No incoming calls when everything else is ok

Asterisk Asterisk SIP
1

Hello,

I’ve just configured 2 asterisk servers and i can call outside and use the iax communication between the 2 servers.

when i call asterisk from outside, no response.

i called with my smartphone 0796684485

Here my pjsip_wizard and my extensions files

(i cannot paste the PJSIP log, this forum says no more than 2 links for new users…)

[peoplefoneO77]
type = wizard
sends_auth = yes
sends_registrations = yes
remote_hosts = sips.peoplefone.ch
; 0215105046
outbound_auth/username = 90791027947
outbound_auth/password = xxxxxxxxxxx

my extensions.conf

[from-external]
exten => s,1,Dial(PJSIP/201&PJSIP/202&PJSIP/203&PJSIP/204&PJSIP/205&PJSIP/206&PJSIP/207&PJSIP/208&PJSIP/209&PJSIP/211&PJSIP/2>
same => n,Voicemail(200@voicemailLS,u)
same => n,Hangup()

;exten => _X.,1,Goto(s,1)

[from-internal]

exten => 200,1,Goto(from-external,s,1)

exten => 210,1,Dial(PJSIP/211&PJSIP/212&PJSIP/213,20)
same => n,Hangup()

exten => _2XX,1,Dial(PJSIP/${EXTEN},20)
same => n,Hangup()

exten => _3XX,1,Dial(IAX2/ge/${EXTEN},20)
same => n,Hangup()

exten => _1X.,1,Dial(PJSIP/${EXTEN}@peoplefoneO77,60,r)
exten => _0XXXXX.,1,Dial(PJSIP/${EXTEN}@peoplefoneO77,60,r)

exten => 299,1,Answer
exten => 299,2,VoiceMailMain(200@voicemailLS,s)
exten => 299,hint,Custom:groupmwi

I’m lost… Can anyone help me please ?

2

I’ve raised your level so you should be able to post more links. Even with out that, if you see no SIP INVITE on an incoming call attempt then it’s not getting to Asterisk which can point to a firewall issue outside of Asterisk, such as a NAT mapping expiring.

3

Thanks !

i’ve just found an intersting thing

if i call with my smartphone (0796684485), it doesn’t work

But if i call with a softphone, it works….

Here my PJSIP log when calling with my smartphone

PJSIP Logging enabled
<— Received SIP request (294 bytes) from UDP:95.128.80.3:5060 —>
OPTIONS sip:s@192.168.0.79:5060 SIP/2.0
Via: SIP/2.0/UDP 95.128.80.3:5060;branch=z9hG4bK6111031
From: sip:ping@peoplefone.com;tag=uloc-678d7a18-6b88-b0fff671-c82dc757-be216871
To: sip:s@192.168.0.79:5060
Call-ID: ca55c7f6-2b83b181-8cef6f6@95.128.80.3
CSeq: 1 OPTIONS
Content-Length: 0

<— Received SIP request (300 bytes) from UDP:95.128.80.3:5060 —>
OPTIONS sip:s@188.154.136.126:5060 SIP/2.0
Via: SIP/2.0/UDP 95.128.80.3:5060;branch=z9hG4bK5921831
From: sip:ping@peoplefone.com;tag=uloc-678d7a18-6b82-04650771-c82dc757-ce216871
To: sip:s@188.154.136.126:5060
Call-ID: ca55c7f6-3b83b181-8cef6f6@95.128.80.3
CSeq: 1 OPTIONS
Content-Length: 0

