"silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer")

FreePBX Installation / Upgrade
1

I had done installation and configure trunk and in/out call option but still incoming and outgoing not working.

== Starting D-Channel on span 1
== Registered channel type ‘DAHDI’ (DAHDI Telephony Driver w/PRI & SS7 & MFC/R2)
== Registered application ‘DAHDIAcceptR2Call’
== Manager registered action DAHDITransfer
== Manager registered action DAHDIHangup
== Manager registered action DAHDIDialOffhook
== Manager registered action DAHDIDNDon
== Manager registered action DAHDIDNDoff
== Manager registered action DAHDIShowChannels
== Manager registered action DAHDIRestart
== Manager registered action PRIShowSpans
Loaded chan_dahdi.so => (DAHDI Telephony Driver w/PRI & SS7 & MFC/R2)
– Remote UNIX connection disconnected
== Primary D-Channel on span 1 up
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Executing [99900039877@from-internal:1] ResetCDR(“SIP/39877-00000014”, “”) in new stack
– Executing [99900039877@from-internal:2] NoCDR(“SIP/39877-00000014”, “”) in new stack
– Executing [99900039877@from-internal:3] Progress(“SIP/39877-00000014”, “”) in new stack
– Executing [99900039877@from-internal:4] Wait(“SIP/39877-00000014”, “1”) in new stack
> 0x7f69f4288610 – Probation passed - setting RTP source address to 192.168.10.76:5004
– Executing [99900039877@from-internal:5] Progress(“SIP/39877-00000014”, “”) in new stack
– Executing [99900039877@from-internal:6] Playback(“SIP/39877-00000014”, “silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer”) in new stack
– <SIP/39877-00000014> Playing ‘silence/1.ulaw’ (language ‘en’)
– <SIP/39877-00000014> Playing ‘cannot-complete-as-dialed.gsm’ (language ‘en’)
== Spawn extension (from-internal, 99900039877, 6) exited non-zero on ‘SIP/39877-00000014’
– Executing [h@from-internal:1] Hangup(“SIP/39877-00000014”, “”) in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/39877-00000014’

2

Did you configure outbound route?

3

outbound route as follows
9-0NXXXXXXXXXX
9-NXXXXXXX
9-NXXXXXXXXXXX

4

I don’t know what 9- is, where did you put that and why don’t you guys post your versions? The post labeled “read this before posting” is very clear on this point. No version, no answer.

Also assuming you mean the 9 is a prefix, none of your dial patterns match the 11 digit string you dialed.

5

9 is the prefix for outcall ,whereas i had given 12 digit including NX
[root@bsgi-pbx /]# uname -a
Linux bsgi-pbx 2.6.32-431.17.1.el6.x86_64 #1 SMP Wed May 7 23:32:49 UTC 2014 x86_64 x86_64 x86_64 GNU/Linux
Asterisk 11.9.0, Copyright © 1999 - 2013 Digium, Inc. and others.
Created by Mark Spencer [email protected]
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core

6

Ypou have two matches for 12 digit nubers

0NXXXXXXXXXX
NXXXXXXXXXXX

which is equivalent to

XXXXXXXXXXXX

X = 0-9
N = 1-9
0 = 0

N or 0 = X

you need to RTFM again :slight_smile:

Either way, none of the Dial patterns to defined match the eleven digit number you dialed on the channel:-

99900039877

Did you actually read Skyking’s answer and adjust your rules to match, or will you need a third correction to your logic?

7

FreePBX version?

8

hello,
i tried to configure with same combination as mention for 12digit
0NXXXXXXXXXX
NXXXXXXXXXXX
but still error response.
please help

9

Start with X. and work your way up :slight_smile:

10

then don’t expect dialing 11 numbers to match (from your post 99900039877@from-internal. . . ).

11

latest error message.
[root@bridgesgi ~]# asterisk -rvvv
Asterisk 11.9.0, Copyright © 1999 - 2013 Digium, Inc. and others.
Created by Mark Spencer [email protected]
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for detail s.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.

Connected to Asterisk 11.9.0 currently running on bridgesgi (pid = 26114)
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Executing [99900039877@from-internal:1] ResetCDR(“SIP/9802-0000000e”, “”) in new stack
– Executing [99900039877@from-internal:2] NoCDR(“SIP/9802-0000000e”, “”) in new stack
– Executing [99900039877@from-internal:3] Progress(“SIP/9802-0000000e”, “”) in new stack
– Executing [99900039877@from-internal:4] Wait(“SIP/9802-0000000e”, “1”) in new stack
– Executing [99900039877@from-internal:5] Progress(“SIP/9802-0000000e”, “”) in new stack
– Executing [99900039877@from-internal:6] Playback(“SIP/9802-0000000e”, “si lence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer”) in new stac k
– <SIP/9802-0000000e> Playing ‘silence/1.gsm’ (language ‘en’)
– <SIP/9802-0000000e> Playing ‘cannot-complete-as-dialed.gsm’ (language 'en ')
== Spawn extension (from-internal, 99900039877, 6) exited non-zero on ‘SIP/980 2-0000000e’
– Executing [h@from-internal:1] Hangup(“SIP/9802-0000000e”, “”) in new stac k
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/9802-0000000e

can anybody face such problem.

12

You still don’t have an outbound route that matches 99900039877

Post your outbound route.

13

this is define outbound route
please correct me if this is wrong .
9|NXXXXXXX
9|NXXXXXXXXXXX
9|XNXXXXXXXXXX

14

does anybody found any solution

15

You have been told several times what the solution is. You need an outbound route that matches the digits you are dialing. You are dialing 11 digits and have no route that matches 11 digits.

16

if this is the case what about incoming ,it is also not working.
even inbound is configure.