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Hi Justin,<br>
<br>
There's absolutely no detail here regarding the SIP messages going
out and back. You'll need to include the asterisk-side sip debug.<br>
<br>
<a class="moz-txt-link-freetext" href="https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information">https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information</a><br>
<a class="moz-txt-link-freetext" href="https://support.digium.com/s/article/How-to-collect-an-Asterisk-Debug-Capture">https://support.digium.com/s/article/How-to-collect-an-Asterisk-Debug-Capture</a><br>
<br>
If you're using pjsip, you'll need to use it's specific logging.<br>
<a class="moz-txt-link-freetext" href="https://www.asterisk.org/new-pjsip-logging-functionality/">https://www.asterisk.org/new-pjsip-logging-functionality/</a><br>
<br>
<br>
<br>
<div class="moz-cite-prefix">On 12/2/22 12:22, Justin Piszcz wrote:<br>
</div>
<blockquote type="cite"
cite="mid:CAO9zADwCPd5+KMZfSUrM0Lg6ri=2wUXw+iG9Zh35uQNJoUEKWQ@mail.gmail.com">
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<div>Hello,</div>
<div><br>
</div>
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.
<div><br>
</div>
<div>asterisk versions: (old -> new)<br>
18.14.0~dfsg+~cs6.12.40431414-1+b1<br>
20.0.0~dfsg+~cs6.12.40431414-2<br>
</div>
<div><br>
</div>
<div>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?</div>
<div><br>
</div>
<div>asterisk18_sip_success.txt (inbound call success) from the
SPA3102 (with asterisk 18 installed)<br>
</div>
<div>Dec 1 17:34:55 system1 local3 fs: 11707:11782:65536<br>
Dec 1 17:34:55 system1 local3 fls: af:1:0:0<br>
Dec 1 17:34:55 system1 local3 fbr:
0:3000:3000:03d6a:0008:0007:5.1.10(GW)<br>
Dec 1 17:34:55 system1 local3 fhs:
01:0:0001:upg:app:0:3.3.6(GW)<br>
Dec 1 17:34:55 system1 local3 fhs:
02:0:0002:upg:app:1:3.3.6(GW)<br>
Dec 1 17:34:55 system1 local3 fhs:
03:0:0003:upg:app:2:3.3.6(GW)<br>
Dec 1 17:34:55 system1 local3 fhs:
04:0:0004:upg:app:0:5.1.10(GW)<br>
Dec 1 17:34:55 system1 local3 fhs:
05:0:0005:upg:app:1:5.1.10(GW)<br>
Dec 1 17:34:55 system1 local3 fhs:
06:0:0006:upg:app:2:5.1.10(GW)<br>
Dec 1 17:34:56 system1 local3 fu: 0:3d91, 0003 0001<br>
Dec 1 17:35:19 system1 local2 FXO: Start CNDD<br>
Dec 1 17:35:21 system1 local2 FXO: CNDD name=11234567890,
number=1234567890<br>
Dec 1 17:35:21 system1 local2 FXO: Stop CNDD<br>
Dec 1 17:35:21 system1 local3 FXO: CNDD Name=11234567890
Phone=1234567890<br>
Dec 1 17:35:22 system1 local2 AUD: Stop PSTN Tone<br>
Dec 1 17:35:22 system1 local2 AUD: Stop PSTN Tone<br>
Dec 1 17:35:22 system1 local2 Calling: <a
href="http://123@system1.int.com:0" target="_blank"
moz-do-not-send="true">123@system1.int.com:0</a><br>
