${FILTER(0-9,${cid})} works.
Thanks.
Jonas.
On 03/20/2012 06:02 PM, Danny Nicholas wrote:
Since you never know when you will actually use one of these, I tried
it this way in 10.1.3
exten = 1238,1,answer
exten = 1238,n,Set(cid=+9600)
exten = 1238,n,saynumber(${cid2)
exten =
I am still getting the same error
On 3/21/12, white hat whitehat...@gmail.com wrote:
Just a guess here, but it looks like you are dialing a 10 digit phone
number but the dial pattern in your outbound route does not handle that.
Try using a different dial pattern in your outbound route such
my sip traces are below
Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 192.168.9.250 port 17722
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to
[0K
--- SIP read from UDP:192.168.9.251:5060 ---
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.9.250:5060;rport=5060;branch=z9hG4bK111ef687
To:
sip:0722490994@192.168.9.251;tag=1332328154302b4aa3-f15d-4eb7-beee-97782e7cbd06
From: pbxserver sip:Unknown@192.168.9.250;tag=as66c75bd7
I am setting up asterisk-fxo gw.
404 Not Found (User not found) means the user is not found, but I
don't need to have extensions or authentication on the fxo gw
On 3/21/12, Michael L. Young myo...@acsacc.com wrote:
[0K
--- SIP read from UDP:192.168.9.251:5060 ---
SIP/2.0 404 Not Found
Via:
Hi
I was asked by our development departement to setup asterisk in a
manner that if someone calls an extension in the department that was
was only configured, but a handset was never attached to it to fall
back to a default extension. For example: Someone calls extension
2408, but there's no
On Wed, Mar 21, 2012 at 8:27 AM, Paolo Supino paolo.sup...@gmail.com wrote:
Hi
I was asked by our development departement to setup asterisk in a
manner that if someone calls an extension in the department that was
was only configured, but a handset was never attached to it to fall
back to a
Paolo
You can use exten - i This will catch any invalid extensions that are
sent into a context. You could than route the flow as you see fit.
Thanks
Bryant
From: Paolo Supino paolo.sup...@gmail.com
Sent: Wednesday, March 21, 2012 8:24 AM
To:
On Mar 21, 2012, at 08:36 , Andrew Latham wrote:
On Wed, Mar 21, 2012 at 8:27 AM, Paolo Supino paolo.sup...@gmail.com wrote:
Hi
I was asked by our development departement to setup asterisk in a
manner that if someone calls an extension in the department that was
was only configured, but a
Over the years I have experienced a few interrupt issues when using some
of the Digium E1/T1 cards with Zaptel drivers, and usually resolved them
by disabling USB devices in the motherboard BIOS settings.
Now more and more systems are coming without PS/2 connections, so USB is
needed for the
On Wed, Mar 21, 2012 at 8:45 AM, Tony Mountifield t...@softins.co.uk wrote:
Over the years I have experienced a few interrupt issues when using some
of the Digium E1/T1 cards with Zaptel drivers, and usually resolved them
by disabling USB devices in the motherboard BIOS settings.
Now more and
404 NOT FOUND means that they were unable to find any
destination/route/rule/prefix match corresponding to your dialled number.
See your FXO gateway configuration Web-UI for outbound patterns OR verify
that the FXO has its outbound line configured and working properly.
On Wed, Mar 21, 2012 at
Hi,
I have configured a route on the fxo to send all incoming sip traffic
to the fxo ports.
I will try set the specific digits and see.
On 3/21/12, SamyGo govoi...@gmail.com wrote:
404 NOT FOUND means that they were unable to find any
destination/route/rule/prefix match corresponding to your
My question is so complex and I try to explain well.
We have a customer that he wants limits incoming calls to his extensions
to only one. That's not complicated with GROUPCOUNT, DEVICE_STATE or
SIPPEER with curcalls option.But the problem is when you want implement
CCBS service.
If we have
Extension i only works for IVRs and other things like Background and
WaitExten, it does not work to match incoming calls to an invalid extension.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant
Hey,
I would also recommend to use SIPPEER and with that verify the status of said
peer. Based on that status, make the dialling decision.
If you want more help, contact me directly.
Rennes Neps
Elion Ettevõtted AS
tel: +372 6402183
mob: +372 56490388
rennes.n...@elion.ee
-Original
Am 20.03.12 10:15, schrieb Olivier:
Hi,
I would like to test the following COLP use case :
Alice and Bob are both using a SIP phone registered on a Asterisk 10 server.
Alice dials Bob's extension.
