GOautodial Open Source Omni-channel Contact Center Suite

Intel(R) Xeon(R) CPU L5520 @ 2.27GHz running goautodial 3.3, Asterisk 1.8, Vicidial 2.7RC1

HI there can anyone tell me how I have mananged to mess up my calls? - I recently followed all the steps installed Asterisk 1.8/Vicidial 2.7RC1 and updated from 3.0 to 3.3 and now there is no connection between my calls, I get the usual welcome message at the agent login screen and when I try to dial next for a live call the status stays on waiting for dial...,

This what i get on CLI
Verbosity is at least 21
[Mar 2 16:50:21] NOTICE⁸⁰⁸¹: res_rtp_asterisk.c:2361 ast_rtp_read: Unknown RTP codec 126 received from '109.144.202.26:29344'
[Mar 2 16:50:31] NOTICE⁸⁰⁸¹: res_rtp_asterisk.c:2361 ast_rtp_read: Unknown RTP codec 126 received from '109.144.202.26:29344'
[Mar 2 16:50:38] Manager 'sendcron' logged on from 127.0.0.1
[Mar 2 16:50:38] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-00000009;2", "8600051,F") in new stack
[Mar 2 16:50:38] > Channel Local/8600051@default-00000009;1 was answered.
[Mar 2 16:50:38] -- Executing [4441246861496@default:1] AGI("Local/8600051@default-00000009;1", "agi://127.0.0.1:4577/call_log") in new stack
[Mar 2 16:50:38] -- <Local/8600051@default-00000009;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Mar 2 16:50:38] -- Executing [4441246861496@default:2] Dial("Local/8600051@default-00000009;1", "SIP/441246861496@Gradwell,,tTo") in new stack
[Mar 2 16:50:38] Using SIP RTP CoS mark 5
[Mar 2 16:50:38] WARNING⁸³⁶³: chan_sip.c:6033 sip_call: No audio format found to offer. Cancelling call to 441246861496
[Mar 2 16:50:38] -- Couldn't call SIP/441246861496@Gradwell
[Mar 2 16:50:38] Everyone is busy/congested at this time (0:0/0/0)
[Mar 2 16:50:38] -- Executing [4441246861496@default:3] Hangup("Local/8600051@default-00000009;1", "") in new stack
[Mar 2 16:50:38] Spawn extension (default, 4441246861496, 3) exited non-zero on 'Local/8600051@default-00000009;1'
[Mar 2 16:50:38] -- Executing [h@default:1] AGI in new stack
[Mar 2 16:50:38] -- <Local/8600051@default-00000009;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----CHANUNAVAIL---------- completed, returning 0
[Mar 2 16:50:38] Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-00000009;2'
[Mar 2 16:50:38] -- Executing [h@default:1] AGI("Local/8600051@default-00000009;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Mar 2 16:50:38] -- &lt;Local/8600051@default-00000009;2&gt;AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Mar 2 16:50:39] Manager 'sendcron' logged on from 127.0.0.1
[Mar 2 16:50:39] -- Executing [58600051@default:1] MeetMe("Local/58600051@default-0000000a;2", "8600051,Fmq") in new stack
[Mar 2 16:50:39] > Channel Local/58600051@default-0000000a;1 was answered.
[Mar 2 16:50:39] -- Executing [8309@default:1] Answer("Local/58600051@default-0000000a;1", "") in new stack
[Mar 2 16:50:39] -- Executing [8309@default:2] Monitor("Local/58600051@default-0000000a;1", "wav,20140302-215037_1246861496_91306432_agent003") in new stack
[Mar 2 16:50:39] -- Executing [8309@default:3] Wait("Local/58600051@default-0000000a;1", "3600") in new stack
[Mar 2 16:50:40] Manager 'sendcron' logged off from 127.0.0.1
[Mar 2 16:50:41] Manager 'sendcron' logged off from 127.0.0.1
[Mar 2 16:50:41] NOTICE⁸⁰⁸¹: res_rtp_asterisk.c:2361 ast_rtp_read: Unknown RTP codec 126 received from '109.144.202.26:29344'
[Mar 2 16:50:51] NOTICE⁸⁰⁸¹: res_rtp_asterisk.c:2361 ast_rtp_read: Unknown RTP codec 126 received from '109.144.202.26:29344'

A codec problem maybe?
would love some help in figuuring this out as I need to be back online asap