Hello
I'm new in this list, but I have some experience working with asterisk and
we are located in Bogota, Colombia.
At now I'm having some problems configuring an E1 MFC/R2.
I have configured Unicall as indicated using the following versions, and
everything seems to be correct,
On Tue, 16 Sep 2008, Roberts Klotins wrote:
Hi Gordon,
The test would be call your home BT line, hang up the home end and THEN
wait for how long time the line will stay open listening to the handset
that initiated the call.
That's exactly what I did.
If it is the caller (call initiator)
Thanks a lot Nhadie. I appreciate your help.
Could you also suggest some brands or models of the FXO+FXS card that
are seamlessly compatible to Asterisk? Also what hardphone I should go
for as there are so many in the market?
What should be the configuration of the system running this kind of
Great!
How stable was the Digium appliance?
Did it ever lock up or reboot without explanation?
Did you have any issues with phones locking up or rebooting?
I need to get a feel for how stable the appliances are so I can make a
decision on which to go with
Thanks
Femi
-Original Message-
Hi there!
Sorry, I should have started this as a separate thread. Here we go:
I wonder if anyone has set up Cellroute or Cellroute 3G mobile network
gateway (see http://www.gsmsave.com/acatalog/CellRoute-3G.pdf ) with
asterisk.
I am about to do that soon, therefore any experience would be
Hi Julien,
the soxmix (or sox in Asterisk 1.4 as default choice) is used by
Asterisk to record queues calls when you ask it to mix the in and out
calls, so I do not have control on it. Asterisk 1.2.x uses soxmix while
Asterisk 1.4.x uses sox instead but the command sox launched by Asterisk
Hi
I have set up voicemail, when i call an ext and voicemail kicks in i can leave
a message. The problem is that the message is in the tmp directory of the
extensions voicemail and when i call to check if there is any new vm there is
no messages at all.
Even if i move the message to inbox or
Hi
I have enabled realtime queue in asterisk, but when i enter a queue i get this
and then asterisk crashes.
Any clues?
-- Executing [EMAIL PROTECTED]:1] Answer(SIP/Ralf-08207de0, ) in new stack
-- Executing [EMAIL PROTECTED]:2] Ringing(SIP/Ralf-08207de0, ) in new
stack
-- Executing
On Wed, Sep 17, 2008 at 11:57 AM, Ralf Träskman [EMAIL PROTECTED] wrote:
Hi
I have enabled realtime queue in asterisk, but when i enter a queue i get
this and then asterisk crashes.
Any clues?
-- Executing [EMAIL PROTECTED]:1] Answer(SIP/Ralf-08207de0, ) in new
stack
-- Executing
Hi all,
I am having a problem with sip uri incoming calls. I have 2 asterisk servers
both are 1.4.2. i dial sip uri from one asterisk server which sends the call
to the other asterisk server by seeing its domain name in the uri. Invite
reaches the recieving asterist server but the call is not
hi all,
Is there an option of dtmf passthru mode in asterisk. If yes, how can i do
it?
--
Best Regards
Rizwan Hisham
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AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register
Look at the canreinvite option.
- Original Message
From: Rizwan Hisham [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, September 17, 2008 3:20:40 PM
Subject: [asterisk-users] dtmf passthru
hi all,
Is
ooopss.. I was in a hurry, wasn't i!??
what is DTMF pass thru??
As far as I know, there's nothing specific for just DTMF as pass through.. its
for the entire call that is established.. for the codecs being used within the
call.
What is the requirement anyway?
- Ben.
- Original Message
If there is a secret= on the receiving peer, the sending peer needs to
provide that secret. Along with a username.
Rizwan Hisham wrote:
Hi all,
I am having a problem with sip uri incoming calls. I have 2 asterisk
servers both are 1.4.2. http://1.4.2. i dial sip uri from one asterisk
What is DTMF passthru?
