No sound between bridged calls

Asterisk Asterisk Support
1

Hello all,

I have various soft phone extensions which all authenticate properly with an Ubuntu 10.10 box running Asterisk 1.8.

The following calls work correctly:

  • Extension to extension
  • Extension to external
  • DID to dialplan (Adhearsion)

The problem is that there is no sound from any incoming DID calls to any extension when bridged through Asterisk.

I can’t tell if this a codec or NATing issue.

For what should I be looking in the logs?

Any help would be greatly appreciated!!

sip.conf:

[general]
context=incoming
format=gsm               

externhost=liffysbox.dyndns.org
externrefresh=600

localnet=192.168.1.0/255.255.255.0

nat=yes

udpbindaddr=0.0.0.0             

bindaddr=0.0.0.0

tcpbindaddr=0.0.0.0             
srvlookup=yes                   

[basic-options](!)                ; a template
        dtmfmode=rfc2833
        context=local
        type=friend
        host=dynamic

[100](basic-options)
dtmfmode=inband
relaxdtmf=yes
md5secret=abd81fb84a2f01a02ba644d9c170008a

[121](basic-options)
md5secret=153e1b0ab3d13dee9af60d0ca02a1302

[122](basic-options)
md5secret=1fe889629a57abf811944049a6968155

[123](basic-options)
md5secret=3ab72802f01f6d3b84ece8292958a676

Asterisk codecs:

apollo*CLI> core show codecs audio
Disclaimer: this command is for informational purposes only.
   It does not indicate anything about your configuration.
                INT    BINARY                  HEX   TYPE       NAME   DESCRIPTION
-----------------------------------------------------------------------------------
                  1 (1 <<  0)                (0x1)  audio       g723   (G.723.1)
                  2 (1 <<  1)                (0x2)  audio        gsm   (GSM)
                  4 (1 <<  2)                (0x4)  audio       ulaw   (G.711 u-law)
                  8 (1 <<  3)                (0x8)  audio       alaw   (G.711 A-law)
                 16 (1 <<  4)               (0x10)  audio   g726aal2   (G.726 AAL2)
                 32 (1 <<  5)               (0x20)  audio      adpcm   (ADPCM)
                 64 (1 <<  6)               (0x40)  audio       slin   (16 bit Signed Linear PCM)
                128 (1 <<  7)               (0x80)  audio      lpc10   (LPC10)
                256 (1 <<  8)              (0x100)  audio       g729   (G.729A)
                512 (1 <<  9)              (0x200)  audio      speex   (SpeeX)
               1024 (1 << 10)              (0x400)  audio       ilbc   (iLBC)
               2048 (1 << 11)              (0x800)  audio       g726   (G.726 RFC3551)
               4096 (1 << 12)             (0x1000)  audio       g722   (G722)
               8192 (1 << 13)             (0x2000)  audio     siren7   (ITU G.722.1 (Siren7, licensed from Polycom))
              16384 (1 << 14)             (0x4000)  audio    siren14   (ITU G.722.1 Annex C, (Siren14, licensed from Polycom))
              32768 (1 << 15)             (0x8000)  audio     slin16   (16 bit Signed Linear PCM (16kHz))
         4294967296 (1 << 32)        (0x100000000)  audio       g719   (ITU G.719)
         8589934592 (1 << 33)        (0x200000000)  audio    speex16   (SpeeX 16khz)
...
    140737488355328 (1 << 47)     (0x800000000000)  audio    testlaw   (G.711 test-law)

Ekiga is enabled with (in order of preference) the following codecs:
PCMU
PCMA
Speex 16 kHz
gsm
Speex 8 kHz
G726-32
G722

Output of “sip set debug on”: pastie.org/1571130

Server side (Asterisk) “tcpdump -i eth0 -n -s0 -v udp port 5060”: pastie.org/1571150

Client side (softphone) “tcpdump -i eth0 -n -s0 -v udp port 5060”: pastie.org/1571161