On Thu, Jun 16, 2011 at 10:58 PM, asterisk asterisk aster...@ck-lee.comwrote:
Can this non gmail.com GV number be terminated at some sip accounts so
that I can bridge to it via asterisk as client?
Yes, I've setup some GV numbers on my google apps accounts (@selbytech.com,
for example), and
both show transfercapability DIGITAL
Regards
Robb
On 16 June 2011 23:40, Richard Mudgett rmudg...@digium.com wrote:
Hi All
Just upgraded from 1.6? to 1.8.4.1
I ised to be able to get a digital call working across a bridged isdn
channel in 1.6 and 1.4 using the following;-
Could you elaborate on how you can associate those non-gmail accounts with
gchat account?
On Fri, Jun 17, 2011 at 2:38 PM, Warren Selby wcse...@selbytech.com wrote:
On Thu, Jun 16, 2011 at 10:58 PM, asterisk asterisk
aster...@ck-lee.comwrote:
Can this non gmail.com GV number be terminated
On Thu, 2011-06-16 at 19:12 -0300, Henrique Fernandes wrote:
It is possible to log queue in mysql without turning on realtime
asterisk?
Thanks!
[]'sf.rique
--
Hi
Yes, you can pick and choose which things you want to use your DB by
defining them in your extconfig.conf
so, in
Is there a SIP header I can set (for Snom and Yealink phones if that's
relevant) or any other mechanism to tell a phone to ignore a particular
call from it's missed call list?
I have bits of the dialplan that ring groups of phones eg:
exten = 200,1,Dial(Sip/112SIP/113SIP/114)
and I don't want
You could use the c option in the dial command which sends a call answered
elsewhere reason to the phone and then the phone won't record it in the missed
list (I know it works on the snom I didn't check it on the yealink )
But you'll have to send that only with the dial command which you don't
I have a free google apps account (http://www.google.com/a I think) setup for
SelbyTech.com. Basically it is a gmail account, just with a different domain.
Thanks,
--Warren Selby, dCAP
On Jun 17, 2011, at 2:43 AM, asterisk asterisk aster...@ck-lee.com wrote:
Could you elaborate on how you can
Hi Users,
I would like to know about the RTP audio streaming. I am taking the example
as youtube, in youtube if bandwidth is less the application will buffer and
will stream the video; likewise how to do with audio buffering and play the
file using RTP in asterisk. Any guide of clue will make me
I'm not trying to be a jerk or anything. But have you played with this at all
or are you just looking for someone to write you a dialplan/config that already
works. There are some great pointers in the sample configs and if you look
around on google for gtalk asterisk. Also, read the asterisk
hello list
i need to create a call files in order to do a click to call with
asterisk1.4
i want to use sip 223 in order to call phone number
i have created a file.call in var/spool/asterisk/tmp and i move it to
var/spool/asterisk/outgoing
but there is no call
please tell me if there
Is there an easy way to feed an audio file (think background music,
ever so softly) to the inbound leg of a bridged call (and not send /
mix it to the outbound leg)?
exten = blah,1,Answer()
exten = blah,2,StartSomeAudio(foo)?
exten = blah,3,Dial(SIP/bar)
Where the audio would continue
On Fri, Jun 17, 2011 at 05:20:39PM +, salaheddine elharit wrote:
i want to use sip 223 in order to call phone number
Is that meant to be the originator or the destination?
Channel: gets the originator; Extension: gets the destination.
Roger
--
Is there any to have asterisk record a file then send that file to
a distribution list of voicemail boxes?
What I'm trying to accomplish is a prompt for a user to record/listen to
their message and then choose to send the recording to multiple voicemail
box's inboxes
--
Or is there anyway to have a message copied from a mailbox to a list of
other mailboxes everytime a message is left in it?
On Fri, Jun 17, 2011 at 1:54 PM, vip killa vipki...@gmail.com wrote:
Is there any to have asterisk record a file then send that file to
a distribution list of voicemail
On Fri, Jun 17, 2011 at 1:05 PM, vip killa vipki...@gmail.com wrote:
Or is there anyway to have a message copied from a mailbox to a list of
other mailboxes everytime a message is left in it?
On Fri, Jun 17, 2011 at 1:54 PM, vip killa vipki...@gmail.com wrote:
Is there any to have asterisk
The way I play a sound file into a bridged call is to use chanspy w option. I
do this with an application that does AMI commands.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jun 17, 2011, at 10:25 AM, Tom Browning wrote:
Is there an easy way to feed an audio
On Fri, 2011-06-17 at 13:54 -0400, vip killa wrote:
Is there any to have asterisk record a file then send that file to
a distribution list of voicemail boxes?
What I'm trying to accomplish is a prompt for a user to record/listen
to their message and then choose to send the recording to
Hi All
Just upgraded from 1.6? to 1.8.4.1
I ised to be able to get a digital call working across a bridged
isdn
channel in 1.6 and 1.4 using the following;-
exten = _X.,1,gotoif($[${TRANSFERCAPABILITY}=DIGITAL]?5:)
exten = _X.,2,dial(DAHDI/g1/${EXTEN})
exten
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Carlos Chavez
Sent: Friday, June 17, 2011 2:53 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] asterisk voicemail distribution groups
Hi, When is the next release planned for as very keen to get it into Production
but require the call pickup fix.
--
Thanks, Phil
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join
any reason why this would happen, should I report a bug on the issue
tracker?
Robb
On 17 June 2011 19:55, Richard Mudgett rmudg...@digium.com wrote:
Hi All
Just upgraded from 1.6? to 1.8.4.1
I ised to be able to get a digital call working across a bridged
isdn
Just upgraded from 1.6? to 1.8.4.1
I ised to be able to get a digital call working across a bridged
isdn
channel in 1.6 and 1.4 using the following;-
[snip]
Could be a problem in the media stream handling not being setup for
digital mode.
..., should I report a bug on
On 2011-06-14 15:51, Robert Huddleston wrote:
I only need 4 fxs... I looked at the IAD2431 but it uses T1/E1 as
WAN... If I could assign Fast Ethernet to WAN that would be great...
Budget is not that great
I've done that on a 2431. There's nothing special about the T1 port.
I've made it
From http://www.voip-info.org/wiki/view/Asterisk+presence
Polycom Phones (updated for 3.2.X firmware with asterisk 1.6.1 Jan/2010) With
SIP 3.2.X firmware (available on the Polycom download site) and Asterisk
1.6.1, Polycom phones now support a full featured BLF showing statuses of
Ringing, Inuse
Inviato da iPhone
Il giorno 16/giu/2011, alle ore 16:37, Eric Wieling ewiel...@nyigc.com ha
scritto:
We experience the same thing. The solution we use is to not change codecs in
the middle of a call. I assumed it was an issue with our upstream.
Hi Eric,
this behavior is an asterisk
I don't know.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Matteo Campana
Sent: Friday, June 17, 2011 5:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing
Hi All;
Please I need a help in the ooh323.
First of all, the only way to have h323 working in asterisk 1.8.3 or 1.8.4 is
to use ooh323? There is no way to get the normal h323 channel that come with
asterisk to work fine !!
Now, let us see the ooh323 problem that I am facing:
Already I
sir its done
thank you A.J
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax
On 18/06/2011 5:36 AM, Matteo Campana wrote:
Inviato da iPhone
Il giorno 16/giu/2011, alle ore 16:37, Eric Wielingewiel...@nyigc.com ha
scritto:
We experience the same thing. The solution we use is to not change codecs in
the middle of a call. I assumed it was an issue with our
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