UK Outbound rule using E.164 format


(Chris) #1

Hey there

i’d like users to dial 9 to get an outside line
reading the instuctions of my sip provider they require the format to be E.164

eg
phone number: 07955931634
freepbx translated: 447955931634

attached is what i thought would match the patter: 907955931634 , remove the 90, and add 44 sending 447955931634 to the trunk.


(Lorne Gaetz) #2

Remove the 90 from the final field so it’s just X’s. The ‘match pattern’ field is for dialed digits that are unchanged.


(Chris) #3

i get a message saying it’s busy

here’s my log
[2020-06-17 13:14:21] VERBOSE[30434][C-00000001] pbx.c: Executing [907950931734@from-internal:13] Macro(“PJSIP/90501-00000000”, “outisbusy,”) in new stack
[2020-06-17 13:14:21] VERBOSE[30434][C-00000001] pbx.c: Executing [s@macro-outisbusy:1] Progress(“PJSIP/90501-00000000”, “”) in new stack
[2020-06-17 13:14:21] VERBOSE[30434][C-00000001] pbx.c: Executing [s@macro-outisbusy:2] GotoIf(“PJSIP/90501-00000000”, “0?emergency,1”) in new stack
[2020-06-17 13:14:21] VERBOSE[30434][C-00000001] pbx.c: Executing [s@macro-outisbusy:3] GotoIf(“PJSIP/90501-00000000”, “0?intracompany,1”) in new stack
[2020-06-17 13:14:21] VERBOSE[30434][C-00000001] pbx.c: Executing [s@macro-outisbusy:4] Playback(“PJSIP/90501-00000000”, “all-circuits-busy-now&please-try-call-later, noanswer”) in new stack
[2020-06-17 13:14:21] ERROR[8825] pjproject: icess0x7f992c019ca8 …Error sending STUN request: Network is unreachable
[2020-06-17 13:14:21] VERBOSE[30434][C-00000001] file.c: <PJSIP/90501-00000000> Playing ‘all-circuits-busy-now.ulaw’ (language ‘en’)
[2020-06-17 13:14:23] VERBOSE[30434][C-00000001] file.c: <PJSIP/90501-00000000> Playing ‘please-try-call-later.ulaw’ (language ‘en’)
[2020-06-17 13:14:25] VERBOSE[30434][C-00000001] pbx.c: Executing [h@from-internal:1] Macro(“PJSIP/90501-00000000”, “hangupcall”) in new stack
[2020-06-17 13:14:25] VERBOSE[30434][C-00000001] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“PJSIP/90501-00000000”, “1?theend”) in new stack
[2020-06-17 13:14:25] VERBOSE[30434][C-00000001] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2020-06-17 13:14:25] VERBOSE[30434][C-00000001] pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“PJSIP/90501-00000000”, “0?Set(CDR(recordingfile)=)”) in new stack
[2020-06-17 13:14:25] VERBOSE[30434][C-00000001] pbx.c: Executing [s@macro-hangupcall:4] NoOp(“PJSIP/90501-00000000”, " montior file= ") in new stack
[2020-06-17 13:14:25] VERBOSE[30434][C-00000001] pbx.c: Executing [s@macro-hangupcall:5] GotoIf(“PJSIP/90501-00000000”, “1?skipagi”) in new stack
[2020-06-17 13:14:25] VERBOSE[30434][C-00000001] pbx_builtins.c: Goto (macro-hangupcall,s,7)
[2020-06-17 13:14:25] VERBOSE[30434][C-00000001] pbx.c: Executing [s@macro-hangupcall:7] Hangup(“PJSIP/90501-00000000”, “”) in new stack
[2020-06-17 13:14:25] VERBOSE[30434][C-00000001] app_macro.c: Spawn extension (macro-hangupcall, s, 7) exited non-zero on ‘PJSIP/90501-00000000’ in macro ‘hangupcall’
[2020-06-17 13:14:25] VERBOSE[30434][C-00000001] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on ‘PJSIP/90501-00000000’
[2020-06-17 13:14:25] VERBOSE[30434][C-00000001] app_stack.c: PJSIP/90501-00000000 Internal Gosub(crm-hangup,s,1) start
[2020-06-17 13:14:25] VERBOSE[30434][C-00000001] pbx.c: Executing [s@crm-hangup:1] NoOp(“PJSIP/90501-00000000”, “Sending Hangup to CRM”) in new stack
[2020-06-17 13:14:25] VERBOSE[30434][C-00000001] pbx.c: Executing [s@crm-hangup:2] NoOp(“PJSIP/90501-00000000”, “HANGUP CAUSE: 21”) in new stack
[2020-06-17 13:14:25] VERBOSE[30434][C-00000001] pbx.c: Executing [s@crm-hangup:3] ExecIf(“PJSIP/90501-00000000”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
[2020-06-17 13:14:25] VERBOSE[30434][C-00000001] pbx.c: Executing [s@crm-hangup:4] NoOp(“PJSIP/90501-00000000”, “MASTER CHANNEL: 1592396061.0 = 1592396061.0”) in new stack
[2020-06-17 13:14:25] VERBOSE[30434][C-00000001] pbx.c: Executing [s@crm-hangup:5] GotoIf(“PJSIP/90501-00000000”, “0?return”) in new stack
[2020-06-17 13:14:25] VERBOSE[30434][C-00000001] pbx.c: Executing [s@crm-hangup:6] Set(“PJSIP/90501-00000000”, “__CRM_HANGUP=1”) in new stack
[2020-06-17 13:14:25] VERBOSE[30434][C-00000001] pbx.c: Executing [s@crm-hangup:7] AGI(“PJSIP/90501-00000000”, “agi://127.0.0.1/sangomacrm.agi”) in new stack
[2020-06-17 13:14:25] VERBOSE[30434][C-00000001] res_agi.c: <PJSIP/90501-00000000>AGI Script agi://127.0.0.1/sangomacrm.agi completed, returning 0
[2020-06-17 13:14:25] VERBOSE[30434][C-00000001] pbx.c: Executing [s@crm-hangup:8] Return(“PJSIP/90501-00000000”, “”) in new stack
[2020-06-17 13:14:25] VERBOSE[30434][C-00000001] app_stack.c: Spawn extension (from-internal, h, 1) exited non-zero on ‘PJSIP/90501-00000000’
[2020-06-17 13:14:25] VERBOSE[30434][C-00000001] app_stack.c: PJSIP/90501-00000000 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=


