Did's route in but terminate

WARNING[25828][C-0000014b]: func_channel.c:538 func_channel_read: Unknown or unavailable item requested: ‘reversecharge’

sip show peers:

206/206 192.168.1.109 D Yes Yes A 5060 OK (9 ms)
207/207 192.168.1.108 D Yes Yes A 5060 OK (7 ms)
208/208 192.168.1.107 D Yes Yes A 5060 OK (8 ms)
209/209 192.168.1.104 D Yes Yes A 5060 OK (8 ms)
210/210 192.168.1.90 D Yes Yes A 5060 OK (8 ms)
211/211 192.168.1.131 D Yes Yes A 5060 OK (9 ms)
212/212 192.168.1.132 D Yes Yes A 5060 OK (8 ms)
213/213 192.168.1.133 D Yes Yes A 5060 OK (9 ms)
215/215 192.168.1.135 D Yes Yes A 5060 OK (9 ms)
216/216 192.168.1.134 D Yes Yes A 5060 OK (6 ms)
250/250 192.168.1.111 D Yes Yes A 5060 OK (7 ms)
251/251 192.168.1.30 D Yes Yes A 56452 OK (7 ms)
252/252 192.168.1.130 D Yes Yes A 5060 OK (15 ms)
300/300 192.168.3.158 D Yes

sip show channels has:

Peer User/ANR Call ID Format Hold Last Message Expiry Peer
192.168.1.132 (None) 3fd51eeb-ae2d7f (nothing) No Rx: REGISTER
192.168.1.134 (None) 180f617-28712ce (nothing) No Rx: REGISTER
192.168.1.130 (None) 233bf24d-2ebe9a (nothing) No Rx: REGISTER
3 active SIP dialogs
localhostCLI> sip show channels
Peer User/ANR Call ID Format Hold Last Message Expiry Peer
192.168.1.132 (None) 3fd51eeb-ae2d7f (nothing) No Rx: REGISTER
192.168.1.134 (None) 180f617-28712ce (nothing) No Rx: REGISTER
192.168.1.131 (None) 7b6eedeb-c77a8f (nothing) No Rx: REGISTER
192.168.1.109 (None) e2089807-2d4c28 (nothing) No Rx: REGISTER
192.168.1.107 (None) e2089807-2d4c28 (nothing) No Rx: REGISTER
192.168.1.130 (None) 233bf24d-2ebe9a (nothing) No Rx: REGISTER
6 active SIP dialogs
localhost
CLI> sip show channels
Peer User/ANR Call ID Format Hold Last Message Expiry Peer
192.168.1.132 (None) 3fd51eeb-ae2d7f (nothing) No Rx: REGISTER
192.168.1.134 (None) 180f617-28712ce (nothing) No Rx: REGISTER
192.168.1.131 (None) 7b6eedeb-c77a8f (nothing) No Rx: REGISTER
192.168.1.104 (None) 94f3b110-5b8d41 (nothing) No Rx: REGISTER
192.168.1.101 (None) 94f3b110-5b8d41 (nothing) No Rx: REGISTER
192.168.1.109 (None) e2089807-2d4c28 (nothing) No Rx: REGISTER
192.168.1.107 (None) e2089807-2d4c28 (nothing) No Rx: REGISTER
192.168.1.130 (None) 233bf24d-2ebe9a (nothing) No Rx: REGISTER
192.168.1.112 (None) 1b2b0498-f6b242 (nothing) No Rx: REGISTER

Sorry Freepbx framework 12.0.76.2 Enabled

they were down all day to day just need some kind of suggestion
here is additional sip.conf
[805]
deny=0.0.0.0/0.0.0.0
secret=*********
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
mediaencryption=no
sendrpid=pai
type=friend
nat=force_rport,comedia
port=5060
qualify=yes
qualifyfreq=60
transport=udp,tcp,tls
avpf=no
force_avp=no
icesupport=no
encryption=no
callgroup=
pickupgroup=
dial=SIP/805
permit=0.0.0.0/0.0.0.0
callerid=423alarm <805>
callcounter=yes
faxdetect=no
cc_monitor_policy=generic

