* You are viewing the archive for May 23rd, 2012

Detecting Fax Tones over IAX2

Hello All,
I use IAX2 as the incoming connection from my DID provider. For whatever
reason, this works best for me, SIP connections lag very frequently and
only have about a 50% success rate for incoming calls (they get dropped

I’m trying to implement a fax/voice switch. I have faxdetect=both in my
sip.conf, and when I use sip, it works well. However, from what I can
tell, there’s no such option for IAX2 connections.

Any ideas on what I can do here, or am I out of luck?


Unable to execute ‘dahdi_scan > /etc/asterisk/dahdi_scan.conf’

On Wed, May 23, 2012 at 02:02:51PM +0500, p070075 Muhammad Atif Ramzan wrote:
> Hi
> Can anyone help me with this error
> Unable to execute ‘dahdi_scan > /etc/asterisk/dahdi_scan.conf’
> i am using asterisk-gui 2.0 , asterisk 1.8, dahdi 2.6 and also my call
> reached the destination but no voice is coming from destination my voice
> reflects back

Have you verified the user asterisk is running as can execute
dahdi_scan? This was asked not too long ago on the forums as well:


twenty thousands (20, 000) users, which asterisk and how many servers

the solution lies in kamailio/opensips’s despatcher module.

Sent from my iPhone

On 23 maj 2012, at 20:46, bilal ghayyad wrote:

> Dear;
> So it is a hardware issue and not software?
> I am afraid that asterisk software it self is not able to support 20 000 users and 2000 concurrent calls.
> About the high availability: is there a method that if the first asterisk server down, then the call will stay connected and failover to second asterisk server?
> Regards
> Bilal
> ————–
>> 20.000 users is really a big number, as big as 2000
>> concurrent calls.
>> As previously stated on this list, it depends… it depends
>> by the type of
>> calls for example. If all media is offloaded from the server
>> letting the
>> phones to reinvite each other, than your server CAN support
>> the call
>> volume. If instead even a tiny portion of the call volume
>> uses service on
>> the pbx, like IVR, music on hold, conferences, queues or
>> even worst,
>> transcoding, then the server is obviously underpowered. From
>> my point of
>> view, servicing 20.000 users with a single piece of hardware
>> is highly
>> risky. It can broke in the middle of the day, leaving all
>> your users
>> without service. I think a better approach will be to have
>> more less
>> powered servers working all together to serving your users.
>> If a day one or
>> two of them broke, you have not to worry because the other
>> will continue to
>> serve your users and nobody notice the little decrease in
>> power.
>> There are a lots of way to achieve the high availability,
>> load sharing,
>> each with its pros and cons.
>> Right now I am building a pbx with high availability and
>> load sharing in
>> mind, for a client who wants to achieve numbers you have
>> just said. Let’s
>> see how it works in few months.
>> Leandro
>> 2012/5/23 bilal ghayyad
>>> Hi All;
>>> I need to use Asterisk for 20 000 users, so which
>> asterisk version to be
>>> used? Is there asterisk version that supports 20,000
>> users on one hardware
>>> machine?
>>> Can I use one strong hardware server i7 with 64 GB RAM
>> and fast hard desk
>>> to handle 20 000 users, and concurrent calls 2000? Or I
>> need multiple
>>> servers, how much?
>>> If I am going to use multiple servers (until now I do
>> not know how much,
>>> and I do not know if the barrier will be the asterisk
>> software or the
>>> hardware), then do I have to use special SIP proxy or I
>> have to use load
>>> balancer)? In this case, I have to use asterisk
>> Database (so all the
>>> servers will read/write from the database)?
>>> What about AsteriskNow, can it support?
>>> Regards
>>> Bilal
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SIP endpoints CANCEL when PRI receives Cause Code 31

We have an Asterisk server which connects to another Asterisk server
acting as a PSTN gateway. This gateway machine has Digium TE210P card
connected to a pair of PRIs.

For the most part, all is working well, however there are some specific
telephone numbers that my users have attempted to call, but we unable to.

I set debugging on and determined that when the the gateway machine
dials one of the numbers in question, we receive from the PSTN an ISDN
cause code 31, which in my understanding is not an error. This is then
passed back to the originating Asterisk server via IAX as progress. It
is then sent to the originating endpoint as a sip message 183 ‘Session
Progress’. 2 seconds after this 183 progress message is sent, the
endpoint sends a SIP CANCEL message and the channel is torn down.

I have the prematuremedia=yes and progressinband=never in the sip.conf
file which looks like it could be a solution, however I believe that
because we are getting ISDN Call Proceeding and a corresponding SIP 100
Trying message that this setting has no effect.

I have tried from several different endpoint types with the same
results. I have verified that the numbers in question are in fact

Any suggestions?

Asterisk version is 1.8.7 on both hosts
Dahdi version 2.5.0
libpri version 1.4.12


Disable All Asterisk Features (blind xfer, disconnect, etc)

Hi Guys,

is there any way to disable all Asterisk Features? We are having false dtmf
detections and randon calls being put on-hold and suspect that dtmf
features is the cause.

Changing features.conf aparently keeps the default options. Since we dont
use it, is there any way to disable it?



No caller id when using cadence with DAHDI

Hello everyone,

Just thought to let you know of a weird issue in Asterisk 1.8.? + Dahdi
2.6.? (and 2.5.?).

When you specify any cadence in an app (Dial, Queue) then caller id does
not work.

For instance with the default cadences (everything commented out in
chan_dahdi.conf) :

Dial(DAHDI/54) caller id works

Dial(DAHDI/54r1) caller id does not work (even for r1)

I just found this issue did not have time to investigate further. Can
anyone else verify that this is true for tonezones other than 13 (gr) which
I am using?