Any idea what is causing this error? Drops a call

General Help
1

[2012-09-20 08:47:14] WARNING[653] app_dial.c: Unable to create channel of type ‘SIP’ (cause 20 - Unknown)

########################### Log Details before and after the error #############
[2012-09-20 08:47:14] VERBOSE[653] res_agi.c: – dialparties.agi: Extension 6103 do not disturb is disabled
[2012-09-20 08:47:14] VERBOSE[653] res_agi.c: – dialparties.agi: dbset CALLTRACE/6100 to XXXXXXXXX
[2012-09-20 08:47:14] VERBOSE[653] res_agi.c: – dialparties.agi: dbset CALLTRACE/6101 to XXXXXXXXX
[2012-09-20 08:47:14] VERBOSE[653] res_agi.c: – dialparties.agi: dbset CALLTRACE/6102 to XXXXXXXXX
[2012-09-20 08:47:14] VERBOSE[653] res_agi.c: – dialparties.agi: dbset CALLTRACE/6103 to XXXXXXXXX
[2012-09-20 08:47:14] VERBOSE[653] res_agi.c: – dialparties.agi: Filtered ARG3: 6100-6101-6102-6103
[2012-09-20 08:47:14] VERBOSE[653] res_agi.c: – <SIP/in-XXXXXXXXX-000038f8>AGI Script dialparties.agi completed, returning 0
[2012-09-20 08:47:14] VERBOSE[653] pbx.c: – Executing [s@macro-dial:7] Dial(“SIP/in-XXXXXXXXX-000038f8”, “SIP/6100&SIP/6101&SIP/6102&SIP/6103,20,trM(auto-blkvm)”) in new stack
[2012-09-20 08:47:14] VERBOSE[653] netsock2.c: == Using SIP RTP TOS bits 184
[2012-09-20 08:47:14] VERBOSE[653] netsock2.c: == Using SIP RTP CoS mark 5
[2012-09-20 08:47:14] VERBOSE[653] app_dial.c: – Called SIP/6100
[2012-09-20 08:47:14] VERBOSE[653] netsock2.c: == Using SIP RTP TOS bits 184
[2012-09-20 08:47:14] VERBOSE[653] netsock2.c: == Using SIP RTP CoS mark 5
[2012-09-20 08:47:14] VERBOSE[653] app_dial.c: – Called SIP/6101
[2012-09-20 08:47:14] WARNING[653] app_dial.c: Unable to create channel of type ‘SIP’ (cause 20 - Unknown)
[2012-09-20 08:47:14] VERBOSE[653] netsock2.c: == Using SIP RTP TOS bits 184
[2012-09-20 08:47:14] VERBOSE[653] netsock2.c: == Using SIP RTP CoS mark 5
[2012-09-20 08:47:14] VERBOSE[653] app_dial.c: – Called SIP/6103
[2012-09-20 08:47:14] VERBOSE[653] app_dial.c: – SIP/6100-000038f9 connected line has changed. Saving it until answer for SIP/in-XXXXXXXXX-000038f8
[2012
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I am getting the above error when someone in the office makes a call just before a new call comes in. The person making the new call will drop the call while the new call will be received.

I verify all aAstra handset configurations are not giving the incoming call the priority.

I just cannot figure this out. Throw me a bone, please?

System Information:
FreePBX 2.10.0.1

2

Aastra’s will drop the outbound call for an inbound call by default.

google for the Aastra admin guide to change their behavior.

3

I found:

Enhancements to “Incoming Call Interrupts Dialing” Feature
The “Incoming Call Interrupts Dialing” feature introduced in Release 2.1 has
been enhanced to work more efficiently.
In previous releases, if the “Incoming Call Interrupts Dialing” feature was
disabled, and you were dialing out on your phone, and you received an incoming
call at the same time, the incoming call would go to an available line and the LED
would blink to let you know where the call was placed by the phone. The Caller
ID would display on the LCD and the number you were dialing disappears. If you
wanted to continue dialing out, you would have to press the Line key for which
you were originally dialing out on. If “Incoming Call Interrupts Dialing” is
enabled, the incoming call interrupts your dialing sequence and displays to the
phone’s LCD for you to answer.
In Release 2.1.1, the behavior for “Incoming Call Interrupts Dialing” has been
modified. Now when this feature is disabled, the incoming call goes to an
available line, the LED blinks, but the LCD screen still displays the number you
were dialing. When this feature is enabled, it performs the same as in previous
releases.

The issue is close to what was described above, however, the phone is dropping the call, not notifying the user of the new call. Also, the “Incoming Call Interrupts Dialing” is disabled.

That said, is there anything else that could be causing the issue below?

2012-09-20 08:47:14] WARNING[653] app_dial.c: Unable to create channel of type ‘SIP’ (cause 20 - Unknown)

Thanks,
Mike

4

Not really the phone has already terminated the call. Do you have call waiting enabled?

5

Yes. It was enabled. I tried to disable the feature today to see if it helps.

Thanks,
Mike