One Way Voice - Dialing out via SIP trunk

Asterisk Asterisk Support
1

Dear community

I’m currently running an Asterisk version 13.1.0 on a Ubuntu 16.04 server.

I’m having issues with the transmission of voice by certain calls.

Let’s start from scratch and list the participants in this specific scenario:

  • Asterisk PBX
  • SIP trunk registered on VoIP provider
  • Extension A -> User agent registered on Asterisk
  • Extension B -> User agent registered on Asterisk

“Extensions A” dials the direct dial of “Extension B”. Asterisk dials the call via SIP trunk from VoIP provider.

VoIP Provider dials incoming call to the direct dial of “Extension B” and builds up the call.

As soon the VoIP user agent “Extension B” picks up the call I hear no voice. The RTP proxy is a server from the VoIP provider and routes the RTP packets from the RTP proxy to the Asterisk server.

Extensions A <-> Asterisk <-> VoIP Provider RTP Proxy <-> Asterisk <-> Extension B

The VoIP user agents of the extensions don’t receive any RTP packets.

Here are the settings of the extensions and sip trunk:

[Extensions A]
username = Extensions A
transfer = yes

call-limit = 100
type = friend
host = dynamic
fullname = Extensions A
cid_number = Extensions A
context = DLPN_X
hasvoicemail = no
hassip = yes
hasiax = no
secret=secret
nat = yes
canreinvite = nonat
dtmfmode = rfc2833
hasagent = yes
pickupgroup = 1

disallow = all
allow = ulaw,alaw

[Extensions B]
username = Extensions B
transfer = yes

call-limit = 100
type = friend
host = dynamic
fullname = Extensions B
cid_number = Extensions B
context = DLPN_X
hasvoicemail = no
hassip = yes
hasiax = no
secret=secret
nat = yes
canreinvite = nonat
dtmfmode = rfc2833
hasagent = yes
pickupgroup = 1

disallow = all
allow = ulaw,alaw

[trunk_test02]
host = voip provider
username = Trunk-X
secret = secret
type = friend
trunkname = trunk_test02
context = incoming_test
insecure = port,invite
directmedia = yes
disallow = all
allow = alaw,ulaw,gsm

Does anyone has a hint how to handle these kind of calls?

Any help would be greatly appreciated.

Thank you and kind regards
Michel Rios

3

What does rtp debug show for RTP coming from the proxy?

Have you checked that the RTP has not had its port numbers rewritten?

(You have the normal mix of deprecated and bad practice settings that result from copying other configurations, but none of it should be relevant, on the assumption that canreinvite is parsed as though it were directmedia, with all the options of directmedia.)

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