You needed to do "asterisk -g" or "amportal start" after your install. The
configs didn't apply because Asterisk wasn't running so there was no
connection to AMI. But when you updated module you Fpbx did an "amportal
restart" or "start" automatically and hence it worked. Anyhow, but the FPBX
rpm is
hello:
please check callerid= in chan_dahdi.conf
Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri<->SIP).
website: www.voipviews.com
From: kaushalshri...@gmail.com
Date: Sat, 17 Dec 2011 06:08:05 +0530
To: asterisk-users@lists.digium.com
Subje
Hello
Did you use callerid(num) in your dial plan?
On Dec 16, 2011 7:38 PM, "Kaushal Shriyan" wrote:
> Hi,
>
> I am having an E1 PRI ISDN Lines with 30 bearer channels and 1 D Channel
> with hundred DIDs (Direct Inward Dialing) numbers attached to a Sangoma PRI
> Card on the server,
> I am using
Hi,
I am having an E1 PRI ISDN Lines with 30 bearer channels and 1 D Channel
with hundred DIDs (Direct Inward Dialing) numbers attached to a Sangoma PRI
Card on the server,
I am using asterisk 1.8.5 on CentOS 5.6.
How can i configure DIDs so that if i make an outgoing call the DID number
should g
Thanks Matt!
It's a multi-tenant system and basically I just wanted to customize the
look of the From info in the emails.
For everyone's reference, I think I have at least found a direction to go
on this.
I am thinking I will redirect the job of mailing the file to a custom
script at some point
Did you run your old configurations thru the Polycom script to convert them to
work with 3.3+?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marco Mooijekind
Sent: Friday, December 16, 2011 4:41 PM
To: Aster
Hello Gord,
the line icon is solid black, which should indicate the lines are
registered.
Marco.
On Fri, Dec 16, 2011 at 10:24 PM, Gord Urquhart wrote:
> Does the phone show the line as registered? The little phone icon on the
> display should be solid for a registered line and just a outline
Does the phone show the line as registered? The little phone icon on the
display should be solid for a registered line and just a outline for a
unregistered line. Using wireshark to watch the SIP traffic is a easy way
to ensure the REGISTER signally is complete.
On Fri, Dec 16, 2011 at 1:02 PM,
Dear all,
I'm using serveral Polycom 335 and 650 phones on Asterisk 1.8.
All worked well. After applying the new Polycom UC 4.0.1 software update to
the phones I notice the following:
When dialing an extension, either on- or off hook, the phone immediately
displays "SIP URL:..".
This does not all
Answering my own question, which is probably bad form.
Updated the modules to current (from 2.7.0.0), applied config, now it works.
Odd.
EKG
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Germann
Sent
Thanks. Checked.
Both running as 'asterisk'
EKG
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Friday, December 16, 2011 3:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussi
Confirm your web server user is running as the same user as asterisk.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Germann
Sent: Friday, December 16, 2011 3:06 PM
To: asterisk-users@lists.digium.com
Sub
Brand new instance on Centos 5.7
Installed asterisk18 via yum from RPM distribution from Digium
Installed FreePBX via yum from Digium distribution.
Asterisk is up. FreePBX is up. However, the changes made in FreePBX aren't
written out to the config files in /etc/asterisk nor does asterisk rec
It sounds like the phone is not getting enough info to do a directed
pickup, have you turned on NotifyCID in sip.conf? If that does'nt work try
using the extended BLF stuff (described here
http://www.excaliburtech.net/archives/147 and here
http://www.voip-info.org/wiki/view/Asterisk+presence)
gor
- Original Message -
> From: "Matthew Jordan"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Friday, December 16, 2011 12:57:43 PM
> Subject: Re: [asterisk-users] fromstring in voicemail.conf
>
>
> - Original Message -
>
> > From: "Todd Routhier"
- Original Message -
> From: "Todd Routhier"
> To: asterisk-users@lists.digium.com
> Sent: Friday, December 16, 2011 11:32:31 AM
> Subject: [asterisk-users] fromstring in voicemail.conf
> I have attempted to set the fromstring option on a per context basis
> in voicemail.conf but it do
Perfect, patched my install and it fixed the issue!
