IVR Breaks SIP trunks

strongbow242

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#1
Until two days ago, I had an IVR set up and working well on my Elastix system. Sometime during the night of 5/12 it broke. I did not make any changes to Elastix at all, NONE! I have time conditions set up so that calls can be routed to extensions or voicemail based on time of day. As of right now, If during normal working hours, I route inbound calls to an IVR, my provider Callcentric gets "SIP/2.0 404 Not Found." If I route calls to an extension, they are completed successfully. I have deleted the IVR and rebuilt it with a different name with no success. Also, if instead of an extension, I route calls to the Phonebook, calls break. Tried an IVR without the phonebook option, still broke. Any ideas?
 

jgutierrez

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#2
What are the destinatinos of your time condition settings? Also, howa re they configured?
 

telecomtechnician

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#3
Hi there, try the following

Create a ring group, include an extension that is not registered to the server, verify it is correctly configured and in the part it says destination if no answer, select the IVR that does not work. Call from any extension to the ring group and it should go to the IVR. If this is succesfull then the problem could be to the inbound route or the sip trunk.
If the IVR does not answer, that means that the message of the IVR is broken or the entire IVR is corrupted.

Waiting for your comments

David Medina
 

cyclopesz

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#4
Hi,
I had the same problem.
am getting error 130T for incoming calls using FREEDIGITS.
my outgoing calls is working okay.

"sip shows peers" shows Freedigits is OK.

any fix?

Thanks
 

jgutierrez

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#5
cyclopesz,
Paste the logs from the cli when it happens...
 

cyclopesz

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#6
jgutierez,
Thanks for the reply.
when dialing my freedigits #. am getting "your call cannot go thru. pls try again 130T"

it looks like my calls were drop before going to IVR

after few seconds am getting cli logs:

