Incomming extern call ringing at intern and extern number at same time

Hi all,

How I will manage that incomming call from external number will ring on both mobile and SIP accout at same time?
Fe. - I out of offfice. Some one calling on my landline number in office. That incomming call is ringing on my landline and my mobile at same time. I was trying to add in Dialplan this, but it doesnt work:

Dialplan:
exten => 123,1,Dial(SIP/456,20,t,k&SIP/789,20)

Here is my Sip config (land line only):
[456]
type=peer
secret=password
fullname=my name
regexten=456
context=default
host=dynamic
callerid=<123>

For explanation
123 - my landline number
456 - SIP
789 - my mobile number

Not so sure If I do have to register my mobile number in Sip config or if its possible to manage it without registration?
Fe. - register => 789:password@sip.provider.com/789

I’m new here and not so experienced, so please have a bit patience with me :slight_smile: Fell free to ask for more information.

Your dialplan is incorrect, it can’t have different options or timeouts per target:

exten => 123,1,Dial(SIP/456&SIP/789,20,tk)

Fixed, but still no result. Only land line is ringing

Your ITSP may be falsely returning ANSWER. You would need more detailed logging to see this.

What logs you will need?

If you can dial from your extension your mobile phone then use this

exten => 123,1,Dial(SIP/456&Local/789@default,20,tk)

If not post your extensions file to check your dialplan.

I can call on my phone from land line. Im not so sure what I should give instead of that “Default”. I expect that this is “provider”. Setting are writen as they are in config. Only numbers in [incoming] were changed.

Here is extension config:
[general]
static=yes
writeprotect=no
clearglobalvars=no

[globals]

[outgoing]

;exten => _X.,1,NoOp()
;same => n,Verbose(1,Outgoing Caller ID:{$CALLERID(num)})
;same => n,Dial(SIP/${EXTEN}@provider)
;same => n,Hangup()

exten => _X.,1,NoOp()
same => n,SipAddHeader(P-Asserted-Identity: sip:${CALLERID(num)}@provider)
same => n,SIPAddHeader(Privacy:off)
same => n,Dial(SIP/${EXTEN}@provider)
same => n,Hangup()

[incoming]
exten => 123,1,Dial(SIP/456&Local/789@provider,20,t,k)

[default]
include => parkedcalls
include => outgoing

exten => _4XX,1,Dial(SIP/${EXTEN},20)
same => n,Hangup()

Did you tried the dial that I posted?

If you are not sure what logging you need, use:

core set verbose 5
core set debug 5
sip set debug on

and enable the full log in logger.conf.

Yes, but it doesnt work. Land line is ringing, but mobile not.

Post the log of your tests. Use the commands that david wrote to have the proper verbosity/debug levels.

I will post it tomorrow, I’m out of office

I made call from extern number, it was ringing and the I hang up

07yyyyyyyyy17 - Extern number from which I was calling
07xxxxx78471 - land line
015zzzzzzzzz99 - mobile number

Scheduling destruction of SIP dialog '4e4cb3b53c6baef3f13b1cb76000o1u0@176.xxx.xxx.72' in 32000 ms (Method: OPTIONS)
asterisk*CLI> core set debug 5
Core debug was OFF and is now 5.
asterisk*CLI> sip set debug on
SIP Debugging re-enabled
Reliably Transmitting (no NAT) to 176.xxx.xxx.72:5060:
OPTIONS sip:176.xxx.xxx.72 SIP/2.0
Via: SIP/2.0/UDP 192.xxx.xxx.13:5060;branch=z9hG4bK5419b8d7
Max-Forwards: 70
From: "asterisk" <sip:07xxxxx78/$%7BEXTEN%7D@192.xxx.xxx.13>;tag=as7d78f056
To: <sip:176.xxx.xxx.72>
Contact: <sip:07xxxxx78/$%7BEXTEN%7D@192.xxx.xxx.13:5060>
Call-ID: 1f45555f15aa6c131fae89d8798b1d9d@192.xxx.xxx.13:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.1-cert2
Date: Wed, 22 Jun 2016 06:26:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:176.xxx.xxx.72:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.xxx.xxx.13:5060;received=176.xxx.xxx.14;branch=z9hG4bK5419b8d7;rport=5060
From: "asterisk" <sip:07xxxxx78/$%7BEXTEN%7D@192.xxx.xxx.13>;tag=as7d78f056
To: <sip:176.xxx.xxx.72>;tag=SD0nfld99-0689fdae-0017-0192-0000-0000
Call-ID: 1f45555f15aa6c131fae89d8798b1d9d@192.xxx.xxx.13:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Allow: INVITE, ACK, PRACK, CANCEL, BYE, OPTIONS, MESSAGE, NOTIFY, UPDATE, REGISTER, INFO, REFER, SUBSCRIBE
Contact: <sip:null.iIiIiI.ac138088.@172.xxx.xxx.80:5060>
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '1f45555f15aa6c131fae89d8798b1d9d@192.xxx.xxx.13:5060' Method: OPTIONS

