On Thursday 16 Jul 2015, Thyda ENG wrote:
I would like to see how can we config the asterisk to enable calling to
multiple SIP number at the same time?
If you want to have a number that will call several phones when dialled, you
can do it in the Dial() command. The following example refers to
Dear Sir,
I would like to see how can we config the asterisk to enable calling to
multiple SIP number at the same time?
Thank,
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Hi list!
I'm trying to configure Asterisk to record incoming calls, if the called
press *3.
I added in features.conf:
automixmon = *3
then, in my dialplan:
exten = 1,n,Dial(SIP/004935,20,RcxX)
Well, if I **CALL** a number I'm able to record the call, but if I'll be
called, and press
On 15 July 2015 at 20:51, Ethy H. Brito ethy.br...@inexo.com.br wrote:
Hi all
Any of you guys could point me in the right direction?
I need to make that a blind transfer to return to the transferrer when the
transferee does not answer.
Scenario:
. Miss Jane Doe, our front desk
On Wednesday 15 Jul 2015, Luca Bertoncello wrote:
But it seems, that I found the problem, adding:
disallow=all
allow=g729
to the configuration of the peer for this number...
You need the following;
disallow=all
allow=alaw
in the configuration for *every* device. There is literally no
Hi Pete.
No problem!
Maybe I will use only OpenSIPS, because it may be enough for me. But I still
have to investigate some points.
As I was learning the past few days, due to the fact that Asterisk is not a SIP
Proxy, it might cause some more difficult in my project.
Best regards.
Dear Asterisk-Users,
By means of Asterisk 11 and sip.conf, I got success implementing early media.
That is, all information that come from callee (SIP 183 message/ SDP) is passed
to the caller without any modification in the SDP body.
However, in Asterisk 13 and using pjsip.conf I'm still
Rodrigo Pimenta Carvalho wrote:
Dear Asterisk-Users,
By means of Asterisk 11 and sip.conf, I got success implementing early
media. That is, all information that come from callee (SIP 183 message/
SDP) is passed to the caller without any modification in the SDP body.
PJSIP does not support
I'm trying to configure my Asterisk machine to work with Vitelity's
vMobile service. I can place calls to the vMobile device and it rings
as expected. However, I have no audio in either direction. There's
no NAT involved though. My asterisk machine has a public IP address
with port 5060 and
Thank you Joshua!
In this case I finally decide to use SIP Proxy. I have to start testing the SIP
Proxy today.
Best Regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979 (Brasil)
De:
On Thu, 16 Jul 2015 09:51:54 +0100
Ishfaq Malik i...@pack-net.co.uk wrote:
On 15 July 2015 at 20:51, Ethy H. Brito ethy.br...@inexo.com.br wrote:
Hi all
Any of you guys could point me in the right direction?
I need to make that a blind transfer to return to the transferrer when the
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