Unable to create channel

Asterisk Asterisk Support
1

My Sip phones are not registering here is the output

US-IRV-01CLI>
== Using SIP RTP CoS mark 5
– Executing [555@cbmin:1] GotoIfTime(“SIP/voipvoip_in-00000001”, "07:00-19:00,mon-fri,,?cbmin,s,1") in new stack
– Goto (cbmin,s,1)
– Executing [s@cbmin:1] Read(“SIP/voipvoip_in-00000001”, “digito,/home/astconf/sounds/newcbm,1,10”) in new stack
– Accepting a maximum of 1 digits.
– <SIP/voipvoip_in-00000001> Playing ‘/home/astconf/sounds/newcbm.slin’ (language ‘en’)
– User entered ‘2’
– Executing [s@cbmin:2] NoOp(“SIP/voipvoip_in-00000001”, “2”) in new stack
– Executing [s@cbmin:3] GotoIf(“SIP/voipvoip_in-00000001”, “0?sales”) in new stack
– Executing [s@cbmin:4] GotoIf(“SIP/voipvoip_in-00000001”, “1?custser”) in new stack
– Goto (cbmin,s,10)
– Executing [s@cbmin:10] Goto(“SIP/voipvoip_in-00000001”, “cbmin,6804,1”) in new stack
– Goto (cbmin,6804,1)
– Executing [6804@cbmin:1] Dial(“SIP/voipvoip_in-00000001”, “SIP/6804,10”) in new stack
[2013-08-13 09:04:45] WARNING[1843]: app_dial.c:2345 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [6804@cbmin:2] VoiceMail(“SIP/voipvoip_in-00000001”, “6804,u”) in new stack
– <SIP/voipvoip_in-00000001> Playing ‘/var/spool/asterisk/voicemail/default/6804/unavail.slin’ (language ‘en’)
== Spawn extension (cbmin, 6804, 2) exited non-zero on 'SIP/voipvoip_in-00000001’
US-IRV-01CLI>

Below is my sip.conf

SIP.CONF

[general]
context=cbmin ; Default context for incoming calls
allowguest=no ; Allow or reject guest calls (default is yes)
allowoverlap=no ; Disable overlap dialing support.
udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
tcpenable=no ; Enable server for incoming TCP connections (default is no)
tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
srvlookup=no ; Enable DNS SRV lookups on outbound calls
qualify=3000
alwaysauthreject=yes

;
register => 7145551212:jdhfw9ehww48r2ipwqofj@10.10.10.10/7145551212
;
[6800]
type=peer
host=dynamic
dtmfmode=rfc2833
username=6800
secret=JJHksj$765938H
canreinvite=no
reinvite=no
callerid=
disallow=all
;allow=g729
allow=ulaw
allow=alaw
allow=gsm
;mailbox=6801
nat=yes
qualify=yes

[voip_in]
username=7145551212
type=peer
secret=jdhfw9ehww48r2ipwqofj
nat=yes
insecure=port,invite
qualify=yes
host=10.10.10.10
fromdomain=10.10.10.10
dtmfmode=rfc2833
context=cbmin
disallow=all
;allow=g729
allow=ulaw
allow=alaw
allow=ilbc
;allow=gsm

[voip_out]
username=7145551212
type=peer
secret=jdhfw9ehww48r2ipwqofj
nat=yes
qualify=yes
insecure=port,invite
host=10.10.10.10
fromuser=7145551212
fromdomain=10.10.10.10
dtmfmode=rfc2833
disallow=all
;allow=g729
allow=ulaw
allow=alaw
allow=ilbc
;allow=gsm

2

[quote]-- Goto (cbmin,6804,1)
– Executing [6804@cbmin:1] Dial(“SIP/voipvoip_in-00000001”, “SIP/6804,10”) in new stack
[2013-08-13 09:04:45] WARNING[1843]: app_dial.c:2345 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)
== Everyone is busy/congested at this time (1:0/0/1)[/quote]

where is the extension 6804 in your sip.conf file.

set qualify=yes

run sip show peers and paste the output

3

Sorry that is just an example 6804 is another extension

US-IRV-01CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
6800/6800 (Unspecified) D N 0 UNKNOWN
6801/sales (Unspecified) D N 0 UNKNOWN
6802/6802 (Unspecified) D N 0 UNKNOWN
6803/support (Unspecified) D N 0 UNKNOWN
6804/6804 (Unspecified) D N 0 UNKNOWN
6805/6805 (Unspecified) D N 0 UNKNOWN
6806/6806 (Unspecified) D N 0 UNKNOWN
6807/6807 (Unspecified) D N 0 UNKNOWN
6808/6808 (Unspecified) D N 0 UNKNOWN
6809/6809 (Unspecified) D N 0 UNKNOWN
voipvoip_in/7145551212 xx.xx.xx.xx N 5060 OK (43 ms)
voipvoip_out/7145551212 xx.xx.xx.xx N 5060 OK (45 ms)
12 sip peers [Monitored: 2 online, 10 offline Unmonitored: 0 online, 0 offline]
US-IRV-01CLI>

4

BTW I did set qualify=yes no change same result.