Dec 1 17:35:22 system1 local2 [1:0]AUD ALLOC CALL
(port=16458)<br>
Dec 1 17:35:22 system1 local2 [1:0]RTP Rx Up<br>
Dec 1 17:35:22 system1 local2 CC: pc(0)=18 not in codec list<br>
Dec 1 17:35:22 system1 local2 [0:0]AUD ALLOC CALL
(port=16460)<br>
Dec 1 17:35:22 system1 local2 [0:0]RTP Rx Up<br>
Dec 1 17:35:22 system1 local2 CC: Ringback<br>
Dec 1 17:35:22 system1 local2 [1:0]RTP Rx Dn<br>
Dec 1 17:35:22 system1 local2 AUD: Play PSTN Tone 9<br>
Dec 1 17:35:23 system1 local3 IDBG: sc-0<br>
Dec 1 17:35:23 system1 local3 IDBG: rs:10<br>
Dec 1 17:35:26 system1 local3 IDBG: sc-0<br>
Dec 1 17:35:26 system1 local3 IDBG: rs:8<br>
Dec 1 17:35:32 system1 local2 AUD: Stop PSTN Tone<br>
Dec 1 17:35:32 system1 local3 FXO: On Hook<br>
Dec 1 17:35:32 system1 local2 AUD: Stop PSTN Tone<br>
Dec 1 17:35:32 system1 local2 FXO: Stop CNDD<br>
Dec 1 17:35:32 system1 local3 [0]FM Alert Stop RxTx
(c=002550b0;a=0)<br>
Dec 1 17:35:32 system1 local2 [1:0]AUD Rel Call<br>
Dec 1 17:35:32 system1 local3 [0]FM Alert Stop RxTx
(c=0024e5e8;a=0)<br>
Dec 1 17:35:32 system1 local2 [0:0]AUD Rel Call<br>
Dec 1 17:35:32 system1 local2 CC: Ended<br>
<br>
<br>
asterisk20_sip_error.txt
(inbound call failure) from the SPA3102 (with asterisk 20
installed)<br>
Dec 1 17:23:21 system1 local2 FXO: Start CNDD<br>
Dec 1 17:23:23 system1 local2 FXO: CNDD name=11234567890,
number=1234567890<br>
Dec 1 17:23:23 system1 local2 FXO: Stop CNDD<br>
Dec 1 17:23:23 system1 local3 FXO: CNDD Name=11234567890
Phone=1234567890<br>
Dec 1 17:23:24 system1 local2 AUD: Stop PSTN Tone<br>
Dec 1 17:23:24 system1 local2 AUD: Stop PSTN Tone<br>
Dec 1 17:23:24 system1 local2 Calling: <a
href="http://123@system1.int.com:0" target="_blank"
moz-do-not-send="true">123@system1.int.com:0</a><br>
Dec 1 17:23:24 system1 local2 [1:0]AUD ALLOC CALL
(port=16418)<br>
Dec 1 17:23:24 system1 local2 [1:0]RTP Rx Up<br>
Dec 1 17:23:24 system1 local2 [1]SIP:ICMP Error -1
(a000001:5060, 2)<br>
Dec 1 17:23:24 system1 local3 RSE_DEBUG: getting alternate
from domain:<a href="http://system1.int.com" target="_blank"
moz-do-not-send="true">system1.int.com</a><br>
Dec 1 17:23:24 system1 local3 [0]FM Alert Stop RxTx
(c=002550b0;a=0)<br>
Dec 1 17:23:24 system1 local2 [1:0]AUD Rel Call<br>
Dec 1 17:23:24 system1 local2 CC: Failed w/ Calling<br>
Dec 1 17:23:24 system1 local2 AUD: Stop PSTN Tone<br>
Dec 1 17:23:24 system1 local2 AUD: Stop PSTN Tone<br>
Dec 1 17:23:39 system1 local2 AUD: Stop PSTN Tone<br>
Dec 1 17:23:39 system1 local3 FXO: On Hook<br>
Dec 1 17:23:39 system1 local2 AUD: Stop PSTN Tone<br>
Dec 1 17:23:39 system1 local2 FXO: Stop CNDD<br>
<br>
</div>
<div>Regards,</div>
<div>Justin</div>
<div><br>
</div>
</div>
<br>
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