While Bob's phone is ringing, Asterisk updates Alice phone screen with
Bob's name, so that
On Wed, Mar 21, 2012 at 12:45:37PM +, Tony Mountifield wrote:
Over the years I have experienced a few interrupt issues when using some
of the Digium E1/T1 cards with Zaptel drivers, and usually resolved them
by disabling USB devices in the motherboard BIOS settings.
Now more and more
On Wed, 21 Mar 2012, Shaun Ruffell wrote:
240 channels in meetme comming from an 8-span digital card? I would have
to measure it...but my guess is a pretty beefy system. In this
configuration, more speed on less cores would serve you better than more
cores.
Would that advice apply if the
On Wed, Mar 21, 2012 at 08:23:48AM -0700, Steve Edwards wrote:
On Wed, 21 Mar 2012, Shaun Ruffell wrote:
240 channels in meetme comming from an 8-span digital card? I
would have to measure it...but my guess is a pretty beefy system.
In this configuration, more speed on less cores would serve
Hello All,
I need to know a way of connecting an Answered call in Asterisk to another
call which was triggered by an AMI. I have a scenario as follows:
1) User dials 123 from a touch screen Polycom phone.
2) Call comes to Asterisk and Asterisk answers the call and asks for PIN
number.
3) Once the
In article 20120321151844.ga11...@digium.com,
Shaun Ruffell sruff...@digium.com wrote:
On Wed, Mar 21, 2012 at 12:45:37PM +, Tony Mountifield wrote:
Over the years I have experienced a few interrupt issues when using some
of the Digium E1/T1 cards with Zaptel drivers, and usually
On 03/21/2012 11:17 AM, Tony Mountifield wrote:
240 channels in meetme comming from an 8-span digital card? I would
have to measure it...but my guess is a pretty beefy system. In this
configuration, more speed on less cores would serve you better than
more cores.
Yes, I wondered about that,
2012/3/21, Stefan Schmidt s...@sil.at:
Am 20.03.12 10:15, schrieb Olivier:
Hi,
I would like to test the following COLP use case :
Alice and Bob are both using a SIP phone registered on a Asterisk 10
server.
Alice dials Bob's extension.
While Bob's phone is ringing, Asterisk updates Alice
Hello,
when generating backtrace I get following output :
/[root@sip ~]# gdb -se asterisk -ex bt full -ex thread apply all
bt --batch -c core.sip-2012-03-21T10\:57\:29+0100 /root/backtrace.txt
asterisk: No such file or directory./
/warning: no loadable sections found in added symbol-file
H Andrew
Your solution is the simplest I received and so I tried implementing
it only to discover that it doesn't work as expected...
TIA
Paolo
On Wed, Mar 21, 2012 at 1:36 PM, Andrew Latham lath...@gmail.com wrote:
On Wed, Mar 21, 2012 at 8:27 AM, Paolo Supino paolo.sup...@gmail.com
On Wed, Mar 21, 2012 at 3:10 PM, Paolo Supino paolo.sup...@gmail.com wrote:
H Andrew
Your solution is the simplest I received and so I tried implementing
it only to discover that it doesn't work as expected...
TIA
Paolo
snip
Check your Dial() options... Verify your options to you
Hi
I've pretty much have it setup properly with the following:
exten = _24XX,1,Dial(SIP/${EXTEN},30)
exten = _24XX,n,GotoIf($${DIALSTATUS}=CHANUNAVAIL?noconn:conn)
exten = _24XX,n(noconn),Dial(SIP/2400)
exten = _24XX,n(conn),hangup()
The only problem is that if 2400 rejects the call asterisk
From: Paolo Supino paolo.sup...@gmail.com
Sent: Wednesday, March 21, 2012 3:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] fallback to default extension
Hi
I've pretty
Minor Correction
Hi
I've pretty much have it setup properly with the following:
exten = _24XX,1,Dial(SIP/${EXTEN},30)
exten = _24XX,n,GotoIf($${DIALSTATUS}=CHANUNAVAIL?noconn:conn)
exten = _24XX,n(noconn),GotoIf($[${EXTEN}=2400]?conn:force)
exten = _24XX,n(force),Dial(SIP/2400)
exten =
Make your user wait in a *Meetme* and then call your destination number
through AMI and once he answers, place him in the same *Meetme*.
e.g. Assuming your destination is SIP extension, have something like...
Action: Originate
Channel: SIP/{your_destination_here}
Application: MeetMe
Data:
Thank you Satish. I was also thinking on similar lines. I was just
wondering if there was any mechanism with which we can bridge a new call
with the existing running call if the Call-ID of the call is known !!
I can definitely use the confbridge application for the same right; as I am
working on
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