For India following are the parameters that i used to configure with sangoma
E1 cards,
coding - HDB3
framing - CRC4 or Non-CRC depends upon the service provider
line type - national
switchtype - EuroISDN
On Thu, Feb 10, 2011 at 11:20 AM, Faisal Hanif fai...@vopium.com wrote:
The settings you
On Thu, 10 Feb 2011 13:08:29 +1000, Da Rock
asterisk-us...@herveybayaustralia.com.au wrote:
I have an asterisk 1.8 server running on FreeBSD 8.1, and another
FreeBSD 8.1 running as a firewall/gateway with PF.
Does it work if you remove the firewall from the equation?
Since Internode is an OZ
This sounds like a job for DISA.
http://www.voip-info.org/wiki/view/Asterisk+cmd+DISA to see if DISA
helps.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius
Smith
Sent: 03 February 2011 19:46
To:
Hello
We have quite some problems with early audio with our asterisk 1.6.2.15
What we observe is:
Asterisk - Carrier PBX
Asterisk:Invite(+sdp) = Carrier
Carrier starts to send RTP Audio (ignored by Asterisk)
Asterisk = Carrier:100 Trying
Asterisk = Carrier:180 Ringing
Asterisk signals
Go here http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
and use proper parameters to dial command to pass early media.
-Original Message-
From: Benoit Panizzon
Sent: Thursday, February 10, 2011 4:08 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Early audio SIP
Hello,
Maybe try that:
In your incoming isdn context:
[isdn-incoming]
exten = s,1,Set(TIMEOUT(digits)=3)
exten = s,2,WaitExten(2)
exten = s,3,Dial(SIP/operator...)
exten = 10,1,Dial(SIP/10)
exten = 20,1,Dial(SIP/20)
So if a call comes in Asterisk waits, 2 seconds for further digits
dialed and
Good morning everyone.
I wonder if it is possible, without touching the source code, to Asterisk
save the cdr with date in unix time instead of the default date. It's
possible?
Thanks in advance,
--
Rodrigo Lang
Opening your mind - Just another Open Source
Hi Faisal
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
Thank you, but I seem to miss the option which tells asterisk to pass audio
even if no 183 or 200 is received.
No, we don't set the 'r' Flag while dialing out.
So, my question ist sill the same.
Sould asterisk pass audio of it
Well. I suggest to use DB function instead of modifying asterisk source. You
can add one additional column and write and after-insert trigger in your cdrs
table which convert dattime to your required format and update the value of
added column.
From: Rodrigo Lang
Sent: Thursday, February 10,
Just use uniqueid, which is exactly what you want. No modification is
necessary.
Regards,
Mindaugas Kezys
Kolmisoft UAB
VoIP Billing Solutions
e-mail: i...@kolmisoft.com
URL: http://www.kolmisoft.com
Find us on Facebook
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christian
Gansberger
Sent: Thursday, February 10, 2011 5:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question
On Thu, Feb 10, 2011 at 6:08 AM, Andrew Thomas a...@datavox.co.uk wrote:
This sounds like a job for DISA.
http://www.voip-info.org/wiki/view/Asterisk+cmd+DISA to see if DISA
helps.
If OP is using Asterisk18, perhaps we should direct him to look here?
...or there :)
Anyway AT sends the call before they finish dialling all 8 digits
means that they don't send all the digits. Conflicting sentence in OP.
Perhaps it would help if the OP could determine if AT actually send 6 or
8 digits in the signalling (I reckon it's 6).
On Wed, Feb 9, 2011 at 6:55 AM, Vieri rentor...@yahoo.com wrote:
[snip]
Since all of the SIP devices in my LAN have static IP addresses, I can keep
track of
everyone on my own. For instance, could I do fake SIP registrations from
localhost
(the * server) and specify a LAN IP address?
On 02/10/2011 06:15 AM, Benoit Panizzon wrote:
Hi Faisal
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
Thank you, but I seem to miss the option which tells asterisk to pass audio
even if no 183 or 200 is received.
No, we don't set the 'r' Flag while dialing out.
So, my question ist
Hi,
I'm having an issue with busy detection, the busy is not being detected.
Asterisk: 1.6.2.13
DAHDI: 2.4.0
Chandahdi: busydetect=yes, busycount=2
Indications zone = us, with the modifications for my country for busy:
425Hz Pattern(0.2ms on, 0.2ms off, 0.2ms on, 0.6ms off)
On Thursday 10 February 2011 06:13:38 Rodrigo Lang wrote:
I wonder if it is possible, without touching the source code, to
Asterisk save the cdr with date in unix time instead of the default
date. It's possible?
The answer is, it depends upon the backend version you're using. With
cdr_pgsql
--- On Thu, 2/10/11, Jonathan Thurman jonat...@thurmantech.com wrote:
Have you looked at the 'defaultip' sip configuration
option? Or
setting host=IP for those devices?
I've read that defaultip can only be used on type=peer and when host=dynamic.
I use type=friend.
host=IP seems to be OK
We are using PostgreSQL real-time connector (res_config_pgsql) with Asterisk
1.6.2.15.