DTMF is regenerated by default. If the DTMF mode is inband, it's simply
part of the audio stream. If it uses named RTP events, those are
regenerated on the other call leg.
Rizwan Hisham wrote:
hi all,
Is there an option of dtmf passthru mode in asterisk. If yes, how
Dear,
is any command to show the codecs of channels , in asterisk 1.4?
Best
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AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
solved
with
sip show channels
best
--- On Wed, 9/17/08, Pezhman Lali [EMAIL PROTECTED] wrote:
From: Pezhman Lali [EMAIL PROTECTED]
Subject: [asterisk-users] codec of channels
To: asterisk-users@lists.digium.com
Date: Wednesday, September 17, 2008, 5:42 PM
Dear,
is any command to show the
On Tue, Sep 16, 2008 at 2:21 AM, Michiel van Baak [EMAIL PROTECTED] wrote:
On 22:46, Mon 15 Sep 08, hugolivude wrote:
I have two Asterisk servers running on the same LAN. One starts fine,
but when I start the other I get:
pbx_dundi.c:4582 load_module: Unable to bind to 0.0.0.0 port 4520:
That means someone else has already open the zap device, most likely
Asterisk. Just one application at a given time can open a zap device.
You cannot run testcall and Asterisk at the same time unless you make
sure they don't try to open the same channels.
Moy
On Wed, Sep 17, 2008 at 1:27 AM, Dae
Just a quick question
---cut---
[Sep 17 15:52:14] WARNING[8232] app_dial.c: Unable to create channel of type
'IAX2' (cause 34 - Circuit/channel congestion)
[Sep 17 15:52:14] WARNING[8232] chan_iax2.c: No more space
[Sep 17 15:52:14] WARNING[8232] chan_iax2.c: Unable to create call
[Sep 17
Check on this link,
http://www.moythreads.com/wordpress/
I have working asterisk+mfc/r2 with 4 E1 on Venezuela and work fine!
Regards,
Luis Morales
On Thu, Sep 18, 2008 at 1:57 AM, Dae Yeung Um [EMAIL PROTECTED] wrote:
Hello
I'm new in this list, but I have some experience working with
Alex,
So u mean DTMF is by default passed thru already.
Benjamin,
Well i dont want to enforce my asterisk dtmf setting on any call. What i
want is whatever the user has set the dtmf mode in his ata or softphone,
that should be used to pass the dtmf signals on to the callee.
On Wed, Sep 17, 2008
thats what i am passing
exten= 456,1,Dial(SIP/adf:[EMAIL PROTECTED]:9060
adf is username and 123 is the password
On Wed, Sep 17, 2008 at 6:01 PM, Alex Balashov [EMAIL PROTECTED]wrote:
If there is a secret= on the receiving peer, the sending peer needs to
provide that secret. Along with a
Hi everybody,
Is there any way to remove dialtone from DISA? If yes, how to do it?
--
Zeeshan A Zakaria
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Register Now:
Hi everybody,
I am having DTMF detection problem on DISA with my callback system. For many
users, it keeps playing the dialtone even after they have input their
number. I have trunk setup to both g729 and ulaw. What could be the reason
for this problem. Some users have to dial a few times before
On Wednesday 17 September 2008 11:25:36 Zeeshan Zakaria wrote:
I am having DTMF detection problem on DISA with my callback system. For
many users, it keeps playing the dialtone even after they have input their
number. I have trunk setup to both g729 and ulaw. What could be the reason
for this
I had it setup on rfc2833. Now I've set it up to auto. Will see how it will
work. But I was thinking is it possible that DTMF tones get distorted on
their way from my server to the provider's server, which cause this problem?
On Wed, Sep 17, 2008 at 12:43 PM, Tilghman Lesher
[EMAIL PROTECTED]
You can use the variable MONITOR_EXEC in your extensions.conf to specify
the shell command to be invoked by the Monitor application to mix the
voice files.
The shell command will be invoked with three command lines arguments
appended: the file recording for the in leg (created by Asterisk); the
Hello, all.
