Inbound connection over H323 problem

Asterisk Asterisk Support
1

Sorry for me English:

Internal phones - SIP, but i need connect to provider over H323 for inbound connection (now normally compeled and fine work for outbound connection). And there a small problem:

  1. externall phone call to my phone number and provider send it call over H323 to my Asterisk-server
  2. Asterisk listen my call and send to one internal SIP phone.
  3. When i answer - continue hear long bill in my external phone and hear nothin in internal phone
  4. If i hang up the receiver - in external phone i hear ringing off.

What problem? Logs and configs below:

fastStart=yes
h245Tunneling=yes
disallow=all
allow=alaw
allow=ulaw
allow=gsm
dtmfmode=rfc2833
gatekeeper=DISABLE
AllowGKRouted = yes
UserByAlias=no; used fo identifu inbound provider - i hav more tha one, but all of them not work!

[74956420775] ; Try many variants - without effect
type=friend
;dtmfcodec=86
;e164=74956420775
host=10.199.198.1
context=apstudio-inc
incominglimit=2

And logs:

pbxCLI>== New H.323 Connection created.
pbxCLI>–Received SETUP message
– Setting up Call
– Call token: [ip$10.199.198.1:33954/12126]
– Calling party name: []
– Calling party number: [84956737715]
– Called party name: [74956420775]
– Called party number: [74956420775]
– Calling party IP: [10.199.198.1]
1:47:30.201 H225 Answer:88bac48 h323.cxx(2401) H225
Set remote application name: "Cisco IOS 12.2 181/18"
Setting capabilities to 0xe (gsm|ulaw|alaw) Capabilities in preference order is (alaw|ulaw|gsm) Allowed Codecs:
pbxCLI> Table:
G.711-ALaw-64k <1>
G.711-uLaw-64k <2>
GSM-06.10 <3>
UserInput/hookflash <4>
UserInput/RFC2833 <5>
UserInput/dtmf <6>
Set:LI>
0:LI>
0:>
G.711-ALaw-64k <1>
G.711-uLaw-64k <2>
GSM-06.10 <3>
1:>
UserInput/hookflash <4>
2:>
UserInput/RFC2833 <5>
UserInput/dtmf <6>
pbxCLI>
1:47:30.239 H225 Answer:88bac48 h323.cxx(4702) H323
CreateLogicalChannel - unknown data type
1:47:30.239 H225 Answer:88bac48 h323.cxx(4702) H323
CreateLogicalChannel - unknown data type pbxCLI>=-= In OnAnswerCall for call 12126
pbxCLI> - Progress Indicator: 0
pbxCLI> - Inserting PI of 0 into ALERTING message
1:47:30.255 H225 Answer:88bac48 h323.cxx(2792) H323
Answering call: AnswerCallDeferredWithMedia
1:47:30.256 H225 Answer:88bac48 h323.cxx(1816) H225
Reading PDUs: callRef=12126
pbxCLI>Answering call ip$10.199.198.1:33954/12126
1:47:30.284 ThreadID=0x05f0fbb0 h323.cxx(2792) H323
Answering call: AnswerCallNow
1:47:30.285 ThreadID=0x05f0fbb0 h323.cxx(3592) H245
Started control channel
pbxCLI>-- Transmitting RFC2833 on payload 101
pbxCLI>-- Received RELEASE COMPLETE message…
– ClearCall: Request to clear call with token ip $10.199.198.1:33954/12126, cause EndedByQ931Cause
1:47:30.363 H225 Answer:88bac48 h323.cxx(1647) H225
Sending release complete PDU: callRef=12126
– Sending RELEASE COMPLETE
1:47:30.363 H225 Answer:88bac48 h323.cxx(3557) H245
Write PDU fail: no control channel.
– ClearCall: Request to clear call with token ip $10.199.198.1:33954/12126, cause EndedByTransportFail
1:47:30.363 H225 Answer:88bac48 h323.cxx(1866) H225
Signal channel closed.
1:47:30.421 H323 Cleaner h323.cxx(1757) H323
Connection ip$10.199.198.1:33954/12126 terminated.
– Call with 84956737715 [10.199.198.1] completed (EndedByQ931Cause) pbx*CLI>== H.323 Connection deleted.

Any idea?

2

It’s very important! Anyone know how fix it?

3

Enable higher level of debugging (h.323 trace 5), and you will be able to see exact packets you get from VoIP provider. I think they provide support of G.729/G.723 codecs only, while you accepts G.711a/u and GSM codecs.