Just FYI how I solved this:
I figured out that JACK_HOOK`ing for open channel does not connect
input and output ports. So instead of
*CLI core set chanvar SIP/poly1-ab23jadf234 JACK_HOOK(manipulate) on
you shoud use:
*CLI core set chanvar SIP/poly1-ab23jadf234
Hello list,
I would like to seek your expert opinion on a setup I am trying as part of my
research. I have not been able to successfully make a call so far.
In my setup, I use two laptops that are interconnected by means of a
stand-alone IS1581 switch. Thus there is no LAN involved.
I have
I am running asterisk 1.6.2.6 and have configured hints for our
extensions and have a couple of Aastra 6755i test phones. The phones
register fine but 'core show hints' shows the lines as idle even if they
are in use.
I read the wiki and see mention about needing to set call-limit in
asterisk
On Sun, 2010-05-02 at 09:52 -0400, Dan Journo wrote:
Hi Bob,
Thanks for that. Is there any way I can make the task run in the
background and free up the console? Also so that I can disconnect my
ssh session without losing the task.
Thanks
Dan
Matthieu NICAISE mentioned screen which
I read the wiki and see mention about needing to set call-limit in
asterisk 1.4 but that has been depreciated in 1.6 so what is the way it
should be done in 1.6?
I set
callcounter=yes
in sip.conf.
--
_
-- Bandwidth and
On Thu, May 6, 2010 at 8:14 PM, Vineet Bhojnagarwala vbho...@gmail.com wrote:
Alternatively, if using normal vlans, this can also be achieved by enabling
access list on the switch and restrict traffic flows. Generally this is done
on a layer 3 switch, don't think it will support on your switch
Hi,
Has anyone tried to use gnokii to send/receive SMS messages via serial or USB
with AT commands while running Asterisk?
Some of my calls have a scratchy sound once in a while. It doesn't seem to be
due to packet loss but some kind of interference (CPU is ok, etc.). I've
noticed some
On May 7, 2010, at 8:03, James Lamanna jlama...@gmail.com wrote:
On Thu, May 6, 2010 at 8:14 PM, Vineet Bhojnagarwala vbho...@gmail.com
wrote:
Alternatively, if using normal vlans, this can also be achieved by
enabling
access list on the switch and restrict traffic flows. Generally
On 05/07/2010 12:23 AM, Richard Kenner wrote:
Is there anything special that has to be done to make video calls work?
Yeah... Skype needs to add video support to the Skype engine that SFA uses.
It doesn't seem to work for me (no video).
That's right. It's not supported.
What CODECS are
stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI(SIP/3000-ca1c,
recordingcheck|20100507-082747|1273235267.398) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20100507-082747|1273235267.398: Outbound recording not
enabled
-- AGI
|3000|OUT) in new
stack
-- Executing GotoIf(SIP/3000-ca1c, 0 0?2:4) in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI(SIP/3000-ca1c,
recordingcheck|20100507-082747|1273235267.398) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
Richard Kenner wrote:
I read the wiki and see mention about needing to set call-limit in
asterisk 1.4 but that has been depreciated in 1.6 so what is the way it
should be done in 1.6?
I set
callcounter=yes
in sip.conf.
Thanks that works perfectly.
--
In which future release of Asterisk are we (since it is open-source, we
theoretically have some control) going to stop renaming and deprecating
features?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gareth
On Friday 07 May 2010 08:25:17 Danny Nicholas wrote:
In which future release of Asterisk are we (since it is open-source, we
theoretically have some control) going to stop renaming and deprecating
features?
I doubt that will ever happen. In the case of callcounter, that's not a
rename,
Do you see the creation of a buyer beware repository that host deprecated
features (like agentcallbacklogin) that aren't happy for current release but
might be desired for backward compatibility? Or is that just a port that
we would bring forward ourselves outside of the norm?
-Original
On Fri, 2010-05-07 at 08:25 -0500, Danny Nicholas wrote:
In which future release of Asterisk are we (since it is open-source, we
theoretically have some control) going to stop renaming and deprecating
features?
It's obviously more complicated that you make it seem with your comment.
Let me try
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me credentials to register to.
Connected to Asterisk 1.6.2.5 currently running on watermelon (pid = 10365)
watermelon*CLI sip show registry
Host dnsmgr Username
On 05/07/10 11:52, Gareth Blades wrote:
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me credentials to register to.
Connected to Asterisk 1.6.2.5 currently running on watermelon (pid = 10365)
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me credentials to register to.
Connected to Asterisk 1.6.2.5 currently running on watermelon (pid = 10365)
watermelon*CLI sip show registry
Host
Mike A. Leonetti wrote:
On 05/07/10 11:52, Gareth Blades wrote:
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me credentials to register to.
Connected to Asterisk 1.6.2.5 currently running on
On 05/07/10 12:14, Gareth Blades wrote:
Mike A. Leonetti wrote:
On 05/07/10 11:52, Gareth Blades wrote:
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me credentials to register to.
Hi Folks,
Is there a generally accepted Asterisk bible for current versions? I
poked around the forums and there didn't seem to be a real consensus,
and there are lots of options out there. I need something that focuses
on Asterisk dialplans and config files, not a linux primer. I'm looking
Mike A. Leonetti wrote:
On 05/07/10 12:14, Gareth Blades wrote:
Mike A. Leonetti wrote:
On 05/07/10 11:52, Gareth Blades wrote:
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me
I personally own (or have owned) about six different asterisk books, and
this one was far the most instrumental.
