Hey Remco,
exten => s,1,Wait(1)
exten => s,2,Answer()
exten => s,3,Dial(SIP/phone)
A bit more simplified then what i have in my config, but exactly the same
order.
Regards,
Evert
On Tue, 19 Jan 2010 10:46:52 +0100 (CET), Remco Barendse
wrote:
> On Mon, 18 Jan 2010, ev...@disrup
Hey Tzafrir,
In my first mail i already said that i receive the clid strings, only
sometimes numbers were scrambled aka missing numbers from the complete
number.
Regards,
Evert
>> Try'd the lines you mentioned below also, with exactly the same result
>> still.
>
> Best
Hey Doug,
According to the commands its:
Answer(ms)
Wait(s)
So it seems not the same.
Regards,
Evert
> Ira wrote:
>> At 09:13 AM 1/18/2010, you wrote:
>>
>> Add a WAIT(1) as the first line of the incoming context and see if
>> that helps.
>>
>
> An
Hey Ira,
It seems after a several testing, that the wait(1) seems to solve the issue.
Only now weirdly enough the phone keeps ringing if the caller hangs up
before i picked up the phone (pstn call)
Regards,
Evert
>
> Add a WAIT(1) as the first line of the incoming context and see i
Hey Gordon,
Those settings are set to =nl in my config for dahdi.
Im assuming that would be correct :)
Regards,
Evert
>loadzone=uk
>defaultzone=uk
>
> Work out what's right for .nl and it'll be a
Ok after the fxotune, it still does it.
i hear no weird hum or wicked echo's
Regards,
Evert
> You may have a gain issue. Since the Caller ID information on an
> 'analog' line is FSK it is sensitive to distortion. How are the quality
> of your lines, do you have a
I haven't try'd fxotune yet, going to lookup how i need to run that properly.
Thanks for the advice!
Regards,
Evert
> You may have a gain issue. Since the Caller ID information on an
> 'analog' line is FSK it is sensitive to distortion. How are the quality
> of you
Hey Danny,
Yes there is an analog phone parallel to the connection going into the
digium card.
And it shows the numbers correctly.
Try'd the lines you mentioned below also, with exactly the same result still.
Regards,
Evert
> In theory, Answer(5) should be the same as this
> - e
Ok try'd 5, but still its missing numbers sometimes.
Regards,
Evert
> Try Answer(5). Don't know how long the Dutch system takes to connect a
> call, but it should not take over 5 seconds. If that doesn't solve it,
> timing is not your issue.
>
> -Original
Ok just try'ed the Answer(1) but nope, still its only a few numbers, still
sometimes it goes ok, and sometimes its just a few numbers.
Regards,
Evert
> Stupid that i didnt think of that :)
>
> i'll try that:)
>
> Regards,
>
> Evert
>
>> ev...@disruptor
Stupid that i didnt think of that :)
i'll try that:)
Regards,
Evert
> ev...@disruptor.nl wrote:
>> Except:
>> Sometimes the callerid from the caller is not the complete number, but
>>
>
> Just a
Maybe someone has an idea, i really dont atm.
Its btw a dutch phoneline.
Regards,
Evert
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y it with 2 or 3 w's instead of 1...
>>
>
> Regards,
>
> Evert
Regards,
Evert
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before dialing, ww1234 a 1 second delay, etc.
>
> Try it with 2 or 3 w's instead of 1...
>
Regards,
Evert
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To U
s4all-out,60,tTwWkK)
But then the other peer says:
-- Called *31#w06123456...@xs4all-out
-- SIP/xs4all-out-0234 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'SIP/evert-0233' status is 'CONGESTION'
Anyon
No one...?
This problem is really bugging me... :-/
Regards,
Evert
Evert wrote:
> Hi all!
>
> Working on the following brain-scratcher. I am setting up a Trixbox
> system for someone who uses 'provider A'. Everything works fine, except
> for the IVR: keyp
em.
And then the IVR works!
Is there any possibility that the config on the provider-side is causing
this difference? If yes, what could it be, and is there a way for me to
fix this?
Regards,
Evert
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. However, when the called phone anwers the
call, the calling phone does not notice this, and keeps on ringing...
No connection is established.
I am not sure whether this is caused by Asterisk or SER, so I am sending
this to the SER-group as well... :-)
Regards,
Evert
William Piper wrote:
> D
Does it really? Most programs stay clear from reserved ports (<1023),
unless the ports are reserved for the particular program of course...
Regards,
Evert
Matt wrote:
> Ok,
> Then why does Asterisk use 1,000-2,000 by default? I see Vonage uses
> 10,000 - 20,000 also.
>
&g
I also tried with Forward instead of Dial, but that seems to make
matters only worse... :-?
