New v15 distro with v14 restore. Now need to move a SIP trunk over to PJSIP but very noisy logs with warnings and errors. PJSIP configuration setup pretty correct. Where do I begin?
By giving us more info?
2 Likes
[2020-04-01 11:24:22] WARNING[22034] loader.c: Some non-required modules failed to load.
[2020-04-01 11:24:22] ERROR[22034] loader.c: Error loading module âchan_local.soâ: /usr/lib64/asterisk/modules/chan_local.so: cannot open shared object file: No such file or directory
[2020-04-01 11:24:22] ERROR[22034] loader.c: Error loading module âres_pjsip_phoneprov_provider.soâ, missing dependency: res_phoneprov
[2020-04-01 11:24:22] ERROR[22034] loader.c: res_config_sqlite declined to load.
[2020-04-01 11:24:22] ERROR[22034] loader.c: res_stun_monitor declined to load.
[2020-04-01 11:24:22] ERROR[22034] loader.c: res_xmpp declined to load.
[2020-04-01 11:24:22] ERROR[22034] loader.c: Declined modules which depend on res_xmpp: chan_motif
[2020-04-01 11:24:22] ERROR[22034] loader.c: chan_console declined to load.
[2020-04-01 11:24:22] ERROR[22034] loader.c: chan_skinny declined to load.
[2020-04-01 11:24:22] ERROR[22034] loader.c: chan_ooh323 declined to load.
[2020-04-01 11:24:22] ERROR[22034] loader.c: chan_mobile declined to load.
[2020-04-01 11:24:22] ERROR[22034] loader.c: res_hep declined to load.
[2020-04-01 11:24:22] ERROR[22034] loader.c: Declined modules which depend on res_hep: res_hep_rtcp, res_hep_pjsip
[2020-04-01 11:24:22] ERROR[22034] loader.c: app_agent_pool declined to load.
[2020-04-01 11:24:22] ERROR[22034] loader.c: res_calendar declined to load.
[2020-04-01 11:24:22] ERROR[22034] loader.c: Declined modules which depend on res_calendar: res_calendar_caldav, res_calendar_ews, res_calendar_exchange, res_calendar_icalendar
[2020-04-01 11:24:22] ERROR[22034] loader.c: cdr_csv declined to load.
[2020-04-01 11:24:22] ERROR[22034] loader.c: cdr_manager declined to load.
[2020-04-01 11:24:22] ERROR[22034] loader.c: cdr_odbc declined to load.
[2020-04-01 11:24:22] ERROR[22034] loader.c: app_alarmreceiver declined to load.
[2020-04-01 11:24:22] ERROR[22034] loader.c: app_festival declined to load.
[2020-04-01 11:24:22] ERROR[22034] loader.c: app_followme declined to load.
[2020-04-01 11:24:22] ERROR[22034] loader.c: chan_phone declined to load.
[2020-04-01 11:24:22] ERROR[22034] loader.c: chan_unistim declined to load.
[2020-04-01 11:24:22] ERROR[22034] loader.c: pbx_ael declined to load.
[2020-04-01 11:24:22] ERROR[22034] loader.c: pbx_dundi declined to load.
[2020-04-01 11:24:22] ERROR[22034] loader.c: pbx_lua declined to load.
[2020-04-01 11:24:22] ERROR[22034] loader.c: func_odbc declined to load.
[2020-04-01 11:24:22] ERROR[22034] loader.c: res_pktccops declined to load.
[2020-04-01 11:24:22] VERBOSE[22034] asterisk.c: Asterisk Ready.
