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Looks like not only my sip phones have issues with pbxes...
E164 hasn't got a route as well??
Wasn't able to test your route, your system returned the following information: 0|Error 404 - An attempt to verify your SIP address was rejected because the user doesn't exist or the domain is invalid
I tried both <accountname>@pbxes.org
and <accountname>-<extensionnr>@pbxes.org
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Thread: |
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16.06.2007 20:58 |
Forum: Bugs |
Zitat: |
Originally posted by Diafora
What string do you have in the Dial Patterns field of the Outbound route? |
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outbound route:
XXXXXXXXX
XXXXXXXXXXXX
and in the trunk I strip the leading 0 and add 0032
but this doesn't seem to work anymore....
and since today none of my sip phones can connect to pbxes.org anymore as well... they are using lifeline (costly)
status shows everything is down?
what's going on???
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Thread: |
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Today I suddenly notices when i do
01658xxxx that voipbuster is used (while voipcheap is my default)... I get a voice that says i have to dial 0032....
I checked... voipcheap should have been taken
and both voipcheap & voipbuster should have added 0032?
when i dial 003216.. it chooses voipcheap correctly
what's going on?
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Thread: |
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is it possible to support flite (text to speech)
and have it dictate text from a dynamic webpage?
something like
-> digital receptionist
"please insert your postcode"
-> run "http://myserver/index.php?id=".postcode
play result (text to speech)
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Thread: RE: FAX switch? |
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I've enabled fax switch on my incoming route for DID trunk 3starsnet...
destination = Analog_fax...
but they all come through my "fax" at home (behind a sipura 3000)
I've got a 5second delay configured in pbxes...and that it should email the fax
I'm faxing with a Canon FaxPrinter...
Date Time Caller ID Number Destination Trunk Context Duration Recording
2007-02-12 17:44:08 "050230300" <050230300> analog_fax 85.119.188.3 ext-fax 23 sec
2007-02-12 17:38:25 skynetbbs-302777 739 ext-local 0 sec
2007-02-12 17:38:24 skynetbbs-302777 0486739739 from-internal-cont 0 sec
2007-02-12 17:31:32 "050230300" <050230300> 1 sip.3starsnet.com ext-group 20 sec
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Thread: RE: forwarding from pbxes to pbxes |
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A few collegues have their own pbx on pbxes...
and they wanted to be able to call me...
Problem 1) Nokia E70 refuses to connect to 2 different accounts on thesame server.
so I created a sip on their account (artacen-725@pbxes.org)
which has to forward to sip:\\skynetbbs-739@pbxes.org
but that doesn't seem to work?
skynetbbs-739 does not receive any calls
Is that possible?
did I forget something ?
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Thread: |
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Is there a method where I can get an "inbound"call from trunk A going to an "outbound" call on trunk B?
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Thread: RE: sipgate.de |
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Zitat: |
Originally posted by i-p-tel
Avoid "sipgate.de" as trunk name for your sipgate trunks. |
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I've learned to use the "sipserver" name as trunkname so that the "status" resolves correctly ?
Should I give it a random name to correct the issue?
Could that be my problem with dialnet.pl as well?
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Thread: RE: sipgate.de |
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I recently acquired a german DID via sipgate.de
it works with the x-lite ;
i've copied servername, username, password for the sip config in pbxes;
I submitted my personal data
I never get "connected" using pbxes (status shows the trunk always "grey")
Is there a blacklist for pbxes?
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Thread: RE: Callback from the web and connect to extension |
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Zitat: |
i-p-tel, how about that? or anyone else has a similar, already working set? If so, how?
Yves |
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on www.voipbuster.com, www.voipbuster.com you can create an account which enables a webcallback feature... you define yourself how much it might cost... eg 0,01 euro to just enable fixed lines and not paylines or mobiles...
you can put the voipbuster sip info in the trunk of pbxes (incoming call!) and define that the calls go to one of your sip phones and/or voicemail...
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Thread: RE: export calling logs? |
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It doesn't seem anyone has asked for it yet?
But is it possible to have an Export button for the call logs, so we can import it in our Excell (CSV is fine!) and do some math on it?
Mvg,
Steven
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Thread: |
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>Now the part which I don't understand. Where and how did you use "sipsak" and "sip:\\"?
I just tried that in a linux box with sipsak compiled...
I'd figure that with sip & presence server & 'email'-like adressing you can do a sort of nslookup on someones sipadress and find out where he's registered... and to perform a direct sip-call from A to B...
But Pbxes is some kind of Back to Back sip user agent and will always give his ip adress and not that of B... so you always have to go by Pbxes... only the RTP stream could be bypassed if configured on pbxes...and perhaps then I could locate someones ipadres if he answers the phone...(or his fax does :-) instead of the voicemail system of pbxes...
Something like they had in icq/msn ... in current stadium you have to send a 2mbyte file to get their direct ipadress...all others are forwarded via Microsoft servers
The implementation, the why/how?
I can't see my ipadress on my gprs/wifi mobile for instance; and some/alot of people have an extension at home...dynamic ip's... and to find their traffic on eg firewall logs?
Dunno if you can "fork" a connection on pbxes? I believe sip was able to have 1 sipaccount registered by multiple devices which could ring/register/reinvate as one... but in practice... the one that registered last shall be used?
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Thread: |
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That's correct...
I tried sjphone, eyebeam, xlite 3, xlite 2
their client is an xlite 2 version;
So i've tried to replicate their settings in xlite 2...
there was only 1 variable I couldn't replicate which was
Use X-Nat to choose SIP/RTP Ports: Default
in xlite you can not change this variable and this is "NEVER"...
According to their support pages it should work on any type of ATA adaptor.. but they didn't want to give any asterisk assurance...
Any hunch?
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Thread: RE: trunk name in calling log? |
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I notice that in some cases not the trunk name...but my login name is used?
Is this normal behaviour or a bug?
Impact
Hence I have to check all my trunks 1 by 1 to see which one was used ...
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Thread: |
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In the status page you only see "green" rectangles mentioning registered ip devices...
is it possible to know their ipadresses?
I tried using their "sip:\\"... but I always ended up with the ip of the pbxes itself and not the device that registered itself as such
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Thread: |
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I've received a polish did on www.newfon.pl
I've successfully called myself using the sipsettings & voipcheap to perform a free call between my fixed line & the free did... quality was ok
I've copied those settings in my pbxes trunk...
the trunk goes green in status...
but my call doesn't seem to end on pbxes (RINGING....) and nothing in the Call Log of pbxes?
outbound proxy : dialnet.pl
user : the DID number
password: ...
nothing special ?
when i configure Xten lite behind my natted router... the call comes through.... is pbxes in some kind of blacklist?
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Thread: |
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is there somehow a method to know what the last known ip was of a registered extension?
I tried with "sipsak accountname-extension@pbxes.org" but I only get timeouts
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Thread: |
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I've found his problem...
he had no outbound routing that worked...
every outbound route had a specific callerid extension limitation...
problem solved :-)
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Thread: |
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I've recently found a cheap DID provider (sip.3starsnet.com) for belgian DID's...
and it works... I've created a callthru with a pin.
and it gets correctly translated...
eg
caller:my callerid -> called: destination @ trunk: voipcheap
caller:my callerid -> called:1 @ trunk: sip.3starsnet.com
a collegue of mine (bennydm) configured the same thing...
he has another didnumber offcoarse... but somehow his sip.3stasnet.com is not correctly seen and is shown as 88.198.18.239 ...he is asked also a password... but it won't give him a second dialtone ?
Could you look into it?
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