Since dahdi is it’s own process and Asterisk just talks to it and FreePBX talks to Asterisk, it might be a good experiment to “hand” stop and start DAHDI, then to an “fwconsore restart”.
It’s possible that the dahdi process has locked itself up or that the card and dahdi are “arguing” somehow.
@methenothing - as we move forward, this answer is going to become more and more true. With the deprecation of Chan-SIP and the inexorable drive to improve the system, more and more features are just going to get left behind.
I hope that you’ll join us in this improvement process by posting what you end up getting working so that other can benefit from your experience and hard work.
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I requested any logfiles from the PRI adtran to see if we might glean any info there. The issues certainly seem to reside between the system and the PRI.
Hi,
I’ve got a few pjsip extensions registered using freepbx.
Here’s the (partial) output from pjsip show endpoints:
Endpoint: 100/100 Not in use 0 of inf
InAuth: 100-auth/100
Aor: 100 1
Contact: 100/sip:100@192.168.1.12:12410;transport=T 127383ff5c Avail 17.425
Endpoint: 101/101 Not in use 0 of inf
InAuth: 101-auth/101
Aor: 101 1
Contact: 101/sip:101@192.168.1.36:37249;transport=T c01a62a4d0 Avail 3.575
For some reason, internal calls do not work.
I have so far:
from-internal
X. to XXXX., because previously these internal calls would be routed out via a trunk101 and (with prefix)*101
I can see in asterisk that the following dialplan function is triggered:
-- Executing [100@from-internal:5] Playback("PJSIP/101-0000000f", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
I’d appreciate any ideas how to fix this.
Can you provide the full call trace of a failed call via pastebin?
https://wiki.freepbx.org/display/SUP/Providing+Great+Debug#ProvidingGreatDebug-AsteriskLogs-PartII
Hello guys,
Thanks for your reply, and sorry for my late reply.
The problem seems to be resolved by changing DTFM mode from “auto” to “rfc2833”. Really weird, but was fixed using it (for now, and I hope it’ll stay like that for the next 2 years, haha).
Thanks for your help and good luck for the future.
In Asterisk SIP Settings, try setting Local Networks to:
192.168.10.0 / 24
111.111.64.148 / 30
111.111.3.10 / 32
After changing these, restart (not just reload) Asterisk and test. If no luck, post a new SIP trace.
Doing this kind of stuff with a relatively ancient Asterisk without pjsip is tough, because you can’t set up different transports for the networks.
Here you go:
[2020-06-05 16:41:31] VERBOSE[25494] pbx_variables.c: Setting global variable 'SIPDOMAIN' to 'asterisk.xxx.org'
[2020-06-05 16:41:31] VERBOSE[25494] netsock2.c: Using SIP RTP Audio TOS bits 184
[2020-06-05 16:41:31] VERBOSE[25494] netsock2.c: Using SIP RTP Audio TOS bits 184 in TCLASS field.
[2020-06-05 16:41:31] VERBOSE[25494] netsock2.c: Using SIP RTP Audio CoS mark 5
[2020-06-05 16:41:31] VERBOSE[25732][C-00000002] pbx.c: Executing [101@from-internal:1] ResetCDR("PJSIP/100-00000001", "") in new stack
[2020-06-05 16:41:31] VERBOSE[25732][C-00000002] pbx.c: Executing [101@from-internal:2] NoCDR("PJSIP/100-00000001", "") in new stack
[2020-06-05 16:41:31] VERBOSE[25732][C-00000002] pbx.c: Executing [101@from-internal:3] Progress("PJSIP/100-00000001", "") in new stack
[2020-06-05 16:41:31] VERBOSE[25732][C-00000002] pbx.c: Executing [101@from-internal:4] Wait("PJSIP/100-00000001", "1") in new stack
[2020-06-05 16:41:32] VERBOSE[25732][C-00000002] pbx.c: Executing [101@from-internal:5] Playback("PJSIP/100-00000001", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
[2020-06-05 16:41:32] VERBOSE[25732][C-00000002] file.c: <PJSIP/100-00000001> Playing 'silence/1.slin' (language 'de_DE')
[2020-06-05 16:41:33] VERBOSE[25732][C-00000002] file.c: <PJSIP/100-00000001> Playing 'cannot-complete-as-dialed.slin' (language 'de_DE')
[2020-06-05 16:41:35] VERBOSE[25732][C-00000002] file.c: <PJSIP/100-00000001> Playing 'check-number-dial-again.slin' (language 'de_DE')
[2020-06-05 16:41:37] VERBOSE[25732][C-00000002] pbx.c: Executing [101@from-internal:6] Wait("PJSIP/100-00000001", "1") in new stack
