Solved call between 3CX and Asterisk : 3CX add a 0 on caller id

[amart]

Sorry for my English.

I have a Asterisk server and a 3CX server.
Phones on asterisk can call phones on 3CX but 3CX phones can't call Asterisk extensions...

On Asterisk i see :
SIP/15792792-000215b9' sent to invalid extension but no invalid handler: context,exten,priority=ext-did,0140,1

I have this message because 3CX or my Asterisk add a 0 on destination extension, for example when i call 140, Asterisk receive 0140 and don't find this extension.

It's just for call extension with a length of 3 number, with 4 number extension it's work

 

[Rhys Haskell]

Can you run a trace from both sides to see what is being sent, I suspect 3CX is sending the correct extension unless it is set to prepend a 0 in the outbound rules. A trace will prove this.

 

[amart]

On 3CX i have this :

07/15/2019 3:30:32 PM - Leg L:96.1[Extn:502] is terminated: Cause: BYE from PBX
07/15/2019 3:30:32 PM - Leg L:96.2[Line:15792792>>140] is terminated: Cause: 603 Declined/INVITE from IP_ASTERISK:5060
07/15/2019 3:30:32 PM - [CM503020]: Call(C:96): Normal call termination. Call originator: Extn:502. Reason: Forbidden
07/15/2019 3:30:32 PM - L:96.1[Extn:502] failed to reach Line:15792792>>140, reason Forbidden
07/15/2019 3:30:32 PM - Call to T:Line:15792792>>140@[Dev:sip:15792792@IP_ASTERISK:5060] from L:96.1[Extn:502] failed, cause: Cause: 603 Declined/INVITE from IP_ASTERISK:5060
07/15/2019 3:30:32 PM - [CM503003]: Call(C:96): Call to <sip:140@IP_ASTERISK:5060> has failed; Cause: 603 Declined/INVITE from IP_ASTERISK:5060
07/15/2019 3:30:32 PM - [CM503025]: Call(C:96): Calling T:Line:15792792>>140@[Dev:sip:15792792@IP_ASTERISK:5060] for L:96.1[Extn:502]
07/15/2019 3:30:32 PM - [CM503027]: Call(C:96): From: Extn:502 ("alexis alexis" <sip:[email protected]:5060>)  to  T:Line:15792792>>140@[Dev:sip:15792792@IP_ASTERISK:5060]
07/15/2019 3:30:32 PM - [CM503004]: Call(C:96): Route 1: from L:96.1[Extn:502] to T:Line:15792792>>140@[Dev:sip:15792792@IP_ASTERISK:5060]
07/15/2019 3:30:32 PM - Line limit check: Current # of calls for line Lc:15792792(@15792792[<sip:15792792@IP_ASTERISK:5060>]) is 0; limit is 10
07/15/2019 3:30:32 PM - Call(C:96): Call from Extn:502 to 140 matches outbound rule 'FR_call'
07/15/2019 3:30:32 PM - [Flow] Call(C:96): has built target endpoint: Out#:>>Rule{FR_call}>>140 for call from L:96.1[Extn:502]
07/15/2019 3:30:32 PM - [Flow] Target endpoint for 140 is Out#:>>Rule{FR_call}>>140
07/15/2019 3:30:32 PM - [CM503010]: Call(C:96): Making route(s) from Extn:502 to <sip:[email protected]:5060>
07/15/2019 3:30:32 PM - [CM505001]: Endpoint Extn:502: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXPhone for Android 16.0.0.35] PBX contact: [sip:[email protected]:5060]
07/15/2019 3:30:32 PM - [CM500002]: Call(C:96): Info on incoming INVITE from Extn:502:
Invite-IN Recv Req INVITE from 10.0.0.152:42679 tid=Pja0c75ca8-42ce-45a8-bd6f-e832ba85e2e8 Call-ID=a0f1e6f0-cda9-4a7b-b9f2-f6ed2db74ac2:
INVITE sip:[email protected];tag3cx=a15b3549-8ac7-4ec5-af0d-ed67f613176f SIP/2.0
Via: SIP/2.0/UDP 10.0.0.152:42679;rport=42679;branch=z9hG4bKPja0c75ca8-42ce-45a8-bd6f-e832ba85e2e8
Max-Forwards: 70
Contact: "alexis alexis" <sip:[email protected]:42679;inst="1b9448fc";rinstance=0-32f0a28c-d52a-4a65-8214-5ad4947aaade;ob>
To: <sip:[email protected];tag3cx=a15b3549-8ac7-4ec5-af0d-ed67f613176f>
From: "alexis alexis"<sip:[email protected]>;tag=06e5b180-2845-4150-b2e9-e8ef6181c2a2
Call-ID: a0f1e6f0-cda9-4a7b-b9f2-f6ed2db74ac2
CSeq: 7897 INVITE
Session-Expires: 1800
Min-SE: 90
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Type: application/sdp
Proxy-Authorization: Digest username="30U6Hre4tz",realm="3CXPhoneSystem",nonce="414d53595d2c7ff876:3c6c9f7aaa45e44de1b509cead058e09",uri="sip:[email protected];tag3cx=a15b3549-8ac7-4ec5-af0d-ed67f613176f",response="900d0a4e22fd35a59590a14f175074da",algorithm=MD5
Supported: replaces, 100rel, timer, norefersub
User-Agent: 3CXPhone for Android 16.0.0.35
Content-Length: 403

