On Tue, Jan 22, 2013 at 1:24 PM, Sakharam Thorat
sakharam.tho...@einfochips.com wrote:
Can anybody send me Detailed process to configure Asterisk in CENTOS ??
Detailed description highly appreciated.
Best Regards,
Sakharam Thorat.
How about this link?
Hi,
I want to check the status of a blind transfer (only sip endpoint)
between various phones. Transfer is working perfectly, using ## from
features.conf or using transfer key from phone, here SNOM320.
My problem is that if party to transfer to is busy, the transfer fail
and the call is
On 01/22/2013 08:54 AM, Sakharam Thorat wrote:
Can anybody send me Detailed process to configure Asterisk in CENTOS ??
Detailed description highly appreciated.
Start by reading the Asterisk book, check asterisk.org and Google around
to see if your question has already been answered.
Please forget this message, BLINDTRANSFER is working, I had a typo in
the dialplan when using this variable.
Apologize
Le 22/01/2013 10:40, Administrator TOOTAI a écrit :
Hi,
I want to check the status of a blind transfer (only sip endpoint)
between various phones. Transfer is working
Mitch Claborn mitch...@claborn.net writes:
Shouldn't asterisk somehow know when the agent disappears?
You are a bit out of luck since SIP session timers, the obvious
solution, cannot be set lower than 90 seconds.
rtptimeout set to e.g. 10 seconds may work, but you need to then set
On Mon, Jan 21, 2013 at 9:59 PM, Frank fr...@efirehouse.com wrote:
Actually, the funny thing is that it works randomly.
This may be due to the fact that voice.google.com actually resolves to a
range of IP addresses. When you set up your firewall, it may not be
including all of the possible
On Tue, Jan 22, 2013 at 1:57 AM, Muhammad mohammad.ghaz...@gmail.comwrote:
Dear All.
When I calling a number from web, my softphone show me Answer and
Decline bottoms, and then I have to click Answer to call the number. it
seems it is two step to calling the number. If I type the number
On Mon, Jan 21, 2013 at 12:03 PM, Mitch Claborn mitch...@claborn.netwrote:
How can I accomplish my goal?
Since nobody seems to have come up with an Asterisk-specific solution, it
sounds like the real approach here is something more generic.
You can set up Nagios to fire off an event if it
Chris,
I covered the whole 74.125.225.* subnet.
Even if I open the ports mentioned below for all (not limited to IP
addresses) I still have the same issue.
Have anyone ever succeeded in such configuration? :
Digium phones on 2 different private networks (2 different buildings)
Asterisk
You are obviously getting the call connected, so the subnet issue is moot.
What this sounds like (pardon the pun) to me is an rtp skip issue. The
working calls are generating rtp connections in the allowed range; the
other calls have one or more ports outside of your rtp range. Verify that
all
Danny,
Thanks for the trick, that made all outgoing calls working.
Now, the issue is with incoming calls. Even if I turn off all other
phones in google voice configuration and have the calls routed to my
Google Chat only, this is what happens:
The Asterisk receives the call.
The D70 rings.
Do a netstat -anp during the call. This will (hopefully) show you where
the out of range condition is occurring.
-Original Message-
From: Frank [mailto:fr...@efirehouse.com]
Sent: Tuesday, January 22, 2013 10:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Danny
Danny,
I tried netstat -anp on a working outgoing call, and non working
incomgin, and I see that the working has CONNECTED status, while the
other one has nothing like that at all. Any other idea ?
Thanks
On 1/22/13 11:36 AM, Danny Nicholas wrote:
Do a netstat -anp during the call. This
Each asterisk call uses 3 ports; 5060 is used to initiate the connection
(5222 for chan_motif/google voice), then 2 consecutive ports from the
10001-2 range are used for voice. Since GV uses TLS, I'm wondering if
5061 also comes into play. I assume you started from this link:
Can you please post a dialplan excerpt about using these variables. I just
tried using them, but they are all empty. Maybe I am making the same
mistake of you.