<— Transmitting SIP response (843 bytes) to UDP:95.128.80.3:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 95.128.80.3:5060;rport=5060;received=95.128.80.3;branch=z9hG4bK5921831
Call-ID: ca55c7f6-3b83b181-8cef6f6@95.128.80.3
From: sip:ping@peoplefone.com;tag=uloc-678d7a18-6b82-04650771-c82dc757-ce216871
To: sip:s@188.154.136.126;tag=z9hG4bK5921831
CSeq: 1 OPTIONS
Accept: application/xpidf+xml, application/cpim-pidf+xml, application/pidf+xml, application/simple-message-summary, application/dialog-info+xml, application/simple-message-summary, application/pidf+xml, application/dialog-info+xml, application/sdp, message/sipfrag;version=2.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, INFO
Supported: 100rel, timer, replaces, norefersub
Accept-Encoding: identity
Accept-Language: en
Server: Asterisk PBX 22.5.1
Content-Length: 0

<— Transmitting SIP response (840 bytes) to UDP:95.128.80.3:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 95.128.80.3:5060;rport=5060;received=95.128.80.3;branch=z9hG4bK6111031
Call-ID: ca55c7f6-2b83b181-8cef6f6@95.128.80.3
From: sip:ping@peoplefone.com;tag=uloc-678d7a18-6b88-b0fff671-c82dc757-be216871
To: sip:s@192.168.0.79;tag=z9hG4bK6111031
CSeq: 1 OPTIONS
Accept: application/xpidf+xml, application/cpim-pidf+xml, application/pidf+xml, application/simple-message-summary, application/dialog-info+xml, application/simple-message-summary, application/pidf+xml, application/dialog-info+xml, application/sdp, message/sipfrag;version=2.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, INFO
Supported: 100rel, timer, replaces, norefersub
Accept-Encoding: identity
Accept-Language: en
Server: Asterisk PBX 22.5.1
Content-Length: 0

<— Received SIP request (988 bytes) from UDP:95.128.80.3:5060 —>
INVITE sip:s@192.168.0.79:5060 SIP/2.0
Record-Route: sip:95.128.80.3;r2=on;lr=on;did=76.fae9
Record-Route: sip:95.128.80.5;r2=on;lr=on;did=76.fae9
Call-ID: 30cd0fbf93dcf886fa76f4e3600825f5@95.128.84.2
CSeq: 1 INVITE
From: “0796684485” sip:0796684485@95.128.84.2;tag=1761649906779
To: sip:90791027947@95.128.80.5:5060
Via: SIP/2.0/UDP 95.128.80.3;branch=z9hG4bKe90e.d2f40d70298e2a27754211203f31155a.0
Via: SIP/2.0/UDP 95.128.84.2:5060;branch=z9hG4bK-3630-6ad01305717588620f3af14ff493f988
Max-Forwards: 69
Contact: sip:41796684485@95.128.84.2:5060
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,INFO,REFER,NOTIFY
X-CID: 1812341810_122422166@213.215.176.143
Content-Type: application/sdp
Content-Length: 266

v=0
o=Sonus_UAC 24004 19664 IN IP4 95.128.80.6
s=SIP Media Capabilities
c=IN IP4 95.128.80.6
t=0 0
m=audio 33564 RTP/AVP 9 8 97
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=sendrecv
a=rtcp:33565
a=ptime:20

<— Received SIP request (991 bytes) from UDP:95.128.80.3:5060 —>
INVITE sip:s@188.154.136.126:5060 SIP/2.0
Record-Route: sip:95.128.80.3;r2=on;lr=on;did=76.fae9
Record-Route: sip:95.128.80.5;r2=on;lr=on;did=76.fae9
Call-ID: 30cd0fbf93dcf886fa76f4e3600825f5@95.128.84.2
CSeq: 1 INVITE
From: “0796684485” sip:0796684485@95.128.84.2;tag=1761649906779
To: sip:90791027947@95.128.80.5:5060
Via: SIP/2.0/UDP 95.128.80.3;branch=z9hG4bKe90e.d2f40d70298e2a27754211203f31155a.1
Via: SIP/2.0/UDP 95.128.84.2:5060;branch=z9hG4bK-3630-6ad01305717588620f3af14ff493f988
Max-Forwards: 69
Contact: sip:41796684485@95.128.84.2:5060
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,INFO,REFER,NOTIFY
X-CID: 1812341810_122422166@213.215.176.143
Content-Type: application/sdp
Content-Length: 266

v=0
o=Sonus_UAC 24004 19664 IN IP4 95.128.80.6
s=SIP Media Capabilities
c=IN IP4 95.128.80.6
t=0 0
m=audio 33564 RTP/AVP 9 8 97
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=sendrecv
a=rtcp:33565
a=ptime:20

