- Joined
- Jan 22, 2020
- Messages
- 28
- Reaction score
- 4
I have a bridge established between 3cx (Master) and Asterisk (Slave) Outbound routing to a separate trunk works without issue. Likewise, extension dialing works without issue from Asterisk to 3cx. This is good enough, as extension dialing from 3cx is not needed (Special use bridge).
The issue I am having is routing traffic from the trunk to the bridge. Most documentation I have reviewed provides that a simple "forward to outside number" plus the extension bridge number should route the inbound calls. In testing I can see the call hit, but it does not provide a route.
Do I need an outbound prefix to hit the bridge even though it is assigned the bridge number? If so, what format does it need to be in? If the external [bridge] number is 10012, should I place 910012 as the external number?
07/20/2021 3:33:25 PM - [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification: Invite-UNK Recv Req INVITE from 162.252.250.42:7000 tid=56fb.c0b40812b7026f71621b449b43c09f8a.0 Call-ID=458254b2-a466-4bcd-905d-d8bfca847814:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 162.252.250.42:7000;branch=z9hG4bK56fb.c0b40812b7026f71621b449b43c09f8a.0
Via: SIP/2.0/UDP 162.252.251.66:11000;received=162.252.251.66;rport=11000;branch=z9hG4bKrpegDmFS7KBDB
Max-Forwards: 48 Record-Route: <sip:162.252.250.42:7000;lr=on;ftag=avtK0ZKDSUygS>
Contact: <sip:[email protected]:11000>
To: <sip:[email protected]>
From: "CHAMPAIGN IL" <sip:[email protected]>;tag=avtK0ZKDSUygS
Call-ID: 458254b2-a466-4bcd-905d-d8bfca847814 CSeq: 38852810 INVITE
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Content-Disposition: session
Content-Type: application/sdp
Supported: path, replaces
User-Agent: Atlas
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Privacy: none
P-Asserted-Identity: "CHAMPAIGN IL" <sip:[email protected]>
Content-Length: 275
v=0 o=FreeSWITCH 1626789265 1626789266 IN IP4 162.252.251.66
s=FreeSWITCH c=IN IP4 162.252.251.66
t=0 0 m=audio 23940 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16 a=silenceSupp
ff - - - -
a=ptime:20
The issue I am having is routing traffic from the trunk to the bridge. Most documentation I have reviewed provides that a simple "forward to outside number" plus the extension bridge number should route the inbound calls. In testing I can see the call hit, but it does not provide a route.
Do I need an outbound prefix to hit the bridge even though it is assigned the bridge number? If so, what format does it need to be in? If the external [bridge] number is 10012, should I place 910012 as the external number?
07/20/2021 3:33:25 PM - [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification: Invite-UNK Recv Req INVITE from 162.252.250.42:7000 tid=56fb.c0b40812b7026f71621b449b43c09f8a.0 Call-ID=458254b2-a466-4bcd-905d-d8bfca847814:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 162.252.250.42:7000;branch=z9hG4bK56fb.c0b40812b7026f71621b449b43c09f8a.0
Via: SIP/2.0/UDP 162.252.251.66:11000;received=162.252.251.66;rport=11000;branch=z9hG4bKrpegDmFS7KBDB
Max-Forwards: 48 Record-Route: <sip:162.252.250.42:7000;lr=on;ftag=avtK0ZKDSUygS>
Contact: <sip:[email protected]:11000>
To: <sip:[email protected]>
From: "CHAMPAIGN IL" <sip:[email protected]>;tag=avtK0ZKDSUygS
Call-ID: 458254b2-a466-4bcd-905d-d8bfca847814 CSeq: 38852810 INVITE
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Content-Disposition: session
Content-Type: application/sdp
Supported: path, replaces
User-Agent: Atlas
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Privacy: none
P-Asserted-Identity: "CHAMPAIGN IL" <sip:[email protected]>
Content-Length: 275
v=0 o=FreeSWITCH 1626789265 1626789266 IN IP4 162.252.251.66
s=FreeSWITCH c=IN IP4 162.252.251.66
t=0 0 m=audio 23940 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16 a=silenceSupp
a=ptime:20