Op 12-06-11 15:38, Anders Fudali schreef:
Hi again,
In my environment, I have my phones configured with the same username
as the ActiveDirectory login and in order to map an incoming call to a
username, I simply do a SQL query against a user and phone
provisioning system that I've built.
Ther
Hy all of you,
I successfully installed an AsterikNow fo a client and almost everything
works fine.
I can call/recieve in internal with perfect quality.
I can do the same with external EXCEPT the little delay in the
communication. I have almost 0,5 s of delay between speakers only with
outsid
Hy,
I've jsut upgraded to the 2.9.0.6 framework of freepbx.
I used to have announce-frequency=15 that is repected.
Unfortunately, after 15 sec, no announce is pronounced.
I had to wait *_1minute and 15_* seconds forearing the announce!!
Is there a solution to this problem here?
Regards
--
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Hi All;
Can anyone advise if using Cisco IP Phones in skinny protocol is fine or not?
Or it is better to use it in SIP protocol?
Regards
Bilal
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to A
Hi List,
I have faced a problem in asterisk queue implementation.
I configured a queue with 'ringall' strategy and 'ringinuse=yes' in
queues.conf. If three calls come to this queue in parallel, the logged in queue
agent used to get only one call (may be the first one), not all the calls
waitin
Quite simply: don't use a queue. Simply ring all phones at the same time
using Dial(SIP/phone1&SIP/phone2&..)
A queue will only send the first call until it is answered, then move on to
the second one (I may be simplifying a bit)
Mike
From: asterisk-users-boun...@lists.digium.com
2011/6/13 bilal ghayyad
> Hi All;
>
> Can anyone advise if using Cisco IP Phones
Which model(s) are you planning to use ?
> in skinny protocol is fine or not? Or it is better to use it in SIP
> protocol?
>
>
> Regards
> Bilal
>
> --
> ___
I connected to port 5038,logged in and was expecting to receive events
about
an extension being placed on HOLD. I did not receive any.
I saw all other events for hangup etc... but not for on hold.
Is there something else to enable to see that?
Thanks
Jerry
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Hi,
On 06/13/2011 01:04 PM, bilal ghayyad wrote:
> Can anyone advise if using Cisco IP Phones in skinny protocol is fine or not?
> Or it is better to use it in SIP protocol?
SCCP works better than SIP in my opinion as there are more features.
Check
Yesterday I rebooted the server and it seems to be working again. Not sure
what the reboot might have changed. Hopefully it doesn't happen again but I
can't be sure. To answer your question I have the sip.conf in my mysql
database and in MySQL I have callcounter set to yes. I don't have a colum
hello list
i have 1 server installed with asterisk centos and digium card
i have installed the same configuration in another unit but in this unit
there is no card installed
i have created a sip trunk between the 2 servers like that
in the server 1 with card
sip.conf
[asterisk1]
type=freind
h
On Mon, 13 Jun 2011, salaheddine elharit wrote:
[configuration details snipped]
i can call the 2000 from my unit and 1000 from my server without issue
my question how i can i do in order to call an external number from my
unit(server 2)
with my server 1 i can call the external number wi
Hi all,
we have a problem with a reinvite sent by our SIP provider to change audio
codec due to the recognition of a fax tone.
After that the SIP call session has been established (INVITE and 200 OK) we
have the following codec situation:
UACASTERISK UAS | A
Hi
I am having a problem with asterisk 1.8.4.2 and chan skinny. I have 10 7961
phones and 2 7920 phones. The 2 7920 phones register without a problem, but the
7961 phones do not. the console message is:
chan_skinny.c:6438 get_input: Skinny Client sent less data then expected.
The it tries
Hi all
i have a scenario where i have 2 DSL lines (i know its not that reliable but
it fits the bill) connected to 1 box and would like my isp to round robin
between both dsl (to allow for more capacity - each dsl could get me thru
about 16-18 calls and i need about 30
incoming sip gets routed co
Hi all,
I'm trying to provision my PAP2T's to use a SVR lookup to find the Asterisk
server. I'm using a provisioning file that contains an element like:
_sip._udp.example.com
However, the PAP doesn't seem to be able to find my server with this hostname.
