Tjyylli

Increase performance creating Mandelbrot set in python

What does this 7 mean above the f flat

How was Earth single-handedly capable of creating 3 of the 4 gods of chaos?

If you attempt to grapple an opponent that you are hidden from, do they roll at disadvantage?

Should my PhD thesis be submitted under my legal name?

Bash method for viewing beginning and end of file

Trouble understanding overseas colleagues

Is a roofing delivery truck likely to crack my driveway slab?

Failed to fetch jessie backports repository

Why Were Madagascar and New Zealand Discovered So Late?

How do I rename a LINUX host without needing to reboot for the rename to take effect?

Can criminal fraud exist without damages?

Time travel short story where a man arrives in the late 19th century in a time machine and then sends the machine back into the past

apt-get update is failing in debian

Is there a problem with hiding "forgot password" until it's needed?

Valid Badminton Score?

Is it correct to write "is not focus on"?

when is out of tune ok?

Was the picture area of a CRT a parallelogram (instead of a true rectangle)?

Where in the Bible does the greeting ("Dominus Vobiscum") used at Mass come from?

Can a monster with multiattack use this ability if they are missing a limb?

Is expanding the research of a group into machine learning as a PhD student risky?

Is HostGator storing my password in plaintext?

The plural of 'stomach"

Asterisk audio isn't transmitted properly, rtp packets sent to public IP

0

I've set up an asterisk server in a VM and set the networking mode to Bridged. I'm trying to make a call using sipML5 demo from another computer in my LAN. It's receiving the SIP packets as desired but the RTP packets are sent to the public IP instead. I've set nat=no but the problem still persists.

Here's my sip.conf -

[general]

udpenable=yes

tcpenable=yes

preferred_codec_only=yes

disallow=all

allow=alaw

allow=ulaw

sipdebug=yes

canreinvite=no

invite=no

encryption=yes

avpf=yes

context=public



[sipML5]

type=friend

context=outgoing

;host=dynamic

dtmfmode=auto

disallow=all

allow=ulaw,alaw

directmedia=yes

nat=no

host=dynamic

canreinvite=no

invite=no

encryption=yes

avpf=yes

transport=wss

dtlsenable=yes

dtlscertfile=/etc/asterisk/keys/asterisk.crt

dtlsprivatekey=/etc/asterisk/keys/asterisk.key

And after setting rtp debug to on in the cli , I see this -

       > 0x7fbed8006b40 -- Strict RTP learning after remote address set to: x.w.y.z:33726

    -- Executing [100@outgoing:1] Answer("SIP/sipML5-0000001f", "") in new stack

    -- Executing [100@outgoing:2] Wait("SIP/sipML5-0000001f", "1") in new stack

    -- Executing [100@outgoing:3] Playback("SIP/sipML5-0000001f", "please-try-again") in new stack

Sent RTP packet to      x.w.y.z:33726 (type 00, seq 014611, ts 000160, len 000160)

    -- <SIP/sipML5-0000001f> Playing 'please-try-again.gsm' (language 'en')

Sent RTP packet to      x.w.y.z:33726 (type 00, seq 014612, ts 000320, len 000160)

Sent RTP packet to      x.w.y.z:33726 (type 00, seq 014613, ts 000480, len 000160)

Where x.w.y.z is my public IP. This just sends the packets and I don't receive any audio.
Appreciate any kind of suggestions.

Thanks.

asterisk sip freepbx

share

asked 1 min ago

Shaksham JaiswalShaksham Jaiswal

1

New contributor

Shaksham Jaiswal is a new contributor to this site. Take care in asking for clarification, commenting, and answering.
Check out our Code of Conduct.

add a comment | 

0

I've set up an asterisk server in a VM and set the networking mode to Bridged. I'm trying to make a call using sipML5 demo from another computer in my LAN. It's receiving the SIP packets as desired but the RTP packets are sent to the public IP instead. I've set nat=no but the problem still persists.

Here's my sip.conf -

[general]

udpenable=yes

tcpenable=yes

preferred_codec_only=yes

disallow=all

allow=alaw

allow=ulaw

sipdebug=yes

canreinvite=no

invite=no

encryption=yes

avpf=yes

context=public



[sipML5]

type=friend

context=outgoing

;host=dynamic

dtmfmode=auto

disallow=all

allow=ulaw,alaw

directmedia=yes

nat=no

host=dynamic

canreinvite=no

invite=no

encryption=yes

avpf=yes

transport=wss

dtlsenable=yes

dtlscertfile=/etc/asterisk/keys/asterisk.crt

dtlsprivatekey=/etc/asterisk/keys/asterisk.key

And after setting rtp debug to on in the cli , I see this -

       > 0x7fbed8006b40 -- Strict RTP learning after remote address set to: x.w.y.z:33726

    -- Executing [100@outgoing:1] Answer("SIP/sipML5-0000001f", "") in new stack

    -- Executing [100@outgoing:2] Wait("SIP/sipML5-0000001f", "1") in new stack

    -- Executing [100@outgoing:3] Playback("SIP/sipML5-0000001f", "please-try-again") in new stack

Sent RTP packet to      x.w.y.z:33726 (type 00, seq 014611, ts 000160, len 000160)

    -- <SIP/sipML5-0000001f> Playing 'please-try-again.gsm' (language 'en')

Sent RTP packet to      x.w.y.z:33726 (type 00, seq 014612, ts 000320, len 000160)

Sent RTP packet to      x.w.y.z:33726 (type 00, seq 014613, ts 000480, len 000160)

Where x.w.y.z is my public IP. This just sends the packets and I don't receive any audio.
Appreciate any kind of suggestions.

