* You are viewing Posts Tagged ‘support’

Asterisk 1.8.13 PlayTones App

i’m trying to implement a Playtones App into my IVR. If a invalid numer is entered, Asterisk should play the info tone defined at the indications.conf.

My Extensions.conf

exten => i,1,Set(CHANNEL(language)=de)
exten => i,2,Progress()
exten => i,3,PlayTones(info)
exten => i,n,Playback(fettefinger)
exten => i,n,Wait(2)
exten => i,n,StopPlayTones()
exten => i,n,Goto(i,3)

The Console shows

Invalid extension ‘5’ in context ‘support’ on SIP/200-0000000a
== CDR updated on SIP/200-0000000a
— Executing [i@support:1] Set(“SIP/200-0000000a”,
“CHANNEL(language)=de”) in new stack
— Executing [i@support:2] PlayTones(“SIP/200-0000000a”, “info”) in new stack
— Executing [i@support:3] Playback(“SIP/200-0000000a”,
“fettefinger”) in new stack
— <SIP/200-0000000a> Playing ‘fettefinger.alaw’ (language ‘de’)
— Executing [i@support:4] Wait(“SIP/200-0000000a”, “2”) in new stack

But the only thing i can hear is my Soundfile fettefinger.

BR Jakob

Does the Eicon/Dialogic 4BRI-8M 800-665 support the NT modus?

On 27.06.2012 18:46, Michelle Konzack wrote:
> Does the Eicon/Dialogic 4BRI-8M 800-665 support the NT modus?

Which PCI-ID is that?


Does Asterisk support AMR and AMR-WB

Hi all, I have a project for the 3G related, AMR and AMR-WB support.

I’m using the client develop suite from the PortSIP(http://www.portsip.com),
as their said
support the AMR, AMR-WB with RFC4867.

Now I have to setup a SIP server/SIP PBX in our Lab for test, does the
support these codecs and RFC4867 ? If no, there has any plugin to support
this ?

Also, any other Server/PBX which support AMR, AMR-WB recommended are

Best regards,

SCCP Questions

Hi List,

Has anyone been running SCCP with a larger number of phones? Im looking to
deploy like 75+ phones and I want to keep SCCP so I don’t have to upgrade
them and for the SLA, some phones also have no SIP software for them so im
forced to keep SCCP. Does anyone have any experience with this? From what
ive read the SCCP support works and works well, im just worried about
trying to run this many phones and if im missing any sort of issues that
could come up.


app_swift beta release

Hi folks,

Just a note to let everyone know I’ve finally finished up the new BETA release of app_swift (now v3.0.1 b1).

This release introduces some pretty major changes to app_swift such as:

– The entire code-base has now been unified and the build system auto detects which Asterisk version you’re using (yay! one branch!)

– Auto-detection and support for both the Cepstral 5.0 and 6.0 engines

– Support for Asterisk 1.4, 1.6, 1.8, 1.8 Certified, and 10

– Asterisk 1.2 support has been dropped.

I have only been able to do some basic testing with all these permutations of Asterisk and the Cepstral engines on a few of my machines here at the house and need some volunteers to help out and be guinea-pigs.

Please email me directly with any feedback you might have.

I’ve updated my github repo with the new app_swift code which can be downloaded using git.

git clone git://github.com/dmsessions/app_swift.git


– D

T.38 debug logs

On 05/24/2012 09:54 AM, Arstan wrote:
> Dear list,
> I have a project where I have:
> Asterisk 10 < -->AudioCodes < --> E1< --> Provider
> AudioCodes supports T.38 and passes the faxes through E1 to the
> provider. From what I read, Asterisk 10 has the most stable(full) T.38
> among other releases.

Asterisk 10 has T.38 gateway support, but you won’t be using it here
because your AudioCodes device will be performing that function. Outside
of gateway support, the T.38 functionality in Asterisk 1.8 and Asterisk
10 are very close to identical.

> My Question: Can I somehow see in the logs if T.38 packets sending and
> see somehow its debugs? Or I should just be better off with capturing
> sip data through tcpdump?

This will depend on what you are asking the Asterisk 10 system to *do*
with T.38. Are you sending FAXes from it, or receiving FAXes into it, or
something else entirely?