No audio on incoming calls from messagenet

Hi all,

i have an issue with incoming calls from messagenet with asterisk+freepbx.

My situation is:

A server with archlinux, with asterisk 1.8.4 and freepbx 2.9 installed on it.
This server is behind a nat.

There are configured some extensions (100, 101, 102…)
There are some trunks, voipcheap, megavoip, messagenet

I use voipcheap and megavoip for outgoing calls, and messagenet for incoming calls…

In “Tools”->“Asterisk sip settings” i have configured the nat settings with dynamic dns:
NAT: yes
Ip configuration: Dynamic ip
Dynamic host: mydyndns_account - Refresh Rate: 10
Local Networks:

In audio settings i’ve enabled three codecs in this order:
ulaw - alaw - gsm

Then, in the nat i have forwarded the ports 5060-5061 and 10000-20000 both TCP and UDP to the asterisk server, i set the range of RTP (10000-20000) in according to the range explaned in /etc/asterisk/rtp.conf

Then with a Linksys PAP2T, LINE1 is registered to “100” extension of this server, and LINE2 is registered with messagenet directly.

There is another sip client (sipdroid) registered to “101” extension of this server.

The result is:

calls extension 100 to extension 101 and viceversa works perfectly
outgoing calls using Megavoip WORKS
outgoing calls using Voipcheap WORKS
outgoing calls using Messagenet WORKS

incoming calls using Messagenet from server, to both 100 and 101 extensions DON’T WORK.
The phone attached at both LINE of pap2t ring, but if i answer on the LINE1 there is no audio, and after few seconds the call drop.

If i answer on the LINE2 (that is directly connected with messagenet) the call WORKS perfectly…

I haven’t understand how to debug this issue… I’ve searched for similar posts in this forums, but no-one help me…

Some information from asterisk CLI are:

sip show settings

Global Settings:

UDP Bindaddress:
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: No
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promsic. redir: No
Enable call counters: No
SIP domain support: No
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: FPBX-2.9.0(1.8.4)
SDP Session Name: Asterisk PBX 1.8.4
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Caller ID: Unknown
From: Domain:
Record SIP history: Off
Call Events: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: -1
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No

Network QoS Settings:

IP ToS RTP audio: EF
IP ToS RTP video: AF41
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No
Jitterbuffer forced: No
Jitterbuffer max size: -1
Jitterbuffer resync: -1
Jitterbuffer impl:
Jitterbuffer log: No

Network Settings:

SIP address remapping: Enabled using externhost
externaddr: PUBLIC_IP(IT IS CORRECT):0
Externrefresh: 10

Global Signalling Settings:

Codecs: 0xe (gsm|ulaw|alaw)
Codec Order: ulaw:20,alaw:20,gsm:20
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: Yes
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 60
RTP Hold Timeout: 300
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: No
Pedantic SIP support: Yes
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Notify ringing state: Yes
Include CID: No
Notify hold state: Yes
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy:
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70

Default Settings:

Allowed transports: UDP
Outbound transport: UDP
Context: from-sip-external
Force rport: Yes
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: *97

sip show registry
Host dnsmgr Username Refresh State Reg.Time N xxxxxxxx 105 Registered Sat, 23 Jul 2011 13:30:26 N yyyyyyy 105 Registered Sat, 23 Jul 2011 13:30:19 N zzzzzzzz 105 Registered Sat, 23 Jul 2011 13:30:26
3 SIP registrations.

Any ideas?

Have a nice day!


I have inverted the sip clients 100 and 101 and now it works :s…


Linksys PAP2T - Line 1 was registered with 100
Sipdroid client was registered with 101


Linksys PAP2T - Line 1 was registered with 101
Sipdroid client was registered with 100

I can hear audio on incoming calls, in both extensions, and everything works smooth… It’s very strange, becouse the configurations of the extensions are the same! But if i go back to the old configuration, the audio of incoming calls don’t works, if i invert the registration of the sip client, it works!