-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Executing [s@from-trunk:1] Set("SIP/67.55.159.156-08720f58", "__FROM_DID=s") in new stack
-- Executing [s@from-trunk:2] Gosub("SIP/67.55.159.156-08720f58", "app-blacklist-check|s|1") in new stack
-- Executing [s@app-blacklist-check:1] LookupBlacklist("SIP/67.55.159.156-08720f58", "") in new stack
-- Executing [s@app-blacklist-check:2] GotoIf("SIP/67.55.159.156-08720f58", "0?blacklisted") in new stack
-- Executing [s@app-blacklist-check:3] Return("SIP/67.55.159.156-08720f58", "") in new stack
-- Executing [s@from-trunk:3] ExecIf("SIP/67.55.159.156-08720f58", "0 |Set|CALLERID(name)=310xxxxxxx") in new stack
-- Executing [s@from-trunk:4] Set("SIP/67.55.159.156-08720f58", "__CALLINGPRES_SV=allowed_not_screened") in new stack
-- Executing [s@from-trunk:5] SetCallerPres("SIP/67.55.159.156-08720f58", "allowed_not_screened") in new stack
-- Executing [s@from-trunk:6] Goto("SIP/67.55.159.156-08720f58", "ivr-3|s|1") in new stack
-- Goto (ivr-3,s,1)
-- Executing [s@ivr-3:1] Set("SIP/67.55.159.156-08720f58", "MSG=connected") in new stack
-- Executing [s@ivr-3:2] Set("SIP/67.55.159.156-08720f58", "LOOPCOUNT=0") in new stack
-- Executing [s@ivr-3:3] Set("SIP/67.55.159.156-08720f58", "__DIR-CONTEXT=") in new stack
-- Executing [s@ivr-3:4] Set("SIP/67.55.159.156-08720f58", "_IVR_CONTEXT_ivr-3=") in new stack
-- Executing [s@ivr-3:5] Set("SIP/67.55.159.156-08720f58", "_IVR_CONTEXT=ivr-3") in new stack
-- Executing [s@ivr-3:6] GotoIf("SIP/67.55.159.156-08720f58", "0?begin") in new stack
-- Executing [s@ivr-3:7] Answer("SIP/67.55.159.156-08720f58", "") in new stack
-- Executing [s@ivr-3:8] Wait("SIP/67.55.159.156-08720f58", "1") in new stack
== Spawn extension (ivr-3, s, 8) exited non-zero on 'SIP/67.55.159.156-08720f58'
-- Executing [h@ivr-3:1] Hangup("SIP/67.55.159.156-08720f58", "") in new stack
== Spawn extension (ivr-3, h, 1) exited non-zero on 'SIP/67.55.159.156-08720f58'
-- Executing [s@from-trunk:1] Set("SIP/67.55.159.156-08720f58", "__FROM_DID=s") in new stack
-- Executing [s@from-trunk:2] Gosub("SIP/67.55.159.156-08720f58", "app-blacklist-check|s|1") in new stack
-- Executing [s@app-blacklist-check:1] LookupBlacklist("SIP/67.55.159.156-08720f58", "") in new stack
-- Executing [s@app-blacklist-check:2] GotoIf("SIP/67.55.159.156-08720f58", "0?blacklisted") in new stack
-- Executing [s@app-blacklist-check:3] Return("SIP/67.55.159.156-08720f58", "") in new stack
-- Executing [s@from-trunk:3] ExecIf("SIP/67.55.159.156-08720f58", "0 |Set|CALLERID(name)=310xxxxxx") in new stack
-- Executing [s@from-trunk:4] Set("SIP/67.55.159.156-08720f58", "__CALLINGPRES_SV=allowed_not_screened") in new stack
-- Executing [s@from-trunk:5] SetCallerPres("SIP/67.55.159.156-08720f58", "allowed_not_screened") in new stack
-- Executing [s@from-trunk:6] Goto("SIP/67.55.159.156-08720f58", "ivr-3|s|1") in new stack
-- Goto (ivr-3,s,1)
-- Executing [s@ivr-3:1] Set("SIP/67.55.159.156-08720f58", "MSG=connected") in new stack
-- Executing [s@ivr-3:2] Set("SIP/67.55.159.156-08720f58", "LOOPCOUNT=0") in new stack
-- Executing [s@ivr-3:3] Set("SIP/67.55.159.156-08720f58", "__DIR-CONTEXT=") in new stack
-- Executing [s@ivr-3:4] Set("SIP/67.55.159.156-08720f58", "_IVR_CONTEXT_ivr-3=") in new stack
-- Executing [s@ivr-3:5] Set("SIP/67.55.159.156-08720f58", "_IVR_CONTEXT=ivr-3") in new stack
-- Executing [s@ivr-3:6] GotoIf("SIP/67.55.159.156-08720f58", "0?begin") in new stack
-- Executing [s@ivr-3:7] Answer("SIP/67.55.159.156-08720f58", "") in new stack
-- Executing [s@ivr-3:8] Wait("SIP/67.55.159.156-08720f58", "1") in new stack
== Spawn extension (ivr-3, s, 8) exited non-zero on 'SIP/67.55.159.156-08720f58'
-- Executing [h@ivr-3:1] Hangup("SIP/67.55.159.156-08720f58", "") in new stack
== Spawn extension (ivr-3, h, 1) exited non-zero on 'SIP/67.55.159.156-08720f58'
 

jgutierrez

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#7
cyclopesz,
Are you setting on the IVR the announcement? I can see from the logs that it answers, waits one second, and then hangups, it doesn't play any audio. If you have set it up, is asterisk able to play it??

To test it, you may create a misc application, and choose your announcement
 

cyclopesz

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#8
after the call was hangup on the caller and hearing "your call cannot go thru. pls try again 130T", a few seconds you can see on the logs that the call is answered by asterisk.

the call was answered by asterisk after the actual call drops on the caller.

looks like asterisk is not answering the calls in timely manner.

here is my freedigits config:

FREE DIGITS
trunk name: free
canreinvite=no
context=from-pstn
defaultip=proxy.freedigits.net
dtmfmode=rfc2833
fromdomain=proxy.freedigits.net
fromuser=515xxxxxxx
host=freedigits.net
insecure=very
nat=yes
qualify=yes
secret=xxxxxx
type=friend

user context: digits
context=from-pstn
host=freedigits.net
insecure=very
nat=yes
secret=xxxx
type=user
username=xxxxxxx

username:paswd@freedigits.net


but when using gizmo5 for incoming calls work perfectly!
 

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