<— Transmitting SIP response (556 bytes) to UDP:95.128.80.3:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 95.128.80.3;rport=5060;received=95.128.80.3;branch=z9hG4bKe90e.d2f40d70298e2a27754211203f31155a.0
Via: SIP/2.0/UDP 95.128.84.2:5060;branch=z9hG4bK-3630-6ad01305717588620f3af14ff493f988
Record-Route: sip:95.128.80.3;lr;r2=on;did=76.fae9
Record-Route: sip:95.128.80.5;lr;r2=on;did=76.fae9
Call-ID: 30cd0fbf93dcf886fa76f4e3600825f5@95.128.84.2
From: “0796684485” sip:0796684485@95.128.84.2;tag=1761649906779
To: sip:90791027947@95.128.80.5
CSeq: 1 INVITE
Server: Asterisk PBX 22.5.1
Content-Length: 0

<— Transmitting SIP response (610 bytes) to UDP:95.128.80.3:5060 —>
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 95.128.80.3;rport=5060;received=95.128.80.3;branch=z9hG4bKe90e.d2f40d70298e2a27754211203f31155a.0
Via: SIP/2.0/UDP 95.128.84.2:5060;branch=z9hG4bK-3630-6ad01305717588620f3af14ff493f988
Record-Route: sip:95.128.80.3;lr;r2=on;did=76.fae9
Record-Route: sip:95.128.80.5;lr;r2=on;did=76.fae9
Call-ID: 30cd0fbf93dcf886fa76f4e3600825f5@95.128.84.2
From: “0796684485” sip:0796684485@95.128.84.2;tag=1761649906779
To: sip:90791027947@95.128.80.5;tag=516f5512-432a-40eb-888b-379afc30337a
CSeq: 1 INVITE
Server: Asterisk PBX 22.5.1
Content-Length: 0

<— Transmitting SIP response (556 bytes) to UDP:95.128.80.3:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 95.128.80.3;rport=5060;received=95.128.80.3;branch=z9hG4bKe90e.d2f40d70298e2a27754211203f31155a.1
Via: SIP/2.0/UDP 95.128.84.2:5060;branch=z9hG4bK-3630-6ad01305717588620f3af14ff493f988
Record-Route: sip:95.128.80.3;lr;r2=on;did=76.fae9
Record-Route: sip:95.128.80.5;lr;r2=on;did=76.fae9
Call-ID: 30cd0fbf93dcf886fa76f4e3600825f5@95.128.84.2
From: “0796684485” sip:0796684485@95.128.84.2;tag=1761649906779
To: sip:90791027947@95.128.80.5
CSeq: 1 INVITE
Server: Asterisk PBX 22.5.1
Content-Length: 0

<— Transmitting SIP response (610 bytes) to UDP:95.128.80.3:5060 —>
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 95.128.80.3;rport=5060;received=95.128.80.3;branch=z9hG4bKe90e.d2f40d70298e2a27754211203f31155a.1
Via: SIP/2.0/UDP 95.128.84.2:5060;branch=z9hG4bK-3630-6ad01305717588620f3af14ff493f988
Record-Route: sip:95.128.80.3;lr;r2=on;did=76.fae9
Record-Route: sip:95.128.80.5;lr;r2=on;did=76.fae9
Call-ID: 30cd0fbf93dcf886fa76f4e3600825f5@95.128.84.2
From: “0796684485” sip:0796684485@95.128.84.2;tag=1761649906779
To: sip:90791027947@95.128.80.5;tag=ec8f9bc7-a74f-462d-951c-934ac8b2a193
CSeq: 1 INVITE
Server: Asterisk PBX 22.5.1
Content-Length: 0