#4

not enough X’s


(Chris) #5

i’ve tried from 10 x’s to 14 x’s and still no joy…


#6

with the exception of Brampton, phone numbers are 10 digits long, adding the 0 trunk prefix if needed by your vsp.


(Chris) #7

i dial:
907950931333

i want the below to convert it to
447950931333

doesn’t work and here’s the logs

[2020-06-17 15:42:45] VERBOSE[18211][C-00000005] pbx.c: Executing [907950931734@zulu-from-internal:1] Set(“PJSIP/90501-00000004”, “ZULU_ID=8c921b92-d4b4-4567-a8e5-92cbb26f3f61”) in new stack
[2020-06-17 15:42:45] VERBOSE[18211][C-00000005] pbx.c: Executing [907950931734@zulu-from-internal:2] Goto(“PJSIP/90501-00000004”, “from-internal,907950931734,1”) in new stack
[2020-06-17 15:42:45] VERBOSE[18211][C-00000005] pbx_builtins.c: Goto (from-internal,907950931734,1)
[2020-06-17 15:42:45] VERBOSE[18211][C-00000005] pbx.c: Executing [907950931734@from-internal:1] ResetCDR(“PJSIP/90501-00000004”, “”) in new stack
[2020-06-17 15:42:45] VERBOSE[18211][C-00000005] pbx.c: Executing [907950931734@from-internal:2] NoCDR(“PJSIP/90501-00000004”, “”) in new stack
[2020-06-17 15:42:45] VERBOSE[18211][C-00000005] pbx.c: Executing [907950931734@from-internal:3] Progress(“PJSIP/90501-00000004”, “”) in new stack
[2020-06-17 15:42:45] VERBOSE[18211][C-00000005] pbx.c: Executing [907950931734@from-internal:4] Wait(“PJSIP/90501-00000004”, “1”) in new stack
[2020-06-17 15:42:45] ERROR[17938] pjproject: icess0x7f998815b348 …Error sending STUN request: Network is unreachable