Thanks.

asterisk sip freepbx

share

asked 1 min ago

Shaksham JaiswalShaksham Jaiswal

1

New contributor

Shaksham Jaiswal is a new contributor to this site. Take care in asking for clarification, commenting, and answering.
Check out our Code of Conduct.

add a comment | 

I've set up an asterisk server in a VM and set the networking mode to Bridged. I'm trying to make a call using sipML5 demo from another computer in my LAN. It's receiving the SIP packets as desired but the RTP packets are sent to the public IP instead. I've set nat=no but the problem still persists.

Here's my sip.conf -

[general]

udpenable=yes

tcpenable=yes

preferred_codec_only=yes

disallow=all

allow=alaw

allow=ulaw

sipdebug=yes

canreinvite=no

invite=no

encryption=yes

avpf=yes

context=public



[sipML5]

type=friend

context=outgoing

;host=dynamic

dtmfmode=auto

disallow=all

allow=ulaw,alaw

directmedia=yes

nat=no

host=dynamic

canreinvite=no

invite=no

encryption=yes

avpf=yes

transport=wss

dtlsenable=yes

dtlscertfile=/etc/asterisk/keys/asterisk.crt

dtlsprivatekey=/etc/asterisk/keys/asterisk.key

And after setting rtp debug to on in the cli , I see this -

       > 0x7fbed8006b40 -- Strict RTP learning after remote address set to: x.w.y.z:33726

    -- Executing [100@outgoing:1] Answer("SIP/sipML5-0000001f", "") in new stack

    -- Executing [100@outgoing:2] Wait("SIP/sipML5-0000001f", "1") in new stack

    -- Executing [100@outgoing:3] Playback("SIP/sipML5-0000001f", "please-try-again") in new stack

Sent RTP packet to      x.w.y.z:33726 (type 00, seq 014611, ts 000160, len 000160)

    -- <SIP/sipML5-0000001f> Playing 'please-try-again.gsm' (language 'en')

Sent RTP packet to      x.w.y.z:33726 (type 00, seq 014612, ts 000320, len 000160)

Sent RTP packet to      x.w.y.z:33726 (type 00, seq 014613, ts 000480, len 000160)

Where x.w.y.z is my public IP. This just sends the packets and I don't receive any audio.
Appreciate any kind of suggestions.

Thanks.

asterisk sip freepbx

share

asked 1 min ago

Shaksham JaiswalShaksham Jaiswal

1

New contributor

Shaksham Jaiswal is a new contributor to this site. Take care in asking for clarification, commenting, and answering.
Check out our Code of Conduct.

[general]

udpenable=yes

tcpenable=yes

preferred_codec_only=yes

disallow=all

allow=alaw

allow=ulaw

sipdebug=yes

canreinvite=no

invite=no

encryption=yes

avpf=yes

context=public



[sipML5]

type=friend

context=outgoing

;host=dynamic

dtmfmode=auto

disallow=all

allow=ulaw,alaw

directmedia=yes

nat=no

host=dynamic

canreinvite=no

invite=no

encryption=yes

avpf=yes

transport=wss

dtlsenable=yes

dtlscertfile=/etc/asterisk/keys/asterisk.crt

dtlsprivatekey=/etc/asterisk/keys/asterisk.key

       > 0x7fbed8006b40 -- Strict RTP learning after remote address set to: x.w.y.z:33726

    -- Executing [100@outgoing:1] Answer("SIP/sipML5-0000001f", "") in new stack

    -- Executing [100@outgoing:2] Wait("SIP/sipML5-0000001f", "1") in new stack

    -- Executing [100@outgoing:3] Playback("SIP/sipML5-0000001f", "please-try-again") in new stack

Sent RTP packet to      x.w.y.z:33726 (type 00, seq 014611, ts 000160, len 000160)

    -- <SIP/sipML5-0000001f> Playing 'please-try-again.gsm' (language 'en')

Sent RTP packet to      x.w.y.z:33726 (type 00, seq 014612, ts 000320, len 000160)

Sent RTP packet to      x.w.y.z:33726 (type 00, seq 014613, ts 000480, len 000160)

share

asked 1 min ago

Shaksham JaiswalShaksham Jaiswal

1

New contributor

Shaksham Jaiswal is a new contributor to this site. Take care in asking for clarification, commenting, and answering.
Check out our Code of Conduct.

share

asked 1 min ago

Shaksham JaiswalShaksham Jaiswal

1

New contributor

Shaksham Jaiswal is a new contributor to this site. Take care in asking for clarification, commenting, and answering.
Check out our Code of Conduct.

asked 1 min ago

Shaksham JaiswalShaksham Jaiswal

1

New contributor

Shaksham Jaiswal is a new contributor to this site. Take care in asking for clarification, commenting, and answering.
Check out our Code of Conduct.

asked 1 min ago

Shaksham JaiswalShaksham Jaiswal

1