Unable to hear calls


(Alejandro) #1

Im not able to hear calls in another network it was working fine but now i just received the calls but im not able to hear them although they are able to hear me. The 5060 port is open on my firewall and modem.
It might be a NAT configuration on my freepbx because if i try from different networks im getting the same issue.
This are my configuration:
Nat config
NAt=yes
Ip Configuration= static ip
Override External IP = my public ip

Sip show settings
Global Settings:

UDP Bindaddress: 0.0.0.0:5060
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: Off
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promisc. redir: No
Enable call counters: No
SIP domain support: No
Path 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-14.0.13.4(13.27.1)
SDP Session Name: Asterisk PBX 13.27.1
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Trust RPID: No
Send RPID: No
Legacy userfield parse: No
Send Diversion: Yes
Caller ID: Unknown
From: Domain:
Record SIP history: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: 4294967295
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No
Store SIP_CAUSE: No

Network QoS Settings:

IP ToS SIP: CS3
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

Network Settings:

SIP address remapping: Enabled using externaddr
Externhost:
Externaddr: mypublicip:0
Externrefresh: 10
Localnet: mypublicip/255.255.255.248

Global Signalling Settings:

Codecs: (ulaw|alaw|gsm|g726|g722)
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: Yes
Compact SIP headers: No
RTP Keepalive: 500
RTP Timeout: 30
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
Sub. min duration 60 secs
Sub. max duration: 3600 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Outbound reg. retry 403:No
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
Record on feature: automon
Record off feature: automon
Force rport: Yes
DTMF: rfc2833
Qualify: 0
Keepalive: 0
Use ClientCode: No
Progress inband: No
Language: en
Tone zone:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: *97
RTCP Multiplexing: No


(Molly Mae) #2

Are you forwarding your RTP ports? By default, it’s ports 10000-20000.

The media of the call is sent over random ports in this range.


(Alejandro) #3

No im not, I created a IAX2 account and looks like that protocol is working. This is weird.


(Dave Burgess) #4

Not really. If you don’t port forward the RTP ports, the outside call cant initiate the RTP stream.

IAX2 doesn’t use separate SIP and RTP streams, so that makes sense.


(Alejandro) #5

How should forward the ports? And the weird part to me is that i haven’t make any changes on that modem all that i change is my freepbx. And that freepbx works fine with another networks no issues at all.
How I can make sure the problem is not coming with a miss-configuration on my pbx.
Thanks in advance…


(system) closed #6

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