Your call cannot be completed as dialed! Why?

General Help
1

I installed and configured the freepbx embedded into raspberry PI asterisk distribution.
I added one extension, and two truncks, only for outgoing calls, the other for incoming.
I think to have correctly configured the outgoing trunck:
trunck name, outgoing CID in the format 0039…
in the peer details:
host=secret=sip.freevoipdeal.com
username=myuserID
secret=mypwd
type=peer

user context: myuserID
USER details:
secret=mypwd
type=user
context=from-trunk

in the outbound route:
only Route name
and combobox Trunk Sequence for Matched Routes

The phone/extension is correctly registered, I think:
Displayname and secret.

But when I try to call it says:
Your call cannot be completed as dialog. Please check the number and dial again.
The Voip provider is freevoipdeal.
What I’m wrong?

Thank you very much

2

host=secret=sip.freevoipdeal.com

won’t work

3

no, my mistake.
Now I’ve checked that is correctly registered on Report section, but still the same message.
I tried also with another provider (eutelia)

4

Next step is spend some time on the wiki, particularly to the troubleshooting bits. Setting sip debug will show the low level conversations betwenn the two endpoints.

5

I tried to search but I not found nothing. What can I do?

6

Perhaps that it is mandatory to set dial patterns of outcoming routes?
I basically want to allow every number to be called.
Simple configuration

7

There is always the paid support above.

8

Hi there.
I’ve got a similar configuration (RasPBX) with a Gigaset C450IP and a Swedish VoIP provider “Teletek”. I have managed to get a “green light” on both my trunk and extension; I can call between internal SIP extensions (tried with a CSip on my android), but as soon as I try to reach the outside World, I get that “Your call cannot be completed as dialed”.

Most annoying. I do understand that there is the paid support option, but it would be nice to at least “get up on my feet”.

My dialplan is a simple “.” and this is what I see in my logs:

[2014-02-02 15:30:08] VERBOSE[2887][C-00000077] netsock2.c: == Using SIP RTP TOS bits 184
[2014-02-02 15:30:08] VERBOSE[2887][C-00000077] netsock2.c: == Using SIP RTP CoS mark 5
[2014-02-02 15:30:08] VERBOSE[12617][C-00000077] pbx.c: – Executing [[email protected]:1] Macro(“SIP/202-00000077”, “user-callerid,LIMIT,EXTERNAL,”) in new stack
[2014-02-02 15:30:08] VERBOSE[12617][C-00000077] pbx.c: – Executing [[email protected]:1] Set(“SIP/202-00000077”, “TOUCH_MONITOR=1391351408.119”) in new stack
[2014-02-02 15:30:08] VERBOSE[12617][C-00000077] pbx.c: – Executing [[email protected]:2] Set(“SIP/202-00000077”, “AMPUSER=202”) in new stack
[2014-02-02 15:30:08] VERBOSE[12617][C-00000077] pbx.c: – Executing [[email protected]:3] GotoIf(“SIP/202-00000077”, “0?report”) in new stack
[2014-02-02 15:30:08] VERBOSE[12617][C-00000077] pbx.c: – Executing [[email protected]:4] ExecIf(“SIP/202-00000077”, “1?Set(REALCALLERIDNUM=202)”) in new stack
[2014-02-02 15:30:08] VERBOSE[12617][C-00000077] pbx.c: – Executing [[email protected]:5] Set(“SIP/202-00000077”, “AMPUSER=202”) in new stack
[2014-02-02 15:30:08] VERBOSE[12617][C-00000077] pbx.c: – Executing [[email protected]:6] Set(“SIP/202-00000077”, “AMPUSERCIDNAME=202”) in new stack
[2014-02-02 15:30:08] VERBOSE[12617][C-00000077] pbx.c: – Executing [[email protected]:7] GotoIf(“SIP/202-00000077”, “0?report”) in new stack
[2014-02-02 15:30:08] VERBOSE[12617][C-00000077] pbx.c: – Executing [[email protected]:8] Set(“SIP/202-00000077”, “AMPUSERCID=202”) in new stack
[2014-02-02 15:30:08] VERBOSE[12617][C-00000077] pbx.c: – Executing [[email protected]:9] Set(“SIP/202-00000077”, “__DIAL_OPTIONS=tr”) in new stack
[2014-02-02 15:30:08] VERBOSE[12617][C-00000077] pbx.c: – Executing [[email protected]:10] Set(“SIP/202-00000077”, “CALLERID(all)=“202” <202>”) in new stack
[2014-02-02 15:30:08] VERBOSE[12617][C-00000077] pbx.c: – Executing [[email protected]:11] GotoIf(“SIP/202-00000077”, “0?limit”) in new stack
[2014-02-02 15:30:08] VERBOSE[12617][C-00000077] pbx.c: – Executing [[email protected]:12] ExecIf(“SIP/202-00000077”, “1?Set(GROUP(concurrency_limit)=202)”) in new stack
[2014-02-02 15:30:08] VERBOSE[12617][C-00000077] pbx.c: – Executing [[email protected]:13] GotoIf(“SIP/202-00000077”, “1?continue”) in new stack
[2014-02-02 15:30:08] VERBOSE[12617][C-00000077] pbx.c: – Goto (macro-user-callerid,s,26)
[2014-02-02 15:30:08] VERBOSE[12617][C-00000077] pbx.c: – Executing [[email protected]:26] Set(“SIP/202-00000077”, “CALLERID(number)=202”) in new stack
[2014-02-02 15:30:08] VERBOSE[12617][C-00000077] pbx.c: – Executing [[email protected]:27] Set(“SIP/202-00000077”, “CALLERID(name)=202”) in new stack
[2014-02-02 15:30:08] VERBOSE[12617][C-00000077] pbx.c: – Executing [[email protected]:28] Set(“SIP/202-00000077”, “CDR(cnum)=202”) in new stack
[2014-02-02 15:30:08] VERBOSE[12617][C-00000077] pbx.c: – Executing [[email protected]:29] Set(“SIP/202-00000077”, “CDR(cnam)=202”) in new stack
[2014-02-02 15:30:08] VERBOSE[12617][C-00000077] pbx.c: – Executing [[email protected]:30] Set(“SIP/202-00000077”, “CHANNEL(language)=en”) in new stack
[2014-02-02 15:30:08] VERBOSE[12617][C-00000077] pbx.c: – Executing [[email protected]:2] NoCDR(“SIP/202-00000077”, “”) in new stack
[2014-02-02 15:30:08] VERBOSE[12617][C-00000077] pbx.c: – Executing [[email protected]:3] Progress(“SIP/202-00000077”, “”) in new stack
[2014-02-02 15:30:08] VERBOSE[12617][C-00000077] pbx.c: – Executing [[email protected]:4] Wait(“SIP/202-00000077”, “1”) in new stack
[2014-02-02 15:30:09] VERBOSE[12617][C-00000077] pbx.c: – Executing [[email protected]:5] Progress(“SIP/202-00000077”, “”) in new stack
[2014-02-02 15:30:09] VERBOSE[12617][C-00000077] pbx.c: – Executing [[email protected]:6] Playback(“SIP/202-00000077”, “silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer”) in new stack
[2014-02-02 15:30:09] VERBOSE[12617][C-00000077] file.c: – <SIP/202-00000077> Playing ‘silence/1.ulaw’ (language ‘en’)
[2014-02-02 15:30:10] VERBOSE[12617][C-00000077] file.c: – <SIP/202-00000077> Playing ‘cannot-complete-as-dialed.ulaw’ (language ‘en’)
[2014-02-02 15:30:12] VERBOSE[12617][C-00000077] pbx.c: == Spawn extension (from-internal, 0890510, 6) exited non-zero on ‘SIP/202-00000077’
[2014-02-02 15:30:12] VERBOSE[12617][C-00000077] pbx.c: – Executing [[email protected]:1] Hangup(“SIP/202-00000077”, “”) in new stack
[2014-02-02 15:30:12] VERBOSE[12617][C-00000077] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/202-00000077’