<--- SIP read from UDP:176.xxx.xxx.72:5060 --->
INVITE sip:07xxxxx78871@176.xxx.xxx.14:5060 SIP/2.0
Via: SIP/2.0/UDP 176.xxx.xxx.72:5060;branch=z9hG4bK7tp6k600e0sh32v5s5m0.1;origin=172.xxx.xxx.80
To: <sip:07xxxxx78871@saxxx.provider.com;user=phone>
From: "07yyyyyyyyy17" <sip:07yyyyyyyyy17@provider.com;user=phone>;tag=SD8r2d301-e6d57baf
Call-ID: SD8r2d301-2c4b7b303257d8086a3e4699d580d5aa-ct46o30
CSeq: 1 INVITE
Max-Forwards: 69
Contact: <sip:07yyyyyyyyy17@176.xxx.xxx.72:5060;transport=udp>
Date: Wed, 22 Jun 2016 08:26:36 GMT
Allow: INVITE, ACK, PRACK, CANCEL, BYE, OPTIONS, MESSAGE, NOTIFY, UPDATE, REGISTER, INFO, REFER, SUBSCRIBE
Supported: resource-priority,100rel
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 237

v=0
o=- 0 22692959 IN IP4 176.xxx.xxx.72
s=IMSS
c=IN IP4 176.xxx.xxx.72
t=0 0
m=audio 55030 RTP/AVP 8 101 18 106
a=rtpmap:101 G726-32/8000
a=rtpmap:106 telephone-event/8000
a=sendrecv
a=ptime:20
a=sqn:0
a=cdsc: 1 image udptl t38
<------------->
--- (14 headers 12 lines) ---
Sending to 176.xxx.xxx.72:5060 (no NAT)
Sending to 176.xxx.xxx.72:5060 (no NAT)
Using INVITE request as basis request - SD8r2d301-2c4b7b303257d8086a3e4699d580d5aa-ct46o30
Found peer 'provider' for '07yyyyyyyyy17' from 176.xxx.xxx.72:5060
  == Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 101
Found RTP audio format 18
Found RTP audio format 106
Found audio description format G726-32 for ID 101
Found audio description format telephone-event for ID 106
Capabilities: us - (alaw|ulaw|g722|gsm|ilbc|speex|opus), peer - audio=(alaw|g729|g726)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 176.xxx.xxx.72:55030
Looking for 07xxxxx78871 in incoming (domain 176.xxx.xxx.14)
sip_route_dump: route/path hop: <sip:07yyyyyyyyy17@176.xxx.xxx.72:5060;transport=udp>

<--- Transmitting (no NAT) to 176.xxx.xxx.72:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 176.xxx.xxx.72:5060;branch=z9hG4bK7tp6k600e0sh32v5s5m0.1;origin=172.xxx.xxx.80;received=176.xxx.xxx.72
From: "07yyyyyyyyy17" <sip:07yyyyyyyyy17@provider.com;user=phone>;tag=SD8r2d301-e6d57baf
To: <sip:07xxxxx78871@saxxx.provider.com;user=phone>
Call-ID: SD8r2d301-2c4b7b303257d8086a3e4699d580d5aa-ct46o30
CSeq: 1 INVITE
Server: Asterisk PBX 13.1-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:07xxxxx78871@192.xxx.xxx.13:5060>
Content-Length: 0