We have an Asterisk 1.4.17 installation and we have setup the dialplan
such that all calls to/from a set of phones (SIP accounts) get
recorded. We do this by ensuring the Monitor application gets
invoked at the start of all calls. We also have canreinvite=no on
the general section of
I'm running 1.4.22-rc5, and now see the following codec format listed:.
What is 0x80004 (ulaw|h) ?
192.168.1.14 101 YjVlYzYwODd 00101/2 0x4 (ulaw) No Rx:
ACK
172.16.1.1 102 7b213e4762c 00102/0 0x80004 (ulaw|h No
Init: INVITE
I'm having some voice quality
Maybe a bit silly question, but why doesn't Asterisk accept if you set
both a usernamepassword as well as an ip address for a phone?
My fixed phones in my home all have a fixed ip address, but i also have 2
Nokia GSM phones that can talk sip wich i would like to use from public
wifi.
It's
On Wednesday 17 September 2008 12:02:44 Zeeshan Zakaria wrote:
I had it setup on rfc2833. Now I've set it up to auto. Will see how it will
work. But I was thinking is it possible that DTMF tones get distorted on
their way from my server to the provider's server, which cause this
problem?
Anybody knows how to get a Coupon Code for the discount on the Asterisk
training classes??? I am interested on taking that upcoming Asterisk
Advance course, and 3K is kinda steep and considering I am still a college
student paying this training out of my pocket, every bit helps.
Maybe a bit silly question, but why doesn't Asterisk accept if you set
both a usernamepassword as well as an ip address for a phone?
but it does accept!
in a peer definition:
[user]
type=user (or better friend)
username=user
secret=secret
host=10.0.0.1
[snip]
It's obvious that the more
# asterisk -rvv
IPPBX*CLI show codecs
Do i understand wrong _?
On Wed, Sep 17, 2008 at 4:12 PM, Pezhman Lali [EMAIL PROTECTED]wrote:
Dear,
is any command to show the codecs of channels , in asterisk 1.4?
Best
___
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On Wed, 2008-09-17 at 19:58 +0200, Remco Barendse wrote:
Why doesn't Asterisk allow both usernamepass as well as setting an ip
adress on a sip.extension?
It does. To enforce ACLs on a SIP user or peer or friend, simply use
permit and deny statements to allow and disallow various IP
addresses
Can anyone explain parked calls?
I've run so many tests over the last few hours I'm totally confused. Half the
time the call times out and returns back to the user that dialed it, through
the same context it was originated from.
The other half it returns to the park-dial context with a
Forgot to mention, I'm running asterisk 1.4.21.2
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Mann
Sent: Wednesday, September 17, 2008 2:32 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Parked
Dear,
I have a little problem with app_conference,
the very low power voices, were amplified, too much,
and normal voices were destroyed.
codec=g729
asterisk=1.4.19
app_conference =last released
best
Mani
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On 17 Sep 2008, at 14:57, Philipp Kempgen wrote:
Just a quick question
---cut---
[Sep 17 15:52:14] WARNING[8232] app_dial.c: Unable to create channel
of type 'IAX2' (cause 34 - Circuit/channel congestion)
[Sep 17 15:52:14] WARNING[8232] chan_iax2.c: No more space
[Sep 17 15:52:14]
Why are you doing that?
Pezhman Lali wrote:
Dear,
i am using asterisk 1.19 and app_conference 2.0.1 .
--- On *Thu, 9/18/08, Anthony Francis /[EMAIL PROTECTED]/* wrote:
From: Anthony Francis [EMAIL PROTECTED]
Subject: Re: [asterisk-users] app_confrence with loud voices
To:
It will syntactically take that definition, but it's nearly pointless. I
suspect he is encountering a problem I have found:
It doesn't do full support static and registration simultaneously.