[900]
deny=0.0.0.0/0.0.0.0
secret=*******
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
mediaencryption=no
sendrpid=pai
type=friend
nat=force_rport,comedia
port=5060
qualify=yes
qualifyfreq=60
transport=Auto
avpf=no
force_avp=no
icesupport=no
encryption=no
callgroup=
pickupgroup=
dial=SIP/900
permit=0.0.0.0/0.0.0.0
callerid=900 <900>
callcounter=yes
faxdetect=no
cc_monitor_policy=generic

[admin]
host=192.168.1.18
defaultuser=admin
secret=*********
type=friend
fromdomain=localhost
context=from-pstn
insecure=port,invite
trustrpid=yes
sendrpid=yes
directmedia=no
qualify=yes
keepalive=45
nat=force_rport,comedia
dtmfmode=rfc2833
allow=ulaw

[sangoma]
host=192.168.1.18
defaultuser=admin
secret=******
type=friend
fromdomain=localhost
context=from-pstn
insecure=port,invite
trustrpid=yes
sendrpid=yes
directmedia=no
qualify=yes
keepalive=45
nat=force_rport,comedia
dtmfmode=rfc2833
allow=ulaw

the sangoma is the sip gateway

Hi!

That message usually indicates that you chose to set “Reject Reverse Charges” to “Yes” on a non PRI channel (SIP, DAHDI, etc…).

There was however a bug that was causing it to appear for non PRI channels even if you had set it “Reject Reverse Charges” to “No”.

It was corrected for FreePBX 13 (I submitted the fix for this) but I am not sure if it was ever backported by Andrew (one of the FreePBX developers) to FreePBX 12.

If you have a problem it’s not related to this, this was simply a very annoying error message… You need to provide both more information on your problem and more logs…

Good luck and have a nice day!

Nick
"

thank you for the reply what do u need?
All calls in bound come in and ring the correct phone but then get disconnected.

where would the setting be? on the sip gateway (vega 100g)

Hi!

The setting that could cause this error message is on the inbound routes of your FreePBX box…

It’s now however the cause of your problem as it’s unable to get it from the type of channel you are using (which appear to be solely SIP).

If you had been using PRI channels and had erroneously set that setting to Yes if could have cause calls which had their charges reversed to be dropped but it’s not the case as it’s not even able to retrieve the variable because it’s a non-PRI channel…

Good luck and have a nice day!

Nick

i went there but nothing was checked

Hi!

I almost exclusively use DAHDI here (except for the connection to the outside world which is SIP) so I am definitely not the best person to help you with this…

That said it sounds like it’s neither your SIP gateway nor your FreePBX box which is dropping your calls but actually your phones.

Did this ever worked?

If it did, what changed before it stopped working?

I am definitely no expert on SIP but it sounds like your phones are unable to negotiate with your FreePBX box something on something they can use…

Could it be possible that it could be the codec or something similar (like telling your FreePBX box to use ulaw but your phone to exclusively use g.729 (which you must have a license to use by the way and for which you must pay if you chose this)?

What I would try to get to the bottom of this is something like

asterisk -rvvvvvvvvv

on the command line and issue something like

sip set debug on

or (if you don’t want to be flooded by too much logging)

sip set debug IP_ADDRESS (where IP_ADDRESS is the IP address of one of your phones)

and try to see if it tells you what it dislikes…

Good luck and have a nice day!