<------------>
    -- Executing [07xxxxx78871@incoming:1] Dial("SIP/provider-0000039d", "SIP/871&Local/015xxxxxxxxx99@provider,30,t,k") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 10074
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding codec g722 to SDP
Adding codec gsm to SDP
Adding codec ilbc to SDP
Adding codec speex to SDP
Adding codec opus to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.xxx.xxx.172:7000:
INVITE sip:871@192.xxx.xxx.172:7000 SIP/2.0
Via: SIP/2.0/UDP 192.xxx.xxx.13:5060;branch=z9hG4bK6df6a7f9
Max-Forwards: 70
From: "07yyyyyyyyy17" <sip:07yyyyyyyyy17@192.xxx.xxx.13>;tag=as40eb4482
To: <sip:871@192.xxx.xxx.172:7000>
Contact: <sip:07yyyyyyyyy17@192.xxx.xxx.13:5060>
Call-ID: 64881a0b2e6a2a1f3317f53457ec5bb7@192.xxx.xxx.13:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.1-cert2
Date: Wed, 22 Jun 2016 06:26:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "07yyyyyyyyy17" <sip:07yyyyyyyyy17@192.xxx.xxx.13>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 430

v=0
o=root 1058146667 1058146667 IN IP4 192.xxx.xxx.13
s=Asterisk PBX 13.1-cert2
c=IN IP4 192.xxx.xxx.13
t=0 0
m=audio 10074 RTP/AVP 8 0 9 3 97 110 107 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:110 speex/8000
a=rtpmap:107 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:30
a=sendrecv

---
    -- Called SIP/871
[Jun 22 08:26:36] NOTICE[2667][C-000001a5]: core_local.c:701 local_call: No such extension/context 015xxxxxxxxx99@provider while calling Local channel
    -- Couldn't call Local/015xxxxxxxxx99@provider
Retransmitting #1 (no NAT) to 192.xxx.xxx.172:7000:
INVITE sip:871@192.xxx.xxx.172:7000 SIP/2.0
Via: SIP/2.0/UDP 192.xxx.xxx.13:5060;branch=z9hG4bK6df6a7f9
Max-Forwards: 70
From: "07yyyyyyyyy17" <sip:07yyyyyyyyy17@192.xxx.xxx.13>;tag=as40eb4482
To: <sip:871@192.xxx.xxx.172:7000>
Contact: <sip:07yyyyyyyyy17@192.xxx.xxx.13:5060>
Call-ID: 64881a0b2e6a2a1f3317f53457ec5bb7@192.xxx.xxx.13:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.1-cert2
Date: Wed, 22 Jun 2016 06:26:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "07yyyyyyyyy17" <sip:07yyyyyyyyy17@192.xxx.xxx.13>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 430

v=0
o=root 1058146667 1058146667 IN IP4 192.xxx.xxx.13
s=Asterisk PBX 13.1-cert2
c=IN IP4 192.xxx.xxx.13
t=0 0
m=audio 10074 RTP/AVP 8 0 9 3 97 110 107 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:110 speex/8000
a=rtpmap:107 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:30
a=sendrecv

---

<--- SIP read from UDP:192.xxx.xxx.172:7000 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.xxx.xxx.13:5060;branch=z9hG4bK6df6a7f9
From: "07yyyyyyyyy17" <sip:07yyyyyyyyy17@192.xxx.xxx.13>;tag=as40eb4482
To: <sip:871@192.xxx.xxx.172:7000>
Call-ID: 64881a0b2e6a2a1f3317f53457ec5bb7@192.xxx.xxx.13:5060
CSeq: 102 INVITE
Supported: replaces, path, timer
User-Agent: Grandstream GXP2140 1.0.7.25
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:192.xxx.xxx.172:7000 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.xxx.xxx.13:5060;branch=z9hG4bK6df6a7f9
From: "07yyyyyyyyy17" <sip:07yyyyyyyyy17@192.xxx.xxx.13>;tag=as40eb4482
To: <sip:871@192.xxx.xxx.172:7000>;tag=1580625090
Call-ID: 64881a0b2e6a2a1f3317f53457ec5bb7@192.xxx.xxx.13:5060
CSeq: 102 INVITE
Contact: <sip:871@192.xxx.xxx.172:7000>
Supported: replaces, path, timer
User-Agent: Grandstream GXP2140 1.0.7.25
Allow-Events: talk, hold
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:871@192.xxx.xxx.172:7000>
    -- SIP/871-0000039e is ringing