<— Received SIP request (371 bytes) from UDP:95.128.80.3:5060 —>
ACK sip:s@192.168.0.79:5060 SIP/2.0
Call-ID: 30cd0fbf93dcf886fa76f4e3600825f5@95.128.84.2
CSeq: 1 ACK
From: “0796684485” sip:0796684485@95.128.84.2;tag=1761649906779
To: sip:90791027947@95.128.80.5;tag=516f5512-432a-40eb-888b-379afc30337a
Via: SIP/2.0/UDP 95.128.80.3;branch=z9hG4bKe90e.d2f40d70298e2a27754211203f31155a.0
Max-Forwards: 69
Content-Length: 0

<— Received SIP request (374 bytes) from UDP:95.128.80.3:5060 —>
ACK sip:s@188.154.136.126:5060 SIP/2.0
Call-ID: 30cd0fbf93dcf886fa76f4e3600825f5@95.128.84.2
CSeq: 1 ACK
From: “0796684485” sip:0796684485@95.128.84.2;tag=1761649906779
To: sip:90791027947@95.128.80.5;tag=ec8f9bc7-a74f-462d-951c-934ac8b2a193
Via: SIP/2.0/UDP 95.128.80.3;branch=z9hG4bKe90e.d2f40d70298e2a27754211203f31155a.1
Max-Forwards: 69
Content-Length: 0

<— Received SIP request (989 bytes) from UDP:95.128.80.3:5060 —>
INVITE sip:s@192.168.0.79:5060 SIP/2.0
Record-Route: sip:95.128.80.3;r2=on;lr=on;did=c65.1527
Record-Route: sip:95.128.80.5;r2=on;lr=on;did=c65.1527
Call-ID: 4ad43a5f356a762051e954638ef2954b@95.128.84.2
CSeq: 1 INVITE
From: “0796684485” sip:0796684485@95.128.84.2;tag=1761649908623
To: sip:90791027947@95.128.80.5:5060
Via: SIP/2.0/UDP 95.128.80.3;branch=z9hG4bK3a55.43ea7af7f82e7b6de869db6d4a2109a5.0
Via: SIP/2.0/UDP 95.128.84.2:5060;branch=z9hG4bK-3630-0b8a0f867b0b9cf205d58ca4ebc792fb
Max-Forwards: 69
Contact: sip:41796684485@95.128.84.2:5060
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,INFO,REFER,NOTIFY
X-CID: 252497407_75793784@212.23.246.82
Content-Type: application/sdp
Content-Length: 269

v=0
o=Sonus_UAC 18535 19196 IN IP4 95.128.80.6
s=SIP Media Capabilities
c=IN IP4 95.128.80.6
t=0 0
m=audio 58680 RTP/AVP 9 8 100
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=sendrecv
a=rtcp:58681
a=ptime:20

<— Received SIP request (992 bytes) from UDP:95.128.80.3:5060 —>
INVITE sip:s@188.154.136.126:5060 SIP/2.0
Record-Route: sip:95.128.80.3;r2=on;lr=on;did=c65.1527
Record-Route: sip:95.128.80.5;r2=on;lr=on;did=c65.1527
Call-ID: 4ad43a5f356a762051e954638ef2954b@95.128.84.2
CSeq: 1 INVITE
From: “0796684485” sip:0796684485@95.128.84.2;tag=1761649908623
To: sip:90791027947@95.128.80.5:5060
Via: SIP/2.0/UDP 95.128.80.3;branch=z9hG4bK3a55.43ea7af7f82e7b6de869db6d4a2109a5.1
Via: SIP/2.0/UDP 95.128.84.2:5060;branch=z9hG4bK-3630-0b8a0f867b0b9cf205d58ca4ebc792fb
Max-Forwards: 69
Contact: sip:41796684485@95.128.84.2:5060
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,INFO,REFER,NOTIFY
X-CID: 252497407_75793784@212.23.246.82
Content-Type: application/sdp
Content-Length: 269

