SIP->h323 in a AS5850 - Not able to send h323 calls coming from a SIP Phone

[olazcano]

Dear All!

I have an AS5850 configured as a SIP Gateway and as a H323 Gateway. I'm planning to use this equipment as an interconnection point between PSTN,SIP and H323.

I already have a functional H323 Network with ISDN trunks to the pstn and it is working fine. I added SIP configuration to the AS5850 in order to be able to route calls out to the PSTN or H323 remote ends coming from a SIP Phone registered with a third-party SIP Proxy.

When the calls coming from the SIP Phone goes to a PSTN destination the calls completes properly, but i am having problems trying to send calls coming from the SIP phone to a remote h323 gateway(also cisco)

Attached is my configuration and the error i'm getting in my cdr. It seems that the "ext" number of the phone is being used as destination string in the last call leg, but i'm not sure.

Please Help!

!

dial-peer voice 100 pots

application session

destination-pattern 5T

port 2/6:D

forward-digits all

!

dial-peer voice 102 pots

application session

destination-pattern 044T

port 2/6:D

forward-digits all

!

dial-peer voice 103 voip

application session

incoming called-number 001T

destination-pattern 001T

session protocol sipv2

session target ipv4:20X.21X.17X.1X

tech-prefix 10511

!

sip-ua

sip-server ipv4:20X.6X.14X.18X

!

CDR ERROR:

.Mar 24 2004 18:31:42.620 GMT: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 2, ConnectionId 9F74CE17 7D2A11D8 82A09B41 D2C3D418, SetupTime .18:31:42.470 GMT Wed Mar 24 2004, ***PeerAddress 2006***, PeerSubAddress , DisconnectCause 3 , DisconnectText no route to destination (3), ConnectTime .18:31:42.620 GMT Wed Mar 24 2004, DisconnectTime .18:31:42.620 GMT Wed Mar 24 2004, CallOrigin 2, ChargedUnits 0, InfoType 2, TransmitPackets 0, TransmitBytes 0, ReceivePackets 0, ReceiveBytes 0

Thanks.

Attached you can find the debug ccsip messages output.

[ngss]

You need SIP-H323 Protocol Converter to communicate between the two world. AS53xx alone won't work.

A quick and dirty way is using PSTN switch act the middlman, ie SIP<->AS5300<->PSTN Switch(E1-ISPN/PRI back to back)<->AS5300<->H.323, but it will be a waste of resources

Thanks

SSng

[olazcano]

Thank yoy very much for your answer. If it is the situation, then how the SIP to VOIP Transfer calls works?

I found a lot of documentation about this feature, and i though this is the required feature to perform SIP to H323 Termination.

Please, let me know your comments about this feature, and any other idea to do this.

Right now, the situation you describe its happening.

In the same network i have an IP PBX connected through an ISDN Link to the AS5850. This IP PBX sends calls for H323 termination on a remote end and calls coming into the AS5850 from other ISDN Link terminate whether in a IP PBX Extension or in a H323 remote end (like access numbers).

If i make a call from the SIP Phone dialing a number that "should" be routed immediatly through the VoIP dial-peer containing the ip address of the remote end, instead of doing this, it goes first to the IP PBX, then the IP PBX takes the call and forwards the call through a voip dial-peer that also contains the ip addr of the remote as if the IP PBX generates the call and then the call completes from the sip phone to the h323 remote destination.

This is first of all, because the dial-peers already configured for the H323 current network matches the digits dialed from the SIP Phone, but before i got your answer i though it was an error on my dialplan design. Now, i wonder if it is first due to the match of the digits and then because this is the only way to go sip->h323??

Please let me know any idea or suggestion about this situation.

Thanks!

[ngss · Accepted solution]

There are 2 solutions here.

1. Use of SIP/H.323 Signalling Gateway as the protocol convertor. Search google will yield heaps of hits on this subject. Product available both commercial and open source, trial, etc. Using this method means that the SIP End Point will communicate with H.323 End Point without going out the PSTN. I believe this is what you want to achieve in the long term. You are trying the AS5xxx as the protocol convertor for you, which it will not work. A call flow will be something like SIP IP Phone->SIP Server->SIP-to-H.323 Gateway->H.323 Gatekeeper->H.323 End Point. Of couse there is a SIP server that do the protocol convertor in the same box but the functionality is the still the same. Performance and concurrent call setup differ from products to products. Going for this solution would require you to find such products and test it on the your network.

2. If you do not wish to try on Soluton 1, this solution is a workaround way by not getting device but using the existing equipment that you have right now. Onto whether this good long term solution for depends on what you want to achieve both in term of commercially and technically. A call flow will be SIP End Point->SIP Server->Voice Gateway (AS5xxx)->PSTN Switch(ISDN/PRI)->Voice Gateway->H.323 Gatekeeper>H.323 End Point. The key is the Voice session must traverse the ISDN link. In other words your dial pattern must be setup is such as way that will go out thru the dial peer pots to pstn switch then come back to another dial-peer pots. I am not saying this is the most efficient way of doing it, I merely suggesting a workable way to achieve your desired goal without soluton 1.

Hopes you get better understanding now.

Thanks

SSng

[olazcano]

Thank you very much for your attention. Now i understand whats going on, and i will go over the best solution.

Thanks!

Oscar.