Inbound Calls

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KTelesurin

I am new with Issabel and inherited a server already configured, so far I have made a couple of trunks with some providers to make calls and I have not had problems but I can not receive calls, I already put the ip of my provider in Whitelist but they still do not see My PBX, my provider already told me that my firewall does not let the traffic go to their IPs, where should I check that?

benjange

KTelesurin hola, fíjate de haber configurado bien el did en inbound routes. Hay que ver qué está pasando haciendo un debug desde la consola de asterisk, entrando desde Linux con el comando rasterisk y luego ingresar sip set debug peer xxx, donde xxx es el nombre de la troncal. Prueba hacer un llamado entrante y envíanos el resultado del debug para poder visualizar qué ocurre con el llamado. Aguardo tus comentarios.
Saludos.

KTelesurin

benjange Hola, hice lo del debug que me dijiste

issabel*CLI> sip set debug peer trunkname
SIP Debugging Enabled for IP: XX.XX.XX.XX

<--- SIP read from UDP:XX.XX.XX.XX:5060 --->
OPTIONS sip:XX.XX.XX.XX SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK7352e3eb;rport
Max-Forwards: 70
From: "tscsw12" <sip:tscsw12@XX.XX.XX.XX>;tag=as297fb7e4
To: <sip:XX.XX.XX.XX>
Contact: <sip:tscsw12@XX.XX.XX.XX:5060>
Call-ID: 64dd1f3d08eedeec1b9c3893493c04b4@XX.XX.XX.XX
CSeq: 102 OPTIONS
User-Agent: TSCSWITCH
Date: Tue, 09 Nov 2021 00:23:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to XX.XX.XX.XX:5060 (NAT)
Looking for s in from-sip-external (domain XX.XX.XX.XX)

<--- Transmitting (NAT) to XX.XX.XX.XX:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK7352e3eb;received=XX.XX.XX.XX;rport=5060
From: "tscsw12" <sip:tscsw12@XX.XX.XX.XX>;tag=as297fb7e4
To: <sip:XX.XX.XX.XX>;tag=as3e6ab855
Call-ID: 64dd1f3d08eedeec1b9c3893493c04b4@XX.XX.XX.XX
CSeq: 102 OPTIONS
Server: IPBX-2.11.0(13.30.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:XX.XX.XX.XX:5060>
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '64dd1f3d08eedeec1b9c3893493c04b4@XX.XX.XX.XX' in 32000 ms (Method: OPTIONS)
[2021-11-08 16:26:46] NOTICE[2900][C-0000021b]: chan_sip.c:26484 handle_request_invite: Call from 'telesur05' (XX.XX.XX.XX:5060) to extension '##########' rejected because extension not f

<--- SIP read from UDP:XX.XX.XX.XX:5060 --->
INVITE sip:##########@XX.XX.XX.XX SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK4d7c73c8;rport
Max-Forwards: 70
From: "##########" <sip:##########@XX.XX.XX.XX>;tag=as6f846bd4
To: <sip:##########@XX.XX.XX.XX>
Contact: <sip:##########@XX.XX.XX.XX:5060>
Call-ID: 3585fdb555b1f38d299a418e051f2a53@XX.XX.XX.XX
CSeq: 102 INVITE
User-Agent: TSCSWITCH
Date: Tue, 09 Nov 2021 00:23:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
X-TSCCARRIER: BANDWIDTH
X-TSCINPREFIX: 1
Content-Type: application/sdp
Content-Length: 299

v=0
o=root 563020292 563020292 IN IP4 XX.XX.XX.XX
s=TSCSWITCH
c=IN IP4 XX.XX.XX.XX
t=0 0
m=audio 10988 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (16 headers 14 lines) ---
Sending to XX.XX.XX.XX:5060 (NAT)
Sending to XX.XX.XX.XX:5060 (NAT)
Using INVITE request as basis request - 3585fdb555b1f38d299a418e051f2a53@XX.XX.XX.XX
Found peer 'trunkname' for '##########' from XX.XX.XX.XX:5060
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|gsm|alaw|g729), peer - audio=(ulaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port XX.XX.XX.XX:10988
Looking for ########## in trunkinbound (domain XX.XX.XX.XX)