Thanks a ton Richard..
--Todd
On Fri, Dec 16, 2011 at 10:17 AM, Richard Mudgett wrote:
> > OK, read all about the patch, thanks for the fix Richard.
> >
> >
> > I would like to apply this patch to my current 1.8.7.1 but I am
> > afraid I don'
Hi everybody,
I'm having a problem with some of my DAHDI Trunks, it is a strange
thing, above is the output of the core show channels command:
Channel Location State
Application(Data)
DAHDI/11-1 ~~s~~@dial_dahdi:15 Up Dial(DAHDI/12/
91162
I have attempted to set the fromstring option on a per context basis in
voicemail.conf but it doesn't seem to work. I would like to somehow either
set this based on context, number dialed into or some other way.
Would it be possible to set this option in the general section to a channel
variable,
On Fri, 2011-12-16 at 11:12 -0600, Russ Meyerriecks wrote:
> - Original Message -
> > From: "Jeff LaCoursiere"
> > I am seeing errors accumulate on an RBS T1 and am wondering what to
> > make
> > of them:
>
> Could we take a peek at /etc/dahdi/system.conf and /proc/dahdi/1 ?
Sure:
root@
- Original Message -
> From: "Jeff LaCoursiere"
> I am seeing errors accumulate on an RBS T1 and am wondering what to
> make
> of them:
Could we take a peek at /etc/dahdi/system.conf and /proc/dahdi/1 ?
--
_
-- Bandwidth
On Fri, Dec 16, 2011 at 05:02:11PM +0100, Olle E. Johansson wrote:
>
> 16 dec 2011 kl. 02:03 skrev Barry Miller:
>
> > So is there a way for the dialplan to determine which device caused SIP to
> > auto-register an extension?
>
> Not really, unless someone else can come up with something.
>
>
Hi,
I am seeing errors accumulate on an RBS T1 and am wondering what to make
of them:
root@vigw3:/usr/local/bin# date
Fri Dec 16 12:13:33 AST 2011
root@vigw3:/usr/local/bin# dahdi_maint -s 1
Span 1:
>FEC : 0:
>CEC : 0:
>CVC : 0:
>EBC : 0:
>BEC : 0:
>PRBS: 0:
>GES : 12:
root@vigw3:/usr/local/bin#
Richard,
Thanks, I'll give it a whirl.
I upgraded Asterisk to 1.8.8.0 last night and this fixed the original issue
I was having. Now calls to the other extensions continue as normal, even
when one of the SIP extensions is unreachable.
Caller-id is still lost on anything that hits follow me. I w
> OK, read all about the patch, thanks for the fix Richard.
>
>
> I would like to apply this patch to my current 1.8.7.1 but I am
> afraid I don't have a clue how.
https://issues.asterisk.org/jira/browse/ASTERISK-17557
Get the patch by following the reviewboard link in the issue and
download it
15 dec 2011 kl. 19:33 skrev Tarek Sawah:
>
> Hello List,
> I have customer with a 40 Agents call center. and is looking to install a PBX
> switch that can serve those agents.
> As per my experience i suggested Asterisk as i have tested it with Call
> Centers, however he has been advised not to
16 dec 2011 kl. 02:03 skrev Barry Miller:
> Hi all,
>
> In sip.conf:
> [general]
> regcontext = autoreg
>
> [devabc]
> regexten = 543
>
> creates "exten=> 543,1,Noop(devabc)" in context autoreg when devabc
> registers. But I can't use "exten=> _5XX,2,Dial(SIP/${EXTEN})" in the
> dialplan,
Hi,
i'm trying to periodically playback a sound to an existing conference
with ConfBridge on Asterisk 10.0.0-rc3
Previously with MeetMe I generated a callout file and had an matching
local dialplan entry.
