By any chance are you running one of the buggy Asterisk versions?
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Constant crashes / server locks up complety
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Grandstream HT-701 attempting to register on pjsip
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How to setup reliance JIO trunk (SIP provider in INDIA)
Hi,
Dear Community Team members,
How to configure reliance JIO SIP trunk (provider in india)
They are provide following information;
Signal IP: 10.1.1.5
Media IP: 10.1.1.6
Pilat Number: 9999999944
DID Range: 9999999942-43 and 9999999945-65
Username & password is not required
how to setup a sip trunk with the above information, Please find the connectivity diagram
Please suggest a solution (Is there any help doc?)
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Extensions_additional.conf | Where are the include elements located?
So I added my sub-diversion-header related dial plan to the extensions_custom.conf file which looked like this prior to the manipulation:
[ats-internal]
exten => _X.,1,NoOp(Default routing to SIP Endpoint)
same => n,Dial(PJSIP/${EXTEN},20)
…and like this after the manipulation:
[ats-internal]
exten => _X.,1,NoOp(Default routing to SIP Endpoint)
same => n,Dial(PJSIP/${EXTEN},20)
[macro-dialout-trunk-predial-hook]
exten => s,1,Set(DIVERSION_REASON=${IF($[${LEN(${DIVERSION_REASON})}=0]?no-answer:${DIVERSION_REASON})})
exten => s,n,Gosub(func-set-sipheader,s,1(Diversion,tel:+49899876${FROM_DID};reason=${DIVERSION_REASON};screen=no;privacy=off))
exten => s,n,MacroExit()
Result:
On forwarded calls, the diversion header looks exactly like before:
Diversion: <sip:123@mycompany.arcor.de>;reason=unconditional
, lacking the trunk number and the forwarded call will fail due to authentication error.
Interestingly, on direct external calls (e.g. from my sip desk phone to a mobile device), the call will fail and the diversion header looks like this:
Diversion: tel:+49899876
Conclusion:
The [macro-dialout-trunk-predial-hook] has an unwanted effect on outgoing calls, but on forwarded calls, the diversion header is not interested in any way what the predial hook tells him.
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Extensions_additional.conf | Where are the include elements located?
Are you sure your Gosub even works in your dial plan?
Why not try something simple first.
Chansip or pjsip?(SIPAddHeader command is for chansip):
[macro-dialout-trunk-predial-hook]
exten => s,1,SIPAddHeader(Diversion:<sip:+49899876@mycompany.arcor.de>;reason=unconditional)
exten => s,n,MacroExit()
There is also a GUI setting that let’s you generate diversion headers in advanced settings. Maybe this is all that you need:
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Freepbx dependence on external services
I installed freepbx distro on hyper-v windows 10 successfully. When I visit the dashboard, i see some ads and I have noticed from the lower left of the browser, that it is connecting to cdn.schmoozecom.com.
Is it possible to remove ads and load everything locally so that I do not have to depend on external services? For example if somehow cdn.schmoozecom.com goes down or shutsdown completely, my freepbx project goes completely useless.
I am sure I have conveyed my point
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Extensions_additional.conf | Where are the include elements located?
Well, the original dial plan is
[sub-diversion-header]
include => sub-diversion-header-custom
exten => s,1,Set(DIVERSION_REASON=${IF($[${LEN(${DIVERSION_REASON})}=0]?no-answer:${DIVERSION_REASON})})
exten => s,n,Gosub(func-set-sipheader,s,1(Diversion,tel:${FROM_DID};reason=${DIVERSION_REASON};screen=no;privacy=off))
exten => s,n,Return()
;–== end of [sub-diversion-header] ==–;
The line to investigate should be this one:
sipheader,s,1(Diversion,tel:${FROM_DID};reason=${DIVERSION_REASON};screen=no;privacy=off))
And the SIP trace shows this diversion header on forwarded calls:
sip:123@mycompany.arcor.de;reason=unconditional
If I am not totally wrong ${FROM_DID} shall reflect the extension number of the DID the caller dialed. In this case it should be the 123 and the diversion header shows that. So I should assume that the origianl dial plan works, shouldn’t I?
