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H323 Trunk from Panasonic TDE100 to cisco router 2801

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3075062

Technical User
Mar 10, 2011
7
RS
I am trying to make H323 trunk between Panasonic TDE100 and Cisco router 2801. I also have PBX Alcatel oxo attached to cisco BRI card.
The idea is to make call from panasonic subscriber (over h323 voip trunk) across cisco router to Alcatel subscriber. I configured all three systems but there is a problem...
When I start call from panasonic subscriber, on the other side subscriber on alcatel rings and when he lift the handset the call is established but there is no voice (in both direction) and after a while the call is broken.

Can anybody give me a hint or any advice where to look for solutions...
 
yes this is VPN...
I also have other destinations coming to the cisco and they call is establishig well. But from destionation of Panasonic only signalisation came through... no voice...
 
I perform tests on cisco and here is examples of "show call active voice brief" tests. First is test from succesfull call from other destination and second is test from unsuccesfull call from Paanasonic..


First (succesfull call)

120A : 117 272612740ms.1 +14780 pid:1 Answer 6294 active
dur 00:00:03 tx:156/25890 rx:120/18722
Tele 0/2/1 (117) [0/2/1.1] tx:3690/2350/0ms g711ulaw noise:-65 acom:26 i/0:-32/-53 dBm

120A : 118 272612760ms.1 +14750 pid:13 Originate 35924889417 active
dur 00:00:03 tx:120/18722 rx:156/24642
IP 10.251.208.129:19172 SRTP: off rtt:0ms pl:2135/0ms lost:0/1/0 delay:60/60/65ms g711ulaw
media inactive detected:n media contrl rcvd:n/a timestamp:n/a


Second (unsuccesfull call) - Panasonic destination

303 : 121 272779300ms.1 +11180 pid:5000 Answer 0112282904 active
dur 00:00:03 tx:153/24480 rx:0/0
IP 10.50.244.70:12168 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:65/65/65ms g711alaw
media inactive detected:n media contrl rcvd:n/a timestamp:n/a

303 : 122 272779370ms.1 +11110 pid:2 Originate *294 active
dur 00:00:03 tx:0/0 rx:154/24640
Tele 0/2/0 (122) [0/2/0.1] tx:3075/3075/0ms g711alaw noise:-54 acom:6 i/0:-33/-65 dBm




 
are you using a V-ipgw16 or a v-sipgw16 on the panasonic and what are the settings used on that.

 
I am using V-ipgw16 on Panasonic (in Gateway configuration, not Gatekeeper). I think that is well configured because I already use this Panasonic for H323 trunking with another Panasonic PBX on another location.
 
The panasonic gateway card is preset to work to another panasonic once you set in the dial plan and ip address.

looking at your log, the systems are not talking to each other, you have rx 0/0 and on the second log tx 0/0

IP 10.50.244.70:12168 is this to the Panasonic or from the panasonic, the h 323 dynamic port number is 10000

Have you change the codex, the default on the ipgw16 is G729a and on you log it shows g711 alaw
 
Yes... the systems are not talking to each othther (rx 0/0...) but why?.

Ip address 10.50.244.70 is address of Panasonic VoIP-DSP.

I tried different codex and when they are mismatched nothing passed through, but when the same codec is in Panasonic and Cisco then only signalisation passed (the phones ring and when someone picks up the handset hears nothing...silence).
 
What is the ip address of the system. You should be using that and not the dsp ip address

 
hmmm... I am using ip address of the system for panasonic, which is 10.50.244.69. That what you saw above (10.50.244.70) is what Cisco write in his tests...(show call active voice brief)

Below is settings for cisco dial-peers which match in unsuccesseful call

dial-peer voice 2 pots
trunkgroup PBX
description Outgoing to PBX
translation-profile outgoing to_PBX
destination-pattern 8*[2-3]..
direct-inward-dial
forward-digits 3

dial-peer voice 5000 voip
incoming called-number [0-9]T
dtmf-relay h245-signal h245-alphanumeric
fax rate 9600
fax nsf 000000
fax protocol t38 ls-redundancy 5 hs-redundancy 2 fallback none
no vad
 
I don't know much about the cisco side of things, so what is the reference to pot in the first section,should it not all be VOIP

 
Pots = Plain Old Telephony System.

In this case cisco has a VoIP-BRI card inside which is connected to Alcatel PBX. So when the call come from voIP "world" into cisco, it has to know which way to go, and then cisco look for matching dial peer which in this case is "dial-peer voice 2 pots" and transfer the call to the VoIP-BRI card and to the alcatel PBX...
 
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