I use an IP Trade system connected to CUCM version 6.1.2. For this, an IP Trade turret is an advanced SIP phone configured on the CUCM. At CUCM I have configured a H.323 gateway with PSTN access. Also in this setup I have an IP Trade TPO server that serves as a RTP mixer for join functions on the IP Trade turrets. the problem I have is this:
When a call is active between PSTN phone (through H.323 gateway) and SIP turret1, and SIP turret2 wants to join this call, the design within the IP Trade system moves the active call from PSTN to turret1 to the TPO server. this is the flow:
Call is active between PSTN and turret1. turret2 wants to join this call.
Turret2 sends a SIP INVITE to the TPO server.
TPO server sends a SIP INVITE to turret1.
Turret1 sends a SIP INVITE to CUCM.
all SIP INVITE messages indicate the new destination for the call --> TPO server.TPO server will mix all the streams making a 'conference'
the problem is this:
the CUCM receives the SIP INVITE from turret1 and has to update the PSTN caller (H.323 call). this is not happening! CUCM does not send any updates to H.323 gateway for change of destination. this is the error in the tracelog of CUCM:
000010464| 2009/12/08 13:55:23.477| 001| SdlError | H245SessionEstablishedFailure | NA | H245Interface(1,100,156,7) | H245SessionManager(1,100,25,7) | (1,100,25,7).1-(*:192.168.20.251) | Destination process does not exist
Is this a process that is not supported by CUCM? I read alot about H.323 not supporting redirection in an active call..
anybody any info on this issue? help would be appriciated, although also a TAC case has been opened for this.
Re: SIP re-invite translated to H.323 on CUCM fails
thanks, that did actually solve the problem.
Still I think this is an issue of H.323 version 4 not being able to handle re-directed calls. follow this link to the 'release notes' of H.323 version 6, and it states that this might be solved in this version. http://www.packetizer.com/ipmc/h323/whatsnew_v6.html
H.460.15 - Call Signalling Transport Channel Suspension and Redirection
This Recommendation allows an intermediary device, for example, to suspend the H.225.0 call signal channel and have it re-routed to another device or, more commonly, point-to-point between the calling and called devices. This allows, for example, a call to be established through a Gatekeeper that routes call signaling and, once the call is stable, to have the signaling burden moved off of the Gatekeeper to the endpoints.
the result of the MTP solution is that all calls now will be routed to the CUCM. CUCM is now handling all RTP traffic for the whole office. this will have an impact on CUCM and probably MTP is also limited in capacity if I am right.
thanks for this information and quick and dirty workaround..
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