Calls thru Nortel BCM

ok got a fresh install of Slack 12.1 AST + Vicidial. We (Vici guys + I)
are trying to place the calls from the Asterisk to the Nortel VIA TI cards. Linux box has a Sangoma 101D and the nortel is using the Nortel T1 interface card. We couldn't get the Asterisk to place a call through the Nortel using PRI_CPE and PRINET modes. The call would fail right away with reason code 21 right away. However, the Asterisk was able to get a trunk on the Nortel using EM_WINK. It wasn't able to pass any digits to the Nortel from what I can see in the BCM Manager(Tool used to watch live call activity). Im open to any suggestions ..
One alternative here is to just directly connect my other PRI to the Asterisk and just route live callers to the BCM via SIP. But thats teh EZ way out!
Here is a brief clip from the logs:
Oct 5 17:25:19 VERBOSE[9941] logger.c: -- Executing AGI("SIP/cc100-08284538", "agi://127.0.0.1:4577/call_log") in new stack
Oct 5 17:25:19 VERBOSE[9941] logger.c: -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
Oct 5 17:25:19 VERBOSE[9941] logger.c: -- Executing Dial("SIP/cc100-08284538", "Zap/g1/93058888202||To") in new stack
Oct 5 17:25:19 VERBOSE[9941] logger.c: -- Requested transfer capability: 0x00 - SPEECH
Oct 5 17:25:19 DEBUG[9174] channel.c: Avoiding initial deadlock for 'Zap/1-1'
Oct 5 17:25:19 VERBOSE[9941] logger.c: -- Called g1/93058888202
Oct 5 17:25:20 DEBUG[9216] manager.c: Manager received command 'Command'
Oct 5 17:25:20 VERBOSE[9180] logger.c: -- Channel 0/1, span 1 got hangup, cause 21
Oct 5 17:25:20 VERBOSE[9941] logger.c: -- Zap/1-1 is circuit-busy
Oct 5 17:25:20 DEBUG[9941] chan_zap.c: Set option AUDIO MODE, value: ON(1) on Zap/1-1
Oct 5 17:25:20 DEBUG[9941] chan_zap.c: Hangup: channel: 1 index = 0, normal = 14, callwait = -1, thirdcall = -1
Oct 5 17:25:20 DEBUG[9941] chan_zap.c: Already hungup... Calling hangup once, and clearing call
Oct 5 17:25:20 DEBUG[9941] chan_zap.c: disabled echo cancellation on channel 1
Oct 5 17:25:20 DEBUG[9941] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/1-1
Oct 5 17:25:20 DEBUG[9941] chan_zap.c: Updated conferencing on 1, with 0 conference users
Oct 5 17:25:20 DEBUG[9941] chan_zap.c: Set option AUDIO MODE, value: OFF(0) on Zap/1-1
Oct 5 17:25:20 DEBUG[9941] chan_zap.c: disabled echo cancellation on channel 1
Oct 5 17:25:20 VERBOSE[9941] logger.c: -- Hungup 'Zap/1-1'
Oct 5 17:25:20 VERBOSE[9941] logger.c: == Everyone is busy/congested at this time (1:0/1/0)
Oct 5 17:25:20 DEBUG[9941] app_dial.c: Exiting with DIALSTATUS=CONGESTION.
Oct 5 17:25:20 VERBOSE[9941] logger.c: -- Executing Hangup("SIP/cc100-08284538", "") in new stack
Oct 5 17:25:20 VERBOSE[9941] logger.c: == Spawn extension (default, 913058888202, 3) exited non-zero on 'SIP/cc100-08284538'
Oct 5 17:25:20 VERBOSE[9941] logger.c: -- Executing DeadAGI("SIP/cc100-08284538", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----CONGESTION----------") in new stack
Oct 5 17:25:20 VERBOSE[9941] logger.c: -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
Oct 5 17:25:20 DEBUG[9941] chan_sip.c: update_call_counter(cc100) - decrement call limit counter


One alternative here is to just directly connect my other PRI to the Asterisk and just route live callers to the BCM via SIP. But thats teh EZ way out!
Here is a brief clip from the logs:
Oct 5 17:25:19 VERBOSE[9941] logger.c: -- Executing AGI("SIP/cc100-08284538", "agi://127.0.0.1:4577/call_log") in new stack
Oct 5 17:25:19 VERBOSE[9941] logger.c: -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
Oct 5 17:25:19 VERBOSE[9941] logger.c: -- Executing Dial("SIP/cc100-08284538", "Zap/g1/93058888202||To") in new stack
Oct 5 17:25:19 VERBOSE[9941] logger.c: -- Requested transfer capability: 0x00 - SPEECH
Oct 5 17:25:19 DEBUG[9174] channel.c: Avoiding initial deadlock for 'Zap/1-1'
Oct 5 17:25:19 VERBOSE[9941] logger.c: -- Called g1/93058888202
Oct 5 17:25:20 DEBUG[9216] manager.c: Manager received command 'Command'
Oct 5 17:25:20 VERBOSE[9180] logger.c: -- Channel 0/1, span 1 got hangup, cause 21
Oct 5 17:25:20 VERBOSE[9941] logger.c: -- Zap/1-1 is circuit-busy
Oct 5 17:25:20 DEBUG[9941] chan_zap.c: Set option AUDIO MODE, value: ON(1) on Zap/1-1
Oct 5 17:25:20 DEBUG[9941] chan_zap.c: Hangup: channel: 1 index = 0, normal = 14, callwait = -1, thirdcall = -1
Oct 5 17:25:20 DEBUG[9941] chan_zap.c: Already hungup... Calling hangup once, and clearing call
Oct 5 17:25:20 DEBUG[9941] chan_zap.c: disabled echo cancellation on channel 1
Oct 5 17:25:20 DEBUG[9941] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/1-1
Oct 5 17:25:20 DEBUG[9941] chan_zap.c: Updated conferencing on 1, with 0 conference users
Oct 5 17:25:20 DEBUG[9941] chan_zap.c: Set option AUDIO MODE, value: OFF(0) on Zap/1-1
Oct 5 17:25:20 DEBUG[9941] chan_zap.c: disabled echo cancellation on channel 1
Oct 5 17:25:20 VERBOSE[9941] logger.c: -- Hungup 'Zap/1-1'
Oct 5 17:25:20 VERBOSE[9941] logger.c: == Everyone is busy/congested at this time (1:0/1/0)
Oct 5 17:25:20 DEBUG[9941] app_dial.c: Exiting with DIALSTATUS=CONGESTION.
Oct 5 17:25:20 VERBOSE[9941] logger.c: -- Executing Hangup("SIP/cc100-08284538", "") in new stack
Oct 5 17:25:20 VERBOSE[9941] logger.c: == Spawn extension (default, 913058888202, 3) exited non-zero on 'SIP/cc100-08284538'
Oct 5 17:25:20 VERBOSE[9941] logger.c: -- Executing DeadAGI("SIP/cc100-08284538", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----CONGESTION----------") in new stack
Oct 5 17:25:20 VERBOSE[9941] logger.c: -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
Oct 5 17:25:20 DEBUG[9941] chan_sip.c: update_call_counter(cc100) - decrement call limit counter