Upgraded From Asterisk 18.14.0 To 20.0.0 And Inbound Registration(?) Is Now Failing
Hello,
I have been using asterisk for the past decade and never had an issue with upgrades until now. Recently, in November I upgraded from 18.14.0 to
20.0.0 and afterwards my SPA3102 can no longer register with asterisk. I
have not made any asterisk or SPA3102 configuration changes in ~1-2 years.
asterisk versions: (old -> new)
18.14.0~dfsg+~cs6.12.40431414-1+b1
20.0.0~dfsg+~cs6.12.40431414-2
An example of the log from the SPA3102 under asterisk (succeeds) 18 vs. asterisk 20 (fails), kindly inquiring what I may have missed that is causing these failures?
asterisk18_sip_success.txt (inbound call success) from the SPA3102 (with asterisk 18 installed)
Dec 1 17:34:55 system1 local3 fs: 11707:11782:65536
Dec 1 17:34:55 system1 local3 fls: af:1:0:0
Dec 1 17:34:55 system1 local3 fbr: 0:3000:3000:03d6a:0008:0007:5.1.10(GW)
Dec 1 17:34:55 system1 local3 fhs: 01:0:0001:upg:app:0:3.3.6(GW)
Dec 1 17:34:55 system1 local3 fhs: 02:0:0002:upg:app:1:3.3.6(GW)
Dec 1 17:34:55 system1 local3 fhs: 03:0:0003:upg:app:2:3.3.6(GW)
Dec 1 17:34:55 system1 local3 fhs: 04:0:0004:upg:app:0:5.1.10(GW)
Dec 1 17:34:55 system1 local3 fhs: 05:0:0005:upg:app:1:5.1.10(GW)
Dec 1 17:34:55 system1 local3 fhs: 06:0:0006:upg:app:2:5.1.10(GW)
Dec 1 17:34:56 system1 local3 fu: 0:3d91, 0003 0001
Dec 1 17:35:19 system1 local2 FXO: Start CNDD
Dec 1 17:35:21 system1 local2 FXO: CNDD name234567890, number34567890
Dec 1 17:35:21 system1 local2 FXO: Stop CNDD
Dec 1 17:35:21 system1 local3 FXO: CNDD Name234567890 Phone34567890
Dec 1 17:35:22 system1 local2 AUD: Stop PSTN Tone Dec 1 17:35:22 system1 local2 AUD: Stop PSTN Tone Dec 1 17:35:22 system1 local2 Calling: 123@system1.int.com:0
Dec 1 17:35:22 system1 local2 [1:0]AUD ALLOC CALL (port458)
Dec 1 17:35:22 system1 local2 [1:0]RTP Rx Up Dec 1 17:35:22 system1 local2 CC: pc(0) not in codec list Dec 1 17:35:22 system1 local2 [0:0]AUD ALLOC CALL (port460)
Dec 1 17:35:22 system1 local2 [0:0]RTP Rx Up Dec 1 17:35:22 system1 local2 CC: Ringback Dec 1 17:35:22 system1 local2 [1:0]RTP Rx Dn Dec 1 17:35:22 system1 local2 AUD: Play PSTN Tone 9
Dec 1 17:35:23 system1 local3 IDBG: sc-0
Dec 1 17:35:23 system1 local3 IDBG: rs:10
Dec 1 17:35:26 system1 local3 IDBG: sc-0
Dec 1 17:35:26 system1 local3 IDBG: rs:8
Dec 1 17:35:32 system1 local2 AUD: Stop PSTN Tone Dec 1 17:35:32 system1 local3 FXO: On Hook Dec 1 17:35:32 system1 local2 AUD: Stop PSTN Tone Dec 1 17:35:32 system1 local2 FXO: Stop CNDD
Dec 1 17:35:32 system1 local3 [0]FM Alert Stop RxTx (c 2550b0;a=0)
Dec 1 17:35:32 system1 local2 [1:0]AUD Rel Call Dec 1 17:35:32 system1 local3 [0]FM Alert Stop RxTx (c 24e5e8;a=0)
Dec 1 17:35:32 system1 local2 [0:0]AUD Rel Call Dec 1 17:35:32 system1 local2 CC: Ended
asterisk20_sip_error.txt (inbound call failure) from the SPA3102 (with asterisk 20 installed)
Dec 1 17:23:21 system1 local2 FXO: Start CNDD
Dec 1 17:23:23 system1 local2 FXO: CNDD name234567890, number34567890
Dec 1 17:23:23 system1 local2 FXO: Stop CNDD
Dec 1 17:23:23 system1 local3 FXO: CNDD Name234567890 Phone34567890
Dec 1 17:23:24 system1 local2 AUD: Stop PSTN Tone Dec 1 17:23:24 system1 local2 AUD: Stop PSTN Tone Dec 1 17:23:24 system1 local2 Calling: 123@system1.int.com:0
Dec 1 17:23:24 system1 local2 [1:0]AUD ALLOC CALL (port418)
Dec 1 17:23:24 system1 local2 [1:0]RTP Rx Up Dec 1 17:23:24 system1 local2 [1]SIP:ICMP Error -1 (a000001:5060, 2)
Dec 1 17:23:24 system1 local3 RSE_DEBUG: getting alternate from domain:
system1.int.com Dec 1 17:23:24 system1 local3 [0]FM Alert Stop RxTx (c 2550b0;a=0)
Dec 1 17:23:24 system1 local2 [1:0]AUD Rel Call Dec 1 17:23:24 system1 local2 CC: Failed w/ Calling Dec 1 17:23:24 system1 local2 AUD: Stop PSTN Tone Dec 1 17:23:24 system1 local2 AUD: Stop PSTN Tone Dec 1 17:23:39 system1 local2 AUD: Stop PSTN Tone Dec 1 17:23:39 system1 local3 FXO: On Hook Dec 1 17:23:39 system1 local2 AUD: Stop PSTN Tone Dec 1 17:23:39 system1 local2 FXO: Stop CNDD
Regards, Justin
2 thoughts on - Upgraded From Asterisk 18.14.0 To 20.0.0 And Inbound Registration(?) Is Now Failing
Hi Justin,
There’s absolutely no detail here regarding the SIP messages going out and back. You’ll need to include the asterisk-side sip debug.
https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information https://support.digium.com/s/article/How-to-collect-an-Asterisk-Debug-Capture
If you’re using pjsip, you’ll need to use it’s specific logging. https://www.asterisk.org/new-pjsip-logging-functionality/
I had similar issues. It looks like modules related to pjsip
(geolocation?) introduced new prerequisites. There is a script in the source that prepares for an asterisk build. Try running that, then recompile asterisk and see if that fixes things. John
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