From time to time, we need to reset our PostgreSQL server, causing all active
DB connections to close. While other packages in our system re-connect
gracefully when this happens, Asterisk appears to not
OK, im pulling my hair out, everything looks configured right, deleted, and
started over, etc, etc. but can't seem to get this to work
Gtalk.conf
[general]
context=google-in
allowguest=yes
bindaddr=192.168.xxx.xxx
extenip=96.254.xxx.xxx
[guest]
context=google-in
disallow=all
Sorry, Asterisk Build 1.6.2.7
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William
Stillwell
Sent: Thursday, February 10, 2011 6:50 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users]
i have been trying to find a way to accomplish the following but have not
found a method in which to do so:
i am trying to configure the meetme conference (asterisk 1.8) to play a *
random* sound file from a specific directory prior to it dropping the caller
into the conference itself. i am able
On Thu, Feb 10, 2011 at 04:58:05PM -0800, John Jolly wrote:
i am trying to configure the meetme conference (asterisk 1.8) to play a *
random* sound file from a specific directory prior to it dropping the caller
into the conference itself.
Absent an Asterisk-specific solution, how about a
On Thu, 10 Feb 2011, John Jolly wrote:
i am trying to configure the meetme conference (asterisk 1.8) to play a
random sound file from a specific directory prior to it dropping the
caller into the conference itself. i am able to successfully get it to
play a specific file prior to entering the
You've got connection=jp_jabber defined in one file, and [jb_jabber] defined in
the other.
Thanks,
--Warren Selby, dCAP
On Feb 10, 2011, at 5:55 PM, William Stillwell will...@stillwellsoft.com
wrote:
Sorry, Asterisk Build 1.6.2.7
From: asterisk-users-boun...@lists.digium.com
Yeah, that was a typo, but I fixed, still no dice.
The incoming jabber call doesn’t fire the gtalk connection.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Thursday, February 10, 2011 10:16 PM
To:
Ok, stopped asterisk
Backed up all modules
Recompiled asterisk to lastest version.
Same thing… jabber call come in, but no firing of the gtalk/extension..
Now running build 1.6.2.16.1
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
On Thu, Feb 10, 2011 at 11:17 PM, William Stillwell
will...@stillwellsoft.com wrote:
Ok, stopped asterisk
Backed up all modules
Recompiled asterisk to lastest version.
Same thing… jabber call come in, but no firing of the gtalk/extension..
Now running build 1.6.2.16.1
Try
William,
Have you tried outgoing calls? What happens there?
Have you restarted the Asterisk after you fixed the typo?
-Vladimir
On 2/10/2011 10:44 PM, William Stillwell wrote:
Yeah, that was a typo, but I fixed, still no dice.
The incoming jabber call doesn’t fire the gtalk
William,
I have just noticed that you have several configuration statements
commented out.
I would suggest to un-comment the status= in jabber.conf. I would
also suggest to un-comment the timeout=, I am not that concerned of
the keepalive=.
You can reload jabber, no need to restart the
I was getting unable to make channel..
So, this is what I am doing..
Service stop asterisk
Purge modules
Make clean
Remove all traces of iskemel
Recompile that. With , add needed entrée into ldconfig.
Verify iksemel loaded via ldconfig –p | grep semel.
Change to /asterisk source
Still no dice..
This make no since.. ive gone over the config a million times now..
The windows gtalk /voice client works just fine. (incoming and outgoing calls)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Trying to install wanpipe 3.5.18.
No errors during compile. But when I reach the point where wanpipe and
dahdi_cfg is started, I encountered an error.
Starting WAN Router...
Loading WAN drivers: wanpipe done.
Starting up device: wanpipe1
wanconfig: WAN device wanpipe1 driver load failed
William,
I have gone through the similar frustration recently. Everything works
as of early morning yesterday. The big difference, I am on 1.8.2.3.
Have you seen this ticket on the tracker
https://issues.asterisk.org/view.php?id=10512 ? Anything applicable to
your case? The messages are
William,
Another thing. Have you tried calling from GMail? If not please make
sure you can send/receive calls there.
One more test. Go to your GV Account Settings / Phones, Edit Google
Chat, Save Watch for the pink error messages in the upper portion
of the screen.
-Vladimir
On
I don’t’ appear to have an jabber [] OUTGOING packets?
I get just 1 incoming packet, and it just sits there, until it rings to
voicemail.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir Mikhelson
Sent: Friday,
William,
Another thing to exclude is networking. Can you verify that nothing
blocks the specific traffic on your network? Any chance of taking the
packet trace on your gateway?
-Vladimir
On 2/11/2011 1:18 AM, William Stillwell wrote:
I don’t’ appear to have an jabber [] OUTGOING packets?
I only have one gtalk account.
I Double checked the chat settings.
For some reason jabber is not sending any outbound response packets at all..
not sure why. Will need to see if I can stuff some more debug code into
res_jabber.c and figure out whats going on, debug seems limited.
Hi William,
just to know that gtalk/asterisk works in your environment you could
quickly create a virtual server and install an asterisk 1.8 with this
guide
http://highsecurity.blogspot.com/2010/11/googlevoice-asterisk-18-with-freepbx.html
which works fine for me.
this way you know for sure that
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