v=0
o=- 3772186231 3772186231 IN IP4 10.0.0.152
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 42008 RTP/AVP 0 8 104 3 18 9 96
c=IN IP4 10.0.0.152
b=TIAS:64000
b=AS:84
a=rtcp:42009
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:104 iLBC/8000
a=fmtp:104 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
07/15/2019 3:30:32 PM - [CM503001]: Call(C:96): Incoming call from Extn:502 to <sip:[email protected]:5060>

And on Asterisk :

    -- Executing [140@from-trunk:1] Set("SIP/15792792-000216ff", "CALLERID(num)=0502") in new stack
    -- Executing [140@from-trunk:2] Set("SIP/15792792-000216ff", "CALLERID(name)=0alexis alexis") in new stack
    -- Executing [140@from-trunk:3] Goto("SIP/15792792-000216ff", "ext-did,0140,1") in new stack
    -- Goto (ext-did,0140,1)
[2019-07-15 15:33:53] WARNING[10097][C-000262f8]: pbx.c:6492 __ast_pbx_run: Channel 'SIP/15792792-000216ff' sent to invalid extension but no invalid handler: context,exten,priority=ext-did,0140,1
[2019-07-15 15:33:53] WARNING[10097][C-000262f8]: pbx.c:6492 __ast_pbx_run: Channel 'SIP/15792792-000216ff' sent to invalid extension but no invalid handler: context,exten,priority=ext-did,0140,1

On router i have this (asterisk is an external server) :

13:38:10.689631 IP IP_3CX.sip > IP_Asterisk.sip: SIP: INVITE sip:140@IP_Asterisk:5060 SIP/2.0
13:38:10.700575 IP IP_Asterisk.sip > IP_3CX.sip: SIP: SIP/2.0 100 Trying
13:38:10.701888 IP IP_Asterisk.sip > IP_3CX.sip: SIP: SIP/2.0 603 Declined
13:38:10.702426 IP IP_3CX.sip > IP_Asterisk.sip: SIP: ACK sip:140@IP_Asterisk:5060 SIP/2.0

IP_Asterisk and IP_3CX are publics address.
I just see the additional 0 on Asterisk log...

 

[Rhys Haskell]

This would suggest that Asterisk is reformatting this not 3CX. On your inbound route on Asterisk do you have it to prepend a 0?

 

[amart]

I don't prepend a 0 on inbound route, it's why i don't understand this 0.

I use the software FreePBX to manage my Asterisk.
I check on the web interface and on the file extensions.conf (real name on FreePBX : extensions_additional.conf)

To test I create an extensions 0140, and when i call 0140 the phone is ringing ... Also i think 0 is prepend just for call number with a length of 3 number

 

[amart]

And logically if Asterisk prepend a 0, when i call the 140, my phone with 0140 extensions would ring ...
But no, it's when i call 0140 ...

 

[Rhys Haskell]

If you just call 10 for example does it reformat it to 010 or 0010? The Asterisk server may have some sort of minimum length for external calls.

 

[amart]

When i call 10, it remate to 010 :
-- Executing [10@from-trunk:1] Set("SIP/15792792-000217e8", "CALLERID(num)=0502") in new stack -- Executing [10@from-trunk:2] Set("SIP/15792792-000217e8", "CALLERID(name)=0alexis alexis") in new stack -- Executing [10@from-trunk:3] Goto("SIP/15792792-000217e8", "ext-did,010,1") in new stack -- Goto (ext-did,010,1) [2019-07-15 16:32:26] WARNING[17604][C-000263a6]: pbx.c:6492 __ast_pbx_run: Channel 'SIP/15792792-000217e8' sent to invalid extension but no invalid handler: context,exten,priority=ext-did,010,1 [2019-07-15 16:32:26] WARNING[17604][C-000263a6]: pbx.c:6492 __ast_pbx_run: Channel 'SIP/15792792-000217e8' sent to invalid extension but no invalid handler: context,exten,priority=ext-did,010,1

With this result we can conclude than external number isn't reformate with 4 number, no ?

 

[amart]

Now it's work, it's an inbound route we prepend a 0.
Thank you !!!
Now, I just need to find how to call to external number from 3CX and everything is working

 

[JohnS_3CX]

Hi amart

Have you checked out this guide? https://www.3cx.com/docs/

 

[amart]

Hi JohnS,

I begin my configuration with this guide, but don't work, but i try many configuration, and now my configuration is like this guide (with much parameters for my asterisk). So I think it's my fault, maybe a syntax error.

But currently, i just can call external number registered in my asterisk, maybe a parameters to add in my Asterisk i think.