Sip to h323 bridging

below is my config:
pstn cisco as5350 --sip–> asterisk --h323–> as5350 --> pstn

after the call is connected, the calling side could not hear voice or dtmf tone. the called side could hear anything.

codec is g729

Pls advise

— below is the console msg —
*CLI> – Executing [21740000@default:1] Wait(“SIP/192.168.1.19-0882ac00”, “1”) in new stack
– Executing [21740000@default:2] Answer(“SIP/192.168.1.19-0882ac00”, “”) in new stack
– Executing [21740000@default:3] SetCallerPres(“SIP/192.168.1.19-0882ac00”, “unavailable”) in new stack
– Executing [21740000@default:4] Dial(“SIP/192.168.1.19-0882ac00”, “H323/4997270519@192.168.1.19|40|m”) in new stack
– Requested transfer capability: 0x00 - SPEECH
– Called 4927270519@192.168.1.19
– Started music on hold, class ‘default’, on SIP/192.168.1.19-0882ac00
– H323/192.168.1.19-1 is making progress passing it to SIP/192.168.1.19-0882ac00
– H323/192.168.1.19-1 is ringing
– H323/192.168.1.19-1 answered SIP/192.168.1.19-0882ac00
– Stopped music on hold on SIP/192.168.1.19-0882ac00
== Spawn extension (default, 21740000, 4) exited non-zero on ‘SIP/192.168.1.19-0882ac00’

This may be a stupid question, but why are you even using H.323?

It would be best to just either one of them all the way and prevent more possible points of failure.

To me the problem may sit with the protocol negotiations and also CoDec negotiation, if you have all IP traffic working on the one protocol you have less chances of problems happening.

Cheers,

David.

i agree to keep things simple but Reality does not allow things to be simple.

the left side is our corporate network. only sip is allowed. the right is a voip provider. only h323 is allowed.

Some postings said this is workable.

BTW, if i use sip-to-sip, a message appears “native bridging …” after call is connected. But no such message for sip-to-h323.

---- console log----
SIP/192.168.1.19-1 answered SIP/192.168.1.19-0882ac00
Stopped music on hold on SIP/192.168.1.19-0882ac00
native bridging …

[quote=“aclhkaclhk”]
BTW, if i use sip-to-sip, a message appears “native bridging …” after call is connected. But no such message for sip-to-h323.

---- console log----
SIP/192.168.1.19-1 answered SIP/192.168.1.19-0882ac00
Stopped music on hold on SIP/192.168.1.19-0882ac00
native bridging …[/quote]

Well when you look at it, SIP-SIP is a native bridge, where as SIP-H.323 is not, they are entirely different protocols therefore is not a native bridge.

Is it not possible to use another VoIP provider? I suspect if your are connecting to a AS-5350 then you are using a wholesale termination carrier, and they have not setup their AS-5350 to do SIP translation, but that is just an assumption.

By looking at the flowgram you gave, i see you want the Asterisk box to accept incoming SIP calls, but then Spit out H.323 to your carrier, in this event it could be done, however since i have not setup a system like that personally myself i have very little understanding of it, but at a guess i would say that you would have to look at your Dial Plan, and you may also need too do it via AGI, which would mean some programming involved.

But i am sure that someone with more knowledge on this matter on here could answer your questions better then i can.

Cheers,

David.

H.323 can perform native bridging with SIP channel because both uses RTP as media transport. Another question that current H.323 implementation just does not support for native bridging at all. Also, when your VoIP call goes from corporate network to public IP network, they probably uses different IP address space, so direct communication between SIP endpoint inside corporate network and H.323 gateway located in public network usually is not possible.

I had issues with H.323 and G729. Also I had an audio issue and it worked when I put in to h323.conf
faststart=yes
h245tunneling=yes

Playing with H.245/faststart options for each peer like a nightmare, but allows to have work-arounds for incompatibility between OpenH323 and peer’s implementations of H.323 standards.