Hi,
Our server is located at Site A, and we have user at Site B. Both sites are connected via Outdoor wireless.
We don’t have issue with the softphones at Site A with multiple vlan and multiple subnet. But we can’t connect from Site B.
Site A Subnet: 172.16.4.0/255.255.252.0
Site B Subnet: 172.16.10.0/255.255.254.0
And there is a firewall at Site A.
All traffics from Site B to the servers at Site A will have to go through this firewall, a Fortinet Fortigate UTM.
Both sites of the outdoor APs are connected to the router.
The router at Site B will throw all the traffics from Site B to A including accessing to Internet.
I’ve even open the firewall port to any, and somehow X-lite still unable to register. I can see the traffic coming in from the Firewall log.
2013-04-15 18:23:29 notice allowed 2 172.16.10.66 172.16.4.10 5060/udp 24420 0
Traceroute from the Freepbx to the client:
traceroute to 172.16.10.66 (172.16.10.66), 30 hops max, 40 byte packets
1 172.16.4.1 (172.16.4.1) 0.418 ms 0.848 ms 1.063 ms
2 172.16.7.246 (172.16.7.246) 0.572 ms 0.541 ms 0.515 ms
3 172.16.7.254 (172.16.7.254) 0.928 ms 1.104 ms 1.299 ms
4 172.16.254.21 (172.16.254.21) 4.161 ms 4.150 ms 4.141 ms
5 172.16.10.66 (172.16.10.66) 5.868 ms * *
And from the Freepbx end, we also have SIP trunk configured.
I’ve also disable the ALG on both sites’ routers and Fortigate Fw.
Below is the information on “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: No
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-2.10.1(1.8.5.0)
SDP Session Name: Asterisk PBX 1.8.5.0
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Legacy userfield parse: No
Caller ID: Unknown
From: Domain:
Record SIP history: Off
Call Events: On
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 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: xxx.xxx.xxx.xxx
Externrefresh: 10
Localnet: 172.16.4.0/255.255.252.0
172.16.10.0/255.255.254.0
Global Signalling Settings:
Codecs: 0x90e (gsm|ulaw|alaw|g726|g729)
Codec Order: ulaw:20,gsm:20,alaw:20,g726:20,g729:20
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: Yes
Pedantic SIP support: Yes
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Outbound reg. timeout: 30 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: Refuse
Session Refresher: uas
Session Expires: 600 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-pstn
Force rport: Yes
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: *97
Appreciate if someone could help.
Regards
Shanmomo