Opening Door : 405 Method not allowed

[NoFate]

Hi

i have an hikvision intercom (door) , hikvision indoor panel and some linphone sip clients
when i call from doorstation, when i accept call with linphone clients, i can press #1 , that opens the door
from the indoor panel, i have there a key button, but that doesnt open the door, instead i get message 405 not allowed

this used to work on freepbx? how i can allow this type of message , so i can also open door from indoor panels?

sreenshot 1 : log from indoor panel
screenshot 2 : log from linphone, seems it uses dtmf there..

so i need to allow somthing in fusionpbx , no sure how i need to configure it?

thnx

 

[Adrian Fretwell]

I Assume that the Yate device is telling you that whilst it understands that you are wanting use the MESSAGE method, this message type (MESSAGE) is not allowed for the address specified in the request URI.

 

[NoFate]

Yeah, can I overwrite it somehow ? So that fusionpbx rewrites it somehow?

Because it used to work on FreePBX , with same clients

 

[Adrian Fretwell]

The SIP message method type "MESSAGE" is part of a group of SIP message methods called Extension Methods. MESSAGE is used to send Instant Messages, the message being contained in the body as a MIME attachment as it is in your case:

Content Type: text/plain
...
<LocknumXML>
<lockNum>0</lockNum>
</LocknumXML>

There is no substitute for this kind of message for an instant message, if it works on FreePBX, get a packet capture and see if a different method type is being used, maybe a NOTIFY fro example. I say NOTIFY because this method can also carry the XML payload as it does for BLF notifications.

 

[NoFate]

yeah, gonna, setup a freepbx again and make a trace, if it works on freepbx, it must be possible somehowm too with fusion, right?
because there is no setting i can do on the sip client side

 

[Adrian Fretwell]

I would expect the key press to go as an RTP telephone event, I'm trying to understand how this SIP message fits into this jigsaw. There is more going on than just the dialogue between 192.168.0.70 and 0.80 because 192.168.0.71 is also mentioned. Does the DTMF payload between two devices cause a SIP message to be sent to/from a third device? Sometimes difficult to grasp when you don't have the actual SIP trace in front of you.

 

[NoFate]

yes
192.168.0.70 is the door intercom
192.168.0.71 is the SIP indoor client
192.168.0.80 is the PBX

so i press the doorbutton outside , so .70 calls .71

the second screenshot in first post, there i did a call from 0.70 to a linphone client on android

but right now, setting up again a freepbx, gonna make a trace later

 

[NoFate]

ok, setup and tested, it works on freepbx running on 192.168.0.174
linphone client 192.168.0.168
indoor client 192.168.0.71
doorstation (caller) 192.168.0.10

logfile 1 wih linphone android client => chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP

Dropbox - File Deleted - Simplify your life

www.dropbox.com

logfile 1 with indoor (not working on fusion, but works on freepbx)

Dropbox - File Deleted - Simplify your life

www.dropbox.com

[2021-10-14 17:22:51] VERBOSE[2267] chan_sip.c: --- (11 headers 3 lines) ---
[2021-10-14 17:22:51] VERBOSE[2267][C-00000005] chan_sip.c: Receiving message!
[2021-10-14 17:22:51] VERBOSE[2267][C-00000005] chan_sip.c: SIP Text message received: '<locknumXML>
<lockNum>0</lockNum>
</locknumXML>'

...

so why does it work on freepbx and not on fusion? i'm missing somethign? all clients are the same setup, so must be on PBX side

thnx

 

[Adrian Fretwell]

But even in your freepbx log we still see the 405:

<------------>
[2021-10-14 17:22:51] VERBOSE[32145][C-00000005] chan_sip.c: Sending text <locknumXML>
<lockNum>0</lockNum>
</locknumXML> on SIP/2000-00000008
[2021-10-14 17:22:51] VERBOSE[32145][C-00000005] chan_sip.c: Reliably Transmitting (NAT) to 192.168.0.70:5060:
MESSAGE sip:2000@192.168.0.70:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.174:5160;branch=z9hG4bK13745afd;rport
Max-Forwards: 70
From: <sip:2001@192.168.0.174:5160>;tag=as217079d5
To: "2000" <sip:2000@192.168.0.70>;tag=723954776
Call-ID: 1801371064@192.168.0.70
CSeq: 102 MESSAGE
User-Agent: FPBX-15.0.17.34(16.17.0)
Content-Type: text/plain;charset=UTF-8
Content-Length: 47

<locknumXML>
<lockNum>0</lockNum>
</locknumXML>
---
[2021-10-14 17:22:51] VERBOSE[2267] chan_sip.c:
<--- SIP read from UDP:192.168.0.70:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.174:5160;branch=z9hG4bK13745afd;rport=5160;received=192.168.0.174
From: <sip:2001@192.168.0.174:5160>;tag=as217079d5
To: "2000" <sip:2000@192.168.0.70>;tag=723954776
Call-ID: 1801371064@192.168.0.70
CSeq: 102 MESSAGE
Server: YATE/5.5.0
Content-Length: 0

<------------->
[2021-10-14 17:22:51] VERBOSE[2267] chan_sip.c: --- (8 headers 0 lines) ---
[2021-10-14 17:22:51] VERBOSE[2267] chan_sip.c:
<--- SIP read from UDP:192.168.0.70:5060 --->
SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP 192.168.0.174:5160;branch=z9hG4bK13745afd;rport=5160;received=192.168.0.174
From: <sip:2001@192.168.0.174:5160>;tag=as217079d5
To: "2000" <sip:2000@192.168.0.70>;tag=723954776
Call-ID: 1801371064@192.168.0.70
CSeq: 102 MESSAGE
Server: YATE/5.5.0
Allow: ACK, INVITE, BYE, CANCEL, MESSAGE, OPTIONS, INFO
Content-Type: text/plain
Content-Length: 1

So maybe the 405 is not the actual reason it is not working with FusionPBX?

 

[NoFate]

indeed, but what is it then
the door opens on freepbx , thats the most important

 

[Adrian Fretwell]

OK, look at how the DTMF is being transmitted. On your FreePBX logs it looks like the keypad digits are being transmitted using SIP INFO messages as opposed to rfc2833. You may have a mismatch of DTMF types when using FusionPBX.

https://freeswitch.org/confluence/display/FREESWITCH/dtmf_type

 

[NoFate]

Ok, that's something I can change in fusionpbx right? Gonna have a look later, it should be on default settings now , not sure how that's configured

 

[Adrian Fretwell]

Yes, by default is is set to rfc2833 in the SIP profile. You can override it using channel variables as shown in the link above.

 

[NoFate]

ok, gonna try later, already setup like below, fingers crossed

 

[NoFate]

that was it!!
much appreciated

1 issue left, is now to have ealy video on all sip clients before i pickup , it only works when i call a sip client
want to have it too when using a ringgroup
do you think its possible?

 

[Adrian Fretwell]

1 issue left, is now to have ealy video on all sip clients before i pickup , it only works when i call a sip client
want to have it too when using a ringgroup
do you think its possible?

I don't the answer, but I rather suspect that it is not possible. Try starting a new thread with that question and see if you get any useful responses.