Can't call between 2 SPA3000

Asterisk Asterisk Support
1

I have setup asterisk with 3 Softphone, and 2 SPA3000.
extension setup:
SPA3000 - ext 101 (internal network)
SPA3000 - ext 102 (connect over internet)
Softphone - ext 103 (connect over internet)
Softphone - ext 999 (internal network)
Softphone - ext 998 (internal network)

from 101, I can call all other extension except 102 (resp 486 busy here), calling *43 works fine.

from 102, I can call all extension, but calling to 101 no audio.
calling to *43 is works fine thou.

from any softphone I can call any extension and everything working fine.

I have no clue where is the problem when i’m trying make call between the spa3000

*note both spa3000 are having the same firmware (3.1.20)

2

We have very little clue either, as you have’t provided much useful information.

The most common causes of one way audio are firewalls and NATted routers. If that doesn’t help you, you will need to provide details from your sip.conf, details of your network topology, and the output produced by sip set debug on.

3

My sip.conf

[code][101]
deny=0.0.0.0/0.0.0.0
secret=Pass101!@#
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
nat=no
port=5060
qualify=yes
qualifyfreq=60
transport=udp
encryption=no
callgroup=
pickupgroup=
dial=SIP/101
mailbox=101@device
permit=0.0.0.0/0.0.0.0
callerid=101 <101>
callcounter=yes
faxdetect=no

[102]
deny=0.0.0.0/0.0.0.0
secret=Pass102!@#
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
nat=yes
port=5060
qualify=yes
qualifyfreq=60
transport=udp
encryption=no
callgroup=
pickupgroup=
dial=SIP/102
mailbox=102@device
permit=0.0.0.0/0.0.0.0
callerid=102 <102>
callcounter=yes
faxdetect=no

[103]
deny=0.0.0.0/0.0.0.0
secret=Pass103!@#
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
nat=yes
port=5060
qualify=yes
qualifyfreq=60
transport=udp
encryption=no
callgroup=
pickupgroup=
dial=SIP/103
mailbox=103@device
permit=0.0.0.0/0.0.0.0
callerid=103 <103>
callcounter=yes
faxdetect=no

[/code]

Asterisk is at 192.168.1.47
extension 101 (SPA3000) is 192.168.1.71
There is a router to internet from 192.168.1.* network, port 5060 and 10000-20000 has already been forwarded to asterisk.

extension 102 (SPA3000) is connected from internet to my global IP address.

raspbx*CLI> sip show peers Name/username Host Dyn Forcerport ACL Port Status Description 101/101 192.168.1.71 D A 5060 OK (18 ms) 102/102 180.x.x.152 D N A 16473 OK (102 ms) 103/103 (Unspecified) D N A 0 UNKNOWN 111/111 (Unspecified) D A 0 UNKNOWN 998/998 (Unspecified) D N A 0 UNKNOWN 999/999 192.168.1.67 D N A 50754 OK (44 ms)

As you can see above that the registration is ok.

from the console, this is what I saw when I tried to dial 102 from 101

[2013-09-29 01:23:19] VERBOSE[3495][C-00000002] pbx.c: -- Executing [s@macro-dial-one:42] Dial("SIP/101-00000004", "SIP/102,,trI") in new stack [2013-09-29 01:23:19] VERBOSE[3495][C-00000002] netsock2.c: == Using SIP RTP TOS bits 184 [2013-09-29 01:23:19] VERBOSE[3495][C-00000002] netsock2.c: == Using SIP RTP CoS mark 5 [2013-09-29 01:23:19] VERBOSE[3495][C-00000002] app_dial.c: -- Called SIP/102 [2013-09-29 01:23:19] VERBOSE[3495][C-00000002] app_dial.c: -- Connected line update to SIP/101-00000004 prevented. [2013-09-29 01:23:20] VERBOSE[2836][C-00000002] chan_sip.c: -- Got SIP response 486 "Busy Here" back from 180.x.x.x:16473 [2013-09-29 01:23:20] VERBOSE[3495][C-00000002] app_dial.c: -- SIP/102-00000005 is busy [2013-09-29 01:23:20] VERBOSE[3495][C-00000002] app_dial.c: == Everyone is busy/congested at this time (1:1/0/0)