Leandro
2013/1/22 Administrator TOOTAI ad...@tootai.net
Please forget this message, BLINDTRANSFER is working, I had a typo in the
If you needed a MITM, nothing would work now. The incoming call is
connecting, but no voice or no connection at all?
-Original Message-
From: Frank [mailto:fr...@efirehouse.com]
Sent: Tuesday, January 22, 2013 11:56 AM
To: Danny Nicholas
Subject: Re: [asterisk-users] Google voice with
Hi,
No, it's not even connecting.
On the caller side, I do not see anything showing that the called party
picks up.
On the D70 side, when I pick up, I have the counter starting so I can
see the seconds going up, but no audio at all. (and the remote party
still hears ring tone)
On
Does your install have a set of gtalk commands? GV isn't a SIP call per se,
so the incoming line would be a gtalk peer. Try these commands from CLI
Gtalk show peers
Core help gtalk
-Original Message-
From: Frank [mailto:fr...@efirehouse.com]
Sent: Tuesday, January 22, 2013 1:04 PM
To:
*CLI core show help gtalk
gtalk show channels Show GoogleTalk channels
*CLI gtalk show channels
Channel Jabber ID Resource
Read Write
0 active gtalk channels
And that's my jabber.conf
[general]
debug=no
autoprune=no
What about jabber show channels?
-Original Message-
From: Frank [mailto:fr...@efirehouse.com]
Sent: Tuesday, January 22, 2013 1:12 PM
To: Danny Nicholas
Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Google voice with no voice
*CLI core show
*CLI jabber show connections
Jabber Users and their status:
[asterisk] r...@gmail.com - Connected
Number of users: 1
On 1/22/13 2:14 PM, Danny Nicholas wrote:
What about jabber show channels?
-Original Message-
From: Frank [mailto:fr...@efirehouse.com]
Sent:
This is incoming, outgoing or idle (no call)?
-Original Message-
From: Frank [mailto:fr...@efirehouse.com]
Sent: Tuesday, January 22, 2013 1:21 PM
To: Danny Nicholas
Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Google voice with no voice
That's idle.
If I call from D70 (working scenario) the result of the command is the same.
gtalk show channels shows this when I call from D70 (again, working
scenario):
Channel Jabber ID Resource
Read Write
Gtalk/+1x@voice.googl
This sounds like a codec issue. Set your verbose to 10 and retry the
incoming call.
-Original Message-
From: Frank [mailto:fr...@efirehouse.com]
Sent: Tuesday, January 22, 2013 1:26 PM
To: Danny Nicholas
Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re:
Looks like it would be pretty darn easy if I was using a queue. I could
just use the periodic announcement and fiddle with specified sound files.
Sadly, I'm calling the phones with SLATrunk. The hold music will only
be heard by the caller when the user pushes the hold button on their
phone.
OK, so here is the new..
By mistake, when I picked up the D70 , I pushed the 2 button.
I suddenly heard google voice saying Okay, I'll send the caller to
voicemail. So I called again.. picked up.. I could not hear anything on
the D70.. But if I push 1 (which is the google voice option to
Frank wrote:
OK, so here is the new..
By mistake, when I picked up the D70 , I pushed the 2 button.
I suddenly heard google voice saying Okay, I'll send the caller to
voicemail. So I called again.. picked up.. I could not hear anything on
the D70.. But if I push 1 (which is the google voice
Hi ,
So I tried
Answer()
Wait(1)
SendDTMF(1)
But I got an error in the console:
[Jan 22 14:54:13] WARNING[28067]: pbx.c:4458 pbx_extension_helper: No
application 'SendDTMF' for extension (gtalk_incoming, r...@gmail.com, 4)
If I do core show application sendDTMF , nothing comes up.
If
Frank wrote:
Hi ,
So I tried
Answer()
Wait(1)
SendDTMF(1)
But I got an error in the console:
[Jan 22 14:54:13] WARNING[28067]: pbx.c:4458 pbx_extension_helper: No
application 'SendDTMF' for extension (gtalk_incoming, r...@gmail.com, 4)
The app_senddtmf.so module has to be built and loaded.