<--- Transmitting (no NAT) to 176.xxx.xxx.72:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 176.xxx.xxx.72:5060;branch=z9hG4bK7tp6k600e0sh32v5s5m0.1;origin=172.xxx.xxx.80;received=176.xxx.xxx.72
From: "07yyyyyyyyy17" <sip:07yyyyyyyyy17@provider.com;user=phone>;tag=SD8r2d301-e6d57baf
To: <sip:07xxxxx78871@saxxx.provider.com;user=phone>;tag=as7fe2f0f2
Call-ID: SD8r2d301-2c4b7b303257d8086a3e4699d580d5aa-ct46o30
CSeq: 1 INVITE
Server: Asterisk PBX 13.1-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:07xxxxx78871@192.xxx.xxx.13:5060>
Remote-Party-ID: "07xxxxx78871" <sip:07xxxxx78871@provider.com>;party=called;privacy=off;screen=no
Content-Length: 0


<------------>

<--- SIP read from UDP:192.xxx.xxx.172:7000 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.xxx.xxx.13:5060;branch=z9hG4bK6df6a7f9
From: "07yyyyyyyyy17" <sip:07yyyyyyyyy17@192.xxx.xxx.13>;tag=as40eb4482
To: <sip:871@192.xxx.xxx.172:7000>;tag=1580625090
Call-ID: 64881a0b2e6a2a1f3317f53457ec5bb7@192.xxx.xxx.13:5060
CSeq: 102 INVITE
Contact: <sip:871@192.xxx.xxx.172:7000>
Supported: replaces, path, timer
User-Agent: Grandstream GXP2140 1.0.7.25
Allow-Events: talk, hold
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:871@192.xxx.xxx.172:7000>
    -- SIP/871-0000039e is ringing
Really destroying SIP dialog '4e4cb3b53c6baef3f13b1cb76000o1u0@176.xxx.xxx.72' Method: OPTIONS

<--- SIP read from UDP:176.xxx.xxx.72:5060 --->
CANCEL sip:07xxxxx78871@176.xxx.xxx.14:5060 SIP/2.0
Via: SIP/2.0/UDP 176.xxx.xxx.72:5060;branch=z9hG4bK7tp6k600e0sh32v5s5m0.1;origin=172.xxx.xxx.80
CSeq: 1 CANCEL
To: <sip:07xxxxx78871@saxxx.provider.com;user=phone>
From: "07yyyyyyyyy17" <sip:07yyyyyyyyy17@provider.com;user=phone>;tag=SD8r2d301-e6d57baf
Call-ID: SD8r2d301-2c4b7b303257d8086a3e4699d580d5aa-ct46o30
Max-Forwards: 69
Content-Length: 0
Reason: Q.850 ;cause=16 ;text="0"

<------------->
--- (9 headers 0 lines) ---
Sending to 176.xxx.xxx.72:5060 (no NAT)

<--- Reliably Transmitting (no NAT) to 176.xxx.xxx.72:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 176.xxx.xxx.72:5060;branch=z9hG4bK7tp6k600e0sh32v5s5m0.1;origin=172.xxx.xxx.80;received=176.xxx.xxx.72
From: "07yyyyyyyyy17" <sip:07yyyyyyyyy17@provider.com;user=phone>;tag=SD8r2d301-e6d57baf
To: <sip:07xxxxx78871@saxxx.provider.com;user=phone>;tag=as7fe2f0f2
Call-ID: SD8r2d301-2c4b7b303257d8086a3e4699d580d5aa-ct46o30
CSeq: 1 INVITE
Server: Asterisk PBX 13.1-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 176.xxx.xxx.72:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 176.xxx.xxx.72:5060;branch=z9hG4bK7tp6k600e0sh32v5s5m0.1;origin=172.xxx.xxx.80;received=176.xxx.xxx.72
From: "07yyyyyyyyy17" <sip:07yyyyyyyyy17@provider.com;user=phone>;tag=SD8r2d301-e6d57baf
To: <sip:07xxxxx78871@saxxx.provider.com;user=phone>;tag=as7fe2f0f2
Call-ID: SD8r2d301-2c4b7b303257d8086a3e4699d580d5aa-ct46o30
CSeq: 1 CANCEL
Server: Asterisk PBX 13.1-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '64881a0b2e6a2a1f3317f53457ec5bb7@192.xxx.xxx.13:5060' in 32000 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 192.xxx.xxx.172:7000:
CANCEL sip:871@192.xxx.xxx.172:7000 SIP/2.0
Via: SIP/2.0/UDP 192.xxx.xxx.13:5060;branch=z9hG4bK6df6a7f9
Max-Forwards: 70
From: "07yyyyyyyyy17" <sip:07yyyyyyyyy17@192.xxx.xxx.13>;tag=as40eb4482
To: <sip:871@192.xxx.xxx.172:7000>
Call-ID: 64881a0b2e6a2a1f3317f53457ec5bb7@192.xxx.xxx.13:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 13.1-cert2
Content-Length: 0


