Unable to create channel of type 'SIP' (cause 20 - Unknown)

Asterisk Asterisk Support
1

I have s sytem that is having several issues including dropped calls, phantom calls from “unknown unknown” and a Grandstream 2010 phone that loses it’s ability to take a 2nd incoming call after a couple of days.

The system is Asterisk 1.6.2.16 with FreePBX 2.7x and a digium AEX800 card. there are a couple error messages in the logs that I see consistently but the problems are so random I can’t get my head around what to look for or fix.

The unable to create channel of type SIP message shows up about 5 times in an hours worth of logging. What does it indicate? What can I do to fix it?

[Aug 10 08:57:05] VERBOSE[19339] netsock.c: == Using SIP RTP TOS bits 184
[Aug 10 08:57:05] VERBOSE[19339] netsock.c: == Using SIP RTP CoS mark 5
[Aug 10 08:57:05] VERBOSE[19339] netsock.c: == Using UDPTL TOS bits 184
[Aug 10 08:57:05] VERBOSE[19339] netsock.c: == Using UDPTL CoS mark 5
[Aug 10 08:57:05] WARNING[19339] app_dial.c: Unable to create channel of type ‘SIP’ (cause 20 - Unknown)
[Aug 10 08:57:05] VERBOSE[19339] netsock.c: == Using SIP RTP TOS bits 184
[Aug 10 08:57:05] VERBOSE[19339] netsock.c: == Using SIP RTP CoS mark 5
[Aug 10 08:57:05] VERBOSE[19339] netsock.c: == Using UDPTL TOS bits 184
[Aug 10 08:57:05] VERBOSE[19339] netsock.c: == Using UDPTL CoS mark 5
[Aug 10 08:57:05] VERBOSE[19339] app_dial.c: – Called 1016

[Aug 10 09:12:51] VERBOSE[27720] chan_sip.c: == Extension Changed 1002[ext-local] new state Idle for Notify User 1008
[Aug 10 09:21:08] VERBOSE[19468] chan_dahdi.c: – Starting simple switch on ‘DAHDI/4-1’
[Aug 10 09:21:12] NOTICE[19468] chan_dahdi.c: Got event 18 (Ring Begin)…
[Aug 10 09:21:13] NOTICE[19468] chan_dahdi.c: Got event 2 (Ring/Answered)…
[Aug 10 09:21:13] VERBOSE[19468] pbx.c: – Executing [s@from-zaptel:1] NoOp(“DAHDI/4-1”, "Entering from-zaptel with DID == ") in new stack
[Aug 10 09:21:13] VERBOSE[19468] pbx.c: – Executing [s@from-zaptel:2] Ringing(“DAHDI/4-1”, “”) in new stack
[Aug 10 09:21:13] VERBOSE[19468] pbx.c: – Executing [s@from-zaptel:3] Set(“DAHDI/4-1”, “DID=s”) in new stack
[Aug 10 09:21:13] VERBOSE[19468] pbx.c: – Executing [s@from-zaptel:4] NoOp(“DAHDI/4-1”, “DID is now s”) in new stack
[Aug 10 09:21:13] VERBOSE[19468] pbx.c: – Executing [s@from-zaptel:5] GotoIf(“DAHDI/4-1”, “1?zapok:notzap”) in new stack

2

the sip extension you are trying to reach is not available (you probably have a dial(sip/${EXTEN}) somewhere, and when it is called, the phone is not reachable.

3

In that case, it generally means there is no entry in sip.conf with a name that is the expansion of ${EXTEN}

4

There is a phone in the main ring group that is not reachable so that makes sense but what does the 2nd message mean.

[Aug 10 09:21:12] NOTICE[19468] chan_dahdi.c: Got event 18 (Ring Begin)…
[Aug 10 09:21:13] NOTICE[19468] chan_dahdi.c: Got event 2 (Ring/Answered)…

The problem I’m really trying to get to the bottom of is the phone that loses it’s ability to put a call on hold and take a 2nd inbound call. It works for a day or 2 and then has to be rebooted. I blamed Grandstream at 1st, but I replaced the phone and still have the problem. Also there is another phone, same model on the sytem that doesn’t have the problem.

All my FreePBX settings are identical on the phone that works and the one that doesn’t.

I just don’t know what to look for in the logs or how to trap any errors.

5

NOTICE message do not imply a problem.

6

In my case I cannot reach the ext. i have configured an hard phone and sometimes works. I have the message the ext is not available. this is my log

  • == CDR updated on DAHDI/25-1
    [2012-01-06 18:10:48] – Executing [7101@defaultlog:1] AGI(“DAHDI/25-1”, “agi-NVA_recording.agi|BOTH------Y—Y---Y”) in new stack
    [2012-01-06 18:10:48] – Launched AGI Script /var/lib/asterisk/agi-bin/agi-NVA_recording.agi
    [2012-01-06 18:10:48] – AGI Script Executing Application: (Monitor) Options: (wav|/var/spool/asterisk/monitor/MIX/20120106181048__7101)
    [2012-01-06 18:10:48] – AGI Script agi-NVA_recording.agi completed, returning 0
    [2012-01-06 18:10:48] – Executing [7101@defaultlog:2] Goto(“DAHDI/25-1”, “default|7101|1”) in new stack
    [2012-01-06 18:10:48] – Goto (default,7101,1)
    [2012-01-06 18:10:48] – Executing [7101@default:1] Dial(“DAHDI/25-1”, “SIP/7101|60|”) in new stack
    [2012-01-06 18:10:48] WARNING[29405]: app_dial.c:1296 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Unknown)
    [2012-01-06 18:10:48] == Everyone is busy/congested at this time (1:0/0/1)
    [2012-01-06 18:10:48] – Executing [7101@default:2] Goto(“DAHDI/25-1”, “default|850266666666667101|1”) in new stack
    [2012-01-06 18:10:48] – Goto (default,850266666666667101,1)
    [2012-01-06 18:10:48] – Executing [850266666666667101@default:1] Wait(“DAHDI/25-1”, “1”) in new stack
    [2012-01-06 18:10:49] – Executing [850266666666667101@default:2] VoiceMail(“DAHDI/25-1”, “7101|u”) in new stack
    [2012-01-06 18:10:49] – <DAHDI/25-1> Playing ‘vm-theperson’ (language ‘en’)
    [2012-01-06 18:10:50] == Refreshing DNS lookups.
    [2012-01-06 18:10:51] – <DAHDI/25-1> Playing ‘digits/7’ (language ‘en’)
    [2012-01-06 18:10:52] – <DAHDI/25-1> Playing ‘digits/1’ (language ‘en’)
    [2012-01-06 18:10:52] – <DAHDI/25-1> Playing ‘digits/0’ (language ‘en’)
    [2012-01-06 18:10:53] – <DAHDI/25-1> Playing ‘digits/1’ (language ‘en’)
    [2012-01-06 18:10:54] – <DAHDI/25-1> Playing ‘vm-isunavail’ (language ‘en’)
    [2012-01-06 18:10:55] – <DAHDI/25-1> Playing ‘vm-intro’ (language ‘en’)
    [2012-01-06 18:10:59] == Parsing ‘/etc/asterisk/manager.conf’: [2012-01-06 18:10:59] Found
    [2012-01-06 18:10:59] == Manager ‘sendcron’ logged on from 127.0.0.1

any help thanks

7

I don’t think that | is a valid parameter delimiter in that version of Asterisk.