#8

thats a zulu problem, try over with a soft or hard phone firts


(Chris) #9

Hey there
just tried using a softphone

[2020-06-18 06:46:03] VERBOSE[20681][C-00000007] pbx.c: Executing [907950931734@from-internal:1] ResetCDR(“PJSIP/501-00000006”, “”) in new stack
[2020-06-18 06:46:03] VERBOSE[20681][C-00000007] pbx.c: Executing [907950931734@from-internal:2] NoCDR(“PJSIP/501-00000006”, “”) in new stack
[2020-06-18 06:46:03] VERBOSE[20681][C-00000007] pbx.c: Executing [907950931734@from-internal:3] Progress(“PJSIP/501-00000006”, “”) in new stack
[2020-06-18 06:46:03] VERBOSE[20681][C-00000007] pbx.c: Executing [907950931734@from-internal:4] Wait(“PJSIP/501-00000006”, “1”) in new stack
[2020-06-18 06:46:04] VERBOSE[20681][C-00000007] pbx.c: Executing [907950931734@from-internal:5] Playback(“PJSIP/501-00000006”, “silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer”) in new stack
[2020-06-18 06:46:04] VERBOSE[20681][C-00000007] file.c: <PJSIP/501-00000006> Playing ‘silence/1.ulaw’ (language ‘en’)
[2020-06-18 06:46:05] VERBOSE[20681][C-00000007] file.c: <PJSIP/501-00000006> Playing ‘cannot-complete-as-dialed.ulaw’ (language ‘en’)
[2020-06-18 06:46:08] VERBOSE[20681][C-00000007] file.c: <PJSIP/501-00000006> Playing ‘check-number-dial-again.ulaw’ (language ‘en’)


(Edge) #10

Try with

X.

to get rid of your probably wrong numbers of X and then try again.
I always try to minimize possible Errors and then work through my Problem.


#11

As @Edge2020 says. If you still have trouble, at the Asterisk command prompt, type
pjsip set logger on
and make a test call. Paste the relevant section of the Asterisk log at https://pastebin.freepbx.org and post the link here.


(Chris) #12

is there more to the command than the above?

[root@voip ~]# pjsip set logger on
-bash: pjsip: command not found
[root@voip ~]#


(Chris) #13

do you mean like this?


(Edge) #14

Yes, so every number will be matched - no matter how many digits there really are


(Edge) #15

You need to execute this command at the asterisk shell.
Try with:

asterisk -rcvvvv

and then execute the pjsip debug command again.


(Chris) #16

here’s the log with X. used

i’m now getting a ‘all circuits are busy now’

[2020-06-18 07:14:21] VERBOSE[25167][C-00000008] pbx.c: Executing [907950931734@from-internal:13] Macro(“PJSIP/501-00000007”, “outisbusy,”) in new stack
[2020-06-18 07:14:21] VERBOSE[25167][C-00000008] pbx.c: Executing [s@macro-outisbusy:1] Progress(“PJSIP/501-00000007”, “”) in new stack
[2020-06-18 07:14:21] VERBOSE[25167][C-00000008] pbx.c: Executing [s@macro-outisbusy:2] GotoIf(“PJSIP/501-00000007”, “0?emergency,1”) in new stack
[2020-06-18 07:14:21] VERBOSE[25167][C-00000008] pbx.c: Executing [s@macro-outisbusy:3] GotoIf(“PJSIP/501-00000007”, “0?intracompany,1”) in new stack
[2020-06-18 07:14:21] VERBOSE[25167][C-00000008] pbx.c: Executing [s@macro-outisbusy:4] Playback(“PJSIP/501-00000007”, “all-circuits-busy-now&please-try-call-later, noanswer”) in new stack
[2020-06-18 07:14:21] VERBOSE[25167][C-00000008] file.c: <PJSIP/501-00000007> Playing ‘all-circuits-busy-now.ulaw’ (language ‘en’)
[2020-06-18 07:14:22] VERBOSE[25167][C-00000008] file.c: <PJSIP/501-00000007> Playing ‘please-try-call-later.ulaw’ (language ‘en’)


