* You are viewing the archive for May 18th, 2012

BroadVoice Unlimited World PLUS – Dialplan Update (18/May/2012)

We are Broadvoice users some years ago, using Unlimited World PLUS
product that seems reasonably acceptable.

The problem is that we review Broadvoice config updates a lot of times
to make our changes, but they are adding a lot of new countries to the
flat rate (which is what alone is all that you want to use to avoid
overruns), but the instructions installation are not updated on their


It seems that they do in order to use the destinations to which there is
no flat rate billing for the additional traffic minutes.

That is why we have proceeded to update the Dialplan so they can make
use of the new destinations as a flat fee. We will verify the
destinations which are not allowed cell phones so they can improve and
optimize their Asterisk Dialplan.


Hope that you help you


50% of time SendDTMF failed

I suggest you put in dtmfmode=auto (so rfc2833 / inband ) can be selected

Wanted to check with the community if this feature holds true on latest
versions of Asterisk ?

Mitul Limbani,
Chief Architech & Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai – 400 086. India
email: mitul@enterux.in
DID: +91-22-61447605
Cell: +91-9820332422

On Fri, May 18, 2012 at 10:11 PM, Ing CIP. Alejandro Celi > wrote:

> **
> Did you try putting inband parameter in dtmfmode and dtmf of your sip.conf?
> Regards,
> –
> Ing CIP. Alejandro Celi Mariátegui
> http://cipher.pe/web/asterisk.html
> El mié, 16-05-2012 a las 16:07 +0100, Shahid H escribió:
> I am having a problem with SendDTMF() – 50% of time it did not succeed.
> I suspect it is not sending clear DTMF tones to the IVR.
> For example:
> SendDTMF(wwwww3wwwww2wwwwww1wwwww4)
> Sometime digit 3 and 2 work, and failed to do digit 1.
> Sometime digit 3 work and failed to do number 2.
> Sometime all went through fine.
> dtmfmode=rfc2833 are set in the sip.conf file
> How do I debug to see what went wrong and how to fix?
> Asterisk
> Installed on VPS (XEN, CentOS 5.x, 768 MB Ram, 1000 GB B/W – Located in
> UK)
> VOIP Provider in UK.
> Thanks
> –_____________________________________________________________________– Bandwidth and Colocation Provided by http://www.api-digital.com –New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello
> asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-use http://lists.digium.com/mailman/listinfo/asterisk-usersrs
> –
> _____________________________________________________________________
> — Bandwidth and Colocation Provided by http://www.api-digital.com
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users

special digits * # on sip dial string

hi guys.
sorry if this is a silly question.

My recharge application uses * digits if the subscriber wants to send
some aditional information to speed up a process, dialing something
like *777*123*5000
On my old ss7 network works great, but on my new ngn/sip i think it’s
not possible because somewhere the call is rejected.
-On the NGN/Ericsson side engineer say that the call whas deliverd.
-On the asterisk side there is no invite shown on debug.

Can sip one or more * signs in a dial?
What am i doing wrong.

thanks in advance..

Best practices to route calls according holidays


At the moment, I’m mostly using a “Day/Night toggle” button to let
users deal with week-ends, holidays and opening hours.
As Asterisk 1.8 introduces Calendar capabilities, I’m wondering if
better alternatives now exist.

Is it possible, safe, reliable and easy to refer from Asterisk to a
public calendar resource listing holidays, for a given country ?
Should you instead refer to a private resource, to avoid depending on
an externaly managed resource ? If you go this way, which tools would
you recommend to build and update a private calendar ?

Suggestions ?


Transfer CDRs


I’m using attended call transfer in asterisk on a CentOS machine.
Each CDR entry of calls that are transferred is repeated once. Every field
including uniqueid, calldate, billsec, duration, src, dst, channel,
dstchannel is exactly the same.
Besides adding a constraint in the database table, isn’t there any way I
can resolve this call transfer cdr duplication issue in asterisk csv cdrs?

RTP stats explaination

Hi all,
This question is not related to asterisk, but related to voip quality
in general. But i thought there are lot of experienced guys out here
who can help me with this. And our telephony platform is also asterisk
:). May be i can extract some bias over this :)

We are getting very poor quality of voice during testing of a new
filtering application of us.

The application receives packets from kernel using netfilter_queue
library. Then insert the packets into a new user managed queue and
does some transformations on it, like concatenation of udp payload.

The network is healthy. Its inside our lab. And it does not drop
packets or anything .

In our app we do not forward packet immediately. After enough packet
received to increase rtp packetization time (ptime) the we forward the
message over raw socket and set dscp to be 10 so that this time
packets can escape iptable rules.

From client side the RTP stream analysis shows nearly every stream as
problematic. summery for some streams are given below :

Stream 1:

Max delta = 1758.72 ms at packet no. 40506
Max jitter = 231.07 ms. Mean jitter = 9.27 ms.
Max skew = -2066.18 ms.
Total RTP packets = 468   (expected 468)   Lost RTP packets = 0
(0.00%)   Sequence errors = 0
Duration 23.45 s (-22628 ms clock drift, corresponding to 281 Hz (-96.49%)

Stream 2:

Max delta = 1750.96 ms at packet no. 45453
Max jitter = 230.90 ms. Mean jitter = 7.50 ms.
Max skew = -2076.96 ms.
Total RTP packets = 468   (expected 468)   Lost RTP packets = 0
(0.00%)   Sequence errors = 0
Duration 23.46 s (-22715 ms clock drift, corresponding to 253 Hz (-96.84%)

Stream 3:

Max delta = 71.47 ms at packet no. 25009
Max jitter = 6.05 ms. Mean jitter = 2.33 ms.
Max skew = -29.09 ms.
Total RTP packets = 258   (expected 258)   Lost RTP packets = 0
(0.00%)   Sequence errors = 0
Duration 10.28 s (-10181 ms clock drift, corresponding to 76 Hz (-99.05%)

Any idea where should we look for the problem?