Most notably, at least in 1.2 (and probably 1.4) if you create a
registrable peer it is NOT monitorable
How are SIP Info Messages interpreted by asterisk as far as I
understand, there are 17 different DTMF messages possible via sip info
DTMF 1,2,3,4,5,6,7,8,9,0 also A,B,C,D,E,F and Flash
, are any of these interpreted by asterisk, as I cannot seem to get
asterisk to use the Flash information to
Tim Panton schrieb:
On 17 Sep 2008, at 14:57, Philipp Kempgen wrote:
Just a quick question
---cut---
[Sep 17 15:52:14] WARNING[8232] app_dial.c: Unable to create channel
of type 'IAX2' (cause 34 - Circuit/channel congestion)
[Sep 17 15:52:14] WARNING[8232] chan_iax2.c: No more space
Philipp Kempgen schrieb:
Tim Panton schrieb:
On 17 Sep 2008, at 14:57, Philipp Kempgen wrote:
Just a quick question
---cut---
[Sep 17 15:52:14] WARNING[8232] app_dial.c: Unable to create channel
of type 'IAX2' (cause 34 - Circuit/channel congestion)
[Sep 17 15:52:14] WARNING[8232]
Hi,
I have need to measure the end-to-end audio delay in the MeetMe conference
application.
Currently, I have written a python program that connects to an Asterisk
MeetMe conference via SIP, and pumps RTP packets into the conference.
Another instance of the program dials into the same
On Wed, 17 Sep 2008, George Williams wrote:
I have need to measure the end-to-end audio delay in the MeetMe conference
application.
Currently, I have written a python program that connects to an Asterisk
MeetMe conference via SIP, and pumps RTP packets into the conference.
Another instance
Hi. I'm writing a speech recognition module for Asterisk. I'm having
problems with simultaneous SIP and PSTN calls. Sometimes Asterisk crashes in
this scenario. I don't have problem with simultaneous calls using PSTN calls
only.
The implementation is in the file res/res_speech.c
Does someone know
Femi wrote:
Great!
How stable was the Digium appliance?
Solid
Did it ever lock up or reboot without explanation?
No
Did you have any issues with phones locking up or rebooting?
No
I need to get a feel for how stable the appliances are so I can make a
decision on which to go with
On Tue, Sep 16, 2008 at 2:21 AM, Michiel van Baak [EMAIL PROTECTED] wrote:
On 22:46, Mon 15 Sep 08, hugolivude wrote:
I have two Asterisk servers running on the same LAN. One starts fine,
but when I start the other I get:
pbx_dundi.c:4582 load_module: Unable to bind to 0.0.0.0 port 4520:
On Wed, Sep 17, 2008 at 2:37 PM, Pascal Bruno [EMAIL PROTECTED] wrote:
Anybody knows how to get a Coupon Code for the discount on the Asterisk
training classes??? I am interested on taking that upcoming Asterisk
Advance course, and 3K is kinda steep and considering I am still a college
That is good you have all those years of experiences and you might know more
than the instructor. But I dont see the connection, or the point you are
trying to make. The question is that there is a space to apply a coupon
code, and I was wondering how and where one could get one. I don't recall
Thank you for the reply
I shutdown asterisk and tried again and I have to following logs...
OUTGOING TEST :
Testcall.conf
caller yes
destination-no 6055151
originating-no 7309130
protocol-class mfcr2
protocol-variant ar,20,4
circuits 1-2
Log:
It seems to me your lines are blocked.
Execute zttool and if you see 1101 in the rx bits, it means the telco
(or whatever you have in the other end) has blocked their side. If
this is a telco line you need to call them and tell them to unblock
your lines.
On Wed, Sep 17, 2008 at 10:33 PM, Dae
In fact I see 1101 in the rx bits on all channels...
But I have in parallel one old Panasonic Key Phone system (Actually in
production, to be replaced by asterisk), and it's works perfectly and
immediately once I pass the E1 cables to there...
So, the problem is not from Telco...
-Original
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