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IP office to Asterisk PBX SIP Trunks 2

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Bonker1974

Bonker1974

Technical User
Joined
Oct 11, 2011
Messages
359
Location
BS
Hi all,
I have an issue with a sip trunks with IP Office and Asterisk PBX.

When the Asterisk PBX called the IP Office I can see the calls in the IP Office monitor but the call never hit the sip trunks. When a call is made I would monitor the System status and the system Monitor on the IP Office, I can see the call coming into the IP Office from IP Office monitor but it never reach the trunks.

I shave SIP Trunk licenses
Incoming call route is programed


Thanks in advance.

Leadership determines the direction of the company. Organization determines the potential of the company. Personnel determines the success of the company.
 
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102974mS SIP Rx: UDP 172.16.20.254:65476 -> 172.16.20.5:5060
INVITE sip:222@172.16.20.5 SIP/2.0
Via: SIP/2.0/UDP 172.16.20.254:65476;branch=z9hG4bK4dc4a67c
Max-Forwards: 70
From: <sip:309@172.16.10.1;otg=qhcbahamas.local>;tag=as3dddbf44
To: <sip:222@172.16.20.5>
Contact: <sip:309@172.16.20.254:65476>
Call-ID: 048d1c5d5175987f35bde6850543663e@172.16.10.1:5060
CSeq: 102 INVITE
User-Agent: UCX-6.0(13.38.3)
Date: Sat, 23 Apr 2022 22:23:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length:325

v=0
o=root 959114735 959114735 IN IP4 172.16.20.254
s=Asterisk PBX 13.38.3
i=(o=IN IP4 172.16.10.1)
c=IN IP4 172.16.20.254
t=0 0
m=audio 12960 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
102977mS SIP Call Rx: phone
INVITE sip:222@172.16.20.5 SIP/2.0
Via: SIP/2.0/UDP 172.16.20.254:65476;branch=z9hG4bK4dc4a67c
Max-Forwards: 70
From: <sip:309@172.16.10.1;otg=qhcbahamas.local>;tag=as3dddbf44
To: <sip:222@172.16.20.5>
Contact: <sip:309@172.16.20.254:65476>
Call-ID: 048d1c5d5175987f35bde6850543663e@172.16.10.1:5060
CSeq: 102 INVITE
User-Agent: UCX-6.0(13.38.3)
Date: Sat, 23 Apr 2022 22:23:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length:325

v=0
o=root 959114735 959114735 IN IP4 172.16.20.254
s=Asterisk PBX 13.38.3
i=(o=IN IP4 172.16.10.1)
c=IN IP4 172.16.20.254
t=0 0
m=audio 12960 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
103474mS SIP Rx: UDP 172.16.20.254:65476 -> 172.16.20.5:5060
INVITE sip:222@172.16.20.5 SIP/2.0
Via: SIP/2.0/UDP 172.16.20.254:65476;branch=z9hG4bK4dc4a67c
Max-Forwards: 70
From: <sip:309@172.16.10.1;otg=qhcbahamas.local>;tag=as3dddbf44
To: <sip:222@172.16.20.5>
Contact: <sip:309@172.16.20.254:65476>
Call-ID: 048d1c5d5175987f35bde6850543663e@172.16.10.1:5060
CSeq: 102 INVITE
User-Agent: UCX-6.0(13.38.3)
Date: Sat, 23 Apr 2022 22:23:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length:325

v=0
o=root 959114735 959114735 IN IP4 172.16.20.254
s=Asterisk PBX 13.38.3
i=(o=IN IP4 172.16.10.1)
c=IN IP4 172.16.20.254
t=0 0
m=audio 12960 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
103478mS SIP Call Rx: phone
INVITE sip:222@172.16.20.5 SIP/2.0
Via: SIP/2.0/UDP 172.16.20.254:65476;branch=z9hG4bK4dc4a67c
Max-Forwards: 70
From: <sip:309@172.16.10.1;otg=qhcbahamas.local>;tag=as3dddbf44
To: <sip:222@172.16.20.5>
Contact: <sip:309@172.16.20.254:65476>
Call-ID: 048d1c5d5175987f35bde6850543663e@172.16.10.1:5060
CSeq: 102 INVITE
User-Agent: UCX-6.0(13.38.3)
Date: Sat, 23 Apr 2022 22:23:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length:325