<--- Reliably Transmitting (NAT) to XX.XX.XX.XX:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK4d7c73c8;received=XX.XX.XX.XX;rport=5060
From: "##########" <sip:##########@XX.XX.XX.XX>;tag=as6f846bd4
To: <sip:##########@XX.XX.XX.XX>;tag=as20db2480
Call-ID: 3585fdb555b1f38d299a418e051f2a53@XX.XX.XX.XX
CSeq: 102 INVITE
Server: IPBX-2.11.0(13.30.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
[2021-11-08 16:26:47] NOTICE[2900][C-0000021c]: chan_sip.c:26484 handle_request_invite: Call from 'trunkname' (XX.XX.XX.XX:5060) to extension '##########' rejected because extension not f
Scheduling destruction of SIP dialog '3585fdb555b1f38d299a418e051f2a53@XX.XX.XX.XX' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:XX.XX.XX.XX:5060 --->
ACK sip:##########@XX.XX.XX.XX SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK4d7c73c8;rport
Max-Forwards: 70
From: "##########" <sip:##########@XX.XX.XX.XX>;tag=as6f846bd4
To: <sip:##########@XX.XX.XX.XX>;tag=as20db2480
Contact: <sip:##########@XX.XX.XX.XX:5060>
Call-ID: 3585fdb555b1f38d299a418e051f2a53@XX.XX.XX.XX
CSeq: 102 ACK
User-Agent: TSCSWITCH
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '3585fdb555b1f38d299a418e051f2a53@XX.XX.XX.XX' Method: ACK
[2021-11-08 16:26:47] NOTICE[2900][C-0000021d]: chan_sip.c:26484 handle_request_invite: Call from 'telesur05' (XX.XX.XX.XX:5060) to extension '##########' rejected because extension not f

<--- SIP read from UDP:XX.XX.XX.XX:5060 --->
INVITE sip:##########@XX.XX.XX.XX SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK7a6a7bd0;rport
Max-Forwards: 70
From: "##########" <sip:##########@XX.XX.XX.XX>;tag=as117a4405
To: <sip:##########@XX.XX.XX.XX>
Contact: <sip:##########@XX.XX.XX.XX:5060>
Call-ID: 5561c7427753b8d52baeb76947b3bb3a@XX.XX.XX.XX
CSeq: 102 INVITE
User-Agent: TSCSWITCH
Date: Tue, 09 Nov 2021 00:23:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
X-TSCCARRIER: BANDWIDTH
X-TSCINPREFIX: 1
Content-Type: application/sdp
Content-Length: 301

v=0
o=root 1624529026 1624529026 IN IP4 XX.XX.XX.XX
s=TSCSWITCH
c=IN IP4 XX.XX.XX.XX
t=0 0
m=audio 10266 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (16 headers 14 lines) ---
Sending to XX.XX.XX.XX:5060 (NAT)
Sending to XX.XX.XX.XX:5060 (NAT)
Using INVITE request as basis request - 5561c7427753b8d52baeb76947b3bb3a@XX.XX.XX.XX
Found peer 'trunkname' for '##########' from XX.XX.XX.XX:5060
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|gsm|alaw|g729), peer - audio=(ulaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port XX.XX.XX.XX:10266
Looking for ########## in trunkinbound (domain XX.XX.XX.XX)

<--- Reliably Transmitting (NAT) to XX.XX.XX.XX:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK7a6a7bd0;received=XX.XX.XX.XX;rport=5060
From: "##########" <sip:##########@XX.XX.XX.XX>;tag=as117a4405
To: <sip:##########@XX.XX.XX.XX>;tag=as4d77f577
Call-ID: 5561c7427753b8d52baeb76947b3bb3a@XX.XX.XX.XX
CSeq: 102 INVITE
Server: IPBX-2.11.0(13.30.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
[2021-11-08 16:26:48] NOTICE[2900][C-0000021e]: chan_sip.c:26484 handle_request_invite: Call from 'trunkname' (XX.XX.XX.XX:5060) to extension '##########' rejected because extension not f
Scheduling destruction of SIP dialog '5561c7427753b8d52baeb76947b3bb3a@XX.XX.XX.XX' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:XX.XX.XX.XX:5060 --->
ACK sip:##########@XX.XX.XX.XX SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK7a6a7bd0;rport
Max-Forwards: 70
From: "##########" <sip:##########@XX.XX.XX.XX>;tag=as117a4405
To: <sip:##########@XX.XX.XX.XX>;tag=as4d77f577
Contact: <sip:##########@XX.XX.XX.XX:5060>
Call-ID: 5561c7427753b8d52baeb76947b3bb3a@XX.XX.XX.XX
CSeq: 102 ACK
User-Agent: TSCSWITCH
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '5561c7427753b8d52baeb76947b3bb3a@XX.XX.XX.XX' Method: ACK
[2021-11-08 16:26:49] NOTICE[2900][C-0000021f]: chan_sip.c:26484 handle_request_invite: Call from 'telesur05' (XX.XX.XX.XX:5060) to extension '##########' rejected because extension not f

