That is only saying where you want to submit the form data back to. Has nothing to do with freepbx. If left empty according to html RFCs it just goes back to the same page
It’s calling javascript probably for validation but only for ringgroups (checkGRP is just a javascript function)
It does, but again it’s javascript and you are looking at PHP https://github.com/FreePBX/ringgroups/blob/release/13.0/assets/js/ringgroups.js#L53
I dont know what you are asking
It already is permanent.
ok thx , how to configure on free pbx ?
More like an unanticipated edge case. Open an issue with your findings at:
http://issues.freepbx.org/
I just can’t give quantitative numbers on how many extensions would work on a single machine. 1900 seems excessively high. 1000 can be ok depending on hardware but I can’t tell you what the hardware would or should be because then it will be misquoted as “you said” in a couple of months.
As for asking for paid support. No not saying that at all. Trying to work through this thread without having people jump the gun and report it as a bug of “high system load”. In a previous thread on this I documented how you can actually pinpoint functions that are slow and then report back to us completely free. I don’t want to keep having to repost this in thread after thread (and I shouldn’t have to). You can look it up
Thanks for all your help Andrew, I’ve come to live with it
Thanks for your reply, but ftp server is running since I can have ftp access from another pbx to the same server at the same time. and from web too
If you truly have an issue with a machine you bought from Sangoma (Sangoma hardware) then you have a warranty under it and get support for free*… you should utilize it.
As for this thread it was brought to my attention that you have a lot of users on a VM with 2 CPUs. You’re going to run into bottlenecks with two CPUs.
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.
i have only this -
[macro-dialout-trunk]
include => macro-dialout-trunk-custom
exten => s,1,Set(DIAL_TRUNK=${ARG1})
exten => s,n,GosubIf($[$["${ARG3}" != “”] & $["${DB(AMPUSER/${AMPUSER}/pinless)}" != “NOPASSWD”]]?sub-pincheck,s,1)
exten => s,n,GotoIf($[“x${OUTDISABLE_${DIAL_TRUNK}}” = “xon”]?disabletrunk,1)
exten => s,n,Set(DIAL_NUMBER=${ARG2})
exten => s,n,Set(DIAL_TRUNK_OPTIONS=${DIAL_OPTIONS})
exten => s,n,Set(OUTBOUND_GROUP=OUT_${DIAL_TRUNK})
exten => s,n,GotoIf($["${OUTMAXCHANS_${DIAL_TRUNK}}foo" = “foo”]?nomax)
exten => s,n,GotoIf($[ ${GROUP_COUNT(OUT_${DIAL_TRUNK})} >= ${OUTMAXCHANS_${DIAL_TRUNK}} ]?chanfull)
exten => s,n(nomax),GotoIf($["${INTRACOMPANYROUTE}" = “YES”]?skipoutcid)
exten => s,n,Set(DIAL_TRUNK_OPTIONS=${TRUNK_OPTIONS})
exten => s,n,Macro(outbound-callerid,${DIAL_TRUNK})
exten => s,n(skipoutcid),GosubIf($["${PREFIX_TRUNK_${DIAL_TRUNK}}" != “”]?sub-flp-${DIAL_TRUNK},s,1)
exten => s,n,Set(OUTNUM=${OUTPREFIX_${DIAL_TRUNK}}${DIAL_NUMBER})
exten => s,n,Set(custom=${CUT(OUT_${DIAL_TRUNK},:,1)})
exten => s,n,ExecIf($[$["${MOHCLASS}" != “default”] & $["${MOHCLASS}" != “”]]?Set(DIAL_TRUNK_OPTIONS=M(setmusic^${MOHCLASS})${DIAL_TRUNK_OPTIONS}))
exten => s,n(gocall),Macro(dialout-trunk-predial-hook,)
exten => s,n,GotoIf($["${PREDIAL_HOOK_RET}" = “BYPASS”]?bypass,1)
exten => s,n,GotoIf($["${custom}" = “AMP”]?customtrunk)
exten => s,n,Dial(${OUT_${DIAL_TRUNK}}/${OUTNUM},300,${DIAL_TRUNK_OPTIONS})
exten => s,n,Noop(Dial failed for some reason with DIALSTATUS = ${DIALSTATUS} and HANGUPCAUSE = ${HANGUPCAUSE})
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s,n(customtrunk),Set(pre_num=${CUT(OUT_${DIAL_TRUNK},$,1)})
exten => s,n,Set(the_num=${CUT(OUT_${DIAL_TRUNK},$,2)})
exten => s,n,Set(post_num=${CUT(OUT_${DIAL_TRUNK},$,3)})
exten => s,n,GotoIf($["${the_num}" = “OUTNUM”]?outnum:skipoutnum)
exten => s,n(outnum),Set(the_num=${OUTNUM})
exten => s,n(skipoutnum),Dial(${pre_num:4}${the_num}${post_num},300,${DIAL_TRUNK_OPTIONS})
exten => s,n,Noop(Dial failed for some reason with DIALSTATUS = ${DIALSTATUS} and HANGUPCAUSE = ${HANGUPCAUSE})
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s,n(chanfull),Noop(max channels used up)
exten => s-BUSY,1,Noop(Dial failed due to trunk reporting BUSY - giving up)
exten => s-BUSY,n,Playtones(busy)
exten => s-BUSY,n,Busy(20)
exten => s-ANSWER,1,Noop(Call successfully answered - Hanging up now)
exten => s-ANSWER,n,Macro(hangupcall,)
exten => s-NOANSWER,1,Noop(Dial failed due to trunk reporting NOANSWER - giving up)
exten => s-NOANSWER,n,Progress
exten => s-NOANSWER,n,Playback(number-not-answering,noanswer)
I have got the below error and my Voice mails are getting delayed for 1 day not sure weather all are well delivered.
