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02:37.58 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.36.0 (2020/09/03) 16.13.0 (2020/09/03) Standard: 17.7.0 (2020/09/03); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
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16:59.33 | Chris30 | Hi, I am struggling for quite a while now and I can't find an issue with my config. I am not getting asterisk to open up port 5061 with the configuration i have. I am getting this error: Sep 5 16:35:39 ip-10-0-0-183 asterisk[4176]: [2020-09-05 16:35:39] #033[1;31mERROR#033[0m[4176]: #033[1;37mconfig_options.c#033[0m:#033[1;37m746#0 |
16:59.34 | Chris30 | 33[0m #033[1;37maco_process_var#033[0m: Error parsing method=tlsv1_1 at line 20 of /etc/asterisk/pjsip.transports.conf |
17:00.02 | Chris30 | There is not much more in the log file. |
17:02.03 | Chris30 | There is a pastebin of my pjsip config https://pastebin.com/9B6cMSYN |
17:08.06 | igcewieling | Error parsing method=tlsv1_1 |
17:08.18 | igcewieling | Unlike chan_sip, pjsip will not load if there are invalid options. |
17:11.51 | Chris30 | I undersrand. But the option is not invalid |
17:12.20 | igcewieling | show me in the docs where the method= option is documented |
17:13.53 | Chris30 | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_res_pjsip#Asterisk13Configuration_res_pjsip-transport_method |
17:14.04 | Chris30 | Search for tlsv_1_1 |
17:14.41 | file | availability of that method is dependent on a recent version of Asterisk, and a recent version of PJSIP |
17:15.33 | Chris30 | Connected to Asterisk 13.18.3~ |
17:16.00 | file | that version of Asterisk does not have it. |
17:17.15 | Chris30 | ok thanks. So i have to upgrade. Why is it showing that method in the documentation then? |
17:17.28 | file | because the documentation is for the latest version of 13 |
17:18.01 | Chris30 | ok got it. Thank you! |
17:19.10 | Chris30 | where did you look that up so quickly? do you mind sharing the info or is it just experience? |
17:19.47 | igcewieling | file is one of those people who knows almost everything important about Asterisk. |
17:20.10 | Chris30 | :) |
17:20.11 | file | I did "git checkout 13.18.3" in my clone, opened the file that has the documentation and looked |
17:20.53 | Chris30 | ok thx for sharing |
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20:04.44 | wayne47 | Apparently, I was incorrect. The phone IS registered, just can not make calls. To recap, I am trying to move a Cisco 7942 from SIP:5060 to PJSIP:5160. Attached is a pastebin of an attempted call from the extension in question (5147) to another 7942 still running SIP (5143) https://pastebin.pl/view/e5eb6afb |
20:08.39 | file | the call came in over chan_sip, not chan_pjsip |
20:09.19 | wayne47 | It's connecting to port 5160 which is PJSIP |
20:09.33 | file | your log does not show that |
20:10.51 | wayne47 | so...suggestion on how to proceed? |
20:11.00 | file | the phone sent the call to chan_sip |
20:11.50 | file | I don't have any experience to recall configuring Cisco phones, so that is the extent of what I can say |
20:14.34 | wayne47 | what indicates that it's going to SIP? Trying to understand the log file |
20:14.52 | file | the log shows chan_sip handling the call, and the channel name is prefixed with "SIP/" |
20:14.59 | file | and the callerid name/number is your Cisco phone |
20:15.14 | wayne47 | line number? |
20:15.37 | file | 48, 115 |
20:16.07 | file | 49 shows chan_sip got the INVITE |
20:18.18 | wayne47 | I see that. even though line 42 says port 5160. And the extension says This device uses PJSIP technology listening on Port 5160 |
20:19.39 | file | line 42 is a debug message from a configuration parser |
20:20.16 | file | and if you're using FreePBX Endpoint Manager for provisioning the device, then you'd need to seek assistance for that |
20:20.26 | wayne47 | manual provision |
20:20.44 | file | then that's likely your problem |
20:20.54 | file | as the "This device uses PJSIP technology" means that PJSIP is configured for the device |
20:21.11 | file | it does not inherently mean that the device is itself configured properly or using the port |
20:50.59 | wayne47 | Thank you. Progress. One value in the phone was still on 5060. Now, tcpdump shows nothing using 5060. Phone can dial another extension, causing it to ring. But, calls in fail and even once a call is connected, I have one directional sound (5147 can hear other caller but other caller can not hear 5147). |
20:52.18 | Samot | wayne47: That's most likely the SIP Port the phone uses. |
20:52.32 | Samot | If the host is 1.1.1.1 you add 1.1.1.1:5160 |
20:52.47 | wayne47 | The phone now correctly uses 5160 |
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21:00.49 | wayne47 | here is the log from 5147(SIP:5060) trying to call 5147(PJSIP:5160) https://pastebin.pl/view/260339c8 |
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