Am 18.03.12 19:53, schrieb Freddi Hansen:
I have a site that moved to the latest 1.8 revision, and began to
have problems with phones in far away places (South America,
and the MidEast).
What I see is that when a Dial() is issued, the sip channel driver
sends out an INVITE to the phone.
I am not sure whether my PRI / BRI card would detect in virtual machine. I
have to check.
On Sun, Mar 18, 2012 at 8:14 AM, virendra bhati virbh...@gmail.com wrote:
you may installed different version at different virtual machines...
it will be easy and not time consuming as well.
On Wed, Mar
when you installed DAHDI/Zaptel on VM then it will work
On Mon, Mar 19, 2012 at 4:35 PM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
I am not sure whether my PRI / BRI card would detect in virtual machine. I
have to check.
On Sun, Mar 18, 2012 at 8:14 AM, virendra bhati
It appears that each time a call is parked that the CDR billing seconds are
lost and they start again when the parked call is picked back up. The call
duration is correct. What is the best way to address this issue to get
proper bill seconds?
Thanks
Bryant Zimmerman (ZK Tech Inc.)
We are bumping into an interesting issue. We have Dynamically created
parking lots. When the lots were created they had five positions when the
customer wants them changed to say 10 position. We update our database but
since they are already created the new range for PARKINGDYNPOS is being
Dumb question - you did module parkandannounce reload?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant
Zimmerman
Sent: Monday, March 19, 2012 8:25 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Remove
On Fri, Mar 16, 2012 at 11:43 AM, Richard Mudgett
rmudg...@digium.com wrote:
snip
pri intense debug:
TEI: 0 State 7(Multi-frame established)
V(A)=31, V(S)=31, V(R)=42
K=7, RC=0, l3_initiated=0, reject_except=0, ack_pend=0
T200_id=0, N200=3, T203_id=8192
[ 00 01 54 3e
On 03/18/2012 06:25 PM, Ast Coder wrote:
I see that the yum install freepbx from Digium repository actually
installs the latest FreePBX which is nice. However, I don't see the old
FOP in FreePBX anymore. Is there a way to install FOP or FOP2 through
repository?
No, the Digium repository does
On 03/18/2012 08:07 PM, Matt Hamilton wrote:
Kevin,thanks for your response.
Here is the more detailed Wireshark capture of the SIP traffic between
phone (10.0.1.57) and Asterisk (10.0.1.103). The numbers between
parentheses are Request Frames. I don't see an ACK for the 200 OK to the
INVITE
Hello All,
I upgraded Asterisk from 1.8.4 to Asterisk 1.8.10.1 last week. After the
upgrade when I make a call and the other party answers the call a ringing
tone is heard in the background even thought the call was successfully
answered.
This issue is ramdom and is not consistent. I do think
Which Technology are you using for the call (DAHDI/SIP/other)?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty.cruz
Sent: Monday, March 19, 2012 10:42 AM
To: asterisk-users@lists.digium.com
Subject:
Hello I apologize for not being specific.
I'm using SIP. Our provider is call Wiline. I do not have Dahdi install on
this server. The phone I used is Polycom soundpoint ip 450.
Rining continues in the background after successfully answered, is a ramdom
not constantly.
Thanks,
-Motty
It sounds like the random calls are not properly bridging. Calls can
connect and proceed without bridging, but they do funny things like
continuing to ring the line and disconnecting after 20 seconds (just a
random number but I see it pop up a lot). Try and get a CLI shot of a
good and bad call
On Mon, Mar 19, 2012 at 21:24, Danny Nicholas da...@debsinc.com wrote:
Which Technology are you using for the call (DAHDI/SIP/other)?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty.cruz
Sent:
amit anand wrote:
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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Instructions at the bottom of every mail.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Yes I think calls are not bridging properly, I continue to see this error
[Mar 19 09:47:30] NOTICE[5699]: chan_sip.c:16128 check_user_full: From
address missing 'sip:', using it anyway
I will try to get CLI shot of a bad call; unfortunately I can't call the
same number because is a customer
Our current implimenation of dynamic hints is more complex than we would
like. Is it possible to access any other varibles other than the ${EXTEN}
inside of a hint statment?
We are looking for a way to distinguish between orginisationl units.
example
exten =
Hi All;
Is there an admin tool (web based) to check the voicemail and manipulate it
(delete all the voicemail under extension, showing how many voicemails for the
extension, ... etc)?
Can AsteriskNow do this? Or any other recommended tool that it is very good in
this?
Regards
Bilal
--
AFAIK the Asterisk GUI still does this. If not, it is a simple thing to do
in PERL, C, etc.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
Sent: Monday, March 19, 2012 4:35 PM
To:
Hi,
Is there a way to have a distinctive ring for the polycom phones when the
timeout is reached on a parked call? I have google this questions to no
success!
Thanks in advance!
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