Silence prob inbound SIP call auto routed to SIP trunk


#1

If I make an inbound SIP call to my Asterisk server, and then automatically transfer it to an external SIP trunk in the extensions.conf file, I am able to successfully transfer the call to the external SIP number, however when the call connects I cannot hear any sound.

from extensions.conf:
exten => _1XXXX,n,Dial(SIP/HSV/${EXTEN:1})

here is the info I see in the CLI:
– Executing Dial(“SIP/sipxx.xxx.com.au-09f8aae8”, “SIP/HSV/2468”) in new stack
– Called HSV/2468
– SIP/HSV-5df9 is making progress passing it to SIP/sipxx.xxx.com.au-09f8aae8
– SIP/HSV-5df9 is ringing
– SIP/HSV-5df9 is making progress passing it to SIP/sipxx.xxx.com.au-09f8aae8
– SIP/HSV-5df9 answered SIP/sip04.astrasip.com.au-09f8aae8
– Attempting native bridge of SIP/sip04.astrasip.com.au-09f8aae8 and SIP/HSV-5df9

Call is answered and goes silent at this point. After I hang up the following messages appear in CLI:
== Spawn extension (from-pstn, 12468, 2) exited non-zero on ‘SIP/sipxx.xxx.com.au-09f8aae8’
– Executing Hangup(“SIP/sipxx.xxx.com.au-09f8aae8”, “”) in new stack
== Spawn extension (from-pstn, h, 1) exited non-zero on ‘SIP/sipxx.xxx.com.au-09f8aae8’

Does anyone have any idea why this could be occuring? I thought initially it may be codec related, so changed my codecs so that I dial in using g711u, and the dial out to the SIP trunk is also g711u.