<--- SIP read from UDP:XX.XX.XX.XX:5060 --->
INVITE sip:##########@XX.XX.XX.XX SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK7fa25286;rport
Max-Forwards: 70
From: "##########" <sip:##########@XX.XX.XX.XX>;tag=as18e0e361
To: <sip:##########@XX.XX.XX.XX>
Contact: <sip:##########@XX.XX.XX.XX:5060>
Call-ID: 4c5592417e4f7cad6f761e756ca284b8@XX.XX.XX.XX
CSeq: 102 INVITE
User-Agent: TSCSWITCH
Date: Tue, 09 Nov 2021 00:23:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
X-TSCCARRIER: BANDWIDTH
X-TSCINPREFIX: 1
Content-Type: application/sdp
Content-Length: 301

v=0
o=root 2060256663 2060256663 IN IP4 XX.XX.XX.XX
s=TSCSWITCH
c=IN IP4 XX.XX.XX.XX
t=0 0
m=audio 12092 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (16 headers 14 lines) ---
Sending to XX.XX.XX.XX:5060 (NAT)
Sending to XX.XX.XX.XX:5060 (NAT)
Using INVITE request as basis request - 4c5592417e4f7cad6f761e756ca284b8@XX.XX.XX.XX
Found peer 'trunkname' for '##########' from XX.XX.XX.XX:5060
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|gsm|alaw|g729), peer - audio=(ulaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port XX.XX.XX.XX:12092
Looking for ########## in trunkinbound (domain XX.XX.XX.XX)

<--- Reliably Transmitting (NAT) to XX.XX.XX.XX:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK7fa25286;received=XX.XX.XX.XX;rport=5060
From: "##########" <sip:##########@XX.XX.XX.XX>;tag=as18e0e361
To: <sip:##########@XX.XX.XX.XX>;tag=as6fd81948
Call-ID: 4c5592417e4f7cad6f761e756ca284b8@XX.XX.XX.XX
CSeq: 102 INVITE
Server: IPBX-2.11.0(13.30.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
[2021-11-08 16:26:50] NOTICE[2900][C-00000220]: chan_sip.c:26484 handle_request_invite: Call from 'trunkname' (XX.XX.XX.XX:5060) to extension '##########' rejected because extension not f
Scheduling destruction of SIP dialog '4c5592417e4f7cad6f761e756ca284b8@XX.XX.XX.XX' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:XX.XX.XX.XX:5060 --->
ACK sip:##########@XX.XX.XX.XX SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK7fa25286;rport
Max-Forwards: 70
From: "##########" <sip:##########@XX.XX.XX.XX>;tag=as18e0e361
To: <sip:##########@XX.XX.XX.XX>;tag=as6fd81948
Contact: <sip:##########@XX.XX.XX.XX:5060>
Call-ID: 4c5592417e4f7cad6f761e756ca284b8@XX.XX.XX.XX
CSeq: 102 ACK
User-Agent: TSCSWITCH
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '4c5592417e4f7cad6f761e756ca284b8@XX.XX.XX.XX' Method: ACK

KTelesurin

Hola, acabo de descubrir que las ips de mi proveedor están en Banned IPS de mi Issabel, las quite pero en automático se volvieron a poner, como puedo hacer que no se agreguen en automático?

benjange

KTelesurin https://forum.issabel.org/d/3116-fail2ban-bloquea-ip-servidor/6

benjange

KTelesurin hola, el problema es que no reconoce la ruta entrante. No se cual es no qué número intenta enviar tu operador, porque está todo comentado con #.. en el debug se puede ver “ rejected because extension not found” chequea que la ruta entrante tenga el mismo número del que te envía el proveedor y avisa cómo te fue.
Saludos.

KTelesurin

Hola, Gracias ya encontré el detalle. Fue el contexto de la troncal, estaba como trunkinbound y lo cambie a from-trunk, con eso funciono.