The DNS records are in place because
Dears;
The Asterisk version is 1.8.3.2
The Cisco IP Phone is 7942G and it is running now skinny.
The used TFTP is tftp-server which is installed in fedora.
I placed the following two files (but look like it was not taken from the TFTP,
as nothing appeared in the messages), but I am able to to
On Mon, 13 Jun 2011, bilal ghayyad wrote:
I placed the following two files (but look like it was not taken from
the TFTP, as nothing appeared in the messages), but I am able to to ping
from the asterisk box to the vlan that the Phone is connected, so no
problem in the reachability:
SEPB8BEBF
See sip.conf.sample included in Asterisk 1.8.x. On my system the relevant
section starts on line 541. "STATUS NOTIFICATIONS (SUBSCRIPTIONS)"
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> Jerry Geis
In my experience this is usually caused by REINVITES. Disable reinvites (aka
directmedia in recent Asterisks) and see if that helps.
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> Florent THOMAS
> Se
A *great* thanks to you,
I will try it ASAP.
I'll let you know!
In my experience this is usually caused by REINVITES. Disable reinvites (aka
directmedia in recent Asterisks) and see if that helps.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-user
Hello,
I am using 1.8.4.2 and while outgoing seems to work, incoming still
does not route calls in to the appropriate context.
Please advise.
Thank you,
Elliot
On Sat, Apr 16, 2011 at 4:24 PM, William Stillwell
wrote:
> You must have 1.8+ its already been posted the 1.6 didn’t get a backport f
Bilal,
I suggest you turn on logging on your tftp server to see what files are
actually being requested, and if the the tftp server is dishing them out... Try
adding a few v's to your tftp setup:
File: /etc/xinetd.d/tftp
Line to change: server_args = -s /tftpboot -v -v -v
Look in /var/log/mess
I also had trouble w/ these phones at first. There was a DHCP option (?81?)
you'll have to google it.
The phones would not talk to tftp until I set dhcp option.
The console aux cable is easy to build and VERY useful
Sent from my iPhone
On Jun 13, 2011, at 8:31 PM, Mark Engelhardt
wrote:
> Bi
hi:
Please check the status of PRI, i think the channels keeps up and down.
Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri<->SIP).
website: www.voipviews.com
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Wed, 8 Jun 201
I know I've bumped this already now, but I do need to resolve this and
I've only been replying to myself.
I've tried another client now (Jitsi), which was the only one with
tls/srtp support that will run on freebsd, and it suffers the same problem.
I am very confused now as to why the only cl
Problem solved. Just changed G1 to g1
--
Sent from my iPhone
On Jun 13, 2011, at 9:36 PM, James zhu wrote:
hi:
Please check the status of PRI, i think the channels keeps up and
down.
Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/
pri<->SIP)
I am not sure but seems like Agent channel not being released from Asterisk.
Next time when this happens, try 'core show channels' to check whether Agent
channel is released or not.
[SATISH]
On Mon, Jun 13, 2011 at 9:12 PM, Duane Larson wrote:
> Yesterday I rebooted the server and it seems to b
rajib,
You can use DIALGROUP function as well
On Mon, Jun 13, 2011 at 7:36 PM, Mike wrote:
> Quite simply: don’t use a queue. Simply ring all phones at the same time
> using Dial(SIP/phone1&SIP/phone2&….)
>
>
>
> A queue will only send the first call until it is answered, then move on to
> the
Hello,
To help clarify, Jabber is receiving the incoming packets, but
Asterisk does not seem to be associating it with the gtalk
configuration and the call is not routed into any context. The remote
caller only hears continous ringing. However, outgoing, gtalk and
jabber work fine.
What could b
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