Any and all help with this newbee situation is highly appreciated!
Regards,
Eriond

9

Beginners mistake, I suppose…
I made a small adjustment to my Outbound Route: I removed everything from the Dial Patterns fields except for the “match pattern” where I left just one dot (.)
Further, I added my telephone number in national notation (area code + number) and checked the “Override extension” box.
And now it works! I’m SO happy!

/Eriond

10

I have a similar problem. I am using the same provider. Teletek from Sweden. Unfortunately, the problem is not corrected by following your indication. Can you send me any information about your setup (sip trunk etc).
Br, Constantin

11

Right, there are a few things not documented well about Teletek. Maybe one day a support person from that Company will copy/paste the following and post it on their FAQ under “Setting up FreePBX”:

One of my trunks looks like this:

PEER Details:

host=sip4.teletek.se
username=[hexadecimal id].teletek.se
secret=[password]
type=peer
insecure=very
disallow=all
canreinvite=no
canredirect=no
allow=alaw&ulaw&gsm&g729
context=from-pstn

User Context:

[your phonenr in NATIONAL notation, ie. 08nnnnnnnn]

User Details:

secret=[password]
type=user
context=from-trunk

12

The “Register String” at the bottom of the Trunk page:
[hexadecimal id].teletek.se:[password]@sip4.teletek.se/[national phone nr]

(The square brackets should of course not be part of this string, nor the settings in the previous post, but I think You already knew that.)

13

I have added this to the wiki

14

The g.729 should be removed as we can assume most have not purchased that CODEC.

15

Thank you all for your quick replies. Eriond configuration worked as a charm for me.I managed to fix Teletek SIP trunk configuration in 3 minutes. Before I had access to Eriond configuration example I had lost 1 entire day without any success. Thank you again, I am most grateful.

16

When I put it to the wiki I pulled out the g729

17

Great initiative to have operator specific settings readily available in the Wiki!
I have one comment though; In the “USER Context” you have written “country prefix” when it should be “national area code”. 08 is the area code of Stockholm, whereas 46 would be the country prefix for Sweden, and should NOT be included.
Maybe a minor thing, but could cause confusion…
Anyway, great job. It’ll surely save (some swedes) a lot of time.

18