Hello folks,
have anyone have experience with Asterisk IP PBX and Powerbuilder.
I am thinking of receiving and sending voice events from Asterisk to a Power Builder application.
Thank you in advance.
I would prefer two Asterisk for failover capabilities and in order to be able to scale the system in the future.
The Idea of Kamailio is to centralize the location of SIP extensions.
I am stuck because I dont know how to use the same QUEUE / ACD in two different Asterisk.
Thank you again.
Can you post this information:
------------------------
LD 20
REQ: prt
TYPE: TIE
TN Loop 1
DATE
PAGE
-------------------------
LD 21
REQ: prt
TYPE: rdb
CUST 0
ROUT RDB
------------------------
Regards
Thank you for your help, this is the first time I hear this.
Questions:
1 - Why IAX? Why not SIP? What would be the advantage of IAX?
2 - Where do you define the queue? I am thinking in queues that are set up in queues.conf. Are there another kind of queues? I am asking this because you have...
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New Post tikimon7 (TechnicalUser)
4 Oct 13 14:01
see if you have option for: dtmfmode=rfc2833
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I have already tried that but it didnt work.
Thank you for your time
Hello,
I am setting up a trunk between an Avaya s8720 and an Asterisk.
I was able to setup the trunk and send calls but the problem is that DTMF tones are not passing through from Asterisk to Avaya
If I make a call from Avaya to Asterisk everything works like a charm! voice and DTMF tones go...
Hello guys,
I would like to know if someone know the way to use one single queue in multiples asterisk.
The idea is to split the traffic using a Kamailio / openser but the problem is that as far as I knwo on Asterisk queues are setup per server.
Is it possible using asterisk or other software...
Hello,
I have a Nortel CS1000 and im seeing an issue with one ISDN_PRI vendor. Instead of detecting busy tones as CAUSE: #17 - USER BUSY, the PBX is detecting busy tones as CAUSE: #31 - NORMAL UNSPECIFIED.
This is only happening with one vendor.
The rest of PRI circuits work fine (detect the...
Hello,
I am using monitor application in Asterisk. Im seeing that created audio file has this configuration
- 16 bit
- 8000 hz
- 128 kbps
I need to change bit codification from 16 bit to 8 bit.
I tried SOX sox IN-FILE.wav -b 8 OUT-FILE.wav but after the transformation there is a background...
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