Outbound calls error with "all circuits busy" or "congestion":
This is the default configuration of Asterisk regardless of the actual error generated (which is infuriating when you are trying to diagnose the real problem) unless PBX is updated to send back the real error rather than the changed error. This error most commonly occurs when the call is not authenticating properly, at which point check the above in the SIP trunk configuration (If Asterisk, swap username= for defaultuser= to see if this solves the issue. Just because a trunk is showing as registered does not mean it will authenticate correctly.

Outbound calls fail with SIP error 488 (Not Accepted Here) or I-SUP errors 58 (bearer capability not available) or 88 (incompatible destination):
Check the codecs allowed in the SIP trunk configuration above, VoiceHost only supports: alaw, ulaw, gsm
If a codec is defined in Asterisk that is not one of the above, or is offering a differing sample rate or interval rate (e.g. 8000/20i - 8000Hz at 20ms) cannot interwork with 16000/30i - 16000Hz at 30ms) the call will fail and the codecs in the SIP trunk configuration need to be aligned to use one of the above codecs.

Inbound calls fail with SIP error 408 (Request Timeout):
Check the inbound number is mapped in the system correctly, if necessary the SIP trunk on the portal can be configured to strip the plus, e.g. if Asterisk is configured to use plus somewhere else. Check the trunk is registered. Ascertain how long the 408 error took to come back, if it was immediate the trunk is usually unregistered, if it took a few seconds the number is usually not mapped in correctly.

Calls fail with SIP error 503, I-SUP errors 34 or 38:
If our platform replies back with 503 it usually means the gateway trying to process the call can't due to "issues", or the customer has hit their Calls-Per-Second (CPS) limit and is sending too many calls at once. Sometimes the error is passed back from IP Exchange through VoiceHost to the customer's system, at which point the call will usually hunt to another route to try and place the call.

Cause code (ISUP) SIP Equivalent Definition
1 404 Not Found Unallocated (unassigned) number
2 404 Not found no route to network
3 404 Not found no route to destination
16 BYE or CANCEL (*) normal call clearing
17 486 Busy here user busy
18 408 Request Timeout no user responding
19 480 Temporarily unavailable no answer from the user
20 480 Temporarily unavailable subscriber absent
21 403 Forbidden (+) call rejected
22 410 Gone number changed (w/o diagnostic)
22 301 Moved Permanently number changed (w/ diagnostic)
23 410 Gone redirection to new destination
26 404 Not Found (=) non-selected user clearing
27 502 Bad Gateway destination out of order
28 484 Address incomplete address incomplete
29 501 Not implemented facility rejected
31 480 Temporarily unavailable normal unspecified
34 503 Service unavailable no circuit available
38 503 Service unavailable network out of order
41 503 Service unavailable temporary failure
42 503 Service unavailable switching equipment congestion
47 503 Service unavailable resource unavailable
55 403 Forbidden incoming calls barred within CUG
57 403 Forbidden bearer capability not authorized
58 503 Service unavailable bearer capability not presently
65 488 Not Acceptable Here bearer capability not implemented
70 488 Not Acceptable Here only restricted digital avail
79 501 Not implemented service or option not implemented
87 403 Forbidden user not member of CUG
88 503 Service unavailable incompatible destination
102 504 Gateway timeout recovery of timer expiry
111 500 Server internal error protocol error
127 500 Server internal error interworking unspecified