5

Yes based on your CLI output, I knew 6804 is another sip peer. Well all your sip internal sip peers look to be offline, except the 2 sip trunks. your might check your network. And reboot the server it could help

6

If you turn on qualify in the configuration of a SIP device in sip.conf, Asterisk will send a SIP OPTIONS command regularly to check that the device is still online. If the device does not answer within the configured (or default) period (in ms) Asterisk considers the device off-line for future calls. This status can be checked by the SIPPEER function, and inversely this function will only provide status information for peers which have qualify=yes.

voip-info.org/wiki/view/Asterisk+sip+qualify

7

OK found the problem, on the phone it self under sip configuration the outbound proxy address for some reason was set the same as the IP addr of the phone, removed it from all the phones they registered fine.

Now need figure out why there is no music on-hold…

Thanks for your help,
tony

8

Music on-hold internally (calling each other extensions) works fine, when making outbound/inbound calls there is no music on-hold

9

POST the CLI.

10

thanks here is the cli but it started working…

== Using SIP RTP CoS mark 5
– Executing [7145551212@cbmin:1] GotoIfTime(“SIP/voipvoip_in-00000021”, “07:00-19:00,mon-fri,,?cbmin,s,1”) in new stack
– Goto (cbmin,s,1)
– Executing [s@cbmin:1] Read(“SIP/voipvoip_in-00000021”, “digito,/home/astconf/sounds/newcbm,1,10”) in new stack
– Accepting a maximum of 1 digits.
– <SIP/voipvoip_in-00000021> Playing ‘/home/astconf/sounds/newcbm.slin’ (language ‘en’)
– User entered ‘1’
– Executing [s@cbmin:2] NoOp(“SIP/voipvoip_in-00000021”, “1”) in new stack
– Executing [s@cbmin:3] GotoIf(“SIP/voipvoip_in-00000021”, “1?sales”) in new stack
– Goto (cbmin,s,9)
– Executing [s@cbmin:9] Goto(“SIP/voipvoip_in-00000021”, “cbmin,6801,1”) in new stack
– Goto (cbmin,6801,1)
– Executing [6801@cbmin:1] Dial(“SIP/voipvoip_in-00000021”, “SIP/6801,10”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/6801
– SIP/6801-00000022 is ringing
– SIP/6801-00000022 answered SIP/voipvoip_in-00000021
– Locally bridging SIP/voipvoip_in-00000021 and SIP/6801-00000022
– Started music on hold, class ‘default’, on SIP/voipvoip_in-00000021

11

how do I configure this feature when a call comes in to an extension to also ring the cell of the person

12

could be something like this.
[did-usa]
exten=>s,1,Dial(SIP/100&SIP/${cellphone}@${out-bound-trunk},tT)

13

Thanks I asked my sip trunk provider what is my out bound trunk info.

They sent me this:

Your outbound setting should be as explained below;
Outgoing Settings

Peer Details

username=7145551212 (your VoIP VoIP account assigned while signing up)
type=peer
qualify=yes
secret=XXXXX (your VoIP VoIP password)
nat=auto
insecure=very
host= some.domain.com
fromuser=7145551212 (your VoIP VoIP account assigned while signing up)
fromdomain=some.domain.com
dtmfmode=rfc2833
disallow=all
allow=g729
allow=ilbc
allow=ulaw
allow=alaw

So what is the outbound trunk info???

14

Before future question check this asteriskdocs.org/

${out-bound-trunk} it is just a variable that is equal to =voip-out , i mean change the ${out-bound-trunk} for the name of your outbound trunk.

15

insecure=very will not work with current versions of Asterisk. You must spell out which insecurities you want.

Also it was rarely actually needed; insecure=invite is usually sufficient.

With modern versions, remotesecret is a better way of handling this sort of one sided authentication.

16

thanks David this is what actually I have

[voip_in]
username=7145551212
type=peer
secret=jdhfw9ehww48r2ipwqofj
nat=yes
insecure=port,invite
qualify=yes
host=10.10.10.10
fromdomain=10.10.10.10
dtmfmode=rfc2833
context=cbmin
disallow=all
;allow=g729
allow=ulaw
allow=alaw
allow=ilbc
;allow=gsm

[voip_out]
username=7145551212
type=peer
secret=jdhfw9ehww48r2ipwqofj
nat=yes
qualify=yes
insecure=port,invite
host=10.10.10.10
fromuser=7145551212
fromdomain=10.10.10.10
dtmfmode=rfc2833
disallow=all
;allow=g729
allow=ulaw
allow=alaw
allow=ilbc
;allow=gsm