---
Scheduling destruction of SIP dialog '64881a0b2e6a2a1f3317f53457ec5bb7@192.xxx.xxx.13:5060' in 32000 ms (Method: INVITE)
  == Spawn extension (incoming, 07xxxxx78871, 1) exited non-zero on 'SIP/provider-0000039d'

<--- SIP read from UDP:176.xxx.xxx.72:5060 --->
ACK sip:07xxxxx78871@176.xxx.xxx.14:5060 SIP/2.0
Via: SIP/2.0/UDP 176.xxx.xxx.72:5060;branch=z9hG4bK7tp6k600e0sh32v5s5m0.1;origin=172.xxx.xxx.80
CSeq: 1 ACK
To: <sip:07xxxxx78871@saxxx.provider.com;user=phone>;tag=as7fe2f0f2
From: "07yyyyyyyyy17" <sip:07yyyyyyyyy17@provider.com;user=phone>;tag=SD8r2d301-e6d57baf
Call-ID: SD8r2d301-2c4b7b303257d8086a3e4699d580d5aa-ct46o30
Max-Forwards: 69
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 'SD8r2d301-2c4b7b303257d8086a3e4699d580d5aa-ct46o30' Method: ACK

<--- SIP read from UDP:192.xxx.xxx.172:7000 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.xxx.xxx.13:5060;branch=z9hG4bK6df6a7f9
From: "07yyyyyyyyy17" <sip:07yyyyyyyyy17@192.xxx.xxx.13>;tag=as40eb4482
To: <sip:871@192.xxx.xxx.172:7000>;tag=1580625090
Call-ID: 64881a0b2e6a2a1f3317f53457ec5bb7@192.xxx.xxx.13:5060
CSeq: 102 CANCEL
Contact: <sip:871@192.xxx.xxx.172:7000>
Supported: replaces, path, timer
User-Agent: Grandstream GXP2140 1.0.7.25
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---

<--- SIP read from UDP:192.xxx.xxx.172:7000 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.xxx.xxx.13:5060;branch=z9hG4bK6df6a7f9
From: "07yyyyyyyyy17" <sip:07yyyyyyyyy17@192.xxx.xxx.13>;tag=as40eb4482
To: <sip:871@192.xxx.xxx.172:7000>;tag=1580625090
Call-ID: 64881a0b2e6a2a1f3317f53457ec5bb7@192.xxx.xxx.13:5060
CSeq: 102 INVITE
Supported: replaces, path, timer
User-Agent: Grandstream GXP2140 1.0.7.25
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Transmitting (no NAT) to 192.xxx.xxx.172:7000:
ACK sip:871@192.xxx.xxx.172:7000 SIP/2.0
Via: SIP/2.0/UDP 192.xxx.xxx.13:5060;branch=z9hG4bK6df6a7f9
Max-Forwards: 70
From: "07yyyyyyyyy17" <sip:07yyyyyyyyy17@192.xxx.xxx.13>;tag=as40eb4482
To: <sip:871@192.xxx.xxx.172:7000>;tag=1580625090
Contact: <sip:07yyyyyyyyy17@192.xxx.xxx.13:5060>
Call-ID: 64881a0b2e6a2a1f3317f53457ec5bb7@192.xxx.xxx.13:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.1-cert2
Content-Length: 0


