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:

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.

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?

  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.

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?

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.”

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.

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