You mentioned the advanced settings’ Generate Diversion Headers option in the GUI. I already knew about that option and I played around with it. Interestingly, the SIP trace always shows the diversion header no matter if the above mentioned option is set to YES or NO.
I tried the [macro-dialout-trunk-predial-hook] you provided. The extensions_custom.conf lokked lik this having added it:
[ats-internal]
exten => _X.,1,NoOp(Default routing to SIP Endpoint)
same => n,Dial(PJSIP/${EXTEN},20)
[macro-dialout-trunk-predial-hook]
exten => s,1,SIPAddHeader(Diversion:<sip:+49899876@mycompany.arcor.de>;reason=unconditional)
exten => s,n,MacroExit()
Result:
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Outgoing calls are not affected. They work as expected.
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Forwarded Calls still do not work. The diversion header in the SIP trace still looks like this:
sip:123@mycompany.arcor.de;reason=unconditional
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Callback module
I have installed the callback module for testing. Am I right that it does the following:
- I call the freepbx from my mobile
- Do something
- I’m called back by the freepbx and a call is setup using the outbound caller id so a) my mobile bill does not run up and b) it seems as if I’m calling from the office
Is this correct? And what should be done on “Do something”? The system must know which number to call, right?
The workings of the module are not quite clear to me.
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P-Asserted-Identity with caller withheld "hidden" calls
I have just read on another post P-Asserted-Identity on Call Forward problem that I may need to delete the
trustrpid=yes
sendrpid=pai
to get custom code working in a dial plan, is this what I am missing?
I cant test this at the minute but just looking any pointers to help solve this.
Thx
D
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Flashing Sangoma Logo
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Freepbx dependence on external services
Block it at your firewall and see. I do not think anything becomes “useless” except your dashboard’s blog feed.
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How to install lame?
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Extensions_additional.conf | Where are the include elements located?
have you validated that if clause within the context you are attempting to check things?
Also have you validated ${FROM_DID} it sounds like you are simply assuming.
dump the channel and look at things.
exten => s,n,DumpChan()
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Freepbx dependence on external services
Weren’t there reports of slowness on the dashboard a while back if things were not resolving?
Did that get fixed?
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Custom trunk context to remove 1 from DID
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Dropped calls
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Endpoint Manager and HTTP Auth
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Extensions_additional.conf | Where are the include elements located?
Where would I have to add this line in order to dump the channel? To the [macro-dialout-trunk-predial-hook] ?
Like this?
[macro-dialout-trunk-predial-hook]
exten => s,1,SIPAddHeader(Diversion:<sip:+49899876@mycompany.arcor.de>;reason=unconditional)
exten => s,n,DumpChan()
exten => s,n,MacroExit()
And then I could investigate the asterisk log file /var/log/asterisk/full to see the dump’s output?
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How to determine which party hangs up the calls
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Extensions_additional.conf | Where are the include elements located?
OK there are a couple things I’m seeing in this thread that should be addressed.
- The semi-colon
(;)in Asterisk is a comment delimiter and must be escaped. So it needs to be like this:
exten => s,1,SIPAddHeader(Diversion<sip:+49899876@mycompany.arcor.de>\;reason=unconditional)
That will escape the semi-colon. It needs to be done each time a semi-colon is used for these things.
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${FROM_DID} is only set during the Inbound Route and it is not a global variable. Which means it will not be set on child channels that result from the inbound call.
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The actual call flow needs to be outlined here. When is this call being forwarded? I see Dial() code to hit an extension so is this when the extension is a no answer? What is going on here?
The last question would shed some light on at what point in the dialplan this is expected to happen so we can fully understand what variables and processes are being hit. There’s not even a single debug showing how this is being processed.
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