Hello,
I have installed PBX Manager 6.0.1.57 from ISO image, created new tenant and incoming trunk, registered extension and routed incoming DID to it.
When I do call the number I am getting in the asterisk console:
-- Executing [18478792867@from-outside:1] Set("SIP/xx.xx.xx.xx-08887898", "__INCOMINGCLI=9178179202") in new stack
-- Executing [18478792867@from-outside:2] Goto("SIP/xx.xx.xx.xx-08887898", "from-outside-redir|18478792867|1") in new stack
-- Goto (from-outside-redir,18478792867,1)
-- Executing [18478792867@from-outside-redir:1] Set("SIP/xx.xx.xx.xx-08887898", "DIALED_PUBLIC_NUMBER=18478792867") in new stack
-- Executing [18478792867@from-outside-redir:2] Set("SIP/xx.xx.xx.xx-08887898", "DIALED_NUMBER=18478792867") in new stack
-- Executing [18478792867@from-outside-redir:3] Set("SIP/xx.xx.xx.xx-08887898", "status=1") in new stack
-- Executing [18478792867@from-outside-redir:4] GotoIf("SIP/xx.xx.xx.xx-08887898", "1?7") in new stack
-- Goto (from-outside-redir,18478792867,7)
[Jun 30 06:26:45] WARNING[23459]: pbx.c:2525 __ast_pbx_run: Timeout, but no rule 't' in context 'from-outside-redir'
When I looked at inbound.include
exten => 18478792867,1,Set(DIALED_PUBLIC_NUMBER=${EXTEN})
exten => 18478792867,2,Set(DIALED_NUMBER=${EXTEN})
exten => 18478792867,3,Set(status=${DB(TL/TENANT/NextRow/status)})
exten => 18478792867,4,GotoIf($["${status}" != "0"]?7)
exten => 18478792867,5,Playback(ss-noservice)
exten => 18478792867,6,Congestion
There are no 7th priority. and [from-outside-redir] ends there.
I used to configure it before on test server and it was working. I can reinstall it and try again but since I want to use it in production I would like to fix it. Does anybody have an idea how to troubleshoot/fix this issue ?
Thank you!
people, it should never say
people, it should never say SIP/some.ip.address, In my 5yrs of using asterisk I have never seen so many people doing this in my life. I am suspecting something systemic at this point but can't put my finger on how or why. Perhaps a survey will help get to the bottom of this.
If you name a trunk SCHMUCKATELLI, and set it up properly, then the calls would arrive with the channel name
SIP/SCHMUCKATELLI-channelnumber
not
SIP/xxx.xxx.xxx.xxx-channelnumber
I see IP addresses as a red flag. Calls that arrive unmatched to a trunk ignore all your settings for that trunk (codec preferences, t38pt_udptl, dtmfmode, qualify etc).
questions to anyone who's inbound calls exhibit this behavior:
1) WHO are you using for a provider?
2) WHO's advice are you following for setting up a trunk?
3) WHY Are you allowing untrusted sources to deliver calls to your PBX? (allowing guests)
4) WHAT did you name your trunk?
5) WHAT did you set the type=
6) Is your trunk setup as insecure (ie trusting the ip not proxy authentication)
7) WHAT did you set for username?
to the original poster: 6.0.1.57 is over a year old, you will need to run a more recent version. Download the webmin module and install it via webmin under the modules section. This will upgrade your version. You also need to assign a script to your inbound route (like an ivr or something).
Username and password for the trial version
Hi,
What's the username and password for the trial version please !
Thanks,
Josh
BANNED start a new thread, we
BANNED
start a new thread, we don't resurrect 2yr old posts just to ask a complete new, unrelated question. I for one, will be ignoring them for the bad forum etiquette.
I have same problem have you find any solution?