No sound with sip g729

Asterisk Asterisk Support
1

Hi,

I am making outbound calls with sip using a voip provider. When I originate the call the phone rings but I can’t hear any sound.

I have one license of the codec (g729) and is loading ok.

My voip provider told me it might be the rtp rfc, they use rfc2327. How can i see the rfc i’m using for rtp?

Any other suggestions to solve this problem?

Thx!
Laura

2

Hi

When on the call do “show g729” and also “sip show channels” this will show what licences are in use.

Ian

3

I’ve done some more testing, here is what i got:

  1. CLI> show g729

0/0 encoders/decoders of 1 licensed channels are currently in use

  1. Playback(room-service) (the file is room-service.g729)
    • I just hear a second of silence and then hangs up
    • During the call: CLI> show g729
      0/0 encoders/decoders of 1 licensed channels are currently in use

Any ideas?

Thx

4

But looking at this, I can see that it recognizes the g729 codec:

[Feb 12 08:53:55] — (13 headers 9 lines) —
[Feb 12 08:53:55] Found RTP audio format 18
[Feb 12 08:53:55] Found RTP audio format 101
[Feb 12 08:53:55] Peer audio RTP is at port 196.38.238.94:20326
[Feb 12 08:53:55] Found audio description format g729 for ID 18
[Feb 12 08:53:55] Found audio description format telephone-event for ID 101
[Feb 12 08:53:55] Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
[Feb 12 08:53:55] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Feb 12 08:53:55] Peer audio RTP is at port 196.38.238.94:20326

5

Hi

Ok

You say that you cant hear anything when making outbound calls. The test results you post are from dialing an internal call to a file.

Please test when on a outbound call with the issue. also do a sip show channels

Ian

6

Hi Ian,

Thanks for your help!

I’m originating calls using a .call file, but they are outbound calls, through a voip provider and then to a cellphone.

I was testing with g729 files from the asterisk sounds folder, and there was not playback at all. Now I made my own g729 files using file convert, and I hear silence but of the duration of the sound file.

Hope I’m getting closer to the solution… here are the results for show g729 and sip show channels.

[Feb 12 13:07:56] – Attempting call on SIP/is/0766107977 for s@outbound:1 (Retry 1)
[Feb 12 13:08:08] > Channel SIP/is-09761d58 was answered.
[Feb 12 13:08:08] – Executing [s@outbound:1] Answer(“SIP/is-09761d58”, “”) in new stack
[Feb 12 13:08:08] – Executing [s@outbound:2] Wait(“SIP/is-09761d58”, “2”) in new stack
bushidoCLI> show g729
0/0 encoders/decoders of 1 licensed channels are currently in use
[Feb 12 13:08:10] – Executing [s@outbound:3] Playback(“SIP/is-09761d58”, “did/welcome”) in new stack
[Feb 12 13:08:10] – <SIP/is-09761d58> Playing ‘did/welcome’ (language ‘en’)
bushidoCLI> show g729
0/0 encoders/decoders of 1 licensed channels are currently in use
bushidoCLI> show g729
0/0 encoders/decoders of 1 licensed channels are currently in use
bushidoCLI> show g729
0/0 encoders/decoders of 1 licensed channels are currently in use
bushidoCLI> show g729
0/0 encoders/decoders of 1 licensed channels are currently in use
bushidoCLI> show g729
0/0 encoders/decoders of 1 licensed channels are currently in use
bushidoCLI> show g729
0/0 encoders/decoders of 1 licensed channels are currently in use
bushidoCLI> show g729
0/0 encoders/decoders of 1 licensed channels are currently in use
bushidoCLI> show g729
0/0 encoders/decoders of 1 licensed channels are currently in use
bushidoCLI> show g729
0/0 encoders/decoders of 1 licensed channels are currently in use
[Feb 12 13:08:23] – Executing [s@outbound:4] Hangup(“SIP/is-09761d58”, “”) in new stack
[Feb 12 13:08:23] == Spawn extension (outbound, s, 4) exited non-zero on ‘SIP/is-09761d58’
[Feb 12 13:08:23] NOTICE[7368]: pbx_spool.c:351 attempt_thread: Call completed to SIP/is/0766107977
[Feb 12 13:09:25] – Attempting call on SIP/is/0766107977 for s@outbound:1 (Retry 1)
[Feb 12 13:09:39] > Channel SIP/is-09761d58 was answered.
[Feb 12 13:09:39] – Executing [s@outbound:1] Answer(“SIP/is-09761d58”, “”) in new stack
[Feb 12 13:09:39] – Executing [s@outbound:2] Wait(“SIP/is-09761d58”, “2”) in new stack
[Feb 12 13:09:41] – Executing [s@outbound:3] Playback(“SIP/is-09761d58”, “did/welcome”) in new stack
[Feb 12 13:09:41] – <SIP/is-09761d58> Playing ‘did/welcome’ (language ‘en’)
bushidoCLI> sip show channels
Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message
196.38.238.94 0766107977 56c93fa3239 00102/00000 0x100 (g729) No Tx: ACK
1 active SIP channel
bushidoCLI> sip show channels
Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message
196.38.238.94 0766107977 56c93fa3239 00102/00000 0x100 (g729) No Tx: ACK
1 active SIP channel
bushidoCLI> sip show channels
Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message
196.38.238.94 0766107977 56c93fa3239 00102/00000 0x100 (g729) No Tx: ACK
1 active SIP channel
bushidoCLI> sip show channels
Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message
196.38.238.94 0766107977 56c93fa3239 00102/00000 0x100 (g729) No Tx: ACK
1 active SIP channel
[Feb 12 13:09:53] – Executing [s@outbound:4] Hangup(“SIP/is-09761d58”, “”) in new stack
[Feb 12 13:09:53] == Spawn extension (outbound, s, 4) exited non-zero on ‘SIP/is-09761d58’
[Feb 12 13:09:53] NOTICE[7373]: pbx_spool.c:351 attempt_thread: Call completed to SIP/is/0766107977
bushido*CLI>