v=0
o=root 959114735 959114735 IN IP4 172.16.20.254
s=Asterisk PBX 13.38.3
i=(o=IN IP4 172.16.10.1)
c=IN IP4 172.16.20.254
t=0 0
m=audio 12960 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
104474mS SIP Rx: UDP 172.16.20.254:65476 -> 172.16.20.5:5060
INVITE sip:222@172.16.20.5 SIP/2.0
Via: SIP/2.0/UDP 172.16.20.254:65476;branch=z9hG4bK4dc4a67c
Max-Forwards: 70
From: <sip:309@172.16.10.1;otg=qhcbahamas.local>;tag=as3dddbf44
To: <sip:222@172.16.20.5>
Contact: <sip:309@172.16.20.254:65476>
Call-ID: 048d1c5d5175987f35bde6850543663e@172.16.10.1:5060
CSeq: 102 INVITE
User-Agent: UCX-6.0(13.38.3)
Date: Sat, 23 Apr 2022 22:23:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length:325

v=0
o=root 959114735 959114735 IN IP4 172.16.20.254
s=Asterisk PBX 13.38.3
i=(o=IN IP4 172.16.10.1)
c=IN IP4 172.16.20.254
t=0 0
m=audio 12960 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
104478mS SIP Call Rx: phone
INVITE sip:222@172.16.20.5 SIP/2.0
Via: SIP/2.0/UDP 172.16.20.254:65476;branch=z9hG4bK4dc4a67c
Max-Forwards: 70
From: <sip:309@172.16.10.1;otg=qhcbahamas.local>;tag=as3dddbf44
To: <sip:222@172.16.20.5>
Contact: <sip:309@172.16.20.254:65476>
Call-ID: 048d1c5d5175987f35bde6850543663e@172.16.10.1:5060
CSeq: 102 INVITE
User-Agent: UCX-6.0(13.38.3)
Date: Sat, 23 Apr 2022 22:23:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length:325

v=0
o=root 959114735 959114735 IN IP4 172.16.20.254
s=Asterisk PBX 13.38.3
i=(o=IN IP4 172.16.10.1)
c=IN IP4 172.16.20.254
t=0 0
m=audio 12960 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
106474mS SIP Rx: UDP 172.16.20.254:65476 -> 172.16.20.5:5060
INVITE sip:222@172.16.20.5 SIP/2.0
Via: SIP/2.0/UDP 172.16.20.254:65476;branch=z9hG4bK4dc4a67c
Max-Forwards: 70
From: <sip:309@172.16.10.1;otg=qhcbahamas.local>;tag=as3dddbf44
To: <sip:222@172.16.20.5>
Contact: <sip:309@172.16.20.254:65476>
Call-ID: 048d1c5d5175987f35bde6850543663e@172.16.10.1:5060
CSeq: 102 INVITE
User-Agent: UCX-6.0(13.38.3)
Date: Sat, 23 Apr 2022 22:23:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length:325

v=0
o=root 959114735 959114735 IN IP4 172.16.20.254
s=Asterisk PBX 13.38.3
i=(o=IN IP4 172.16.10.1)
c=IN IP4 172.16.20.254
t=0 0
m=audio 12960 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
106477mS SIP Call Rx: phone
INVITE sip:222@172.16.20.5 SIP/2.0
Via: SIP/2.0/UDP 172.16.20.254:65476;branch=z9hG4bK4dc4a67c
Max-Forwards: 70
From: <sip:309@172.16.10.1;otg=qhcbahamas.local>;tag=as3dddbf44
To: <sip:222@172.16.20.5>
Contact: <sip:309@172.16.20.254:65476>
Call-ID: 048d1c5d5175987f35bde6850543663e@172.16.10.1:5060
CSeq: 102 INVITE
User-Agent: UCX-6.0(13.38.3)
Date: Sat, 23 Apr 2022 22:23:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length:325

v=0
o=root 959114735 959114735 IN IP4 172.16.20.254
s=Asterisk PBX 13.38.3
i=(o=IN IP4 172.16.10.1)
c=IN IP4 172.16.20.254
t=0 0
m=audio 12960 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

