Using a Grandstream phone and I had auto updates for firmware turned on
The problems just started a few days ago – and I believe that it may have to do with Grandstream just having released a new firmware – I did not make any other changes on the PBX so again guessing that might be the issue?
Here is the link to the new firmware http://firmware.grandstream.com/Release_Note_GXP16xx_1.0.4.152.pdf
thanks
Someone is using ## or *2 to do a transfer. You have Dial options of Tt set which allows transfers from caller or calling party.
[2020-01-12 00:53:52] VERBOSE[20234][C-0000087c] pbx.c: -- Executing [s@macro-auto-blkvm:8] ExecIf("Local/17172937715@from-internal-0000025a;1", "0?Set(MASTER_CHANNEL(CONNECTEDLINE(name))=)") in new stack
[2020-01-12 00:53:58] VERBOSE[20236][C-0000087c] res_musiconhold.c: -- Started music on hold, class 'default', on DAHDI/1-1
[2020-01-12 00:53:58] VERBOSE[20236][C-0000087c] file.c: -- <Local/17172937715@from-internal-0000025a;2> Playing 'pbx-transfer.ulaw' (language 'en')
[2020-01-12 00:54:05] VERBOSE[20279][C-0000087c] pbx.c: -- Executing [18769263700@from-internal-xfer:1]
This issue was discussed in another thread, which I can’t find at the moment. Depending on the complexity of your dialplan, there are more opportunities for someone to bypass the safeties and use the transfer codes.
RFC7462 I don’t find in the Advanced Settings with latest software 7.6.2.f.
Where exacty do I find this settings?
As @billsimon said.
It appears that the incoming call is being forwarded to an external 717-293-xxxx number, using DAHDI (which lacks answer supervision). So, the call appears almost immediately answered and as an outgoing call, which allows the transfer.
You can confirm the vulnerability by calling in from your mobile and pressing *2 or ## while the forwarded-to number is ringing.
Then, set Asterisk Trunk Dial Options for your trunks to (blank) (remove any T or t from the list). Submit, Apply Config, retest to confirm that *2 and ## are ignored.
Of course, this prevents the use of these codes to transfer calls. Assuming that you have IP phones with transfer buttons or softkeys, you should not need these codes.
-Now it gets “407407 Proxy Authentication Required” from remote SIP trunk. What amI missing in outgoing config?
yep indeed.
[2020-01-12 00:53:48] VERBOSE[20234][C-0000087c] res_agi.c: -- dialparties.agi: Added extension 17172937715# to extension map
[2020-01-12 00:53:48] VERBOSE[20234][C-0000087c] res_agi.c: -- dialparties.agi: Extension 17172937715# cf is disabled
[2020-01-12 00:53:48] VERBOSE[20234][C-0000087c] res_agi.c: -- dialparties.agi: Filtered ARG3: 17172937715
[2020-01-12 00:53:52] VERBOSE[20234][C-0000087c] pbx.c: -- Executing [s@macro-auto-blkvm:8] ExecIf("Local/17172937715@from-internal-0000025a;1", "0?Set(MASTER_CHANNEL(CONNECTEDLINE(name))=)") in new stack
[2020-01-12 00:53:58] VERBOSE[20236][C-0000087c] res_musiconhold.c: -- Started music on hold, class 'default', on DAHDI/1-1
[2020-01-12 00:53:58] VERBOSE[20236][C-0000087c] file.c: -- <Local/17172937715@from-internal-0000025a;2> Playing 'pbx-transfer.ulaw' (language 'en')
[2020-01-12 00:54:05] VERBOSE[20279][C-0000087c] pbx.c: -- Executing [18769263700@from-internal-xfer:1] Macro("Local/18769263700@from-internal-xfer-0000025b;2", "user-callerid,LIMIT,EXTERNAL,") in new stackInbound callers exploiting the in-call transfer feature codes was a known issue a few months ago and resolved in core versions 14.0.28.19 and 15.0.9.46.
There was an update done over the Christmas/New Years break, so I guess this could be a bug.
Thanks for the reply!
This has also popped up in the dashboard. I didn’t change anything manually there and I have restarted asterisk before and this went away, but now it is back. I’m also not sure if it is relevant.
Just after I posted this, I got another message there that Allow Transports Reload was on, but recommended to be off. I turned it off and both of those messages went away, but my phones still aren’t connecting.
Hello, guys.
I want to set up two features:
The question is: can i do that without buying commercial modules or writing some custom extensions/contexts ?
Thank you
Once again, need to update systems as often as possible.
In Advanced Settings
Set Display Readonly Settings to Yes and Override Readonly Settings to Yes, reload the page and look again for it.
Not without custom context AFAIK.
On my Cisco 8961 phones the newest SIP firmware version I can use is 9-2-2SR1-9 (Dec 2011), otherwise „call completed elsewhere“ doesn‘t work.
Yes, you can download old firmware versions from Cisco.
It isn’t really the solution I was looking for, but switching the extensions from pjsip to sip seems to solve the “my users aren’t getting calls” problem, but not the “why did this break” problem. Going to switch the phones over for now and I’ll see if I can figure out why pjsip broke. Anyone have any troubleshooting suggestons?
Hi all
I’m trying to setup a freepbx server on VMWARE 15 but I got one problem. I have read the previous posts but they don’t seem to resolve my problem.
I want to configure two networks on my server in VMWARE : one for the updates of my server and the other to be able to access locally to my server.
So i put the vnet0 on NAT and I’m able to make updates but when I want to put a static address on the other interface(vnet0) I lost the internet connection.
My freepbx server must also be able to connect to a gsm gateway on a external network.
I’ve probably missed something but I don’t know what.
Is someone could help me?
Thanks
In-call transfer (by DTMF) is a specialty function that is needed by very few users. In addition to security issues, the feature sometimes interferes with DTMF control of a remote system or results in loud beeps being played to a caller. Metadata displayed on the transferring phone is often inaccurate.
IMO, these features should be disabled by default for new installs.
If needed, it should be enabled only for the required path. For example, if you use Follow Me to forward some calls to mobiles who need to transfer calls, do that via a ‘special’ trunk with the ‘t’ option, so the authorized mobile user can perform the transfer. Normal outbound calls should not have that option, otherwise the callee could trick the caller into placing him on hold, then make a fraudulent call via *2.
Hello, Sangoma?
Any word on the issue of Hot Desking phones that are currently not logged in?
As well as the issue of users who work in different locations to update the emergency CID?
There is a ticket open with this suggestion; please vote for it. https://issues.freepbx.org/browse/FREEPBX-20583
I might be wrong but is *2 (?? In-Call Transfer) required for Parking Lot? I remember in the past trying to have it work but could not without having *2 enabled.