Iax from one server to the second server no connection?

Hello i have followed a lot of Tut online even ome videos on youtube how to setup a iax server from one freepbx server to another. I can ping both addressses beause they re VPN together. If I’m on freepbx 1 server in the command line i can ping the second freepbx server with internal ip address it goes through. vs versus works as well from second server to first server.

As of today i found this post Here. and followed it but no luck. I’m not sure what is going on. When i looked at the first server and i make a call to the server phone ext nothing goes through. Not sure what to do next. Here is my setup i have used.

first server

trunkname SECOND-SERVER

PEER Details
host=myipaddress
username=myusername
secret=mypassword
type=friend
context=from-internal
transfer=no
allow=alaw&ulaw
disallow=all

Second server

trunk name FIRST-SERVER
PEER:
host=myipaddress
username=myusername
secret=mypassword
type=friend
context=from-internal
transfer=no
allow=alaw&ulaw
disallow=all

Can someone please help me with what I’m missing?

Joseph

Hello here is a update from the asterisk


  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [[email protected]:1] ResetCDR("SIP/761-00001a87", "") in new stack
    -- Executing [[email protected]:2] NoCDR("SIP/761-00001a87", "") in new stack
    -- Executing [[email protected]:3] Progress("SIP/761-00001a87", "") in new stack
    -- Executing [[email protected]:4] Wait("SIP/761-00001a87", "1") in new stack
       > 0x7f69bc14e910 -- Probation passed - setting RTP source address to xx.xx.xx.xx
    -- Executing [[email protected]:5] Progress("SIP/761-00001a87", "") in new stack
    -- Executing [[email protected]:6] Playback("SIP/761-00001a87", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
    -- <SIP/761-00001a87> Playing 'silence/1.alaw' (language 'en')
    -- <SIP/761-00001a87> Playing 'cannot-complete-as-dialed.alaw' (language 'en')
  == Spawn extension (from-internal, 600, 6) exited non-zero on 'SIP/761-00001a87'
    -- Executing [[email protected]:1] Hangup("SIP/761-00001a87", "") in new stack
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/761-00001a87'

This is all that comes up. i know the ext 600 works.

Joseph

Cannot complete as dialed is indicating that there were not any matches on the outbound route for the dialed number. Check your outbound route from Server 1 to Server 2 and viceversa.

… also, if you’re using IP authentication (you only allow connections from their IP address), adding Username and Password just adds overhead without any improvement in security or connection speed. IP Auth is the fastest way to connect two terminus servers on a line like this.

Hello My dialing plan does work. I can dial out phone numbers with no problem. But if i try to dial a Ext that is from the second server to the first server that is when i get that error. Same if i was calling a Ext from the first server to the second one same error. Both servers can dial phone numbers with no problem i followed the the letter as they say it the directions from that link.

To go from one server to the other you need outbound routes.
Show the outbound routes of the two servers.

If you want to dial local (in addition to external) resources on the remote server, the context for the trunk on the remote server should be from-internal and the Outbound Route on the local server should be flagged Intra-Company.

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