Hi there,
I’ve set up the FreePBX Distro 13.0.109. I’ve done the basic configuration as described in First Steps, further more I’ve set up several extensions(pjsip) for my DECT-IP Solution, a SIP Trunk(pjsip), an inbound route and an outbound route. While configuring the SIP Trunk I followed a tutorial, which refers to my provider.
Internal calls are working as well as incoming calls. When I try to make outgoing calls I got the “All circuits are busy” sound.
I check if my Trunk is registrated, which looks well:
`Chan_PJSip Registrations
<Registration/ServerURI…> <Auth…> <Status…>
mynumber/sip:tel.my-provider.de:5060 mynumber Registered`
` ParameterName : ParameterValue
auth_rejection_permanent : true
client_uri : sip:[email protected]:5060
contact_user : mynumber
endpoint :
expiration : 3600
fatal_retry_interval : 0
forbidden_retry_interval : 0
line : false
max_retries : 10
outbound_auth : mynumber
outbound_proxy :
retry_interval : 60
server_uri : sip:tel.my-provider.de:5060
support_path : false
transport : 0.0.0.0-udp`
I’ve tried to get more information logging pjsip in asterisk CLI. I’ve anonymized some information:
(192.168.1.20 → FreePBX Server; 192.168.1.22 DECT-IP-Base)
`<— Received SIP request (1287 bytes) from UDP:192.168.1.22:5060 —>
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.22:5060;rport;branch=z9hG4bKwo1cl2w53vo2baq
Max-Forwards: 70
From: “Extension-Name” sip:[email protected];tag=fvbdpvx1oohv5
To: sip:[email protected];user=phone
Call-ID: [email protected]
CSeq: 21352 INVITE
Contact: sip:[email protected];line=6556
Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK, UPDATE
Authorization: Digest username=“111”, realm=“asterisk”, nonce=“1462981863/da6af1268b8834477e9b55dbfe134d1b”, uri="sip:[email protected];user=phone", response=“1aea9f3fcd919719839c2d286c3e40fc”, opaque=“7fc74192644fe95e”, qop=auth, nc=00000001, algorithm=MD5, cnonce=“ntc9whivigq”
Content-Disposition: session
Min-SE: 90
Session-Expires: 3600
Supported: replaces,100rel,timer
User-Agent: DECTIP-System/03.24.0012 (MAC=000413611533; SER= 00000; HW=255)
Content-Type: application/sdp
Content-Length: 303
v=0
o=111 16876 16876 IN IP4 192.168.1.22
s=-
c=IN IP4 192.168.1.22
t=0 0
a=sendrecv
m=audio 50042 RTP/AVP 0 8 2 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=maxptime:80
a=sendrecv
a=rtcp:50043
<— Transmitting SIP response (342 bytes) to UDP:192.168.1.22:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.22:5060;rport=5060;received=192.168.1.22;branch=z9hG4bKwo1cl2w53vo2baq
Call-ID: [email protected]
From: “Extension-Name” sip:[email protected];tag=fvbdpvx1oohv5
To: sip:[email protected];user=phone
CSeq: 21352 INVITE
Server: FPBX-13.0.109(13.7.1)
Content-Length: 0
[2016-05-11 17:51:03] WARNING[1750]: func_cdr.c:352 cdr_write_callback: CDR requires a value (CDR(variable)=value)
<— Transmitting SIP request (979 bytes) to UDP:IP_of_my_provider:5060 —>
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP My_public_IP:5060;rport;branch=z9hG4bKPjvI-734OdTxBccFv78VGC3WrPDM5P20tu
From: sip:[email protected];tag=zKIc74aB.3waex3PgUpk.zWLJiH8qaqj
To: sip:[email protected]
Contact: sip:asterisk@My_public_IP:5060
Call-ID: XBwzYqWxRPkXIg1unvA2Wpy3cpP9svNZ
CSeq: 15768 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: FPBX-13.0.109(13.7.1)
Content-Type: application/sdp
Content-Length: 312
v=0
o=- 230578407 230578407 IN IP4 192.168.10.20
s=Asterisk
c=IN IP4 My_public_IP
t=0 0
m=audio 12548 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<— Received SIP response (385 bytes) from UDP:IP_of_my_provider:5060 —>
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP My_public_IP:5060;received=My_public_IP;rport=64289;branch=z9hG4bKPjvI-734OdTxBccFv78VGC3WrPDM5P20tu
To: sip:[email protected];tag=h7g4Esbg_hb2dkpqhyv6vez1vcdc3upjjy7zlkw65
From: sip:[email protected];tag=zKIc74aB.3waex3PgUpk.zWLJiH8qaqj
Call-ID: XBwzYqWxRPkXIg1unvA2Wpy3cpP9svNZ
CSeq: 15768 INVITE
Content-Length: 0`
I understand that 403 Forbidden shows me the error and I assume the warning “CDR requires a value” isn’t related to this issue.
I’ve read several topics in this and other forums unfortunatelly I’m not able to solve this issue.
For example, I’ve read that NAT must be enabled within the SIP Settings, I’cant find a NAT Option but set up the SIP Settings as follows:
Anyone who can point me in the right direction to solve this issue?
Best regards,
Pet