Thread: RE: Admin, Please read ASAP!! |
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Business model: you get the software, not the support.
It's the same model as with open source software if you think and it's not so bad.
If you value your time more that the learning time needed then personal support is a smart move.
You complain that you have to pay to post on the forum, but please think about the cost of the configuration of the forum itself, the bandwidth and housing costs.
As per your question, you have different options
call "sip:address@domain"
register a trunk with only outbound calls, create an d outbound route for that trunk and place calls
as for other questions, I suggest to have a look at the fundamentals of the SIP protocol and then paly around with the system log and SIP trace to see what's going on in your pbx.
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Thread: ICMP from pbxes to the router? |
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Hi,
I'm trying to troubleshoot a network issue that make my calls drop.
Looking at the logs from my router I can see that
. the router rstablish udp connection with www1
.www1 try to ping endlessly my router that is givin no response because of security policy
.the connection closes
.the call drops
Now, I would like to know why is pbxes pinging the other endpoint so often?
Could this be related to the connection getting frop?
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Thread: RE: legzzi login does not work |
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sorry but you are logged in, otherwise you couldn't post to the forum.
If you wanted to register an extension, remember to create one first.
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Thread: RE: Digital receptionist without options to press |
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Dont' call me Santa but you can already do what you want.
Upgrade to Pbxpro
Go to Source View
extension.conf
select aa_X (where X is your menu #)
change the line
exten => s,9,ResponseTimeout(7)
to
exten => s,9,ResponseTimeout(0)
save and apply.
Be careful, if you modify the menu in web page again, it returns to default config and you have to edit source again.
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Thread: RE: Outbound routing - dial pattern to identify based on lenght |
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It would be nice if the dial pattern in the outbound routes could identify a number based on lenght as well.
A small bug I found is that if you insert a prefix of 5 number, only calls to numbers 6 or more digits long respect the rule.
Calls with 5 digits are not touched, because the prefix is equal to the whole number and the system expect another figure at the end.
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Thread: RE: Digital receptionist without options to press |
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You can create a queue with a welcome messagge and your destination as an agent.
Once the welcome messagge is over the digital receptionist picks up.
If it's an extension then it starts ringing but you have to figure out a good moh that remind the phone ringing
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Thread: RE: Increase time limit for Skype |
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It is not going to happen because of limitations Skype imposes.
If you need to call skype account have a look at those providing commercial service (the good ones are not cheap).
there is also a free utility to translate from skype to SIP but you have to maintain a server connected
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Thread: RE: Default RTP and SIP Port Ranges |
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if it's about port forwarding, why do you care?
Setup you ata with the range desidered, then let the SIP protocol do the rest.
Pbxes and your ATA chosse the right path.
Configure your router to route udp packets to your device accordingly with his local ports.
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Thread: RE: skype connect |
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no problem here with skypeconnect.
.Setup trunk as usual
.setup incoming rule: TRUNK NAME MUST BE YOUR SKYPE CONNECT USERNAME
no problem so far.
Add contact to skype, restart skype client; you buddy appear online.
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Thread: RE: Interoperability |
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have you tried
username: enter phonenumber
password: enter password:authname
SIP server: enter server:port
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Thread: RE: Reason for Outage 17/07/2012 |
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Dear Pbxes,
it's been 50 minutes now that our pbx is down; we haven't received any notification about the automatic failover; in fact it didn't happen.
As soon as we regain control over the HTTP pages, we switch to another server with not luck.
We are "submit and restart" now, hoping that it would solve the issue.
Looking at the status page, we could see that SIP services are "critical" in every host.
Can you please give us an explanation on what happened and what step you are taking to approach the issue?
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Thread: RE: Sound Distorted - bad quality |
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Dear Pbxes,
IVR and music on hold sounds very bad, like metallic noises being playd along with music or speech.
Recordings are made from human voice and sampled at 8hz 16 bit mono and they play nice on my computer.
Tried the same recording with another pbx. No issue here
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Thread: RE: Extension with only Incoming Calls |
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Let's say that you have ext 1000 in the reception
Let's say that your other extension are from 2000 to 2999
Let's say that you have only 1 trunk and outbound rule configured
- Create a second outbound rule, sepcifying Extension 1000 in the correct field
- Put it at the top of the list by clicking on the up arrow at the end of its name, in the left menu
- Open the other outbound rule
- Edit the Extension Field, adding 2XXX
- Save and apply
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diciamo che l'interno della reception è il 1000
diciamo che gli altri interni che hai vanno da 2000 a 2999
diciamo che hai solo 1 trunke una regola in uscita configurata
- crea una seconda regola in uscita, specificando il campo "Interno" =1000
- poni la nuova regola in cima alla lista, agendo sulle frecce che trovi a lato del nome, nel menu a sinistra
- Apri l'altra regola in uscita
- Modifica il campo "Interni" inserendo 2XXX
- Salva e applica le modifiche
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Thread: RE: not receiving voicemail emails |
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to view voicemail
use these credential
user XXX-YY
password: ZZZZ
XXX : account name
YY: extension number
ZZZZ: voicemail password.
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Thread: RE: My pbxes Account IP address |
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You wouldn't want to use IP authorization.
Every Pbxes user shares the same ip. it means that if you can make calls through your carier, everybody else can do the same.
The only difference is that you will be the one paying our bills.
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Thread: Queue Member penalty |
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It would be nice to have the possibility to specify the penalty for each memeber.
This way one can create a single queue instead of having more than one to reproduce the same feature
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