nik...@gmail.com mailto:nik...@gmail.com
Dear all
i'm planning a migration to asterisk for a high volume IVR service
(from 1000 to 1500 concurrent call)
The IVR service is based only on DTMF tones so the features
On Friday 08 March 2013, Luis H. Forchesatto wrote:
Yes, it worked :D
Thankyou guys for the help.
Glad it worked for you.
Just be very careful if you change anything via the GUI in future, because it
might undo any changes you made manually -- especially if you didn't get the
format of
Hi,
I am using Asterisk 11.2.1. I am logging CDRs to a mysql database (via
odbc). The table contains the fields clid and src. Both fields are
varchar(100). But alls entries are without the leading 0. For example
0211 for Germany-Düsseldorf.
Where can I configure that behaviour, please?
Hi,
I have upgraded vom Atserisk 1.6.1.20 to 11.2.1. Most things went fine.
But 1 thing will not work: IAX. I used the same configuration but
Asterisk will not answer the incoming IAX-Call.
When enabling iax debugging I can see the following:
[Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:
- Original Message -
From: Thorsten Göllner t...@ovm-group.com
I set verbose and debug to 100 but no(!) message was given.
Read through
https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information and
read through the logger.conf sample file.
Collect a full log with
Ok, thanks for the info.
-Bryan Anderson
On Thu, Mar 7, 2013 at 6:07 PM, Chad Wallace cwall...@lodgingcompany.comwrote:
On Thu, 7 Mar 2013 17:12:47 -0800
Bryan Anderson shadow...@gmail.com wrote:
Has any one ever worked with placing idle display images onto the
Polycom SPIP331 phones?
Thank you both
Matthew, I can not do that because core file is not available
Bharat is right, the file was not written because of abrt's issue
https://bugzilla.redhat.com/show_bug.cgi?id=768149
I am turning it off now, I hope asterisk won't crash again but in any case
if it does I will have a
As I recall, there was an IAX2 protocol addition for newer versions of
Asterisk a while ago due to a security issue - which can potentially
trigger IAX2 interop issues if your config file for chan_iax2 is not
setup properly. You can read more about it here:
Hello List,
I have some doubt about direct media settings.
I have an asterisk 1.8.14 instance running on 172.20.255.50, a soft phone
on IP 10.100.210.51 and a gateway at 10.100.210.254
I have set both gateway and peer to directmedia=yes but still on gateway
I see RTP from asterisk's IP, have
Hi All;
How to let the virualbox (ubuntu OS) to be able to see the digium card? Because
when I install elastix or asterisk with dahdi, it is not able to see the digium
card if the installation though the virualbox .. What is the solution?
Regards
Bilal
--
convert the calls from PRI to SIP and throw it inside the VirtualBox
Asterisk, thats the ONLY WAY OUT
Regards,
Mitul Limbani,
Chief Architech Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
How to let the virualbox (ubuntu OS) to be able to see the digium card?
It's called PCI Passthru and from what I've tried, the timing is horrible in a
virtualized environment. VirtualBox and ESXi 5
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little
It's called PCI Passthru and from what I've tried, the timing is horrible in
a virtualized environment. VirtualBox and ESXi 5
Doug
What are your experiences, Doug. I've heard a lot about it but I'm
running Asterisk on ESXi5 Dell boxes without problems. Did you
encouter the timing issues
The Asterisk Development Team has announced the release of:
DAHDI-Linux 2.6.2
DAHDI-Tools 2.6.2
DAHDI-Linux-Complete 2.6.2+2.6.2
This release is available for immediate download at:
http://downloads.asterisk.org/pub/telephony/dahdi-linux
http://downloads.asterisk.org/pub/telephony/dahdi-tools
Hello List,
I have some doubt about direct media settings.
I have an asterisk 1.8.14 instance running on 172.20.255.50, a soft phone
on IP 10.100.210.51 and a gateway at 10.100.210.254
I have set both gateway and peer to directmedia=yes but still on gateway
I see RTP from asterisk's IP, have
If you want to use direcmedia = yes , in order take to effect.You must not set
dtmf = rfc2833 .You should set it dtmf = info.
It should work then.
Regard/chui king man
寄件人︰ Mark Henry markhenry...@gmail.com
收件人︰ asterisk-users@lists.digium.com
傳送日期︰
Gertjan Baarda wrote:
What are your experiences
dahdi_test would produce accuracies of almost 80% Whereas normal
hardware would produce 99.998%
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor
Hi;
If my landline service provider does not provide the ability to send the SMS,
and I need to send SMS from asterisk, then what is the required? How?
Is it possible to send SMS from asterisk using SIM card to be connected via GSM
adaptor connected to FXS ports? Or HOW?
From the other side,
Yes, you can check solutions from sangoma and khomp.
Saludos/Regards
--
Ing. Gerardo Barajas Puente
Proyectos Especiales/Preventa | www.neocenter.com
T:+52 (55) 8590-9000 x 7003
On Fri, Mar 8, 2013 at 6:32 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Hi;
If my landline service provider
hello
regardless the virtual box, just in terms of Ubuntu, I have experience that
Digium TP110p does now work with Ubuntu. it was long time a go I had this
experience, I hardly could remember that what Ubuntu version I was using. my
experience was on the Ubuntu system would not able to load
The solution is, as mentioned before, PCI passthru. This must be supported
by the hardware. Not sure if the Digium cards will eat it.
Sent from my iPhone
On 9 mrt. 2013, at 08:29, longst longst...@gmail.com wrote:
hello
regardless the virtual box, just in terms of Ubuntu, I have experience
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