Greetings,
Evert
Evert wrote:
> SERADDRESS is a variable that points to a SER-server (currently running
> on the same IP as Asterisk, but on a different port).
>
> I noticed that
998,3,Set(TIMEOUT(digit)=15)
exten => 998,4,WaitExten(10)
exten => _,1,Dial(SIP/[EMAIL PROTECTED],60,o)
This does send the entered extension to SER, and the phone rings.
However, when the called phone anwers the call, the calling phone does
not notice this, and keeps on ringing...
Regard
Hi! :)
Thanks for the tip. I'm almost there now, the only problem that I have
left is that I do NOT want Asterisk to check whether the extension
entered is valid. In the current setup Asterisk will refuse to forward
the call since it thinks the extension is invalid... :-/
Regards,
ot;, "TIMEOUT(digit)=5") in new stack
-- Digit timeout set to 5
Nov 6 07:54:17 WARNING[5503]: pbx.c:1700 pbx_extension_helper: No
application 'ResponseTime' for extension (context, 998, 4)
== Spawn extension (context, 69697602, 4) exited non-zero on
'SIP/998-b6600ea0&#
exten => 998,n,Dial(SIP/[EMAIL PROTECTED],60,tr)
WaitExten obviously does not fill EXTEN with its value...
Anyone any suggestions?
Regards,
Evert
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g wrong here? I don't even get the background music while
WaitExten is active. I doubt that it is active anyway, since I get
disconnected before the 20 seconds have passed... :-/
Greetings,
Evert
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hannel 'SIP/asterisk.domain.com-081477a0' status
is 'UNKNOWN'
(the name of the box is here asterisk.domain.com )
What am I doing wrong?
Regards,
Evert
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Maxim Vexler wrote:
> On 7/6/06, Evert Meulie <[EMAIL PROTECTED]> wrote:
>> Hi all!
>>
>> How do I make Asterisk recognize fax calls and disconnect them?
>>
>> Regards,
>> Evert
>>
>> _
Hi all!
How do I make Asterisk recognize fax calls and disconnect them?
Regards,
Evert
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*bump*
Anyone? I still can't find little/no info on DNID... :-/
Regards,
Evert
Evert Meulie wrote:
Hi all!
I'm in the process of configuring an Asterisk server here that, based on
which number was called, should send calls to different extensions:
913 - 1 -> ext.
/tiki-index.php?page=DNID ) is not of much help either.
Anyone here with any suggestions?
Regards,
Evert
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/tiki-index.php?page=DNID ) is not of much help either.
Anyone here with any suggestions?
Regards,
Evert
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Have you checked
http://www.voip-info.org/wiki/view/Asterisk+High+Availability+Solutions
Regards,
Evert
[EMAIL PROTECTED] wrote:
Hi all,
I would like to know if there is a solution to this question.
Scenario:
Two asterisk servers connected across the Intenet ( in SIP or IAX mode, no
Just wondering...
Has someone ever contacted Skype/Ebay and asked them about their point of
view/opinion on interfacing with SIP / Asterisk? 8-)
Regards,
Evert
[EMAIL PROTECTED] wrote:
I sincerely believe that it's completely non-sense to make a channel for
Skype.
Skype
As soon as they port it to Gentoo I'll try it out... ;-)
Evert
Kerry Garrison wrote:
Everyone should simply uninstall Skype and switch to the Gizmo project
because it interfaces quite nicely with Asterisk.
Kerry Garrison
Publisher - http://GeekGazette.com - http://VOIPSpeak.net
(949
.. does it exist already?
Regards,
Evert
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I found the price. $450 :-/
Kevin P. Fleming wrote:
Evert Meulie wrote:
That unit looks VERY promising! Thanks! :-)
Would anyone happen to know an approx. price for a unit like this?
Anyone? I bet the manufacturer of the unit would know a price for it,
and it's probably
That unit looks VERY promising! Thanks! :-)
Would anyone happen to know an approx. price for a unit like this?
Regards,
Evert
BJ Weschke wrote:
On 12/16/05, Evert Meulie <[EMAIL PROTECTED]> wrote:
Hi all!
I am looking for a device that I can stick in a USB-port on my Asterisk
the phone', if you know what I mean... ;-)
Regards,
Evert
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Found it! For some reason [EMAIL PROTECTED] had chosen to build itself without
app_meetme.so!
After building this module by hand, all worked! :-)
Evert
Evert Meulie wrote:
Read before you reply... ;-)
To be 100% clear on zaptel/ztdummy, here's the output of my lsmod:
[EMAIL PROT
0
mii 8641 1 8139too
ext3 118729 2
jbd59481 1 ext3
ata_piix 13253 3
libata 47901 1 ata_piix
sd_mod 20545 4
scsi_mod 116429 2 libata,sd_mod
Kunal Parikh wrote:
Hi Evert,
Do you have the z
is when I dial 8125 from extension 125. 8125 is defined in the
meetme(-additional).conf
And before you ask: yes, ztdummy is loaded...