My bad, I found it not loaded in my modules.conf.
This is now working.
What a pain. Is there a wiki page I can update in order to share the
configuration and how to have this work, with everybody ?
On 1/22/13 2:58 PM, Joshua Colp wrote:
Frank wrote:
Hi ,
So I tried
Answer()
Wait(1)
Frank wrote:
My bad, I found it not loaded in my modules.conf.
This is now working.
What a pain. Is there a wiki page I can update in order to share the
configuration and how to have this work, with everybody ?
A wiki page for using it with the unsupported chan_gtalk / res_jabber
combination
Look at asterisk 11
A option was added to play announcements between music Files and so forth
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar
Hi all,
I registered my Digium D70 using a name (D70) instead of a number.
Is there a way to program Asterisk (or the phone?) so when I press the
MSGS button, it automatically goes to the correct voicemail, with or
without asking me for a password ?
As of now, it asks me for my mailbox
Theoretically you can do this
Exten = 370,1,Voicemailmain(D70@default)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank
Sent: Tuesday, January 22, 2013 2:28 PM
To: Asterisk Users Mailing List -
That worked, thank you.
Is there a way to program the keys of the Digiums D70 from asterisk ?
Or does everything needs to be done on the phone itself ?
On 1/22/13 3:31 PM, Danny Nicholas wrote:
Theoretically you can do this
Exten = 370,1,Voicemailmain(D70@default)
-Original Message-
That one's not in my wheelhouse since I only use Polycom phones. Google
D70 Provisioning
-Original Message-
From: Frank [mailto:fr...@efirehouse.com]
Sent: Tuesday, January 22, 2013 2:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Danny Nicholas
Subject: Re:
On Tue, Jan 22, 2013 at 1:48 PM, Frank fr...@efirehouse.com wrote:
That worked, thank you.
Is there a way to program the keys of the Digiums D70 from asterisk ?
Or does everything needs to be done on the phone itself ?
The whole point of the Digium phones is their tight integration with
You can use XML configuration via a http server or asterisk phoneprov
module, or use the res_digium_phone module on intel platforms and create
config files. Both methods support setting the mailbox (and lots of other
things).
https://wiki.asterisk.org/wiki/display/DIGIUM/DPMA+Configuration
On
Christopher Harrington ch...@acsdi.com writes:
Since nobody seems to have come up with an Asterisk-specific solution, it
sounds like the real approach here is something more generic.
You can set up Nagios to fire off an event if it detects endpoints or
infrastructure are suddenly dead. In
On Tue, Jan 22, 2013 at 2:27 PM, Frank fr...@efirehouse.com wrote:
Hi all,
I registered my Digium D70 using a name (D70) instead of a number.
Is there a way to program Asterisk (or the phone?) so when I press the
MSGS button, it automatically goes to the correct voicemail, with or
without
On Tue, Jan 22, 2013 at 3:01 PM, Benny Amorsen benny+use...@amorsen.dkwrote:
Can a Nagios-based solution provide quicker failover than the 90 seconds
provided by sip timers or the 10-30 seconds provided by rtptimeout?
Certainly; Nagios can detect missed ping responses with a granularity of
I'm going to try to use DPMA.
I think I did everything right, but when comes the time to load the
res_digium module I have an error showing up:
*CLI module load res_digium_phone.so
== Host-ID: a:b:c:d:e
== Found license 'DPMA-xxx'
== Found total of 1 DPMA licenses
Unable to load
Not doubting how quickly Nagios can respond, but if the Nagios solution is
going to place a call using Asterisk, wouldn’t it be just as efficient (or more
so) to depend on Asterisk?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
How do you propose that Asterisk determines that the endpoint has vanished
off the network without waiting for a 10-90 second timeout period?
On Tue, Jan 22, 2013 at 4:07 PM, Danny Nicholas da...@debsinc.com wrote:
Not doubting how quickly Nagios can respond, but if the Nagios solution is
Using qualify=10 ?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christopher
Harrington
Sent: Tuesday, January 22, 2013 5:11 PM
To: Danny Nicholas
Cc: Asterisk Users Mailing List - Non-Commercial
Does that, then, solve Mitch's original issue? I only proposed Nagios
because nobody actually responded to his question.