But this does not work... The local channel gets joined to the
conference, is stuck there u
16 dec 2011 kl. 11:29 skrev James Courtier-Dutton:
> Hi,
>
> I have a situation where unfortunately, I cannot use IAX for trunks,
> and need to instead use SIP trunks.
> Is there any way to fit the voice data from more than one simultaneous
> phone call into a single IP packet over the SIP trunk
thanks, It worked for h!
and if I want in DeadAGI? I want cdr function in the same AGI.
Regards,
Zohair Raza
On Fri, Dec 16, 2011 at 7:08 PM, Eric Wieling wrote:
> From cdr.conf.sample:
>
> ; Normally, CDR's are not closed out until after all extensions are
> finished
> ; executing. By en
>From cdr.conf.sample:
; Normally, CDR's are not closed out until after all extensions are finished
; executing. By enabling this option, the CDR will be ended before executing
; the "h" extension so that CDR values such as "end" and "billsec" may be
; retrieved inside of of this extension.
;endb
Still same, even when I am trying to write in one agi and calling it using
DeadAGI
Regards,
Zohair Raza
On Fri, Dec 16, 2011 at 6:56 PM, Danny Nicholas wrote:
> Try this
>
> exten => _X.,1,Dial(SIP/1*100)
>
> exten => h,1,wait(10)
>
> exten => h,n,AGI(cdr.php,11)
>
> ** **
>
>
No.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James
Courtier-Dutton
Sent: Friday, December 16, 2011 5:30 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] SIP Trunk
Hi,
I have a situatio
Try this
exten => _X.,1,Dial(SIP/1*100)
exten => h,1,wait(10)
exten => h,n,AGI(cdr.php,11)
Don't know how long after hangup this information gets updated, but would be
shocked if 10 seconds doesn't cover it.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@
I am out of the office until 12/16 but I will still be checking my
messages. For immediate assistance, please call Telecommunication Services
at 799-6543. Thank you.
Chet Stevens
Telecommunication Services
Clark County School District
--
___
Yes running from h
exten => _X.,1,Dial(SIP/1*100)
exten => h,1,AGI(cdr.php,11)
Regards,
Zohair Raza
On Fri, Dec 16, 2011 at 6:42 PM, Danny Nicholas wrote:
> You are running the AGI from the h() exten? Otherwise I wouldn’t expect
> CDR(end) to populated or correct.
>
> ** **
>
> *From
You are running the AGI from the h() exten? Otherwise I wouldn't expect
CDR(end) to populated or correct.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zohair Raza
Sent: Friday, December 16, 2011 8:38 AM
To: Asterisk Users Mailing L
Hi,
I've tested 1.8.6.0, 1.8.4.0 and 1.8.0
I can get proper start and answer time but not the end time of call
AGI Rx << GET VARIABLE CDR(start)
AGI Tx >> 200 result=1 (2011-12-16 18:34:48)
AGI Rx << GET VARIABLE CDR(end)
AGI Tx >> 200 result=1 (2011 12-16 18:34:48)
AGI Rx << GET VARIABLE CDR(an
Hi,
I have a situation where unfortunately, I cannot use IAX for trunks,
and need to instead use SIP trunks.
Is there any way to fit the voice data from more than one simultaneous
phone call into a single IP packet over the SIP trunk.
I believe this is possible with IAX trunks, but I don't know ho
I am out of the office until 12/16 but I will still be checking my
messages. For immediate assistance, please call Telecommunication Services
at 799-6543. Thank you.
Chet Stevens
Telecommunication Services
Clark County School District
--
___
Hi all
I'm trying to configure my Asterisk setup to load the musiconhold.conf file
from an ODBC connection to MySQL, working through the example given in the
excellent book "Asterisk: The Definite Guide". I'm using Asterisk 1.4.19 and
MySQL 5.1.58. I've configured the ODBC bit and in my Genera
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