Removing ported out DID

Good afternoon,

I have an interesting issue.
I have an extension that has been ported out successfully, however, trying to call that extension from within the legacy asterisk system, results in a re-direct tone. Where calling from verizon cell network, is successfull.

It seems to me that calling from the legacy asterisk system - asterisk is recognizing that extension and is attempting to place the internal call. Which fails

Iv scanned my extensions.conf and my sip.conf and have removed all entries related to the extension in question.
Then issued a
Also, a

Is there something else I need to do to get asterisk to stop recognizing that extension as an internal call?

Thanks.

Dialplan logic would be in extensions.conf or any included files in it, after which you would issue a “dialplan reload”. If the call still fails you’d need to provide the console output to show exactly what happened - for example it could be due to the upstream provider.

Let me give you some pre-context.

We are a voip provider who uses asterisk based system.
Our company’s ownership has changed hands and the powers that be have signed a contract with Sinch to move forward with in terms of our voip services. The goal is now, to slowly migrating away from our legacy Asterisk based system, into the new Sinch, cloud based system.

Our test DID is 9705221202
Was previously working fine through our asterisk system.

We just ported the extension out to Sinch to test through our new Sinch platform. Which works. I can complete a phone call using my cell phone.

My office phones are all still behind the legacy asterisk system. Its calling from here, within the office, within the legacy asterisk system, where the call results in a re-direct.

I did do a dialplan reload, and also a module reload chan_sip.so after removing all traces of 1202 from extensions.conf and sip.conf - actually they were in included files extensions_exten.conf and sip_exten.conf

Here is the dialog as shown from my voice-switch dialing from my office, to the 1202 DID:

[Jan 21 15:05:32] == Using SIP RTP CoS mark 5
[Jan 21 15:05:32] – Executing [9705221202@incustomerintl:1] NoOp(“SIP/kcioffice-0001fc1f”, “ten digit outgoing”) in new stack
[Jan 21 15:05:32] – Executing [9705221202@incustomerintl:2] Set(“SIP/kcioffice-0001fc1f”, “CHANNEL(amaflags)=DOCUMENTATION”) in new stack
[Jan 21 15:05:32] – Executing [9705221202@incustomerintl:3] Dial(“SIP/kcioffice-0001fc1f”, “SIP/voice-gw-1/9705221202”) in new stack
[Jan 21 15:05:32] == Using SIP RTP CoS mark 5
[Jan 21 15:05:32] – Called SIP/voice-gw-1/9705221202
[Jan 21 15:05:32] == Using SIP RTP CoS mark 5
[Jan 21 15:05:32] NOTICE[1163][C-0001021e]: chan_sip.c:26721 handle_request_invite: Call from ‘voice-switch-2’ (172.31.80.12:5060) to extension ‘9705221202’ rejected because extension not found in context ‘incoming’.
[Jan 21 15:05:32] – Got SIP response 503 “Service Unavailable” back from 172.31.80.12:5060
[Jan 21 15:05:32] – SIP/voice-gw-1-0001fc20 is circuit-busy
[Jan 21 15:05:32] == Everyone is busy/congested at this time (1:0/1/0)
[Jan 21 15:05:32] – Executing [9705221202@incustomerintl:4] NoOp(“SIP/kcioffice-0001fc1f”, “CONGESTION”) in new stack
[Jan 21 15:05:32] – Executing [9705221202@incustomerintl:5] GotoIf(“SIP/kcioffice-0001fc1f”, “0?busycall”) in new stack
[Jan 21 15:05:32] – Executing [9705221202@incustomerintl:6] Congestion(“SIP/kcioffice-0001fc1f”, “”) in new stack
[Jan 21 15:05:32] == Spawn extension (incustomerintl, 9705221202, 6) exited non-zero on ‘SIP/kcioffice-0001fc1f’
[Jan 21 15:05:32] – Executing [h@incustomerintl:1] Hangup(“SIP/kcioffice-0001fc1f”, “”) in new stack
[Jan 21 15:05:32] == Spawn extension (incustomerintl, h, 1) exited non-zero on ‘SIP/kcioffice-0001fc1f’

Thank you for your assistance!

