I’m trying to reset my portal password. It’s asking for a passkey. I have no idea what of my passkey should be. Can any one give me an idea on what a passkey to reset the portal password?
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Sangoma portal passkey
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Quick Question - Intercom style phone system?
Yes you can transfer calls from a desk phone to a softphone if you want.
What model of Cisco phones are they? If they are the SPA series, they are easy to connect, otherwise they will not be easy as the others are meant for Cisco’s platform.
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Sangoma portal passkey
Please ignore this post. I didn’t read the password reset note carefully. the passkey is sent with the email to reset the password form. Sorry…
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FreePBX server compromised
Just an update, setting the UDP timeout to 660 seconds in the USG has so far resolved our issue completely. Even if we don’t change the phone registration expiration.
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Need to pass a valid external CID over an intra-company outbound route
I have a PJSIP trunk setup between two FreePBX 15 systems. trunk works perfectly for all scenarios the users need except one, Find Me/Follow Me.
Normally, calling from extension 121 to extension 5120 shows the CID of 121 on the screen for 5120. But when 5120 has their FMFM active to their cell phone, the call fails because almost no carriers allow invalid CID to be sent any more.
The problem is than a route marked intra-company drops all other CID info except the local CID info.
@lgaetz told me to check out this post for the same thing related to an IAX trunk.
Well that logic makes sense, but what can I do with a PJSIP trunk to pass the info? Setting a SIP header is easy enough, but then I need to know how to read it on the other side during the inbound call handling. Is there some other variable I can use that would hang around like the IAXVAR does?
Knowing we are going out an intra-company trunk is easy. Added a pjsip header was easy. I can confirm on the inbound call I see the header.
[macro-dialout-trunk-predial-hook]
exten => s,1,NoOp(Entering user defined context [macro-dialout-trunk-predial-hook] in extensions_custom.conf)
; Determine if call is on an intra-company route or not
exten => s,n,GotoIf($["${INTRACOMPANYROUTE}"="YES"]?intra_call:pstn_call)
; This is an intracompany call note the outbound CID in the sip header X-OUTBOUND-CID
exten => s,n(intra_call),Noop(This call uses an Intra-Company route)
; If the calling extension has an outbound CID specified, use it
exten => s,n,ExecIf($[${DB_EXISTS(AMPUSER/${AMPUSER}/outboundcid)}]?Set(x_out_cid=${DB(AMPUSER/${AMPUSER}/outboundcid)}))
; If there was no CID found, get the CID specified on the outbound route (yes labeled TRUNKCIDOVERRIDE when found in the route)
exten => s,n,ExecIf("${x_out_cid}"=="")?Set(x_out_cid=${TRUNKCIDOVERRIDE}))
; Add the result as a sip header, change to variable?
exten => s,n,GoSub(func-set-sipheader,s,1(X-Outbound-CID,${x_out_cid}))
exten => s,n,goto(exit_macro)
; This is not an intra-company call. Check for the outbound CID to have been sent along
exten => s,n(pstn_call),NoOp(This call is heading for the PSTN)
;exten => s,n, ;what to do here is the question.
; <snip> unrelated code for setting a header specific to a sip provider </snip>
exten => s,n(exit_macro),MacroExit()
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Need to pass a valid external CID over an intra-company outbound route
Rough guess, not tested, add this to the pstn_call path:
exten => s,n,Set(x_out_cid=${PJSIP_HEADER(read,X-OUTBOUND-CID)})
exten => s,n,ExecIf("${x-out-cid}"!="")?Set(CALLERID(all)=${x-out-cid}))
This is unfortunately after the CDR outbound_cnum and outbound_cnam would have been set, so you may want additional code to deal with that, if you care.
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Grandstream HT502
You’re a real piece of work.
I have the Grandstream HT502, a pair of Cisco ATAs and a Sipura.
None of which I have been able to get line-2 to work.
I had a requirement for three analog phones.
I ended up using The Grandstream, one of the Cisco and the Sipura as single line ATAs.
Both on the forums, and via Google search, the standard results are “Do this” and it doesn’t.
I have all three analog phones working now.
A 1910 Strowger PAX candlestick, my HP 8610 all-in-one printer (FAX) and a Crosley reproduction payphone.
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Grandstream HT502
There should be no problem using Line 2 on these ATAs. The two lines of course need to have different extension numbers. They also need different local ports, though that is the default.
If the second line doesn’t register, please paste a log of the registration attempt at pastebin.freepbx.org, after setting:
pjsip set logger on
or
sip set debug on
according to extension type.
If it registers ok, paste logs for failing inbound and/or outbound calls.
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Grandstream HT502
Well, that’s just it. “Should be no problem” but it was.
They had two different extension numbers, Line-1 was pjsip 5060 and line-2 was sip 5061.
Line-1 would register, and line-2 never did on all four ATAs.
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Grandstream HT502
The screenshot you posted for the 502 had SIP Registration: No, so of course it wouldn’t register. The question is why it wouldn’t register when you set it to Yes (the default).
Please turn that on and provide a log (with SIP trace) of attempted registration. If the extension configuration has any non-default settings other than extension number and secret, please provide details.
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Grandstream HT502
The HT-502 is working as a single line ATA at the moment, I’d rather not mess with it.
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Quick Question - Intercom style phone system?
Oh really, they are CP-7942
Is this going to be a problem?
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Quick Question - Intercom style phone system?
yes, this is the mother of all problems 
This type of Cisco phone uses a Cisco protocol. They offer alternative SIP firmware though…at Cisco. Maybe it is already installed?
These phones are the worst choice for a freePBX beginner, because you have to apply several fixes…to make them work…
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Attended Transfer Recording Options
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.
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Quick Question - Intercom style phone system?
Explains why they are so cheap !
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Possible to use VoIP extension as a Trunk for placing calls?
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Quick Question - Intercom style phone system?
Yes…excellent hardware, but poor SIP support. I still use Cisco 8961 phones with freePBX at one place.
I patched Asterisk (Asterisk Cisco BLF patch), configured an openLDAP server (for a central phonebook) etc
Took me years…literally…but has been working perfectly.
It is an interesting project…Cisco&freePBX…but a no-go for beginners.
And you would have to buy the Endpointmanager…to finally find out that it just gives you very basic functionality…with Ciscos.
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Quick Question - Intercom style phone system?
Well, I now have a headache!
I’m waiting for my PoE switch today, I’m not sure what to do now. lol
I guess I need to go look at getting new phones bought.
Unless i give it my all to get them working… but like you said as a beginner, I will be way out of my depth.
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Quick Question - Intercom style phone system?
You could try to use the EPM, but as I said, it gives you just around 20% of features. The BLF (busy lamp) wont work, no phonebook etc.
The Ciscos require a xml config file in the tftp directory of you freePBX server. The EPM is supposed to create such a file. Check the compatibility list of the EPM. Is your phone listed?
The other problem is…
Which firmware is installed on your phones? SIP? You would have to point the Ciscos to the IP address of your server to pick up the config file.
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Quick Question - Intercom style phone system?
If you have lots of time
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