auto forwarding calls between trunks

[yelgadi]

Hello everyone,

I have a question -might be a noob question- in my setup I need to connect 3cx to voipswitch and to goip

what I am currently trying to achieve is that I want all of the calls coming from voipswitch to be auto forwarded to the GoIP box

voipswitch is connected as a sip trunk (10000)
goip is connected as a PSTN Trunk (90001 , 90002 , 90003 , 90004) 4 ports

currently I can dial external numbers from extensions via GoIP
but whenever I try to forward calls from voipswitch i get the following,

03-Jun-2016 18:54:06.953 Leg L:14.1[Line:10000<<00114-faisal] is terminated: Cause: BYE from PBX
03-Jun-2016 18:54:06.914 [CM503020]: Call(C:14): Normal call termination. Call originator: Line:10000<<00114-faisal. Reason: Not found
03-Jun-2016 18:54:06.914 [CM503014]: Call(C:14): No known route from Line:10000<<00114-faisal to target: <sip:[email protected]:5060>
03-Jun-2016 18:54:06.914 Call(C:14): from Line:10000<<00114-faisal to 9 doesn't match any outbound rule. No outbound rule could be selected.
03-Jun-2016 18:54:06.914 [Flow] Target endpoint for 9 can not be built!
03-Jun-2016 18:54:06.913 [CM505003]: Provider:[voipswitch] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [VoipSwitch 2.0] PBX contact: [sip:[email protected]:5060]
03-Jun-2016 18:54:06.911 [CM503001]: Call(C:14): Incoming call from Line:10000<<00114-faisal to <sip:[email protected]:5060>
03-Jun-2016 18:54:06.911 Line limit check: Current # of calls for line Lc:10000(@voipswitch[<sip:[email protected]:5060>]) is 1; limit is 4
03-Jun-2016 18:54:06.911 NAT/ALG check:L:14.1[Line:10000<<00114-faisal] REQUEST 'INVITE' - basic check passed. No information for extended checks
03-Jun-2016 18:54:06.903 [CM503012]: Inbound any hours rule (00218*) for 10000 forwards to DN:9

I am trying to autoforward to external number 9 -as a test- and my outbound rules are setup to accept a prefix starting with 9

but as I can understand from the log the 3cx is actually trying to call 9 instead of forwarding the whole number,

can anyone assist?

Thank you in advance guys,

 

[leejor]

Direct trunk to trunk may not work. 3CX isn't really designed as a Tandem switch. Incoming calls are directed, in a number of ways, to an extension. With that in mind, an extension can then forward calls back out on a trunk. So... you will have to establish some extensions, send the incoming calls to them, and use the forwarding rules, of the extension to direct the calls back out again. This will also involve the use (creation) of the appropriate outbound rules.
You may also want to investigate the use of a Bridge Trunk to connect to the other PBX, if you haven't already, there may be some advantages to using this connection method.

03-Jun-2016 18:54:06.914 Call(C:14): from Line:10000<<00114-faisal to 9 doesn't match any outbound rule. No outbound rule could be selected.

This tells me that you haven't built an outbound rule, which may not be the only issue.

 

[yelgadi]

thanks for the reply,

well for some reason bridging didnt work with voipswitch , the only option that worked was a sip trunk

anyways Thank you for the info , I will try a little bit more , but i have a feeling that 3CX wont be the best option for my setup

 

[leejor]

Did you set the Bridge trunk on 3CX as the Master, and have the other PBX register to it as if 3CX were simply another VoIP provider?

If you were able to use a VoIP trunk, then how did it register to the other PBX, as an extension on the switch? Or, does it allow a trunk to register to it?

Users that have Bridge trunks to other 3CX installation, can dial over the trunks to an outside call at the far end. It all depends on exactly what you are trying to achieve. You may first need to review the outbound rules you created (or didn't) for the calls.

 

[yelgadi]

Hey Leejor,

thanks for your reply,

I tried to setup the remote voipswitch as a bridge as per your suggestion,

and apparently there is an authentication issue -I am sure a 100% that credentials match-

here is the log from a call:

04-Jun-2016 21:32:12.261 [CM102001]: Authentication failed for AuthFail Recv Req INVITE from 164.132.81.230:443 tid=321219127999125118347771 [email protected]:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 164.132.81.230:443;branch=z9hG4bK321219127999125118347771
Max-Forwards: 70
Contact: <sip:164.132.81.230:443;transport=udp>
To: <sip:[email protected]:5060>
From: <sip:[email protected]:443>;tag=123219118347646
Call-ID: [email protected]
CSeq: 2 INVITE
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, INFO, NOTIFY, MESSAGE, SUBSCRIBE, REFER, PUBLISH, UPDATE
Content-Type: application/sdp
Proxy-Authorization: DIGEST username="10000",realm="3CXPhoneSystem",nonce="414d535c0d63bdbc47:043664e12a1abc6393ce272ab1b36f6c",uri="sip:[email protected]:5060",response="afd7c833ec05e4b5b17a30f07feb97bf"
User-Agent: VoipSwitch 2.0
Content-Length: 282
Remote-Party-ID: <sip:[email protected]:443>;privacy=off;screen=no

v=0
o=- 496502998 496502999 IN IP4 164.132.81.230
s=VoipSIP
c=IN IP4 164.132.81.230
t=0 0
m=audio 6764 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
; Reason: Credentials don't match, check that authorization-ID and password match the ones in extension settings

 

[leejor]

Can you explain, in more detail, your setup (block diagram?), what digits you are sending (from where), and where you expect them to route. Types of trunks you've used, at each end, if the registration was successful...etc.