[2020-06-05 16:41:38] VERBOSE[25732][C-00000002] pbx.c: Executing [101@from-internal:7] Congestion("PJSIP/100-00000001", "20") in new stack
[2020-06-05 16:41:38] VERBOSE[25732][C-00000002] pbx.c: Spawn extension (from-internal, 101, 7) exited non-zero on 'PJSIP/100-00000001'
[2020-06-05 16:41:38] VERBOSE[25732][C-00000002] pbx.c: Executing [h@from-internal:1] Macro("PJSIP/100-00000001", "hangupcall") in new stack
[2020-06-05 16:41:38] VERBOSE[25732][C-00000002] pbx.c: Executing [s@macro-hangupcall:1] GotoIf("PJSIP/100-00000001", "1?theend") in new stack
[2020-06-05 16:41:38] VERBOSE[25732][C-00000002] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2020-06-05 16:41:38] VERBOSE[25732][C-00000002] pbx.c: Executing [s@macro-hangupcall:3] ExecIf("PJSIP/100-00000001", "0?Set(CDR(recordingfile)=)") in new stack
[2020-06-05 16:41:38] VERBOSE[25732][C-00000002] pbx.c: Executing [s@macro-hangupcall:4] NoOp("PJSIP/100-00000001", " montior file= ") in new stack
[2020-06-05 16:41:38] VERBOSE[25732][C-00000002] pbx.c: Executing [s@macro-hangupcall:5] GotoIf("PJSIP/100-00000001", "1?skipagi") in new stack
[2020-06-05 16:41:38] VERBOSE[25732][C-00000002] pbx_builtins.c: Goto (macro-hangupcall,s,7)
[2020-06-05 16:41:38] VERBOSE[25732][C-00000002] pbx.c: Executing [s@macro-hangupcall:7] Hangup("PJSIP/100-00000001", "") in new stack
[2020-06-05 16:41:38] VERBOSE[25732][C-00000002] app_macro.c: Spawn extension (macro-hangupcall, s, 7) exited non-zero on 'PJSIP/100-00000001' in macro 'hangupcall'
[2020-06-05 16:41:38] VERBOSE[25732][C-00000002] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on 'PJSIP/100-00000001'
Its interesting that it only drops the calls when the call is made from a softphone which is using Asterisk SIP chan_pjsip
when the call is started from zulu its ok
How odd, there is no associated dialplan for your local extensions. How were they created? If you edit an extension, submit the page with no changes and apply config, does anything change?
Tom, thanks for your reply, so to restate what you just said,
if there is a reasonable expection that the device is portable and will be used that way…like a laptop, then we have 2 years to figure this out. if the expectation is it will not move regularly, then jan,2021…is that accurate?
thanks again
It appears in the log that Multicast send out several IF statements to cover various ALERTINFO statuses. Since this is Multicast and the receiver is just listening it doesn’t seem to include information from the receiving client in the log. I have however toggle back and forth between enabling Intercom enabled and disabled. But the results are the same.
Zulu doesn’t use SIP as signalling, that’s the reason.
Yesssssssssssss!
Thanks a lot @Stewart1
I installed the api module and performed the authentication operation by insomnia and received access token. I also sent some query by graphQl and got the answer. Now I need an api to be able to manage Inbound Routes from a client or for example through insomnia. So I’m looking for a dedicated module that can create, delete, or edit Inbound Routes. I couldn’t find the perfect documentation to build a module. What solution do you have for this?
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this solution physically connects to the phone with extra wires and I’m not sure it works when on speaker phone? I dont have a problem when I am on the handset its only when I am on speakerphone. I do a lot of technical support and while I am working there are periods of silence and people mistake this for me not being on the phone and they walk in and just start talking, sometimes saying stupid stuff I would rather my clients not hear. I like to be able to move my phone around my desk with the 1 wire it has not and dont want to have to add more wires or worse a handset lifter. Some of the solutions I have seen here with WiFi lights and RPi are very interesting. If I build an RPi we can just hand people a light bulb and ask them their extension. Brilliant concept!
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