You can use following command to check
netstat -an
This will show host and ports in numeric format.*
Regards,*
Amit Patkar
On 2/27/2015 6:33 AM, Rusty Newton wrote:
On Mon, Feb 23, 2015 at 5:51 AM, Raj Roy Ghandhi roy.gan...@gmail.com
mailto:roy.gan...@gmail.com wrote:
Hi Friends,
On Thu, Feb 26, 2015 at 10:34 AM, Salaheddine Elharit
salah.elharit...@gmail.com wrote:
hello liste
i have creat i trunk sip and inboun route for inbound calls the issue whe
i use the DID in inboud route i have a error No DID or CID Match.
but when i leave this DID field blank i can route
2015-02-25 18:23 GMT-06:00 John Kiniston johnkinis...@gmail.com:
I'd recommend using DEVICE_STATE
On your extension 101, Check the DEVICE_STATE of peer SIP/101, If it's not
'NOT_INUSE' then dial it, Otherwise dial SIP/102
exten =
On Wednesday 25 Feb 2015, ricky gutierrez wrote:
I have a gw wiht 4 port gsm , my provider gives me 4 lines and one of
them is the main , the problem is that all my incoming calls using
this number and is always busy , and the other three are always free,
it is possible that the call is
hello liste
i have creat i trunk sip and inboun route for inbound calls the issue whe i
use the DID in inboud route i have a error No DID or CID Match.
but when i leave this DID field blank i can route the call without any issue
how can ido in order to use DID in route inboud i use elastix
Hi,
I would like to do some tasks after the CDR has been closed, and the
CDR(end), CDR(billsec) and CDR(duration) fields are available. I have tried
to do that on the h extension, but it seems the CDR is not yet complete in
the h extension.
When is the CDR closed? How can I trigger some actions
On 26 February 2015 at 11:57, Daniel Gonzalez gonva...@gonvaled.com wrote:
Hi,
I would like to do some tasks after the CDR has been closed, and the
CDR(end), CDR(billsec) and CDR(duration) fields are available. I have tried
to do that on the h extension, but it seems the CDR is not yet
Can anyone recommend a good WebRTC phone to use with Asterisk? I do
not mind if it is commercial or open source. Customers are starting to
ask for web solutions and we need to start testing.
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)9116-91161
--
2015-02-26 10:45 GMT-06:00 A J Stiles asterisk_l...@earthshod.co.uk:
You just need to use call groups.
In your chan_extra.conf (if it's an OpenVox) or chan_dahdi.conf, add
something like
group=1
to the definition for each span.
Now in the [globals] section of your dialplah, have
On Mon, Feb 23, 2015 at 5:51 AM, Raj Roy Ghandhi roy.gan...@gmail.com
wrote:
Hi Friends,
I encountered a strange issue.
I am running Asterisk 11.8.1 on Cent OS with no firewall running.
It has 3 NIC interfaces.
in my sip.conf I have
allowguest=yes
bindaddr=0.0.0.0
udpbindaddr = 0.0.0.0
For the client:
JSSIP and Sipml5.
If you are going to be coding something up yourself I like the JSSIP 0.5.x
javascript interfaces. If you are simply going to use a pre-canned one then
sipml5 works pretty well and remembers your settings in localstorage. I
haven't used any closed source versions
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