Nick

they went from 13 pots lines to a pri,
i am trying the debug now
thanks

<— Transmitting (NAT) to 192.168.1.102:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK7b8f695a3180A85;received=192.168.1.102;rport=5060
From: “Jess-204” sip:[email protected];tag=13FCA52C-CD3B821F
To: sip:[email protected];tag=as50a61940
Call-ID: [email protected]
CSeq: 765 REGISTER
Server: FPBX-12.0.76.2(11.19.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="7cf7ef9c"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.1.102:5060 —>
REGISTER sip:192.168.1.25:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK9dac3f1c6FC33FCF
From: “Jess-204” sip:[email protected];tag=13FCA52C-CD3B821F
To: sip:[email protected]
CSeq: 766 REGISTER
Call-ID: [email protected]
Contact: sip:[email protected];methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.3.2.0413
Accept-Language: en
Authorization: Digest username=“204”, realm=“asterisk”, nonce=“7cf7ef9c”, uri=“sip:192.168.1.25:5060”, response=“1dd943b1f742291d52432f8af33dbbe7”, algorithm=MD5
Max-Forwards: 70
Expires: 120
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Sending to 192.168.1.102:5060 (NAT)
Reliably Transmitting (NAT) to 192.168.1.102:5060:
OPTIONS sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.25:5060;branch=z9hG4bK77a0541a;rport
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as5fb77867
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-12.0.76.2(11.19.0)
Date: Tue, 20 Oct 2015 04:53:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<— Transmitting (NAT) to 192.168.1.102:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK9dac3f1c6FC33FCF;received=192.168.1.102;rport=5060
From: “Jess-204” sip:[email protected];tag=13FCA52C-CD3B821F
To: sip:[email protected];tag=as50a61940
Call-ID: [email protected]
CSeq: 766 REGISTER
Server: FPBX-12.0.76.2(11.19.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 120
Contact: sip:[email protected];expires=120
Date: Tue, 20 Oct 2015 04:53:40 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘[email protected]:5060’ in 6400 ms (Method: NOTIFY)
Reliably Transmitting (NAT) to 192.168.1.102:5060:
NOTIFY sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.25:5060;branch=z9hG4bK722121a5;rport
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as54a205ee
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: [email protected]92.168.1.25:5060
CSeq: 102 NOTIFY
User-Agent: FPBX-12.0.76.2(11.19.0)
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 87

Messages-Waiting: no
Message-Account: sip:*[email protected]
Voice-Message: 0/0 (0/0)


Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.1.102:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.25:5060;branch=z9hG4bK77a0541a;rport
From: “Unknown” sip:[email protected];tag=as5fb77867
To: “Jess-204” sip:[email protected];tag=9C84FD09-AEE8CCB0
CSeq: 102 OPTIONS
Call-ID: [email protected]:5060
Contact: sip:[email protected]
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.3.2.0413
Accept-Language: en
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS

<— SIP read from UDP:192.168.1.102:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.25:5060;branch=z9hG4bK722121a5;rport
From: “Unknown” sip:[email protected];tag=as54a205ee
To: “Jess-204” sip:[email protected];tag=CB3FF982-253AC54D
CSeq: 102 NOTIFY
Call-ID: [email protected]:5060
Contact: sip:[email protected]
Event: message-summary
User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.3.2.0413
Accept-Language: en
Content-Length: 0

<------------->

Hi!

Did you try calling that extension when you logged this?

Good luck and have a nice day!

Nick

it was flying too fast

if i am just at the console and i call in that is when i get the
Ext. 5451:2 @ from-pstn: Friendly Scanner from 192.168.1.18
[2015-10-20 01:01:43] WARNING[32384][C-00000005]: func_channel.c:538 func_channel_read: Unknown or unavailable item requested: 'reversecharge

I wish dicko was around since he is most likely familiar with that kind of hardware but he is most likely sleeping (as I should be and will soon do…)

Good luck and have a nice day!

Nick

thanks have a good night

That’s kind of the problem with doing this…

The times I played with this I remoted my box with SSH and had the scroll back buffer of Putty set to insane values to catch most of it and then I pasted this into and editor to be able to search in it…

Calls between extensions work, right?

Only calls from the outside are rejected, right?

It looks like your SIP gateway is taking care of converting everything to SIP so assuming it’s correctly configured your FreePBX box probably only see it as a SIP trunk…

Have you tried relaxing the codecs used by at least of one your phones to see if it makes any difference?

Sounds to me your box could be having difficulty negotiating something like the codec or other similar thing…

Don’t forget you also have a similar setting on the FreePBX box in the SIP settings…

Good luck and have a nice day!

Nick

nice
i found some of the jabbering, the global setting was set for nat even though everything is on lan