The call was sent to “voice-gw-1” which then seemingly sent the call back to Asterisk, which didn’t work since the dialed number didn’t exist.

Okay, I have another question for you.

My voice-gw-1 is also an asterisk unit, would you have any ideas why gw-1 is pushing it back? Or possibly where and/or what to look for?


Also, I dont know if its related to the issue, but…
voice-switch-1 is where I edited extension.conf and sip.conf to remove 9705221202. and did the dialplan reload… etc…

Im noticing the reference to “voice-switch-2”(172.31.80.12:5060)

Is that line suggesting that voice-switch-2 is at 172.31.80.12?

Im confused, voice-switch-2 sits at 172.31.80.18
voice-gw-1 sits at 172.31.80.12

voice-switch-2 shouldn’t actually have anything to do with this, I didn’t think.
Hopefully this ‘voice-switch-2’ just a typo in a .conf somewhere?

unless my voice-switch-1 and voice-switch-2 are working together somehow. Can asterisk be setup as multiple systems acting as 1 redundant system?

They won’t be actively working together, but a fallback configuration could be configured into the dialplans. You would need to provide logs from the other systems an probably enable protocol logging (“sip set debug on”, as you are using the obsolete channel driver)

All,
Thank you all for your guidance. This has been resolved.

For context,
I made the call again looking at the dialog as displayed by my voice-gw-1:

[Jan 27 10:54:18]     -- Executing [9705221202@outgoing:1] NoOp("SIP/voice-switch-1-0002cbff", "sub-outgoing routing") in new stack
[Jan 27 10:54:18]     -- Executing [9705221202@outgoing:2] Gosub("SIP/voice-switch-1-0002cbff", "checkinternal,9705221202,1") in new stack
[Jan 27 10:54:18]     -- Executing [9705221202@checkinternal:1] NoOp("SIP/voice-switch-1-0002cbff", "trying to route to 9705221202") in new stack
[Jan 27 10:54:18]     -- Executing [9705221202@checkinternal:2] Set("SIP/voice-switch-1-0002cbff", "DEST=voice-switch-1") in new stack
[Jan 27 10:54:18]     -- Executing [9705221202@checkinternal:3] GotoIf("SIP/voice-switch-1-0002cbff", "0?noroute") in new stack
[Jan 27 10:54:18]     -- Executing [9705221202@checkinternal:4] Gosub("SIP/voice-switch-1-0002cbff", "voice-switch-1,9705221202,1") in new stack
[Jan 27 10:54:18]     -- Executing [9705221202@voice-switch-1:1] Dial("SIP/voice-switch-1-0002cbff", "SIP/voice-switch-1/9705221202") in new stack
[Jan 27 10:54:18]   == Using SIP RTP CoS mark 5
[Jan 27 10:54:18]     -- Called SIP/voice-switch-1/9705221202
[Jan 27 10:54:18]   == Everyone is busy/congested at this time (1:0/0/1)
[Jan 27 10:54:18]     -- Auto fallthrough, channel 'SIP/voice-switch-1-0002cbff' status is 'CHANUNAVAIL'

I then checked my dialplan, looking for this “checkinternal”, and found:

[ Context 'checkinternal' created by 'pbx_config' ]
'h' =>            1. Hangup()                                   [extensions.conf:286]
'i' =>            1. Hangup()                                   [extensions.conf:287]
't' =>            1. Hangup()                                   [extensions.conf:288]
'_NXXNXXXXXX' =>  1. NoOp(trying to route to ${EXTEN})        [extensions.conf:280]
                2. Set(DEST=${DB(kci/route/${EXTEN})})        [extensions.conf:281]
                3. GotoIf($["${DEST}" = ""]?noroute)          [extensions.conf:282]
                4. Gosub(${DEST},${EXTEN},1)                  [extensions.conf:283]
                5. Hangup()                                   [extensions.conf:284]
 [noroute]      6. Return()                                   [extensions.conf:285]

Using this I can clearly see that its referencing the database at kci/route/EXT.

database show:
/kci/route/9705221202                             : voice-switch-1           

This, above, is how my gateway is pushing the call back to voice-switch-1

The resolution:

database del kci route/9705221202

My call is now able to continue as an outbound call.
Thanks again for all your assistance!