provided by Star2Billing S.L.

Can anyone help me understand what is causing this "Loop Detected" issue? I am trying to route a DID through A2B and on to an existing a2b customer.

My pbx and a2b are on the same box. Is that an issue that requires special handling?

I sincerely appreciate it. Please let me know what other information would be required to track this down.

CLI dump

-- Executing [12033070000@from-trunk:1] Set("SIP/flowroute-0000003f", "__FROM_DID=12033070000") in new stack
-- Executing [12033070000@from-trunk:2] ExecIf("SIP/flowroute-0000003f", "1 |Set|CALLERID(name)=+12158170000") in new stack
-- Executing [12033070000@from-trunk:3] Set("SIP/flowroute-0000003f", "__CALLINGPRES_SV=allowed_not_screened") in new stack
-- Executing [12033070000@from-trunk:4] SetCallerPres("SIP/flowroute-0000003f", "allowed_not_screened") in new stack
-- Executing [12033070000@from-trunk:5] Goto("SIP/flowroute-0000003f", "custom-a2billing-did|12033070000|1") in new stack
-- Goto (custom-a2billing-did,12033070000,1)
-- Executing [12033070000@custom-a2billing-did:1] DeadAGI("SIP/flowroute-0000003f", "a2billing.php|1|did") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
-- AGI Script Executing Application: (DIAL) Options: (SIP/[email protected]|60|HiL(3600000:61000:30000))
-- Limit Data for this call:
> timelimit = 3600000
> play_warning = 61000
> play_to_caller = yes
> play_to_callee = no
> warning_freq = 30000
> start_sound =
> warning_sound = timeleft
> end_sound =
-- Called [email protected]
-- Got SIP response 482 "Loop Detected" back from 76.99.139.000
-- Now forwarding SIP/flowroute-0000003f to 'Local/2058550718@from-sip-external' (thanks to SIP/76.99.139.000-00000040)
-- Forwarding SIP/flowroute-0000003f to 'Local/2058550718@from-sip-external' prevented.
== Everyone is busy/congested at this time (1:1/0/0)
-- AGI Script Executing Application: (Busy) Options: (1)
-- Playing 'prepaid-isbusy' (escape_digits=#) (sample_offset 0)

Hi

[email protected] is not your own IP is it - because that would cause these symptoms.

Joe

This was solved by removing the @76.99.139.000 and keeping the SIP/account construct.

Suggestion for the help text in the destinations config: Enter the phone number you wish to call, or the VoIP client to reach. (ie: 347894999 or SIP/[email protected]). If the call is VoIP, the VoIP_Call must be set to yes. If the user is on the same system: SIP/<username>

hi robert

i have the same issue did you manage to fix this

thankx

py