Who has any suggestions? I'm stumped... :-/
Regards,
Evert
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Turns out my VoIP provider made a booh-booh... ;-)
Evert Meulie wrote:
> Hi all!
>
> Quite a mystery. The following happened when I was on holiday, and no one
> else has changed any configs of either Asterisk or the Cisco's in the
> building...
>
> The situati
disallow=all
allow=ulaw
allow=alaw
and no allow/disallows at the phones themselves.
This used to work just fine... What could have happened...?
Regards,
Evert
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Hi all!
Who can tell me what the correct/preferred/only DTMFmode setting is for
Windows Messenger SIP clients?
Regards,
Evert
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Thanks for the info! That's exactly the pointed I needed! ;-)
(but I'll implement it myself. Cheaper...) ;-) ;-)
Greetings,
Evert
Alistair Cunningham wrote:
Evert,
The best way to do this is have your PHP code put a control file in the
outgoing directory of Asterisk. This then
s of
course also listed.
How do I commence here? I'm pretty sure others have done this already,
so I was wondering whether there's someone who can point me in the right
direction... :-)
(Preferable in PHP, since that's the flavor of choice of our porta
Hi everyone!
I wonder whether the following would be possible:
Can Asterisk show the country from which a call originates on the
display, along with the phone number?
Regards,
Evert Meulie
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>ping 0.5.0.4
connect: Invalid argument
Nope! ;-)
Andreas Sikkema wrote:
[EMAIL PROTECTED] wrote:
WARNING[245775]: chan_sip.c:597 __sip_xmit: sip_xmit of 0x3ddfa3a8
(len 508) to 0.5.0.4 returned -1: Invalid argument
Who can tell me what causes these, and how to fix it...?
Is that a valid IP
(len 508) to 0.5.0.4 returned -1: Invalid argument
Who can tell me what causes these, and how to fix it...?
Regards,
Evert Meulie
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Is there a page/site where the progress/info on this project is to be
found? :-)
Regards,
Evert Meulie
Jon Radon wrote:
Right now, you'd need an FXS port and a modem for HylaFax to use.
It's not an ideal setup, but more reliable than using an ATA such as
the Sipura. Steve
Hi everyone!
I have an Asterisk server here that also has Hylafax installed on it.
What else do I need to have that server send/receive faxes?
Regards,
Evert Meulie
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Hi!
When I call a colleague of mine from my Cisco (via Asterisk), they get
on their display:
From Evert
asterisk
How do I remove/change the 'asterisk' part?
Regards,
Evert
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,7,ResponseTimeout,30
***
Who can help me a little further on the way?
Regards,
Evert
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To
Maurizio Marini wrote:
On Friday 17 September 2004 11:43, Evert Meulie wrote:
How do I implement this in extensions.conf...?
maybe this may help...
http://lists.digium.com/pipermail/asterisk-users/2004-February/036737.html
Thanks! That works like a charm! The only thing I'd like to do now is
Hi everyone!
I'd like to create the following: a user picks up the phone (gets a dial
tone), dials '0' for an 'outside' line, gets a second (different?)
dialtone, and is able to enter an external phone number.
How do I implement this in extensions.
Found it. It's a Micronet-specific error message. So much for
standards... :-/
Evert Meulie wrote:
From a 'sip debug':
Sip read:
SIP/2.0 100 Trying
From: "Evert";tag=as6e18534e
To:
Call-ID: [EMAIL PROTECTED] ext. IP]
CSeq: 102 INVITE
Via:SIP/2.0/UDP [my ext. IP]:50
Dave Cotton wrote:
On Thu, 2004-09-16 at 12:00 +0200, Evert Meulie wrote:
Dave Cotton wrote:
On Thu, 2004-09-16 at 09:35 +, Murali wrote:
Hi friends,
Redhat9.0 installed in my machine.Asterisk1.0 installed in that machine.
Thereis no mpg123 player. So, I download the mpg123
Dave Cotton wrote:
On Thu, 2004-09-16 at 09:35 +, Murali wrote:
Hi friends,
Redhat9.0 installed in my machine.Asterisk1.0 installed in that machine.
Thereis no mpg123 player. So, I download the mpg123 player and installed it.
My sound card is configured correctly.
When I tried to check
Hi everyone!
I was wondering... Does the musiconhold quality improve if the mpg123
processes run with a negative priority? If so, is there a way to make
them start like that, so I don't have to renice them?
Regards,
Evert
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Hi everyone!
Is it safe to use this (old!) webmin module with asterisk 1.0rc2?
