SIP trunk CM - FreePBX

[Smokeythe2004]

Hi,

I have to connect CM and FreePBX with SIP trunk and I have to do this without Avaya SM.
So far I have managed to make SIP trunk on CM and FreePBX.
Trunk on both side is made with TCP transport method.
Call is working in direction from CM to FreePBX, but from FreePBX to CM does not work.
Calls from FreePBX are routed to the correct SIP trunk to CM.
On wireshark trace (made on FreePBX), I can see FreePBX sending INVITE SIP message, but from CM does not get any SIP response.
In wireshark trace is only SIP Invite from FreePBX and some TCP ACK packets from CM side.
In the same wireshark trace, I can see FreePBX sending SIP Option messages to CM and CM is responding with SIP 200 OK message.
Both CM and FreePBX are in the same network without any firewall between and FreePBX can ping CM and CM can ping FreePBX.

Any idea what did I missed for calls from FreePBX to CM?

Im using CM 6.2 and FreePBX 13.

Regards,
Smokeythe

Here is config for Signal group and trunk group:
signaling group
Group Number: 102 Group Type: sip
IMS Enabled? n Transport Method: tcp
Q-SIP? n
IP Video? n Enforce SIPS URI for SRTP? y
Peer Detection Enabled? y Peer Server: Others



Near-end Node Name: procr Far-end Node Name: FreePBX
Near-end Listen Port: 5060 Far-end Listen Port: 5060
Far-end Network Region: 233

Far-end Domain: 10.99.4.100
Bypass If IP Threshold Exceeded? n
Incoming Dialog Loopbacks: eliminate RFC 3389 Comfort Noise? n
DTMF over IP: rtp-payload Direct IP-IP Audio Connections? n
Session Establishment Timer(min): 3 IP Audio Hairpinning? n
Enable Layer 3 Test? y
Alternate Route Timer(sec): 6




Trunk
Group Number: 902 Group Type: sip CDR Reports: y
Group Name: TEST-FreePBX COR: 995 TN: 1 TAC: *902
Direction: two-way Outgoing Display? n
Dial Access? n Night Service:
Queue Length: 0
Service Type: public-ntwrk Auth Code? n
Member Assignment Method: auto
Signaling Group: 102
Number of Members: 5



TRUNK PARAMETERS

Unicode Name: no

Redirect On OPTIM Failure: 5000

SCCAN? n Digital Loss Group: 18
Preferred Minimum Session Refresh Interval(sec): 600

Disconnect Supervision - In? y Out? y


XOIP Treatment: auto Delay Call Setup When Accessed Via IGAR? n

TRUNK FEATURES
ACA Assignment? n Measured: none
Maintenance Tests? y



Numbering Format: public
UUI Treatment: service-provider

Replace Restricted Numbers? n
Replace Unavailable Numbers? n


Modify Tandem Calling Number: no




Show ANSWERED BY on Display? y

change trunk-group 902 Page 4 of 21
PROTOCOL VARIATIONS

Mark Users as Phone? n
Prepend '+' to Calling Number? n
Send Transferring Party Information? n

Send Diversion Header? n
Support Request History? y
Telephone Event Payload Type: 101

Overwrite Calling Identity? n
Convert 180 to 183 for Early Media? y
Always Use re-INVITE for Display Updates? n
Identity for Calling Party Display: From
Block Sending Calling Party Location in INVITE? n
Enable Q-SIP? n

TRUNK GROUP
Administered Members (min/max): 1/5
GROUP MEMBER ASSIGNMENTS Total Administered Members: 5

Port Name
1: T01068 FreePBX
2: T01069 FreePBX
3: T01070 FreePBX
4: T01071 FreePBX
5: T01072 FreePBX

 

[Arlo555]

Have you tried setting IMS (IP Multimedia Subsystem) to yes in the signaling group form?

 

[Smokeythe2004]

Yes I did, but did not make any difference

 

[kyle555]

PCAP please
* where my thin king is leading me is that CM likes domains. I can't remember the last time I saw IP in far end "domain" but they're not equivalent. You can't just put domain 192.168.1.100 in Session Manager and expect it to treat is like a real domain.

So, make a domain, call it freeswitch.com or something and have a DNS SRV lookup for SIP map to CM's IP.

You're not getting an answer from CM because CM picks incoming sig group like this:
for an invite with a IP/port pair, identify all CM sig groups that match that IP/port pair
Then, answer the lowest numbered signaling group where the entirety of "far end domain name" in that sig group fits into the invite.

So, for invite at steak.cow.farm.com, were sig group 1 farm.com, it would match. Were sig 1 ribs.cow.farm.com, it would not. Were it cow.farm.com, it would match.
Best practice for domain design is most specific subdomains have lowest numbers. sig 999 should be yourrootdomain.com

Anyway, all to say that I don't think CM is assigning it to a sig group at all if you're not getting a 100 trying or 180 ringing back. Just like if you spun up another SIP thing that CM knew nothing about and tossed sip messaging at it - it just won't answer.

 

[Smokeythe2004]

Problem solved.

It was not fault on CM. Problem was in script on FreePBX, script was inserting diversion header in SIP message.
After fixing this script (removed diversion header from SIP header) in FreePBX, calls started to work.

Kyle555 I will try what you said.

Thank for help.
Will be back with complete config after I made whole setup.