app_swift beta release

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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…

Asterisk Users 3.2 years ago 0 Answers

TLS bug in asterisk?

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Hi folks. I've got a problem dialing with my new Snom M9 via TLS on asterisk 1.8.7.1 .
Registration works like a charm - the phone becomes 'AVAILABLE'.
An INVITE is responded by a 401 to be expected, but then asterisk closes the TLS connection upon the Snom's ACK. The authenticated INVITE the Snom sends immediately after the ACK meets a closed socket and merely triggers a TCP RST packet on asterisk's behalf. There's no ERROR or WARNING put out on the asterisk CLI.
The only hint I get is asterisk complaining about not finding the CSeq…

Asterisk Users 3.7 years ago 0 Answers

skype connect & early media

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hi folks. when i use regular PSTN(sip phone -> asterisk -> PRI) to call
certain numbers and when that number is unavailable. i usually
hear an early media message saying "blahblah is unavailable,
please try again". but when i use skype connect(sip phone -> asterisk
-> skype connect). i just hear ring back tone for about 20 seconds
and then become fast busy. is there any setting i'm unaware of
when setting up sip w/ skype connect? any suggestions would be appreciated.

Asterisk Users 3.7 years ago 0 Answers

forwarding early media

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Hi folks, Please discard the e-mail since it seems to be a problem of the E1 provider
and nothign related to asterisk. Apologies for the noise,
Samuel On 17 October 2011 18:32, samuel wrote: > Hi folks,
>
> I'm having an issue with an asterisk 1.4.36 with an E1 card that is not
> forwarfing the early media a remote SIP end-point is creating.
>
> --incoming E1 call-->asterisk 1.4.36---->SIP endpoint (which happens to be
> an asterisk 1.6.20).
>
> I've checked signalling and the remote…

Asterisk Users 3.8 years ago 0 Answers

Digium FFA + Gafachi T38 outgoing issues

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Hi, folks. I'm having a heck of a time trying to get outgoing T38 faxing (I don't
need inbound right now) working with FFA and Gafachi. G711 faxing works
(as well as can be expected over the internet), but I want the higher
reliability of T38. I'm running Asterisk 10-beta1. When I drop my callfile in to make the call, I get this:

Asterisk Users 3.9 years ago 1 Answer

issue with the detection of the call status after sending it using Orginate (DAHDI/1/...., app, ...

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hi folks, i hope that i will get some help about this issue,so my configuration is : X100P card ,with FXO port ,the problem is that when i send a call using
Originate,every thing goes well but what i realy need is to know how can i
detect the status of this channel till the called person hung up or pick up
the phone or may be busy .so how can i know that those dahdi show status and the like those not work for me cause it just
give u information about if the call…

Asterisk Users 4 years ago 0 Answers

asterisk-users Digest, Vol 85, Issue 23

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Numtodial is the variable that will receive the DTMF input - enternum would
be the prompt played before entry (/var/lib/asterisk/sounds/enternum.wav
(gsm, slin, whatever the codec dictates). From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tahar .H
Sent: Saturday, August 13, 2011 7:13 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] asterisk-users Digest, Vol 85, Issue 23 hi folks, can some one please explain to me what this one stands for : Exten => 1234,1,read(numtodial,enternum,10,skip,1,10) that numtodial and enternum !!!!

Asterisk Users 4 years ago 0 Answers

How to use Atxfer in AMI

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Hi folks, I repeat "as is" the title of a post someone did a few months ago,
since I am facing the same problem and did not see one single answer
to his post. Maybe I'll be a little bit more lucky. When I'm trying to issue an Atxfer AMI command, in the asterisk 1.8
branch, what happens is that some DTMF's are sent, like this :
[Mar 22 15:46:27] DTMF[5910]: channel.c:3900 __ast_read: DTMF end '1' received on SIP/bmwgsjrponciuj-0000009f, duration 0 ms
[Mar 22 15:46:27] DTMF[5910]: channel.c:3926 __ast_read: DTMF begin emulation of '1' with duration…

Asterisk Users 4.4 years ago 0 Answers

Handle in dialplan user disconnection

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Hi folks, How handle in dialplan when a user disconnected ? The first try was using the special context h, however all hangups are handled by this context and I only need identify when user's hangup in a Read/Background/Dial application. Can asterisk send to dialplan only when the user hangup? Best Regards,
S idarta Oliveira

Asterisk Users 4.5 years ago 3 Answers