v=0
o=Sonus_UAC 18535 19196 IN IP4 95.128.80.6
s=SIP Media Capabilities
c=IN IP4 95.128.80.6
t=0 0
m=audio 58680 RTP/AVP 9 8 100
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=sendrecv
a=rtcp:58681
a=ptime:20

<— Transmitting SIP response (558 bytes) to UDP:95.128.80.3:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 95.128.80.3;rport=5060;received=95.128.80.3;branch=z9hG4bK3a55.43ea7af7f82e7b6de869db6d4a2109a5.0
Via: SIP/2.0/UDP 95.128.84.2:5060;branch=z9hG4bK-3630-0b8a0f867b0b9cf205d58ca4ebc792fb
Record-Route: sip:95.128.80.3;lr;r2=on;did=c65.1527
Record-Route: sip:95.128.80.5;lr;r2=on;did=c65.1527
Call-ID: 4ad43a5f356a762051e954638ef2954b@95.128.84.2
From: “0796684485” sip:0796684485@95.128.84.2;tag=1761649908623
To: sip:90791027947@95.128.80.5
CSeq: 1 INVITE
Server: Asterisk PBX 22.5.1
Content-Length: 0

<— Transmitting SIP response (612 bytes) to UDP:95.128.80.3:5060 —>
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 95.128.80.3;rport=5060;received=95.128.80.3;branch=z9hG4bK3a55.43ea7af7f82e7b6de869db6d4a2109a5.0
Via: SIP/2.0/UDP 95.128.84.2:5060;branch=z9hG4bK-3630-0b8a0f867b0b9cf205d58ca4ebc792fb
Record-Route: sip:95.128.80.3;lr;r2=on;did=c65.1527
Record-Route: sip:95.128.80.5;lr;r2=on;did=c65.1527
Call-ID: 4ad43a5f356a762051e954638ef2954b@95.128.84.2
From: “0796684485” sip:0796684485@95.128.84.2;tag=1761649908623
To: sip:90791027947@95.128.80.5;tag=2c56dd8d-29f5-4295-b2dc-7774f45cee2a
CSeq: 1 INVITE
Server: Asterisk PBX 22.5.1
Content-Length: 0

<— Transmitting SIP response (558 bytes) to UDP:95.128.80.3:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 95.128.80.3;rport=5060;received=95.128.80.3;branch=z9hG4bK3a55.43ea7af7f82e7b6de869db6d4a2109a5.1
Via: SIP/2.0/UDP 95.128.84.2:5060;branch=z9hG4bK-3630-0b8a0f867b0b9cf205d58ca4ebc792fb
Record-Route: sip:95.128.80.3;lr;r2=on;did=c65.1527
Record-Route: sip:95.128.80.5;lr;r2=on;did=c65.1527
Call-ID: 4ad43a5f356a762051e954638ef2954b@95.128.84.2
From: “0796684485” sip:0796684485@95.128.84.2;tag=1761649908623
To: sip:90791027947@95.128.80.5
CSeq: 1 INVITE
Server: Asterisk PBX 22.5.1
Content-Length: 0

