ARS with SIP Trunk Overflow

[cavagnaro]

Humm...and which message should it return then? Do you know if on AudioCodes you can change the messages??

[root]

Humm...and which message should it return then? Do you know if on AudioCodes you can change the messages??

e.g. 5xx, you can check on OXE which SIP response is mapped to a call handling cause which enable an overflow there is a TC available which describe on which cause an overflow is performed

[MrAnMo]

I guess TC0260?

[root]

I guess TC0260?

yes

[tot3nkopf]

I have tested overflow scenario on my lab h1.301.38c (R 9.0) but only for out of service of external gw:

For ABCF SIP TG:
- if ext gateway 1 is down --> no ARS overflow NOK
- if external gateway not out of service logically (registration timer= 0) but proxy not there --> overflow to 2nd gw OK

For ISDN SIP TG:
- if ext gateway 1 is down --> ARS overflow works well --->OK
- if external gateway not out of service logically (registration timer = 0) but proxy not there --> overflow OK

I will try to test also with busy causes and get back on you Cav, but you can try with ISDN SIP TG

[tot3nkopf]

To complete the solution (tested in R.10, available from R 9.1): as root suggested mapping in SIP to CH ---> 480 Temporarily Unavailable mapped as "No circuit" instead of "No user responding" is the solution.
For me when testing:
OXE 1st ext gw<---->Asterisk
2nd ext gw<--->SJPhone
Asterisk replied with 480 when exceeded TG channel limit (which I have set in Asterisk config).

---> ARS overflow from one ext gw to another, or to other route OK

[cavagnaro]

So many thanks Tot! You rule! Will do my job

[tot3nkopf]

So many thanks Tot! You rule! Will do my job

Don't worry Cav, I was interested in this as I have an ongoing project comprising this configuration

[MrAnMo]

Cool, good to know this. Thanks for sorting this out Tot.

[tot3nkopf]

Cool, good to know this. Thanks for sorting this out Tot.

root was the one who pointed the direction