Hi, I’m having problems getting around the following problem:
One way audio inbound calls / no audio from remote
-
I can place outbound calls without any issue, but if I receive a call, the other party can’t hear my voice…
-
Also the inbound extension seems to ring endless after having picked up an inbound call.
Does anyone have a clue on how to get around this? I’m using freepbx with pfsense and a cisco 7961 phone
My setup:
PFSENSE firewall
(1 LAN & 1 WAN port)
external_sip_servers = all ips of my provider’s servers
- portforwardings
wan interface
Rule1
source external_sip_servers udp 5060,5061,5062
destination pbx udp 5060,5061,5062
Rule2
source external_sip_servers udp 10000:20000
destination pbx udp 10000:20000
- AON - Advanced Outbound NAT
wan interface
Rule1
source pbx udp 10000:20000
destination external_sip_servers udp 10000:20000
nat address wan address
nat port *
static port Yes
Rule2
source pbx udp 5060,5061,5062
destination external_sip_servers udp 10000:20000
nat address wan address
nat port *
static port Yes
FREEPBX
Global Settings:
----------------
UDP Bindaddress: 0.0.0.0:5061
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
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-13.0.74(11.22.0)
SDP Session Name: Asterisk PBX 11.22.0
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
Call Events: 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: Disabled
Externhost: <none>
Externaddr: (null)
Externrefresh: 10
Localnet: 192.168.1.0/255.255.255.0
Global Signalling Settings:
---------------------------
Codecs: (gsm|ulaw|alaw|g726)
Codec Order: ulaw:20,alaw:20,gsm:20,g726:20
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: Yes
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
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 403:0
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: <not set>
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: Never
Language: de
Tone zone: <Not set>
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: *97
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