[2020-04-01 11:24:24] VERBOSE[22087] asterisk.c: Remote UNIX connection
[2020-04-01 11:24:24] VERBOSE[22638] asterisk.c: Remote UNIX connection disconnected
[2020-04-01 11:24:28] VERBOSE[22087] asterisk.c: Remote UNIX connection
[2020-04-01 11:24:28] VERBOSE[22961] asterisk.c: Remote UNIX connection disconnected
[2020-04-01 11:24:32] VERBOSE[22087] asterisk.c: Remote UNIX connection
[2020-04-01 11:24:32] VERBOSE[23284] asterisk.c: Remote UNIX connection disconnected
[2020-04-01 11:24:36] VERBOSE[22087] asterisk.c: Remote UNIX connection
[2020-04-01 11:24:36] VERBOSE[23473] asterisk.c: Remote UNIX connection disconnected
[2020-04-01 11:24:40] VERBOSE[22087] asterisk.c: Remote UNIX connection
[2020-04-01 11:24:40] VERBOSE[23491] asterisk.c: Remote UNIX connection disconnected
[2020-04-01 11:24:44] VERBOSE[22087] asterisk.c: Remote UNIX connection
[2020-04-01 11:24:44] VERBOSE[23598] asterisk.c: Remote UNIX connection disconnected
[2020-04-01 11:24:49] VERBOSE[22087] asterisk.c: Remote UNIX connection
[2020-04-01 11:24:49] VERBOSE[23609] asterisk.c: Remote UNIX connection disconnected
[2020-04-01 11:24:53] VERBOSE[22087] asterisk.c: Remote UNIX connection
[2020-04-01 11:24:53] VERBOSE[23627] asterisk.c: Remote UNIX connection disconnected
[2020-04-01 11:24:57] VERBOSE[22087] asterisk.c: Remote UNIX connection
[2020-04-01 11:24:57] VERBOSE[23654] asterisk.c: Remote UNIX connection disconnected
[2020-04-01 11:25:01] VERBOSE[22087] asterisk.c: Remote UNIX connection
[2020-04-01 11:25:01] VERBOSE[23661] asterisk.c: Remote UNIX connection disconnected
[2020-04-01 11:25:01] VERBOSE[22205] res_pjsip/pjsip_options.c: Contact Flowroute_PJSIP_NJ/sip:[email protected] is now Unreachable. RTT: 0.000 msec
[2020-04-01 11:25:05] VERBOSE[22087] asterisk.c: Remote UNIX connection
[2020-04-01 11:25:05] VERBOSE[23689] asterisk.c: Remote UNIX connection disconnected
[2020-04-01 11:25:09] VERBOSE[22087] asterisk.c: Remote UNIX connection
[2020-04-01 11:25:09] VERBOSE[23696] asterisk.c: Remote UNIX connection disconnected
This is the only relevant thing in there about your PJSIP trunk. It shows it is going unreachable which means itâs not getting responses back from Flowroute for its keepalives. Thatâs generally a firewall/NAT issue or a configuration issue on the PBX side.
Is this PBX behind NAT?
My CSIP connection is working. Since we are originating an outbound connection are there any PBX firewall settings needed? What might be different for the PJSIP from a router or other settings point of view?
Is it connecting over the same network to Flowroute?
You still also havenât answered my question. Is the PBX behind NAT? If the answer is yes, then you need to make sure the PBX has all the proper settings for PJSIP like telling it the local and external details.
Yes, the same route out of my network but to a different POP. Yes it is behind NAT.
Where do I best find documentation to set up PJSIP. I have followed the documentation offered in the wiki and by the vendor. If I need more I need info. No discussion found of settings necessary for router, nat or whatever.
What is all this about?