********** SysMonitor v10.0.0.6.0 build 3 [connected to 172.16.20.5 (Quality Home Cr)] **********
109384mS SIP Rx: UDP 172.16.20.254:65476 -> 172.16.20.5:5060
INVITE sip:221@172.16.20.5 SIP/2.0
Via: SIP/2.0/UDP 172.16.20.254:65476;branch=z9hG4bK2bb2592a
Max-Forwards: 70
From: <sip:309@172.16.10.1;otg=qhcbahamas.local>;tag=as3906493c
To: <sip:221@172.16.20.5>
Contact: <sip:309@172.16.20.254:65476>
Call-ID: 1ec9fde46f1855ef740432505eb50923@172.16.10.1:5060
CSeq: 102 INVITE
User-Agent: UCX-6.0(13.38.3)
Date: Sat, 23 Apr 2022 22:22:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length:327

v=0
o=root 1390448780 1390448780 IN IP4 172.16.20.254
s=Asterisk PBX 13.38.3
i=(o=IN IP4 172.16.10.1)
c=IN IP4 172.16.20.254
t=0 0
m=audio 12956 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
109388mS SIP Call Rx: phone
INVITE sip:221@172.16.20.5 SIP/2.0
Via: SIP/2.0/UDP 172.16.20.254:65476;branch=z9hG4bK2bb2592a
Max-Forwards: 70
From: <sip:309@172.16.10.1;otg=qhcbahamas.local>;tag=as3906493c
To: <sip:221@172.16.20.5>
Contact: <sip:309@172.16.20.254:65476>
Call-ID: 1ec9fde46f1855ef740432505eb50923@172.16.10.1:5060
CSeq: 102 INVITE
User-Agent: UCX-6.0(13.38.3)
Date: Sat, 23 Apr 2022 22:22:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length:327

v=0
o=root 1390448780 1390448780 IN IP4 172.16.20.254
s=Asterisk PBX 13.38.3
i=(o=IN IP4 172.16.10.1)
c=IN IP4 172.16.20.254
t=0 0
m=audio 12956 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
110474mS SIP Rx: UDP 172.16.20.254:65476 -> 172.16.20.5:5060
INVITE sip:222@172.16.20.5 SIP/2.0
Via: SIP/2.0/UDP 172.16.20.254:65476;branch=z9hG4bK4dc4a67c
Max-Forwards: 70
From: <sip:309@172.16.10.1;otg=qhcbahamas.local>;tag=as3dddbf44
To: <sip:222@172.16.20.5>
Contact: <sip:309@172.16.20.254:65476>
Call-ID: 048d1c5d5175987f35bde6850543663e@172.16.10.1:5060
CSeq: 102 INVITE
User-Agent: UCX-6.0(13.38.3)
Date: Sat, 23 Apr 2022 22:23:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length:325

v=0
o=root 959114735 959114735 IN IP4 172.16.20.254
s=Asterisk PBX 13.38.3
i=(o=IN IP4 172.16.10.1)
c=IN IP4 172.16.20.254
t=0 0
m=audio 12960 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
110477mS SIP Call Rx: phone
INVITE sip:222@172.16.20.5 SIP/2.0
Via: SIP/2.0/UDP 172.16.20.254:65476;branch=z9hG4bK4dc4a67c
Max-Forwards: 70
From: <sip:309@172.16.10.1;otg=qhcbahamas.local>;tag=as3dddbf44
To: <sip:222@172.16.20.5>
Contact: <sip:309@172.16.20.254:65476>
Call-ID: 048d1c5d5175987f35bde6850543663e@172.16.10.1:5060
CSeq: 102 INVITE
User-Agent: UCX-6.0(13.38.3)
Date: Sat, 23 Apr 2022 22:23:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length:325

v=0
o=root 959114735 959114735 IN IP4 172.16.20.254
s=Asterisk PBX 13.38.3
i=(o=IN IP4 172.16.10.1)
c=IN IP4 172.16.20.254
t=0 0
m=audio 12960 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

********** Warning: Logging to Screen Stopped **********


Leadership determines the direction of the company. Organization determines the potential of the company. Personnel determines the success of the company.
 
Upvote 0 Downvote
Without seeing the SIP trunk config I can't really tell what the issue is as the Sysmon cuts off and doesn't show any responses to the Invite messages.

I would check the System Status, Under System and Voip Security, just make sure the IP isn't blacklisted.

Then get a sysmon with the SIP tab all checked and default settings on the Call Tab (at least make sure targeting is enabled)

Then it should show on the trace as to why the system is rejecting the Invite, and lead you to the fix.
 
Upvote 0 Downvote
Good day,
Thanks for your reply. For some strange reason it's working now. Not sure what happened but it's working.


Thanks for your reply

Leadership determines the direction of the company. Organization determines the potential of the company. Personnel determines the success of the company.
 
Upvote 0 Downvote
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