* You are viewing the archive for May 11th, 2011

Disabling echo cancellation by software

Dear, I have Asterisk 1.6 with an E1 Digium card with echo
cancellation module. So I need to use just the echo cancellation by
hardware and disable the echo cancellation by software. I use DAHDI
for my telephony hardware.

If the lines involved with echo cancel are:

In /etc/dahdi/system.conf:


In /etc/asterisk/chan_dahdi.conf:


What lines do I have to comment or to change the value if I want to
disable echo cancellation by software ???

Thanks a lot,


CLI – displaying all channel variables

On 11-05-11 12:29 PM, Steve Edwards wrote:
> On Wed, 11 May 2011, Eric Wieling wrote:
>> Generally you should insert a Noop in the dialplan to examine variables.
>> Noop(EXTEN is ${EXTEN}) for example.
> The ‘verbose()’ application would be an example of ‘better practices.’
> It’s function is obvious rather than just a convenient side-effect.
> It has additional functionality in that you can specify the ‘verbosity’ level
> needed.

Agreed. I tend to use NoOp() for an actual No Operation, such as using it on the
first line of an extension:

exten => something_awesome,1,NoOp()
same => n,Verbose(2,Incoming call from ${CALLERID(all)})
same => n,Dial(SIP/someone_awesome)
same => n,Hangup

That way if you want to place things ahead of any line, you can do that without
impunity. Even using Verbose() on the first line can cause problems if you want
to move the Verbose() around and place something before it — now you have to do
some copy/pasting, and extra work that could be avoided :)


Asterisk SIP Trunking with Cisco UC 560

On 05/11/2011 01:30 PM, Darrin Henshaw wrote:

> I’m interested in knowing if anyone out there has successfully
> connected Asterisk to a Cisco UC 560 via SIP trunking? We have a
> client of ours that we put in an Asterisk install, one of their
> sister companies who we don’t control is putting in a Cisco UC 560.
> From my looking I think it can be done, but the vendor is telling
> them it can’t. Thought I’d ask around here and see if anyone has done
> it? Thanks.

I see no reason why it can’t be done in principle; the vendor or VAR is
just being dismissive of open-source, lest the customer suspect they
might be able to do everything the UC560 can do with open-source
components without paying for a UC560.

The only sticking point is whether this endpoint, like Cisco Call
Manager, by default does SDP offer in the 200 OK and answer in the ACK.
This is an SDP offer-answer flow prescribed by RFC 3261 as valid but
not supported by Asterisk, as best as I can tell, even in 1.8.

concurrent call tracking

On 11-05-11 12:57 PM, Skyler wrote:
> I would like to track/store concurrent call usage per user by
> day/week/month and get server totals by day/week/month. Google comes up with
> mostly info regarding concurrent call limits, though my goal is to calculate
> actual concurrent channel usage and add it into reporting. I’m using * 1.6.2
> + mysql – realtime (no gui). Any suggestions / open-source / AGI on where to
> start looking into implementing something like this?

Just use SNMP to get the channel usage. If you don’t want to use SNMP, then just
use something like GROUP(), GROUP_COUNT() and func_odbc to write channel usage
to the database. Something like….

exten => _NXXNXXXXXX,1,NoOp()
same => n,GoSub(subTotalCallCounter,start,1(outgoing))

exten => start,1,NoOp()
same => n,Set(GROUP(totalcalls)=${ARG1})
same => n,Set(ODBC_TOTAL_CALLS(${ARG1})=${GROUP_COUNT(${ARG1}@totalcalls)})
same => n,Return()

exten => 4165551212,1,NoOp()
same => n,GoSub(subTotalCallCounter,start,1(incoming))

exten => _1XX,1,NoOp()
same => n,GoSub(subTotalCallCounter,start,1(internal))


With what options is asterisk compiled in rpm’s


I’m trying to add modules compiled from source into a rpm install of asterisk (from digium) on centos and asterisk complains that its not compiled with same options so it won’t load it

I know I could install the entire thing from source but for other reasons I would like to keep the main things installed from rpm and install whatever else I need from source (or roll my own rpm for those)


no audio with SIP:INFO in meetme

Hello List,

Asterisk is blocking audio if ‘F’ flag is enabled in meetme with DTMF mode enabled as INFO for SIP channel.
If it is a bug in asterisk or something need to be enabled in sip.conf for the same.

Dialplan looks like
Exten => 100,1,MeetMe(100,dmF)



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