[2020-04-01 13:01:10] VERBOSE[14495] loader.c: Reloading module âres_pjproject.soâ (PJPROJECT Log and Utility Support)
[2020-04-01 13:01:10] ERROR[14495] res_sorcery_config.c: Unable to load config file âpjproject.confâ
[2020-04-01 13:01:10] VERBOSE[14495] loader.c: Reloading module âres_pjsip.soâ (Basic SIP resource)
[2020-04-01 13:01:10] ERROR[6379] res_pjsip_config_wizard.c: Unable to load config file âpjsip_wizard.confâ
[2020-04-01 13:01:10] ERROR[6379] res_pjsip_config_wizard.c: Unable to load config file âpjsip_wizard.confâ
[2020-04-01 13:01:10] ERROR[6379] res_pjsip_config_wizard.c: Unable to load config file âpjsip_wizard.confâ
[2020-04-01 13:01:10] ERROR[6379] res_pjsip_config_wizard.c: Unable to load config file âpjsip_wizard.confâ
[2020-04-01 13:01:10] NOTICE[6379] sorcery.c: Type âsystemâ is not reloadable, maintaining previous values
[2020-04-01 13:01:10] ERROR[6379] res_sorcery_config.c: Could not create an object of type âregistrationâ with id âFlowroute_NJâ from configuration file âpjsip.confâ
[2020-04-01 13:01:10] ERROR[6379] res_pjsip_config_wizard.c: Unable to load config file âpjsip_wizard.confâ
[2020-04-01 13:01:10] VERBOSE[14506] res_pjsip/pjsip_options.c: Contact Flowroute_PJSIP_NJ/sip:[email protected] has been deleted
[2020-04-01 13:01:10] VERBOSE[14495] loader.c: Reloading module âres_pjsip_authenticator_digest.soâ (PJSIP authentication resource)
[2020-04-01 13:01:10] VERBOSE[14495] loader.c: Reloading module âres_resolver_unbound.soâ (Unbound DNS Resolver Support)
[2020-04-01 13:01:10] ERROR[14495] config_options.c: Unable to load config file âresolver_unbound.confâ
[2020-04-01 13:01:10] VERBOSE[14495] loader.c: Reloading module âres_pjsip_endpoint_identifier_ip.soâ (PJSIP IP endpoint identifier)
[2020-04-01 13:01:10] ERROR[14495] res_pjsip_config_wizard.c: Unable to load config file âpjsip_wizard.confâ
4/1 Still looking for help and suggestions.
4/2 Having difficulty knowing how to solve this problem. No help so far. Can I get help?
If the PBX is behind NAT then you need to make sure the External Signalling and Media addresses are set along with the Local Networks in the PJSIP transport settings.
Settings -> Asterisk SIP Settings -> PJSIP Tab -> Transports
Outside of that you need to make sure you have the proper NAT rules in your firewall/router for the PBX.
I guess that is an option if the Community has no suggestions on how to debug this?
Did you just ignore my last reply to you? Did you check those things? Confirm that they are setup correctly?
So far you have shown log output that has some Asterisk modules that didnât load. Most of which arenât relevant. You show your PJSIP trunk going UNREACHABLE in that mess of output. Weâve told you what that means and why it would happen. Iâve also told you what to check and where to check it.
So again, have you done anything in regards to my last reply?
2 Likes
Tom **** Actually I did miss your post This does not relate to you, I am sorry. The timing must have been such that I didnât see it wheh I made my post above.
âConfirm that they are setup correctly?â
I have able to locate no instructions on what those settings would be, can you recommend one.
Youâre joking right? This was pretty clear on what to look at, where it is at and what do to navigate to it via the GUI. Can I tell you what to put in there exactly? No, I canât. I donât know your external IP or what your local networks are. Youâre going to have to put that information in yourself since you have it.
You were told that this is a NAT issue and that you need to make sure the PBX and the firewall have your NAT settings properly configured. Therefore you were told TWO places to locate the issue and what you need to look for and do in those locations.
P.S.: And no we canât tell you exact things to do in your PBX and firewall because we donât known what you have, how it is setup, etc. Some things we can only guide and direct you on, the actual data or setting values are going to be dependent on your setup and what you are using. We canât get that granular with vague details.
By âthe firewallâ can I assume you donât mean FreePBXs firewall but, in my case, my router?
The note on the transport section confuses me as to its application. Iâll see if I can find any documentation.
âNote that the interface is only displayed for your information, and is not referenced by asterisk. After you enable/disable a transport, asterisk needs to be restarted , not just reloaded.â
It seems to be the case may be that creating new PJSIP Trunk also requires a restart of asterisk. Possibly only for the first PJSIP trunk created.
Following the restart, the trunks are online but all my channel SIP phones are now disconnected. ???
you probably mixed up ports 5060 and 5160 and confused yourself with 5062 for chan_sip and chan_pjsip, refer to the wiki.
Changed the trunks to 5160. Now I am trying to get the phones back on 5060.
Trying searching the wiki.Need to get running then figure out how to move the phones to PJSIP later as I am mentally shot today.
All is well for now. Thanks, the biggest problem was needing to reboot after adding PJSIP trunks to get them to connect.