Chan_mobile and music on hold


#1

Hi,

I’ve set up chan_mobile to connect my mobile phone to a SIP Phone via Asterisk.

It works well (incoming call, outgoing).

However, when I put on hold a call which is coming from (or going to) chan_mobile, I can’t hear the music on hold (just no sound). Actually, in the CLI the channel isn’t put on hold :

    -- Executing [s@arrivee-mobile:1] Dial("Mobile/NOKIA208-b281",""SIP/sip10",,tTwW") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/sip10
    -- SIP/sip10-0000001e is ringing
    -- SIP/sip10-0000001e answered Mobile/NOKIA208-b281
    -- Channel SIP/sip10-0000001e joined 'simple_bridge' basic-bridge <9400cbe3-fe9b-4d73-ad3e-fafc134155ab>
    -- Channel Mobile/NOKIA208-b281 joined 'simple_bridge' basic-bridge <9400cbe3-fe9b-4d73-ad3e-fafc134155ab>
       > 0x740048f8 -- Probation passed - setting RTP source address to 192.168.101.30:12118
       > 0x740048f8 -- Probation passed - setting RTP source address to 192.168.101.30:12118
       > 0x740048f8 -- Probation passed - setting RTP source address to 192.168.101.30:12118
    -- Channel SIP/sip10-0000001e left 'simple_bridge' basic-bridge <9400cbe3-fe9b-4d73-ad3e-fafc134155ab>
    -- Channel Mobile/NOKIA208-b281 left 'simple_bridge' basic-bridge <9400cbe3-fe9b-4d73-ad3e-fafc134155ab>
  == Spawn extension (arrivee-mobile, s, 1) exited non-zero on 'Mobile/NOKIA208-b281'

The first “0x740048f8 – Probation passed” is when I answer the call, the second when I put the call on hold from the sip phone, and the last when I resume the call.

With a SIP to SIP call the music on hold works :

  == Using SIP RTP CoS mark 5
    -- Executing [10@arrivee-sip:1] Dial("SIP/sip11-00000021", "SIP/sip10,,tTwW") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/sip10
    -- SIP/sip10-00000022 is ringing
    -- SIP/sip10-00000022 answered SIP/sip11-00000021
    -- Channel SIP/sip10-00000022 joined 'simple_bridge' basic-bridge <c680f0a1-cb42-4d2d-9027-325657d59504>
    -- Channel SIP/sip11-00000021 joined 'simple_bridge' basic-bridge <c680f0a1-cb42-4d2d-9027-325657d59504>
       > 0x270d708 -- Probation passed - setting RTP source address to 192.168.101.30:12126
       > 0x74407208 -- Probation passed - setting RTP source address to 192.168.101.13:10000
    -- Started music on hold, class 'default', on channel 'SIP/sip11-00000021'
       > 0x270d708 -- Probation passed - setting RTP source address to 192.168.101.30:12126
    -- Stopped music on hold on SIP/sip11-00000021
       > 0x270d708 -- Probation passed - setting RTP source address to 192.168.101.30:12126
    -- Channel SIP/sip10-00000022 left 'simple_bridge' basic-bridge <c680f0a1-cb42-4d2d-9027-325657d59504>
    -- Channel SIP/sip11-00000021 left 'simple_bridge' basic-bridge <c680f0a1-cb42-4d2d-9027-325657d59504>
  == Spawn extension (arrivee-sip, 10, 1) exited non-zero on 'SIP/sip11-00000021'

Here is my Asterisk version : Asterisk GIT-14-5d6e90c94a.

Do you have any idea ?

Best regards,
Jonathan