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Apply Config error
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Queues rrordered
Hi,
What is meant by “the queue member order from config file” which config file and how do you set the order?
rrordered : same as rrmemory, except the queue member order from config file is preserved
Thanks
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Inbound/Outbound calls using existing cellular service and Cisco phone through FreePBX?
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Inbound calls via same SIP trunk
One aimed at domestic users, one man businesses, and small businesses wanting a centrex type service (i.e. no PABX of their own).
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High CPU usage from schedtc.php
Hi,
on my machine all cores run php /var/lib/asterisk/bin/schedtc.php with full CPU usage. This seems to be a recurring problem. Running it by hand (as user asterisk) resulted in a PHP stacktrace pointing to a SysV semaphore exception.
Shutting down and rebooting freepbx helped, but I’d still like to investigate this further, so any pointers are greatly appreciated.
Thanks,
Matthias
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Illegal string offset 'value' response from GraphQL query
Thanks, I’ll do that.
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Sip Trunk Deutsche Telekom
Many thanks for all the info here. I’m trying to switch our chan_sip trunk to pjsip to circumvent some sporadic Rejection and Forbidden problems, when the Telekom SIP-server’s IP address changes.
Anyway, after setting up the pjsip trunk and disabling the chan_sip trunk I still get the following:
[2019-11-11 13:42:45] WARNING[14701]: pjsip:0 <?>: tsx0x7fde84e78 Failed to send Request msg REGISTER/cseq=6860 (tdta0x7fde84003d70)! err=320053 (DNS “Name Error” (PJLIB_UTIL_EDNS_NXDOMAIN))
[2019-11-11 13:42:45] WARNING[11541]: res_pjsip_outbound_registration.c:751 schedule_retry: No response received from ‘sip:xxxxxxxxxxxx@reg.sip-trunk.telekom.de:5060’ on registration attempt to ‘sip:+49zzzzzzzz@sip-trunk.telekom.de’, retrying in ‘60’
[2019-11-11 13:42:50] WARNING[11541]: pjsip:0 <?>: tsx0x7fde84e78 .Failed to send Request msg OPTIONS/cseq=9121 (tdta0x7fde84003d70)! err=320047 (No answer record in the DNS response (PJLIB_UTIL_EDNSNOANSWERREC))
[2019-11-11 13:42:50] ERROR[11541]: res_pjsip.c:3167 endpt_send_request: Error 320047 ‘No answer record in the DNS response (PJLIB_UTIL_EDNSNOANSWERREC)’ sending OPTIONS request to endpoint
Unfortunately “pjsip set logger on” does not shed more light on this.
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Sipsettings.class.php error on reload
Issue should already be fixed
https://issues.freepbx.org/browse/FREEPBX-20742
Upgrade SIP Settings to edge with:
fwconsole ma upgrade sipsettings --edge↧
Sip Trunk Deutsche Telekom
The linked issue is not applicable to Asterisk 13. You’d need to look at the DNS lookups being done (packet capture would work) and the results, and see where it is falling apart.
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Queues rrordered
The config file is queus.conf or one of its includes. FreePBX writes static queue agents in the same order they are written in the Static Agents field on the Queue Agents tab, so it’s that order which will be used.
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Calendar module
HI.
I’m using FreePBX 14. Current version of the Calendar module is 14.0.3.6.
The problem is - when i create new event in the calendar, make it reoccuring and choose “Repeat every weekday (Monday to Friday)” in fact it generates events from Tuesday to Friday but not on Mondays.
Am i doing something wrong or it just a bug and i need to open an issue on this?
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Calendar module
It is a bug on the iCal parser backend. There is a bug report open because of that.
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Calendar module
And this bug also affecting the local calendars?
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High CPU usage from schedtc.php
It would be nice to fix the issue, but if you need a band-aid, know that the only thing this script does is to update Asterisk hints for Time Conditions BLF buttons. If you don’t have any TC BLFs you can disable maintenance polling in advanced settings.
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Calendar module
I think so, as they are also iCal calendars.
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Sip Trunk Deutsche Telekom
Capturing done, Wireshark and analysis follows.
I have to ask, this makes sense, even though we don’t have DNS resolution problems when using the same trunk with chan_sip?
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Sip Trunk Deutsche Telekom
Two different DNS clients, could also be different configuration resulting in different DNS lookups. As well the SRV support in Asterisk 13 (where the PJSIP built in support is used) is not as complete or predictable as that of Asterisk 16 and higher, so that could be playing a part.
I’ve seen this provider come up multiple times with people using both chan_sip and chan_pjsip and I don’t know if anyone has got it working perfectly for them these days.
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Quesiton about Page Pro
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Sip Trunk Deutsche Telekom
the problem seems to be that the SRV lookup in the pjsip case is using UDP instead of TCP:
| DNS | 94 | Standard query 0x0062 SRV _sip._udp.reg.sip-trunk.telekom.de |
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| DNS | 170 | Standard query response 0x0062 No such name SRV _sip._udp.reg.sip-trunk.telekom.de SOA dns00.dns.t-ipnet.de |
I set the PJSIP trunks transport to 0.0.0.0-tcp and so far I don’t know how to force TCP. Is it maybe possible within the Client and Server URIs?
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Sip Trunk Deutsche Telekom
The “;transport=tcp” parameter likely needs to be added to the server URI. In a .conf file it also needs to be escaped with “” at front thus “;transport=tcp”
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