Connect 2 boxes by sip

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Connect 2 boxes by sip

atux_null

Hi. I have two pbxes that i would like to interconnect, by sip only. Topology is
-Box A 192.168.1.5/24 with extensions 5xxx
-Box B 192.169.1.6/24 with extensions 6xxx

Box A has some Sip providers, while Box B has some others. I would like
- to be able to call all extensions to the other connected box eg Box A ext 5000 to bw able to call 6010 as if it were in the same pbx. The opposite applies as well.
-be able to route a DID from Box A to Box B and the opposite.

Your help is much appreciated.

Alfio

atux_null You can make a troncal SIP, and create outbound routes using the new created troncal.
For example a simple sip Trunks for test:

host=ippbxb
type=friend
context=from-pstn

For the DID part, in the incoming route there is a option to route incoming calls from one DID to a troncal SIP in the other PBX.

Try and keep us inform.

hgmnetwork

Debes crear dos troncales uno en cada servidor apuntando las IP luego creas una ruta de salida hacia el troncal

atux_null

Hi. i created one trunk to each server that points to the other server and in the sip show peers i can see the status. The following is from server A.
issabelCLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
500/500 192.168.1.105 D No No A 5060 OK (31 ms)
pbx1/pbx1 192.168.1.154 Auto (No) No 5060 OK (1 ms)
2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline]
-- Remote UNIX connection
-- Remote UNIX connection disconnected
issabelCLI>

Similar output comes from Server B pointing to server A's IP.

In server A i have extensions 500-599 and in server B i have extensions 600 -699. It does not seem to send the call to each other. This is from server A

-- Remote UNIX connection
-- Remote UNIX connection disconnected
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [600@from-internal:1] ResetCDR("SIP/500-00000000", "") in new stack
-- Executing [600@from-internal:2] NoCDR("SIP/500-00000000", "") in new stack
-- Executing [600@from-internal:3] Progress("SIP/500-00000000", "") in new stack
-- Executing [600@from-internal:4] Wait("SIP/500-00000000", "1") in new stack
-- Executing [600@from-internal:5] Progress("SIP/500-00000000", "") in new stack
-- Executing [600@from-internal:6] Playback("SIP/500-00000000", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
-- <SIP/500-00000000> Playing 'silence/1.gsm' (language 'en')
-- <SIP/500-00000000> Playing 'cannot-complete-as-dialed.gsm' (language 'en')
== Spawn extension (from-internal, 600, 6) exited non-zero on 'SIP/500-00000000'
-- Executing [h@from-internal:1] Hangup("SIP/500-00000000", "") in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/500-00000000'
issabel*CLI>

In each server i created an Outbound route with prefix 9 and match pattern dot(.) to send everything to the trunk that connects to the opposite server.
Any ideas please?