Telephone test with SIP Trunk between Sites. 4

[kecubung dalam bacan]

Hi all colleagues,

I'm having some problems when I make a test call from Site A to Site B

Here are the log results I got:


********** SysMonitor v11.1.2.1.0 build 3 [connected to 10.210.200.1 (IPOSE (Server Edition(P)))] **********
414795755mS PRN: Monitor Status S-Edition Primary 10.1.0.0.0 build 237
414795755mS PRN: Linux Whoo
414795870mS DTM: 14/03/2022 09:19:32 (Mon 14 Mar 2022) [10.210.200.1 (IPOSE (Server Edition(P)))]
414795916mS SIP Call Tx: 3
INVITE sip:8888@192.168.50.11 SIP/2.0
Via: SIP/2.0/UDP 10.210.200.1:5060;rport;branch=z9hG4bK61f9623336788bab6cbe7486028dee59
From: "Helpdesk" <sip:5000@192.168.50.11>;tag=2e863b5a92e0371e
To: <sip:8888@192.168.50.11>
Call-ID: f78f33b7bb9221d5a9934b305431338e
CSeq: 1125703276 INVITE
Contact: "Helpdesk" <sip:5000@10.210.200.1:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
User-Agent: IP Office 10.1.0.0.0 build 237
Content-Type: application/sdp
Content-Length: 250

v=0
o=UserA 1114560140 819480275 IN IP4 10.210.200.1
s=Session SDP
c=IN IP4 10.210.200.1
t=0 0
m=audio 42690 RTP/AVP 8 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
414795916mS SIP Tx: UDP 10.210.200.1:5060 -> 192.168.50.11:5060
INVITE sip:8888@192.168.50.11 SIP/2.0
Via: SIP/2.0/UDP 10.210.200.1:5060;rport;branch=z9hG4bK61f9623336788bab6cbe7486028dee59
From: "Helpdesk" <sip:5000@192.168.50.11>;tag=2e863b5a92e0371e
To: <sip:8888@192.168.50.11>
Call-ID: f78f33b7bb9221d5a9934b305431338e
CSeq: 1125703276 INVITE
Contact: "Helpdesk" <sip:5000@10.210.200.1:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
User-Agent: IP Office 10.1.0.0.0 build 237
Content-Type: application/sdp
Content-Length: 250

v=0
o=UserA 1114560140 819480275 IN IP4 10.210.200.1
s=Session SDP
c=IN IP4 10.210.200.1
t=0 0
m=audio 42690 RTP/AVP 8 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
414797515mS Sip: 0ad2c80100000b15 3.2837.0 613 SIPTrunk Endpoint(f2b33ee0) PacketTimer expired (no response to INVITE), break the call
414797516mS Sip: SIPDialog f2b33ee0 deleted, dialogs 0 txn_keys 0
414797517mS PRN: RTCP collector is NOT initialized
414799487mS PRN: IPOKeepaliveTask::Main sending keepalives at 5000 ms
414800881mS DTM: 14/03/2022 09:19:37 (Mon 14 Mar 2022) [10.210.200.1 (IPOSE (Server Edition(P)))]
414802246mS PRN: RTCP collector is NOT initialized
414804487mS PRN: IPOKeepaliveTask::Main sending keepalives at 5000 ms
414805890mS DTM: 14/03/2022 09:19:42 (Mon 14 Mar 2022) [10.210.200.1 (IPOSE (Server Edition(P)))]
414807383mS Sip: SIP Line (2) cannot find a suitable SIP URI to dial out
414807384mS Sip: SIP Line (3): License, Valid 1, Available 30, Consumed 0
414807384mS Sip: SIP Line (3): sip_trunk_config_items 500a0001, sip_trunk_config_items_2 00000000, voip.flags 00000948
414807384mS Sip: SIPDialog f2b33ee0 created, dialogs 1 txn_keys 0 video 0 presentation 0 camera 0
414807384mS Sip: 0ad2c80100000b1b 3.2843.0 615 SIPTrunk Endpoint(f2b410f8) received CMSetup
414807384mS Sip: 0ad2c80100000b1b 3.2843.0 615 SIPTrunk Endpoint(f2b33ee0) SetLocalRTPAddress to 10.210.200.1:42696
414807384mS SIP Call Tx: 3
INVITE sip:8888@192.168.50.11 SIP/2.0
Via: SIP/2.0/UDP 10.210.200.1:5060;rport;branch=z9hG4bK79f56be10d7d2915f6b04bf557b0a682
From: "Helpdesk" <sip:5000@192.168.50.11>;tag=4405755fa6e492e9
To: <sip:8888@192.168.50.11>
Call-ID: 392de1b0729dc29244537ba62dd4575f
CSeq: 234186592 INVITE
Contact: "Helpdesk" <sip:5000@10.210.200.1:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
User-Agent: IP Office 10.1.0.0.0 build 237
Content-Type: application/sdp
Content-Length: 250