---
Scheduling destruction of SIP dialog '64881a0b2e6a2a1f3317f53457ec5bb7@1xx.xxx.xxx.13:5060' in 32000 ms (Method: INVITE)
Really destroying SIP dialog '64881a0b2e6a2a1f3317f53457ec5bb7@1xx.xxx.xxx.13:5060' Method: INVITE
asterisk*CLI> exit

[quote=“Gwentaur, post:1, topic:67075, full:true”][Jun 22 08:26:36] NOTICE[2667][C-000001a5]: core_local.c:701 local_call:
No such extension/context 015xxxxxxxxx99@provider while calling Local
channel
– Couldn’t call Local/015xxxxxxxxx99@provider
[/quote]

The call wasn’t even attempted because the number invalid in your dialplan.

In your summary, you said both were SIP, but the failing one is a local channel and you have no context called provider.

Also, please mark log content as unformatted text (</> button) and use the log file rather than a screen scrape (log files have time stamps).

You had a retransmission to 781, so you should also look at the performance of your LAN.

Land Line is defined in SIP config. Mobile not. Not sure if mobile have to be also somehow defined in SIP config.

How I will define context “provider”? I thought that this from dial plan and SIP will be enough:

[outgoing]

;exten => _X.,1,NoOp()
;same => n,Verbose(1,Outgoing Caller ID:{$CALLERID(num)})
;same => n,Dial(SIP/${EXTEN}@provider)
;same => n,Hangup()

exten => _X.,1,NoOp()
same => n,SipAddHeader(P-Asserted-Identity: @provider>)
same => n,SIPAddHeader(Privacy:off)
same => n,Dial(SIP/${EXTEN}@provider)
same => n,Hangup()

SIP config

[provider]
context=incoming
type=peer
username=07XXXXX78
fromuser=07XXXXX78/${EXTEN}
authuser=07XXXXX78
domain=07XXXXX78.saxxx.provider.com
host=176.xxx.xxx.72
fromdomain=07xx.saxxx.provider.com
qualify=yes
insecure=port,invite
canreinvite=no

I used screen scrape because, there are another 30 guys calling through asterisk. So i will be for me almost imposible to find begining and end of that test attempt.

LAN performance problems is known to me, Im connected over VPN to main network and here is low branwith line.

You need to explain why you used a local channel before we can tell you how to use it. I rather suspect you coded Local when you should have used sip.

I Used Local, because “Astbox” advised me to use it.

astbox

If you can dial from your extension your mobile phone then use this

exten => 123,1,Dial(SIP/456&Local/789@default,20,tk)

If not post your extensions file to check your dialplan.

You didn’t follow their advice and use the default context. Their advice does assume there is a default context (probably true, and it allows calling the number you want to call (generally not advisable if that number is a chargeable number).

As I wrote on begining, I’m newbe in this topic. So default context should be this:

[default]
include => parkedcalls
include => outgoing

exten => _4XX,1,Dial(SIP/${EXTEN},20)
same => n,Hangup()

This is outgoing context
[outgoing]

;exten => _X.,1,NoOp()
;same => n,Verbose(1,Outgoing Caller ID:{$CALLERID(num)})
;same => n,Dial(SIP/${EXTEN}@provider)
;same => n,Hangup()

exten => _X.,1,NoOp()
same => n,SipAddHeader(P-Asserted-Identity: @provider>)
same => n,SIPAddHeader(Privacy:off)
same => n,Dial(SIP/${EXTEN}@provider)
same => n,Hangup()

and this is provider context

[provider]
context=incoming
type=peer
username=07XXXXX78
fromuser=07XXXXX78/${EXTEN}
authuser=07XXXXX78
domain=07XXXXX78.saxxx.provider.com
host=176.xxx.xxx.72
fromdomain=07xx.saxxx.provider.com
qualify=yes
insecure=port,invite
canreinvite=no

What I’m missing to make it working?

provider isn’t a context.

You cannot use dialplan variables in sip.conf.

You don’t have an incoming context in your dialplan.

It looks to me as though you actually need to buy consultancy here (I’m not in that market myself). Even if someone gave you a complete solution (which they probably cannot, because they would need to do more requirements analysis to work out what you really needed), I don’t think you would be able to maintain it, so you need to pay for a supported solution.