I have a CUCM 8.6.1 server at a client working with multiple Avaya systems. The Avaya systems are setup as h.323 gateways in CUCM. Phones can 4-digit dial between systems with no problem.
If a user sets their phone on the Cisco side to call forward all to their cell phone (outside number) this works without a problem for all calls.
If a user sets their phone on the Cisco side to call forward to their cell phone for no answer, a call from the Avaya side will first hear ringback as the Cisco phone is ringing. When the call gets forwarded to the cell phone, the cell phone rings and the caller hears ringback then, but when the cell phone is answered the Avaya users gets a fast busy and the call is dropped.
This sounds like a codec issue, what is the region/DP setting on the h.323 GW pointing to Avaya trunk and region/DP on the egress GW to PSTN? What protocol are you using for the PSTN GW? What type of circuit are you sending the call out on?
In CUCM, the GW region for the Avaya trunk is set to g.729 when calling the Cisco region, region is DC, DP is DC.
The PSTN gateway for forwarded calls is h.323. That gateway is set to g.729 between the Avaya region.
The circuit is a T1 PRI.
I've attached the debug.
I changed the numbers to protect the innocent.
The called number is 1111111111.
If I had to guess, this is the useful part:
Nov 21 17:01:34.230: ISDN Se0/0/0:23 Q931: TX -> CONNECT_ACK pd = 8 callref = 0x010C
Nov 21 17:01:34.814: ISDN Se0/0/0:23 Q931: TX -> DISCONNECT pd = 8 callref = 0x010C
Cause i = 0x80AF - Resource unavailable, unspecified
Nov 21 17:01:34.886: ISDN Se0/0/0:23 Q931: RX <- RELEASE pd = 8 callref = 0x810C
Nov 21 17:01:34.886: ISDN Se0/0/0:23 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x010C Nov 21 17:01:34.230: ISDN Se0/0/0:23 Q931: TX -> CONNECT_ACK pd = 8 callref = 0x010C
What resource is it looking for? I have things set to require a MTP and that MTP is configured on the same gateway and is registered to CUCM.
The resource it refers to is a B-channel on the PRI, this is either a DSP issue or a codec issue, can you provide 2 more items:
debug voice ccapi inout
I'll grab that info for you shortly. I did see this on the gateway though:
dspfarm profile 2 mtp
maximum sessions software 100
associate application SCCP
If everything is setup for g729 and th MTP can only use g711...stands to reason that's my problem?
Is there any way to view active MTP calls so I can shut down the MTP to add the other codec? Or am I barking up the wrong tree?
If you are invoking that MTP then unless you are also invoking Xcoder then yes that would be a problem. What are you needing MTPs for? This type of integration should not require MTP usage.
The MTP was set to require. I'm not sure if that is necessary but since it was set it seemed like a possible cause.
When I tried to change it it said that the other codc, g729, was set in the profile and I couldn't add it to the MTP config.
I've attached the debug and GW config.
Thanks for helping out with this!
I have tried that. When it is not checked I get the same result with the exception of not hearing any ringback on the calling phone when the call gets forwarded to the cell phone.
So, caller calls and hears 10 seconds of ringing, then the call gets forwarded to the cell and the caller hears nothing when the cell phone is ringing. When the cell phone is answered the caller still hears nothign until the call is dropped.
With the MTP checked the caller hears ringback througout and the call is dropped immediately on answer with the caller getting a fast busy.
Well, the issue is definitely codec related, is your xcoder defined in CUCM to be available by the Avaya H323 and PSTN GW? Since your MTP is defined as G711 you will need transcoders to transcode the codecs, perhaps it is not properly allocated to MRG/MRGL on these devices, debug sccp may be of help here.
Okay, the GW that is making the outbound didn't have a MRGL defined. (oops) So fixed that and now the call doesn't hangup immediately and the caller doesn't get the fast busy. There's about five seconds of silence on both ends and then the call drops.
I've attached the output from debug sccp messages. Should I try another debug?
I spoke with TAC on this and they did a call trace with RTMT. The call setup is working properly including codec negotiation. Once the call is answered the Avaya gateway is sending a (41) Temporary Failure error.
So TAC is getting me the details on this but since this was working before and only happens on FNA calls could there be something else?
I googled the error and is seems to be a generic IDSN message which basically says "try again in a few."