Asterisk: The Future of Telephony, 2nd Edition, dead tree edition
http://www.amazon.com/Asterisk-Telephony-Jim-Van-Meggelen/dp/0596510489/ref=sr_1_1
-Karl
- Original Message
On 05/07/10 12:40, Gareth Blades wrote:
Mike A. Leonetti wrote:
On 05/07/10 12:14, Gareth Blades wrote:
Mike A. Leonetti wrote:
On 05/07/10 11:52, Gareth Blades wrote:
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk
When something happens this will be good...
http://asteriskcookbook.com/wiki/index.php/Main_Page
~
Andrew lathama Latham
lath...@gmail.com
* Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
* Learn more about Linux http://en.wikipedia.org/wiki/Linux
* Learn more about Tux
Hi all,
I'm running Asterisk 1.6.2.7 using the following pseudo-dialplan (not actual
dialplan, because of complexity):
[something]
exten = s,1,Answer()
exten = s,n,AGI(blah,arg1,arg2)
exten = s,n,Playback(blah)
exten = s,n,DoMoreStuff()
exten = s,n,Hangup()
What I'd like to do, is have Asterisk
On 05/07/10 12:40, Gareth Blades wrote:
Mike A. Leonetti wrote:
On 05/07/10 12:14, Gareth Blades wrote:
Mike A. Leonetti wrote:
On 05/07/10 11:52, Gareth Blades wrote:
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk
Why not do System(blah) instead of AGI(blah) unless there is some AGI
specific item you need. To quote Steve Edwards, Do your AGI in C so it will
run 100 times faster and don't worry about when it comes back.
_
From: asterisk-users-boun...@lists.digium.com
I posted a topic about the billsec.
Asterisk around to the lesser int sec.
Exemple: If the call duration is 15.9 seconds, billsec (ANSWEREDTIME) will
be 15
I could change somethings in the source to have a correct calculation.
Exemple: : If the call duration is 15.9 seconds, then billsec
On Thu, May 6, 2010 at 3:11 PM, Steve Totaro
stot...@totarotechnologies.com wrote:
Yes, I purchased licenses for Fax for Asterisk and yes I called tech support
and had the WORST experience I have ever had with any technical support
call.
I am running Asterisk 1.6.2.6 and:
FAX For Asterisk
All:
Still experimenting with the asterisk server for the company.
My local phone company has given me two sip numbers to experiment with,
say 444-456-1234 444-456-5678
Calling in and out works, and I've spread a couple of the phones out
with some co-workers.
My question is this: Do I have
On Fri, May 7, 2010 at 3:41 AM, Jared Smith jsm...@digium.com wrote:
To make it more clear and less cryptic, we split out the callcounter
functionality in sip.conf, so that you could turn on/off the SIP device
state tracking without limiting calls, and encouraged people to use the
GROUP() and
On Friday 07 May 2010 13:59:23 Matt Darnell wrote:
On Fri, May 7, 2010 at 3:41 AM, Jared Smith jsm...@digium.com wrote:
To make it more clear and less cryptic, we split out the callcounter
functionality in sip.conf, so that you could turn on/off the SIP device
state tracking without
The SIP trunking service that I am trying to set up keeps saying that my
registration from Asterisk is invalid.
Asterisk registration:
REGISTER sip:{registration_ip} SIP/2.0
Via: SIP/2.0/UDP {asterisk_ip}:5060;branch=z9hG4bK5c2eb10c;rport
Max-Forwards: 70
From:
Hi,
checkout new open source voipmonitor.org SIP packet sniffer. I've
developed it for my telco company and I've decided to share it.
Testing and contributions are welcome!
VoIPmonitor is open source live network packet sniffer which analyze
SIP and RTP protocol. It can run as daemon or analyzes
Martin-
checkout new open source voipmonitor.org SIP packet sniffer. I've
developed it for my telco company and I've decided to share it.
Testing and contributions are welcome!
VoIPmonitor is open source live network packet sniffer which analyze
SIP and RTP protocol. It can run as daemon
Hi guys,
This has been bugging me for days and I can't figure out what's happening.
Using the same channel on DAHDI, I can't make a call after previous one being
hung up by remote side. However, it works like it should if my end hangs up the
call. To make things interesting, DAHDI, or
I think it is typical to have some limited number of outbound channels to your
SIP provider. You send all calls, up to your limit, to the same place. The
phone numbers your provider gave you are used to route inbound calls to your
asterisk box. You will typically have some limited number of
Does anyone know how to send a text message from Asterisk?
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
On Fri, May 7, 2010 at 2:01 PM, Martin asteriskl...@callthem.info wrote:
On Thu, May 6, 2010 at 3:11 PM, Steve Totaro
stot...@totarotechnologies.com wrote:
Yes, I purchased licenses for Fax for Asterisk and yes I called tech
support
and had the WORST experience I have ever had with any
On Fri, 7 May 2010, Thomas Perron wrote:
Does anyone know how to send a text message from Asterisk?
Carrier specific, but how about:
system(echo foo | mail -s bar 551...@txt.att.net)
--
Thanks in advance,
-
On 05/08/2010 08:15 AM, Steve Totaro wrote:
On Fri, May 7, 2010 at 2:01 PM, Martin asteriskl...@callthem.info
mailto:asteriskl...@callthem.info wrote:
On Thu, May 6, 2010 at 3:11 PM, Steve Totaro
stot...@totarotechnologies.com
mailto:stot...@totarotechnologies.com wrote:
thanks
do i need to have an smtp server somewhere. i tried directly from my
dialplan but no joy! i know you know that i am not a star with this
but any help would be cool
here is my config:
exten = 600,1,Answer()
exten = 600,n,Wait(1)
exten = 600,n,system(echo foo | mail -s bar
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