PJSIP Incoming Calls, Multiple Transports

Asterisk Asterisk Support
1

Hi,

I am trying to configure Asterisk 13 to send and receive calls from multiple VoIP providers over two different internet connections. To do so, I have set up two transports with different external_signaling_address and external_media_address and assigned them to the proper endpoints (see sample below). The problem is, when an INVITE from our ‘secondary’ VoIP provider comes in over the connection to which it is assigned (10.2.2.2 in the example), for some reason the OK response Asterisk is sending uses the IP address from the ‘primary’ transport in the Header Contact and in the Message Body Session Description Protocol entries, even though we haven’t specified that transport anywhere in the ‘secondary’ setup. This causes our provider to end RTP and other signal packets to the wrong IP. I’m sure I’m simply missing some setting, but I haven’t been able to figure out what needs to be done to fix the issue. Any help would be appreciated!

[transport-primary]
type = transport
protocol = udp
bind = 0.0.0.0:5060
local_net = 192.168.0.0/16
external_signaling_address = 10.1.1.1
external_media_address = 10.1.1.1

[transport-secondary]
type = transport
protocol = udp
bind = 0.0.0.0:5061
local_net = 192.168.0.0/16
external_signaling_address = 10.2.2.2
external_media_address = 10.2.2.2

[primary]
type = aor
contact = sip:10.5.5.5:5060

[primary]
type = identify
endpoint = primary
match = 10.5.5.5

[primary]
type = endpoint
context = inbound
dtmf_mode = rfc4733
disallow = all
allow = ulaw
allow = g726
allow = gsm
rtp_symmetric = yes
rewrite_contact = yes
transport = transport-primary
language = en
aors = primary
direct_media = no

[secondary]
type = aor
contact = sip:10.9.9.9

[secondary]
type = identify
endpoint = secondary
match = 10.9.9.9

[secondary]
type = endpoint
context = inbound
dtmf_mode = rfc4733
disallow = all
allow = ulaw
allow = g726
allow = gsm
transport = transport-secondary
inband_progress = no
language = en
aors = secondary
direct_media = no

Thanks,
Dave

2

I may be confused, but this looks like a bug to me.

Can you file this on issues.asterisk.org/jira and include the following:

  • The configuration you provided here.
  • Dialplan used to make/receive the call
  • Asterisk log of a call demonstrating the issue.
  • PCAP (wireshark compatible) of the same call.

Be sure the Asterisk log includes DEBUG channel messages, with the DEBUG level turned up to 5 or above.

wiki.asterisk.org/wiki/display/ … nformation