PJSUA on ASTERISK

Asterisk Asterisk Endpoints
1

Hi Team,

I’m trying to play/record wave file in media state using PJSUA client but getting the below error.

tsx0x7f8bde0606a8  .......Temporary failure in sending Request msg INVITE/cseq=24675 (tdta0x7f8bdd8e0aa8), will try next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT)

Is there any configuration that I’ve to do in the asterisk server?

Below is snippet of the code I’m trying to do

lib.create_player("sample.wav", loop='true')

Thanks in advance :blush:

2

Your problem is with pjsua. There’s no transport to be able to send the SIP INVITE out. Asterisk itself doesn’t use pjsua, so I have no knowledge of what is involved in it.

3

Thanks for your response.

I’m new to Asterisk. Could you please help me with the below queries.

• Can I(my application) make a SIP call using Asterisk to any SIP client?
• If yes, how to manipulate audio so that I can run my customized voice assistant?

4
  1. Yes, Asterisk speaks SIP so it can speak to other SIP devices and implementations.
  2. I don’t understand the question. You’d need to be specific about where such a thing would occur.

I don’t think your problem is really Asterisk, but pjsua, which isn’t really supported here.

5

Once the call is connected, asterisk opens channel to communicate between the devices. In this channel, I wanted to play/record audio real time over the call. Is it possible using asterisk?

6

Asterisk will forward the audio between both sides. You need to be more specific about what you are trying to achieve. For example:

“I want to write a SIP client to call Asterisk and then in Asterisk I want to be able to take the audio in an application and direct it elsewhere.”

7

I think the OP wants to write an EAGI application. I think when he says SIP client he means SIP server (UAS); Asterisk is the UAC.

8

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