v=0
o=UserA 1532930939 754621337 IN IP4 10.210.200.1
s=Session SDP
c=IN IP4 10.210.200.1
t=0 0
m=audio 42696 RTP/AVP 8 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
414807384mS SIP Tx: UDP 10.210.200.1:5060 -> 192.168.50.11:5060
INVITE sip:8888@192.168.50.11 SIP/2.0
Via: SIP/2.0/UDP 10.210.200.1:5060;rport;branch=z9hG4bK79f56be10d7d2915f6b04bf557b0a682
From: "Helpdesk" <sip:5000@192.168.50.11>;tag=4405755fa6e492e9
To: <sip:8888@192.168.50.11>
Call-ID: 392de1b0729dc29244537ba62dd4575f
CSeq: 234186592 INVITE
Contact: "Helpdesk" <sip:5000@10.210.200.1:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
User-Agent: IP Office 10.1.0.0.0 build 237
Content-Type: application/sdp
Content-Length: 250

v=0
o=UserA 1532930939 754621337 IN IP4 10.210.200.1
s=Session SDP
c=IN IP4 10.210.200.1
t=0 0
m=audio 42696 RTP/AVP 8 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
414809384mS SIP Call Tx: 3
INVITE sip:8888@192.168.50.11 SIP/2.0
Via: SIP/2.0/UDP 10.210.200.1:5060;rport;branch=z9hG4bK79f56be10d7d2915f6b04bf557b0a682
From: "Helpdesk" <sip:5000@192.168.50.11>;tag=4405755fa6e492e9
To: <sip:8888@192.168.50.11>
Call-ID: 392de1b0729dc29244537ba62dd4575f
CSeq: 234186592 INVITE
Contact: "Helpdesk" <sip:5000@10.210.200.1:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
User-Agent: IP Office 10.1.0.0.0 build 237
Content-Type: application/sdp
Content-Length: 250

v=0
o=UserA 1532930939 754621337 IN IP4 10.210.200.1
s=Session SDP
c=IN IP4 10.210.200.1
t=0 0
m=audio 42696 RTP/AVP 8 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
414809384mS SIP Tx: UDP 10.210.200.1:5060 -> 192.168.50.11:5060
INVITE sip:8888@192.168.50.11 SIP/2.0
Via: SIP/2.0/UDP 10.210.200.1:5060;rport;branch=z9hG4bK79f56be10d7d2915f6b04bf557b0a682
From: "Helpdesk" <sip:5000@192.168.50.11>;tag=4405755fa6e492e9
To: <sip:8888@192.168.50.11>
Call-ID: 392de1b0729dc29244537ba62dd4575f
CSeq: 234186592 INVITE
Contact: "Helpdesk" <sip:5000@10.210.200.1:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
User-Agent: IP Office 10.1.0.0.0 build 237
Content-Type: application/sdp
Content-Length: 250

