Help with an asterisk to Nortel CS1000 H.323 link one way calls

[ACooperman]

Hi Guys,

I have created an endpoint on our nortel switch using H.323, this is linked to another companies asterisk switch.

I can call them fine and i can call any number they have, below are the TRAK results.

TN 031 0 00 04

KEY 0 SCR MARP ACTIVE VGW TN 031 0 00 04

ORIG TN 031 0 00 04 KEY 0 SCR MARP CUST 0 DN 1312 TYPE 3903

VGW TN 014 0 00 25

MEDIA ENDPOINT IP: 172.17.10.213 PORT: 5250

TERM VTN 099 0 00 29 VTRK IPTI RMBR 120 30 OUTGOING VOIP GW CALL

FAR-END H.323 SIGNALLING IP: 10.1.110.119

FAR-END MEDIA ENDPOINT IP: 10.1.50.5 PORT: 15012

MEDIA PROFILE: CODEC G.711 A-law PAYLOAD 20 ms VAD OFF

DIAL DN 4476

MAIN_PM ESTD

TALKSLOT ORIG 5 TERM 37

EES_DATA:

NONE

QUEU NONE

CALL ID 0 203





---- ISDN ISL CALL (TERM) ----

CALL REF # = 414

BEARER CAP = VOICE

HLC =

CALL STATE = 10 ACTIVE

CALLING NO = 1312 NUM_PLANRIVATE TON:ABBREVIATED ESN:CDP

CALLED NO = 1064476 NUM_PLANRIVATE TON:ABBREVIATED ESN:CDP

These calls work fine when the nortel is calling the asterisk, now when the asterisk system calls the nortel the phone rings but if i try to answer it disconnects immediately. Below is the TRAK from my phone when they try to call me when the phone is ringing (if i answer it hangs up).

TN 031 0 00 04

KEY 0 SCR MARP ACTIVATED



ORIG VTN 099 0 00 00 VTRK IPTI RMBR 120 1 INCOMING VOIP GW CALL

FAR-END H.323 SIGNALLING IP: 0.0.0.0

FAR-END MEDIA ENDPOINT IP: 0.0.0.0 PORT: 0

TERM DN 1312

DIAL DN 1312

MAIN_PM RING AUX_PM

TALKSLOT ORIG 49 TERM 17

EES_DATA:

NONE

QUEU CAD

CALL ID 0 238





---- ISDN ISL CALL (ORIG) ----

CALL REF # = 385

BEARER CAP = VOICE

HLC =

CALL STATE = 7 CALL RCV

CALLING NO = 4476 NUM_PLAN:UNKNOWN TON:UNKNOWN ESN:UNKNOWN

CALLED NO = 1071312 NUM_PLAN:UNKNOWN TON:UNKNOWN ESN:UNKNOWN




Any help would be hugely appreciated as i am under pressure from all angles to get this resolved.

Adam

 

[ACooperman]

Here is the endpoint configuration from the other end ....

gm-ho-voip-02*CLI> ooh323 show config
gm-ho-voip-02*CLI>
Objective Open H.323 Channel Driver's Config:
IPort: 10.1.50.5:1720
H.225 port range: 12030-12230
FastStart yes
Tunneling yes
CallerId Asterisk PBX
MediaWaitForConnect no
Gatekeeper: No Gatekeeper
H.323 LogFile: /var/log/asterisk/h323_log
Context: from-trunk
Capability: 0xe (gsm|ulaw|alaw)
DTMF Mode: rfc2833
AccountCode: ast_h323
AMA flags: Unknown

 

[ACooperman]

Hey Guys,

I have been looking further into this issue and it looks like the call information is ok but when meridian tries to open the channel for the audio when being called by the asterisk it is hanging up the call because it has no route for the audio, if you look at the IP's for the calls above it when i call them i can see all of the IP's and ports used but when the asterisk is calling me I do not get any TCP/IP port information. I think that the endpoints are talking to each other but the audio from the asterisk is being routed to the wrong place on the meridian.

This has completely puzzled me I am not sure if its a setting in the meridian or the asterisk, I am guessing that I either need to change the IP/Ports that the asterisk sends the audio on or change the IP/Ports that the meridian accepts audio on.

Configuring all of this is a little out of my comfort zone and I am really stuggling to get the final step of this working. My vendor *Cough* BT *Cough* have been absolutely useless on helping me with this. And at the other end they do not have a support contract or a vendor.

Please ask if you need any more information to give me nudges in the right direction.

Adam