(Chris) #17

great i’ve now done this

where can i pick up the logs for it once i’ve tried another call using a soft phone


(Chris) #18

ahh i see it prints it in the terminal

<— Transmitting SIP response (314 bytes) to UDP:188.213.136.18:18714 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.153:37145;rport=18714;received=188.213.136.18;branch=z9hG4bK181309047
Call-ID: 847652739-37145-5@BJC.BGI.B.BFD
From: sip:501@VoIP.unit.tv;tag=1802669652
To: sip:447950931734@VoIP.unit.tv
CSeq: 41 INVITE
Server: FPBX-15.0.16.53(16.9.0)
Content-Length: 0

== Using SIP RTP Audio TOS bits 184
== Using SIP RTP Audio TOS bits 184 in TCLASS field.
== Using SIP RTP Audio CoS mark 5
– Executing [447950931734@from-internal:1] ResetCDR(“PJSIP/501-00000008”, “”) in new stack
– Executing [447950931734@from-internal:2] NoCDR(“PJSIP/501-00000008”, “”) in new stack
– Executing [447950931734@from-internal:3] Progress(“PJSIP/501-00000008”, “”) in new stack
– Executing [447950931734@from-internal:4] Wait(“PJSIP/501-00000008”, “1”) in new stack
> 0x7f98fc03a640 – Strict RTP learning after remote address set to: 188.213.136.18:17202
<— Transmitting SIP response (799 bytes) to UDP:188.213.136.18:18714 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.153:37145;rport=18714;received=188.213.136.18;branch=z9hG4bK181309047
Call-ID: 847652739-37145-5@BJC.BGI.B.BFD
From: sip:501@VoIP.unit.tv;tag=1802669652
To: sip:447950931734@VoIP.unit.tv;tag=38b72e60-afbd-4b0d-9f6d-94fe75d97101
CSeq: 41 INVITE
Server: FPBX-15.0.16.53(16.9.0)
Contact: sip:217.138.34.101:5060
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Type: application/sdp
Content-Length: 253

v=0
o=- 8000 8002 IN IP4 217.138.34.101
s=Asterisk
c=IN IP4 217.138.34.101
t=0 0
m=audio 12816 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

   > 0x7f98fc03a640 -- Strict RTP switching to RTP target address 188.213.136.18:17202 as source
-- Executing [447950931734@from-internal:5] Playback("PJSIP/501-00000008", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
-- <PJSIP/501-00000008> Playing 'silence/1.ulaw' (language 'en')
-- <PJSIP/501-00000008> Playing 'cannot-complete-as-dialed.ulaw' (language 'en')
-- <PJSIP/501-00000008> Playing 'check-number-dial-again.ulaw' (language 'en')
   > 0x7f98fc03a640 -- Strict RTP learning complete - Locking on source address 188.213.136.18:17202

<— Received SIP request (333 bytes) from UDP:188.213.136.18:18714 —>
CANCEL sip:447950931734@VoIP.unit.tv SIP/2.0
Via: SIP/2.0/UDP 192.168.1.153:37145;branch=z9hG4bK181309047;rport
Call-ID: 847652739-37145-5@BJC.BGI.B.BFD
From: sip:501@VoIP.unit.tv;tag=1802669652
To: sip:447950931734@VoIP.unit.tv
CSeq: 41 CANCEL
Max-Forwards: 70
User-Agent: Grandstream Wave 1.0.3.34
Content-Length: 0


(Chris) #19

i can see in the above log it’s now converting the mobile number from
07950931734
to
447950931734

…which is what my sip trunk provider requires which is good


(Edge) #20

Is your outgoing Trunk setup correctly?
Seems like you have an emergency outgoing route and one intracompany. None of them matches, so you can’t dial out.