v=0
o=UserA 1532930939 754621337 IN IP4 10.210.200.1
s=Session SDP
c=IN IP4 10.210.200.1
t=0 0
m=audio 42696 RTP/AVP 8 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
414809487mS PRN: IPOKeepaliveTask::Main sending keepalives at 5000 ms
414810899mS DTM: 14/03/2022 09:19:46 (Mon 14 Mar 2022) [10.210.200.1 (IPOSE (Server Edition(P)))]
414810984mS Sip: 0ad2c80100000b1b 3.2843.0 615 SIPTrunk Endpoint(f2b33ee0) PacketTimer expired (no response to INVITE), break the call
414810985mS Sip: SIPDialog f2b33ee0 deleted, dialogs 0 txn_keys 0
414810986mS PRN: RTCP collector is NOT initialized
414814487mS PRN: IPOKeepaliveTask::Main sending keepalives at 5000 ms
414815910mS DTM: 14/03/2022 09:19:52 (Mon 14 Mar 2022) [10.210.200.1 (IPOSE (Server Edition(P)))]
414817979mS PRN: RTCP collector is NOT initialized
414819487mS PRN: IPOKeepaliveTask::Main sending keepalives at 5000 ms
414820918mS DTM: 14/03/2022 09:19:56 (Mon 14 Mar 2022) [10.210.200.1 (IPOSE (Server Edition(P)))]
414824487mS PRN: IPOKeepaliveTask::Main sending keepalives at 5000 ms
414825924mS DTM: 14/03/2022 09:20:02 (Mon 14 Mar 2022) [10.210.200.1 (IPOSE (Server Edition(P)))]
414828532mS PRN: RTCP collector is NOT initialized
414829487mS PRN: IPOKeepaliveTask::Main sending keepalives at 5000 ms
414831330mS DTM: 14/03/2022 09:20:07 (Mon 14 Mar 2022) [10.210.200.1 (IPOSE (Server Edition(P)))]
414834488mS PRN: IPOKeepaliveTask::Main sending keepalives at 5000 ms
414836334mS DTM: 14/03/2022 09:20:12 (Mon 14 Mar 2022) [10.210.200.1 (IPOSE (Server Edition(P)))]
414839488mS PRN: IPOKeepaliveTask::Main sending keepalives at 5000 ms
414841345mS DTM: 14/03/2022 09:20:16 (Mon 14 Mar 2022) [10.210.200.1 (IPOSE (Server Edition(P)))]
414842219mS PRN: Optimizing BTree Lists Completed...Started:414842219 Duration:0 Size:205 Attempted:205

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


And the log results as above for the test call failed.
Can colleagues help me to give their views and suggestions regarding the problems that I found.
Thanks.

 

[sizbut]

Really! You couldn't at least say what the problem appears to be in plain English? No audio, one-way audio, no ringing, no connection,..

Stuck in a never ending cycle of file copying.

 

[kecubung dalam bacan]

Hi sizbut

Ok.
I added the issue of the problem.
When the telephone test is between sites, from site A to site B, the ip phones get an UNOBTAINABLE notification, no ringing at all and no one-way audio.

 

[Ekster]

Would:

414810984mS Sip: 0ad2c80100000b1b 3.2843.0 615 SIPTrunk Endpoint(f2b33ee0) [highlight #FCE94F]PacketTimer expired (no response to INVITE)[/highlight], break the call

Have anything to do with it?

“Some humans would do anything to see if it was possible to do it.
If you put a large switch in some cave somewhere, with a sign on it saying 'End-of-the-World Switch. PLEASE DO NOT TOUCH'.
The paint wouldn't even have time to dry.”

Terry Pratchet

 

[critchey]

As Ekster pointed out in your log you get a "no response to INVITE" so at some point the communication breaks down and the phone doesn't work. First thing to look for is SIP_ALG/SIP Transformations/SIP Helper breaking the communication. Beyond that it is almost for sure something in your network causing the issue.

The truth is just an excuse for lack of imagination.

 

[kecubung dalam bacan]

Hi Ekster and Critchey,

After I analyzed it more deeply, it turned out that there was a notification like the following:

437652379mS Sip: 0ad2c80100000db1 3.3505.0 836 SIPTrunk Endpoint(f2b27510) PacketTimer expired (no response to INVITE), break the call
437652379mS Sip: 0ad2c80100000db1 3.3505.0 -1 SIPTrunk Endpoint(f2b2a078) received CMReleaseComp
437652379mS Sip: SIPDialog f2b27510 deleted, dialogs 0 txn_keys 0
437652380mS PRN: RTCP collector is NOT initialized

 

[kecubung dalam bacan]

Hi critchey,

Thank you for the info you provided.
By the way, it turns out that the location of the problem that I experienced was in the proxy and I should have listed the IP on the ITSP Proxy Address.

After checking the torch feature, it turned out that what I had to put on the ITSP Proxy Address came out as ip 10.xxx.x.xx (microtik eth2 interface) , not as ip 192.xxx.xx.xx

 

[critchey]

Thank you posting the resolution.

The truth is just an excuse for lack of imagination.