Joseph_sys
Advocate
Joined: 08 Jun 2004
Posts: 2712
Location: Edmonton, AB
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 Posted: Thu Feb 01, 2024 11:44 pm Post subject: Call between asterisk servers over iax - can not hear |
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Calling between two asterisk servers over IAX, I can hear the remote phone ringing but we can not hear each other.
When remote asterisk calls me (home server) it works OK.
On a router I have setup port forwarding:
external: 4569 to internal: 4569 protocol Both
dialing remote: asterisk
| Code: | exten => 4,1,Dial(IAX2/home_server:5xxxxxxx7@${clinic_server}/${EXTEN},60,rw)
exten => 4,n,Hangup() |
| Code: | iax2 show registry
Host dnsmgr Username Perceived Refresh State
192.168.143.1:4569 N home_serve 192.168.143.7:4569 60 Registered
1 IAX2 registrations |
| Code: | home: asterisk dial plan:
exten => 5,1,Dial(IAX2/home_server:5xxxxxx7@${clinic_server}/${EXTEN},60,rw)
exten => 5,n,Hangup() |
i | Code: | ax.conf
[clinic_server]
type=friend
host=dynamic
context=internal
disallow=all
allow=ulaw
allow=alaw
requirecalltoken=no
callgroup=1
pickupgroup=1
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remote asterisk dial plan:
| Code: | exten => 4,1,Dial(SIP/4,15,trw)
exten => 4,n,GotoIf($[“${DIALSTATUS}”=“BUSY”]?vmail:line2)
exten => 4,n(line2),Dial(SIP/54,20,rw)
exten => 4,n(vmail),Voicemail(4)
exten => 4,n,Voicemail(4)
exten => 4,n,Hangup()
exten => 5,1,Dial(${FD_L2},25,trw)
exten => 5,n,GotoIf($[“${DIALSTATUS}”=“BUSY”]?line1)
exten => 5,n(line1),Dial(${FD_L1},20,trw)
exten => 5,n,Voicemail(4)
exten => 5,n,Hangup() |
| Code: | iax.conf
[home_server]
type=friend
host=dynamic
secret=5xxxxxxx7
context=extensions
disallow=all
allow=ulaw
allow=alaw
callgroup=1
pickupgroup=1 |
When I call from home server to remote asterisk:
on home server:
| Code: | – Executing [5@internal:1] Dial(“SIP/55-00000001”, “IAX2/home_server:5xxxxxx7@192.168.143.1/5,60,rw”) in new stack
– Called IAX2/home_server:5xxxxx7@192.168.143.1/5
– Call accepted by 192.168.143.1:4569 (format ulaw)
– Format for call is (ulaw)
– IAX2/192.168.143.1:4569-8095 is ringing
– IAX2/192.168.143.1:4569-8095 is ringing
– Hungup ‘IAX2/192.168.143.1:4569-8095’
– No one is available to answer at this time (1:0/0/0)
– Executing [5@internal:2] Hangup(“SIP/55-00000001”, “”) in new stack
== Spawn extension (internal, 5, 2) exited non-zero on 'SIP/55-00000001 |
remote asterisk:
| Code: | – Accepting AUTHENTICATED call from 192.168.143.7:
– > requested format = ulaw,
– > requested prefs = (ulaw|gsm|ilbc|speex|g729|g723|alaw),
– > actual format = ulaw,
– > host prefs = (ulaw|alaw),
– > priority = mine
– Executing [5@extensions:1] Dial(“IAX2/home_server-4355”, “SIP/54,25,trw”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/54
– SIP/54-0000000a is ringing
> 0x7fef8c013f90 – Strict RTP learning after remote address set to: 10.10.0.15:6030
– SIP/54-0000000a answered IAX2/home_server-4355
> 0x7fef8c013f90 – Strict RTP switching to RTP remote address 10.10.0.15:6030 as source
> 0x7fef8c013f90 – Strict RTP learning complete - Locking on source address 10.10.0.15:6030
== Spawn extension (extensions, 5, 1) exited non-zero on ‘IAX2/home_server-4355’
– Hungup ‘IAX2/home_server-4355’ |
I can hear the phone ringing in a remote location but when they pick up a phone, they can not hear me nor can I hear them.
When I call a remote location over standard POT line, the calls go through normally.
I have a backup server that is running same version of asterisk and has same dial plan,
all the file in asterisk directory are the same; compare them with “meld”
On backup server asterisk calls go through without any problems. |
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Networking & Security
