Not Working on asterisk-16.16.2

[masteringvoip]

Hi Sir,

It not working with asterisk 16

Here is the error:

== Using SIP RTP CoS mark 5

0x7f35dc008ad0 -- Strict RTP learning after remote address set to: 202.52.48.210:4002 -- Executing [7000@main-out:1] Answer("SIP/9999-00000000", "") in new stack -- Executing [7000@main-out:2] Verbose("SIP/9999-00000000", "starting audio fork") in new stack starting audio fork -- Executing [7000@main-out:3] AudioFork("SIP/9999-00000000", "ws://localhost:8080/") in new stack == setting wsserver to ws://localhost:8080/ == setting direction to 2 0x7f35dc008ad0 -- Strict RTP learning after remote address set to: 202.52.48.210:4002 asterisk-tts*CLI> Disconnected from Asterisk server Asterisk cleanly ending (0). Executing last minute cleanups Asterisk ending (0). [root@asterisk-tts asterisk-16.16.2]# dmesg [92488.239632] asterisk[14000]: segfault at 88 ip 00007f36b14e7a7d sp 00007f36b73d42e0 error 4 in app_audiofork.so[7f36b14e3000+9000] [92598.178537] asterisk[14178]: segfault at 88 ip 00007fb7910e5a7d sp 00007fb791b462e0 error 4 in app_audiofork.so[7fb7910e1000+9000] [92873.266843] asterisk[17954]: segfault at 88 ip 00007f35e2cfda7d sp 00007f35763882e0 error 4 in app_audiofork.so[7f35e2cf9000+9000]

Thanks

[MattRiddell]

The only difference I can see from a machine that I had this working on was this:

diff --git a/apps/app_audiofork.c b/apps/app_audiofork.c
index 1b3a0174f..12e5b2f81 100644
--- a/apps/app_audiofork.c
+++ b/apps/app_audiofork.c
@@ -598,7 +598,7 @@ static int setup_audiofork_ds(struct audiofork *audiofork,

   ast_verb(2, "Connecting websocket server at %s\n",
-           audiofork_ds->wsserver);
+           audiofork->audiofork_ds->wsserver);

   //check if we're running with TLS
   if (audiofork->has_tls == 1) {

It definitely worked for me before I switched over to using the external media channel via ARI:

globalARI.channels.externalMedia({
    channelId: "EM-"+incoming.id,
    app: "externalMedia",
    external_host: ip+":"+rtpPort,
    format: "slin16",
    encapsulation: "rtp",
    transport: "udp",
    variables: {channel: incoming.id}
})

[masteringvoip]

Hi Sir @MattRiddell ,

Can you share, the specific type of asterisk that you use so that this script https://github.com/nadirhamid/asterisk-audiofork/blob/master/app_audiofork.c worked?

Thanks

[MattRiddell]

It's out of a git fork but just config changes.

Version is 16.10.x

[MattRiddell]

If it keeps crashing do a backtrace on it and see where:

Getting a backtrace

[masteringvoip]

@MattRiddell it working on asterisk 16.10

Thanks Sir

[MattRiddell]

No problems - post back if you have issues and I'll help where I can

[masteringvoip]

Hi Sir @MattRiddell ,

On previous message you said "I switched over to using the external media channel via ARI", can you share it on this github? so I can try it

i'm looking programs for realtime TTS or STT conneted to Google Voice.

Thanks

[MattRiddell]

Hah ok,

So @DanJenkins shared a repository with me that I used:

RTP Audio Server

I'm using it for a different purpose than what I used this repository for.

This repository I used to send off a stream for transcribing.

Dan's repository I used for just dealing directly with the audio so I could feed it into my AI pipeline.

Bear in mind, with Dan's repository you'd be responsible for parsing RTP etc.

Anyway, hope that helps 😊

[MattRiddell]

Dan's repository is probably perfect for you - he connects to Microsoft's dialog voice or whatever it's called.

I still use this repository to split off a stream for transcribing though.

Each has its purpose 😊

[masteringvoip]

Hi Sir @MattRiddell @danjenkins

Thank you very much for the help, you are so kind to reply to my question, I am so happy to be able to try it. Good luck to me.

Thanks

[jayasingh1802]

Hi @MattRiddell @masteringvoip

I tried this with Asterisk 16.6.2 & Asterisk 16.10.0 but no luck.

Here is the error: -- Executing [s@STREAM:1] NoOp("SIP/voda-00000000", ""starting audio fork------"") in new stack -- Executing [s@STREAM:2] Answer("SIP/voda-00000000", "") in new stack

0x7fdc1c00a670 -- Strict RTP switching to RTP target address 10.229.6.148:17800 as source vl073096-app52-pd-obd-vcz-cts-blr-in-vf*CLI> Disconnected from Asterisk server Asterisk cleanly ending (0). Executing last minute cleanups

[MattRiddell]

You'll need to get a back trace.

Instructions here