retrieve_conf failed to sym link:
/var/www/html/recordings/modules/conferencespro.module from conferencespro/ari/modules (Already exists, not a link)
This can result in FATAL failures to your PBX. If the target file exists and not identical,
the symlink will not occur and you should rename the target file to allow the automatic sym link to occur and remove this error, unless this is an intentional customization.
Is anything i need to do to resolve the above error. Is this error causing the delay in Voicemails.
Cisco IP phone comunicator trying to register to freepbx I get this error
Apr 2 15:26:09 pbx in.tftpd[44133]: sending NAK (4, Missing mode) to x.x.x.x
What would be the issue?
do you see anything in the FTP log why it does’t like it?
If you have experience with FreePBX (able to setup a system from start to finish, integrate SIP trunk, configure phones, troubleshoot problems), and are looking for a few hours of work per week, please send me a DM with your hourly rate and resume!
I am managing a handful of FreePBX systems for my customers and running out of time to work on some projects that are coming in, could use an extra hand
About us:
I’m trying to add the following stanza’s to EPM for a Cisco config:
<addOnModules>
<addOnModule uuid="{c42eaca3-d387-58e7-4f3f-46dfea8b7593}" idx="1">
<deviceType>CKEM</deviceType>
<deviceLine>36</deviceLine>
<loadInformation></loadInformation>
<phoneTemplateId></phoneTemplateId>
</addOnModule>
<addOnModule uuid="{c42eaca3-d387-58e7-4f3f-46dfea8b7593}" idx="2">
<deviceType>CKEM</deviceType>
<deviceLine>36</deviceLine>
<loadInformation></loadInformation>
<phoneTemplateId></phoneTemplateId>
</addOnModule>
</addOnModules>
I can’t seem to add anything after the “addOnModule uuid” Any clues for me?
Hello, I have a freepbx 13 on asterisk 13.
What we are trying to accomplish is to have any inbound calls that exceed the call limit to rollover and play a busy tone. Currently they go back to our sip provider as a failed call and they forward calls to a roll over number (typically our cell phone).
Ok I will read these thanks.
Ok, I am on site now. The EXP50 uses the EXP 40 Template just fine
Just a guess here: the AT&T<->Asterisk link is using g729 (which normally uses 30 ms packetization) and Asterisk is transcoding to ulaw on the Riedel side. There are several ways to see what is actually happening.
You can issue
sip set debug on
at the Asterisk command line, which will cause all SIP packets to appear in the Asterisk log (along with what would normally be there).
Or, you can run tcpdump in a root shell to capture all packets (including SIP and RTP) to a file, which you can then move to your workstation e.g. with sftp and view / analyze with Wireshark.
Here is the SDP for a typical INVITE from my trunking provider (not AT&T):
v=0
o=Sonus_UAC 483212 586479 IN IP4 67.231.5.112
s=SIP Media Capabilities
c=IN IP4 67.231.5.79
t=0 0
m=audio 45708 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
The ‘a’ attributes indicate they can handle PCMU (a.k.a. G.711U or ulaw) and G729.
ptime indicates that the provider wants 20 ms of audio in each RTP packet.
You may also see maxptime, which means they will accept up to the amount specified.
When Asterisk responds, the SDP in the 183 or 200 response indicates the codec selected and the amount of audio it wants or will accept per RTP packet.
A similar negotiation occurs for the Asterisk<->Riedel part of the call.
The actual RTP will show the payload type being used and its length. ulaw requires eight payload bytes per millisecond of audio; g729 only one.
Please report what happens in the failing case (with Asterisk 13) as well as on your working system.
Is there a folder/file where user data, such as login names are kept?