<— Transmitting SIP response (612 bytes) to UDP:95.128.80.3:5060 —>
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 95.128.80.3;rport=5060;received=95.128.80.3;branch=z9hG4bK3a55.43ea7af7f82e7b6de869db6d4a2109a5.1
Via: SIP/2.0/UDP 95.128.84.2:5060;branch=z9hG4bK-3630-0b8a0f867b0b9cf205d58ca4ebc792fb
Record-Route: sip:95.128.80.3;lr;r2=on;did=c65.1527
Record-Route: sip:95.128.80.5;lr;r2=on;did=c65.1527
Call-ID: 4ad43a5f356a762051e954638ef2954b@95.128.84.2
From: “0796684485” sip:0796684485@95.128.84.2;tag=1761649908623
To: sip:90791027947@95.128.80.5;tag=17062a6c-83eb-4704-b715-8b897d40357f
CSeq: 1 INVITE
Server: Asterisk PBX 22.5.1
Content-Length: 0

<— Received SIP request (371 bytes) from UDP:95.128.80.3:5060 —>
ACK sip:s@192.168.0.79:5060 SIP/2.0
Call-ID: 4ad43a5f356a762051e954638ef2954b@95.128.84.2
CSeq: 1 ACK
From: “0796684485” sip:0796684485@95.128.84.2;tag=1761649908623
To: sip:90791027947@95.128.80.5;tag=2c56dd8d-29f5-4295-b2dc-7774f45cee2a
Via: SIP/2.0/UDP 95.128.80.3;branch=z9hG4bK3a55.43ea7af7f82e7b6de869db6d4a2109a5.0
Max-Forwards: 69
Content-Length: 0

<— Received SIP request (374 bytes) from UDP:95.128.80.3:5060 —>
ACK sip:s@188.154.136.126:5060 SIP/2.0
Call-ID: 4ad43a5f356a762051e954638ef2954b@95.128.84.2
CSeq: 1 ACK
From: “0796684485” sip:0796684485@95.128.84.2;tag=1761649908623
To: sip:90791027947@95.128.80.5;tag=17062a6c-83eb-4704-b715-8b897d40357f
Via: SIP/2.0/UDP 95.128.80.3;branch=z9hG4bK3a55.43ea7af7f82e7b6de869db6d4a2109a5.1
Max-Forwards: 69
Content-Length: 0

<— Received SIP request (294 bytes) from UDP:95.128.80.3:5060 —>
OPTIONS sip:s@192.168.0.79:5060 SIP/2.0
Via: SIP/2.0/UDP 95.128.80.3:5060;branch=z9hG4bK8460558
From: sip:ping@peoplefone.com;tag=uloc-678d7a18-6b87-bc60b671-a36d4947-cdd26871
To: sip:s@192.168.0.79:5060
Call-ID: ca55c7f6-3a35b181-0def6f6@95.128.80.3
CSeq: 1 OPTIONS
Content-Length: 0

<— Transmitting SIP response (840 bytes) to UDP:95.128.80.3:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 95.128.80.3:5060;rport=5060;received=95.128.80.3;branch=z9hG4bK8460558
Call-ID: ca55c7f6-3a35b181-0def6f6@95.128.80.3
From: sip:ping@peoplefone.com;tag=uloc-678d7a18-6b87-bc60b671-a36d4947-cdd26871
To: sip:s@192.168.0.79;tag=z9hG4bK8460558
CSeq: 1 OPTIONS
Accept: application/xpidf+xml, application/cpim-pidf+xml, application/pidf+xml, application/simple-message-summary, application/dialog-info+xml, application/simple-message-summary, application/pidf+xml, application/dialog-info+xml, application/sdp, message/sipfrag;version=2.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, INFO
Supported: 100rel, timer, replaces, norefersub
Accept-Encoding: identity
Accept-Language: en
Server: Asterisk PBX 22.5.1
Content-Length: 0

asteriskLS*CLI> pjsip set logger off
PJSIP Logging disabled

4

The incoming call was rejected for some reason, usually due to codecs. I don’t see you as having configured any codecs to use so in the “peoplefoneO77” wizard you’d probably want:

endpoint/allow=!all,g722,alaw
endpoint/dtmf_mode=rfc4733
5

THANKS !!!

That solved my problem !!!

You made my day ! Really thanks for your quick reaction and solution !

Best Regards

Tibor