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SIP Trunks to multiple Asterisk servers

siptrunk
freepbx
asterisk
Tags: #<Tag:0x00007fcd22831458> #<Tag:0x00007fcd22831278> #<Tag:0x00007fcd22831138>

(Abdulbaset Alanesi) #1

Hello,

I have 4 asterisk servers and I need to create a SIP trunk between them.
I found different configurations on the internet, and I started with the first two servers, both phones can ring but no voice.

PEER Details:
host=<<>>
type=peer
context=from-trunk
disallow=all
allow=ulaw,alaw
qualify=yes
qualifyfreq=10

Have I missed something?


#2

One way audio or no audio at all are frequently caused by incorrect NAT settings. Are your FreePBX installations behind a NAT device? If not, it can also be caused by a codec mismatch, but IIRC codec mismatch would cause a failed call.


(Abdulbaset Alanesi) #3

Hi arielgrin,

Thanks for your reply,

I think NAT settings are not the same on both servers, and both of them are sitting behind a Pfsense box.
I’ll check the NAT first and come back.


#4

I myself use pfSense without issues, but I read a lot of reports by other members stating that FreePBX behind pfSense is just not recommended. If both FreePBX servers are each behind a pfSense router, why don’t you establish a VPN between them, that way you avoid NAT by sending all SIP traffic down the VPN link. That is how I’ve been using it for more than 5 years and it works great.


(Itzik) #5

Can you perhaps post a guide how to setup the pfSense to be “SIP Friendly”?


#6

Not that I ever left my SIP port opened to the internet other than a couple of minutes for testing, but I managed to make it work by following this link https://doc.pfsense.org/index.php/VoIP_Configuration
In any case, for continuous use, I connect through OpenVPN. Other than correctly configuring OpenVPN server, there is no need for any special SIP related configurations. I don’t like to leave open SIP ports to the internet. Of course I understand this might not be the use case for everybody else.


(Abdulbaset Alanesi) #7

I do have a VPN between all pfsense servers.
But still no voice.


(James Zhu) #8

please show the CLi log and also check the codec as well.


#9

Then there’s definitely something configured incorrectly, probably FreePBX NAT parameters.

I could try to help you figure it out, but you will have to provide more details about your setup, at least network topology and FreePBX SIP settings.


(Abdulbaset Alanesi) #10

That’s great,
So Asterisk is behind the Pfsense, Pfsense is acting as Firewall, VPN, DHCP and local DNS.

SIP server 1:

Global Settings:

UDP Bindaddress: 0.0.0.0:5160
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-13.0.194.2(13.12.1)
SDP Session Name: Asterisk PBX 13.12.1
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: XXX.XXX.XXX.XXX:0
Externrefresh: 10
Localnet: XXX.XXX.XXX.XXX/255.255.255.0
XXX.XXX.XXX.XXX/255.255.255.0
XXX.XXX.XXX.XXX/255.255.255.0

Global Signalling Settings:

Codecs: (ulaw|alaw|gsm|g726)
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: No
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: No
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


END

SIP Server 2

Global Settings:

UDP Bindaddress: 0.0.0.0:5160
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
Videosupport: Yes
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-13.0.192.19(13.12.1)
SDP Session Name: Asterisk PBX 13.12.1
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: xxx.xxx.xxx.xxx:0
Externrefresh: 10
Localnet: xxx.xxx.xxx.xxx/255.255.254.0
xxx.xxx.xxx.xxx/255.255.254.0

Global Signalling Settings:

Codecs: (ulaw|alaw|gsm|g726|h264|mpeg4|vp8|h263p|h263|h261)
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: No
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: No
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


(Abdulbaset Alanesi) #11

Please help I’m stuck here