On Tue, Jan 22, 2013 at 4:12 PM, Eric Wieling ewiel...@nyigc.com wrote:
Using qualify=10 ?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
A qualify value that low would be a resource hog (some phones can't even
re-register in 10 seconds). The Nagios solution would require a custom shell,
so it would less needy to make the shell be a daemon independent of either.
-Original Message-
From: Eric Wieling
My suggestion of Nagios was only to avoid re-implementing the detect
device has gone out to lunch code. Obviously the part interacting with AMI
is still custom and always will be.
On Tue, Jan 22, 2013 at 4:15 PM, Danny Nicholas da...@debsinc.com wrote:
A qualify value that low would be a
I should have actually suggested qualifyfreq=10.
Qualify has nothing whatsoever to do with registration.
-Original Message-
From: Danny Nicholas [mailto:da...@debsinc.com]
Sent: Tuesday, January 22, 2013 5:15 PM
To: Eric Wieling; ch...@acsdi.com; 'Asterisk Users Mailing List -
The Asterisk Development Team has announced the release of Asterisk 1.8.20.1.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 1.8.20.1 resolves several issues reported by the
community and would have not been
The Asterisk Development Team has announced the release of Asterisk 10.12.1.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 10.12.1 resolves several issues reported by the
community and would have not been possible
The Asterisk Development Team has announced the release of Asterisk 11.2.1.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 11.2.1 resolves several issues reported by the
community and would have not been possible
Dears;
Can someone advise me where to find a technology (open source) that let us able
to integrate with social media like whatsapp and facebook? And use this in call
center (queuing the messages and routing it for agent)?
Anyone give me a light to start?
Regards
Bilal
--
For just the messaging part, you should be able to use wget or curl to
interface and create messages. You might have to go a little higher level
like C or Perl, but it sounds very doable.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
What I'm trying to achieve is that a voicemail message should be at
least 3 seconds long for it to be saved, but *after that* a prolonged
silence (e.g. 10 seconds) should terminate the call and recording.
My current settings (Asterisk 10.7.0 and 11.2.1) are:
; Minimum length of a voicemail
Depending on exactly what you want, those may be the right settings. What
the warning is telling you is that with those settings, someone could
reach voicemail, say nothing (silence on the line) and after 10 seconds it
will hang up on them and deliver a 10 second silent message to their
Un-topposted
Eric Wieling ewiel...@nyigc.com writes:
Using qualify=10 ?
qualifyfreq=10 is fine, but Asterisk will not AFAIK do anything to a
call just because the peer goes unreachable qualify-wise. You are still
stuck with running a script that listens to qualify-unreachables and
does the
But with max silence at 2 seconds, couldn't someone leave a 30-second
message, pause for a couple seconds to gather their thoughts or dig up a
phone number, and get hung up on?
I'd think that the 3-second and 10-second settings are sensible. Saving a
message that is known to be total silence
On Tue, Jan 22, 2013 at 4:22 PM, asterisk users ast4...@gmail.com wrote:
What are the right settings for this situation?
We've used the following settings system-wide for about nine years without
one complaint or known issue:
[general]
format = wav49|pcm
maxsecs = 360
minsecs = 4
skipms =
Yes. That is why we don't use this setting.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly
Sent: Tuesday, January 22, 2013 6:44 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
On Tue, Jan 22, 2013 at 4:43 PM, Don Kelly d...@donkelly.biz wrote:
But with max silence at 2 seconds, couldn’t someone leave a 30-second
message, pause for a couple seconds to gather their thoughts or dig up a
phone number, and get hung up on?
Two seems short, but nobody has complained
Hi, Stefan,
Thanks for your reply.
SipRemoveHeader is a good application to remove the header added by
SipAddHeader application. If the header is included in the receiving
message by Asterisk(not added by SipAddHeader), can it be removed?
From my testing it doesn't work.
Ding Peng
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