Regards,
Evert
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16 08:27:47 WARNING[360465]: rtp.c:1382 ast_rtp_bridge: codec0 = 268
is not codec1 = 0, cannot native bridge.
== Spawn extension (sip, , 1) exited non-zero on 'SIP/105-1559'
(123.123.123.123 is the IP of our VoIP-provider, is my cell
phone, and 105 is
Thanks, that did the trick! :-)
Kinda weird though that the mp3's that actually come with Asterisk don't
work correctly 'out of the box'. Or is this a mpg123 bug?
Regards,
Evert Meulie
Andreas Roedl wrote:
Hello!
Am Dienstag, 14. September 2004 19:21 schrieb Evert Meul
When I do a 'asterisk -vc' I get following, but asterisk does NOT stay up:
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
Tried that. Now I get:
Sip read:
SIP/2.0 403 Forbidden (From header is not a Trust host or gateway)
From: "Evert"@>;tag=as0687982f
To: >;tag=87f2a0d5-13c4-4146e66c-1a8baa18-5e5
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Via:SIP/2.0/UDP 217.13.2.82:5060;branch=z9hG4bK3dc10bb5
Cont
From a 'sip debug':
Sip read:
SIP/2.0 100 Trying
From: "Evert";tag=as6e18534e
To:
Call-ID: [EMAIL PROTECTED] ext. IP]
CSeq: 102 INVITE
Via:SIP/2.0/UDP [my ext. IP]:5060;branch=z9hG4bK4fd1045b
Content-Length:0
7 headers, 0 lines
Sip read:
SIP/2.0 403 Forbidden (From header is
0 Trying
From: "Evert";tag=as0aca53fa
To:
Call-ID: [EMAIL PROTECTED] IP]
CSeq: 102 INVITE
Via:SIP/2.0/UDP [my IP]:5060;branch=z9hG4bK00215868
Content-Length:0
7 headers, 0 lines
Sip read:
SIP/2.0 403 Forbidden (From header is not a Trust host or gateway)
From: "Evert";tag=as0aca53f
Hi everyone!
I now have obtained a couple of SIP-accounts at a local VOIP-provider.
How do I specify that ALL outgoing calls to _NXXX go out via one of
these accounts?
Regards,
Evert
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[EMAIL PROTECTED]
http
ection between both systems.
Should bindaddr (iax.conf) or externip (sip.conf) be defined for a setup
like this one?
Regards,
Evert
Do you know where it got the 10.138.3.2 IP from? Is it configured
anywhere on the server? Do you have
externip defined in that config file?
Evert Meulie wrote:
Hi ev
192.168.2.44??
All other traffic going over these lines has no problems with this. The
192.168.2.x & 192.168.11.x networks are fully 'connected' to each other...
Who knows the answer...?
Regards,
Evert Meulie
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Hi everyone!
Is there a way to let Asterisk connect to/interface with the Micronet
SP5210 SIP server ( http://www.micronet.info/Products/voip/SP5210.asp )?
It does not support IAX, but maybe there is another way...?
Greetings,
Evert Meulie
Hi!
It turns out my provider uses the Micronet SIP server. Any possibilies
to let this one interface with Asterisk?
Regards,
Evert
Evert Meulie wrote:
Hi everyone!
I have a problem... We have received a couple of phone numbers for
voip from a local voip-provider. The work fine directly with a
y options then for
routing incoming, outgoing or both via this voip-provider?
Greetings,
Evert
Benjamin on Asterisk Mailing Lists wrote:
On Wed, 08 Sep 2004 10:08:00 +0200, Evert Meulie <[EMAIL PROTECTED]> wrote:
I have a problem... We have received a couple of phone numbers for voip
fr
be all I need to let incoming calls on ring on
extension 106, right?
Regards,
Evert
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Hi!
If I install a CAPI-compatible ISDN-card in my server, will that:
a) enable me to connect that server to the public phone net
b) allow me to connect an ISDN phone to the server and use it as a SIP-phone
c) all of the above?
Regards,
Evert
Addition: the console also has these showing:
Jul 29 09:58:06 WARNING[1142106560]: chan_sip.c:612 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 102 (Non-critical Request)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Evert
Thanks for your swift reply!
It did help me... kind of... ;)
Guess what I had to do to get it working on my system? I had to ADD
dtmfmode=inband to my config! 8-)
But now I have full access to my mailbox! :)
Regards,
Evert
-Original Message-
From: [EMAIL PROTECTED
Date: Wed, 28 Jul 2004 13:44:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0
IP 192.168.2.175 is the phone
IP 192.168.11.6 is Asterisk
(it's not a routing problem, since other phones on the 192.168.2.x IP's do
show up as 'OK')
Regards,
ere to fix this...?
Regards,
Evert
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config on that phone? If so, who can tell
me what?
Regards,
Evert
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