Hello all,
Good day,
I am new on Freepbx config’s. I have got an issue where our calls gets Hung in between for couple of seconds. Not sure where is the problem, weather Network side or in an config in PBX. Currently we have a separate dedicated VLAN for VOIP also PBX server has two Nics one to publi IP configured and another was connected to LAN VOIP Vlan.
Can you guys help me out to diagnosis, below is my sip
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
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-13.0.192.9(13.9.1)
SDP Session Name: Asterisk PBX 13.9.1
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: Disabled
Externhost:
Externaddr: (null)
Externrefresh: 10
Localnet: 12.210.165.176/255.255.255.248
12.194.24.0/255.255.255.0
192.168.10.0/255.255.255.0
Global Signalling Settings:
Codecs: (ulaw|alaw|gsm|g726|g729)
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:
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
==========================
Interface em1 IP: 192.168.10.2
Interface em1 MAC: xx:83:xx:xx:2B:xx
Interface em2 IP: xx.xxx.xxx.178
Interface em2 MAC: xx:xx:xx:xx:2B:xxxxxxx:
[root@pbx01 asterisk]# route -n
Kernel IP routing table
Destination Gateway Genmask Flags Metric Ref Use Iface
12.210.165.176 0.0.0.0 255.255.255.248 U 0 0 0 em2
12.194.24.0 12.210.165.177 255.255.255.0 UG 0 0 0 em2
192.168.10.0 0.0.0.0 255.255.255.0 U 0 0 0 em1
169.254.0.0 0.0.0.0 255.255.0.0 U 1002 0 0 em1
169.254.0.0 0.0.0.0 255.255.0.0 U 1003 0 0 em2
0.0.0.0 192.168.10.1 0.0.0.0 UG 0 0 0 em1
[root@pbx01 asterisk]#