a little
bit you will find lots - each with a different pricing (just like VoIP
providers).
If you are in the UK, I can recommend sendmytxt.
An answer to your question: Yes, this can be done without GSM modem
connected.
Rene Kluwen
Chimit
-Original Message-
From: [EMAIL PROTECTED]
[mailto
.
An external entity (such as us, Chimit) can possibly do that for you.
Rene Kluwen
Chimit
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of B
Sent: donderdag 10 augustus 2006 19:47
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: sms callback
(at least for me).
I don't know how or why this is. But it is my experience. I am not using i4l
anymore.
Rene Kluwen
Chimit
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Alain
Degreffe
Sent: woensdag 5 april 2006 17:18
To: asterisk-users@lists.digium.com
Subject
I have the same problem.
My solution is differentiate in extensions.conf, since all calls are
terminated to different MSISDN's.
So in extensions.conf I have something like:
[incoming]
exten = 9995551212,1,Goto(company1-context,s,1)
exten = 9995551213,1,Goto(company2-context,s,1)
etc.
Rene
Just a long shot: Do you need to Answer() the call first?
exten = 10,1,Answer
exten = 10,2,MP3Player(...)
Like this.
Rene Kluwen
Chimit
-Ursprüngliche Nachricht-
Von: Bayrouni [mailto:[EMAIL PROTECTED]
Gesendet: Dienstag, 7. Februar 2006 21:39
An: [EMAIL PROTECTED]; Asterisk
tips?
Rene Kluwen
Chimit
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I am running Asterisk 1.0.9 and the latest bristuff in combination with a
HFC isdn card, connected to a BRI interface.
For some reason, I am not able to have it dial out (see below). It exits
with DIALSTATUS=CHANUNAVAIL.
One thing that may be misconfigured is that it says: Signalling Type: PRI
Deze meneer:
[EMAIL PROTECTED]
heeft
wat jij wilt.
--
Rene
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of
makaSent: vrijdag 23 september 2005 18:17To:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] ZAP ISDN
losing
Use:
ignorepat = 9
in
your dialplan.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Brian
McEntireSent: vrijdag 23 september 2005 20:29To:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject:
[Asterisk-Users] Continue
This is a limitation of your PSTN provider.
The telco's don't allow you to set your callerid number when dialing out.
They always change it to the one that is allocated to you.
Rene Kluwen
Chimit
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
[EMAIL
I call
to Mexico (fixed and cellular phones) via an IAX2 link. And it gives me no
problems at all.
What
kind of trunk are you using?
Maybe
you should post your dial plan... probably there is a mistake
there...
Rene
Kluwen
Chimit
-Original Message-From:
[EMAIL PROTECTED
countries). I am posting this just like a (IMHO) useful idea.
Any comments?
Rene Kluwen
Chimit
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Nathan
Pralle
Sent: dinsdag 20 september 2005 17:50
To: Asterisk Users Mailing List - Non-Commercial Discussion
I have the same problem. Asterisk always makes two calls, even when the
first one went through successfully.
Rene Kluwen
Chimit
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Remco
Barende
Sent: maandag 19 september 2005 22:09
To: Asterisk Users Mailing
I am running Asterisk 1.0.9 and the latest bristuff in combination with a
HFC isdn card, connected to a BRI interface.
For some reason, I am not able to have it dial out (see below). It exits
with DIALSTATUS=CHANUNAVAIL.
One thing that may be misconfigured is that it says: Signalling Type: PRI
admin?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Joshua
Abbott
Sent: zaterdag 17 september 2005 1:45
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Grandstream
Where do I find or what is the default password for
port work. When an incoming call via SIP
comes in, the phone rings and then the SIP connection is dropped with
status: NO-ANSWER. How do I make it work?
My other question: Does somebody have a working example so a Caller ID
signal is presented on the FXS ports?
TIA,
Rene Kluwen
Chimit
In sip.conf, when using:
bindaddr=0.0.0.0
incoming registrations are denied with a 403 [Unauthorized] response.
Is this not the way to listen on all interfaces?
When using:
bindaddr=213.193.241.43
everything works fine. How do I listen on all available interfaces?
Rene Kluwen
Chimit
iaxcomm: http://iaxclient.sourceforge.net/iaxcomm/
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of
MattSent: woensdag 31 augustus 2005 1:55To:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] free open
source softphone for
I have a C++ wrapper class around the cagi class that is listed on the wiki.
It doesnt implement everything and to tell you the truth is still in beta.
But it works... and yours if you want to help testing it.
Rene Kluwen
Chimit
Being a lazy person, I was wondering if anyone has a c++ class
For Asterisk to play MOH, it will need to have an RTP connection, right?
How otherwise, would you want to play MOH?
Rene Kluwen
Chimit
For canreinvite=yes to work, I think I need to remove the t argument in
the Dial(SIP/ext|60|t) application. Otherwise, Asterisk will allways
stay
This morning, also calls to Mexico were dropped.
Rene Kluwen
Chimit
No problems with US calls. No calls going through to UK though. My
account login on the website worked this morning.
Michael
Original Message
Subject: RE: [Asterisk-Users] VoipJet Problems - anyone
a
second time!
It never connects three times (which I think is weird).
The question: How do I prevent Asterisk from calling twice?
Setting MaxRetries to 0 does not seem to help.
All ideas are welcome
Rene Kluwen
Chimit
___
Asterisk-Users mailing list
This only works if you DONT have:
insecure=very
in your SIP section.
Rene Kluwen
Chimit
In sip.conf you specify a context right?
In extensions.conf, in that context, you route the call...
exten = 12134441234,1,Dial(whatever)
--On Thursday, July 21, 2005 11:30 AM -0600 Olusoji (soji
This only works if you DONT have:
insecure=very
in your SIP section.
Rene Kluwen
Chimit
In sip.conf you specify a context right?
In extensions.conf, in that context, you route the call...
exten = 12134441234,1,Dial(whatever)
--On Thursday, July 21, 2005 11:30 AM -0600 Olusoji (soji
Hmmm...
The first one that can send a message with Kannel (+ SMPP) to a VoIP phone
gets a price.
Kannel is used to connect to SMS gateways to cellular phones and paging
devices. It has no VoIP abilities to send messages to VoIP phones.
Cheers
Rene Kluwen
Chimit
Asterisk + Kannel. When you
Hello Stijn,
This is not a problem on the Asterisk website.
I have a working KPN SMS setup. Just the KPN SMSC is buggy.
Sometimes it does not accept the SMS. Other times it accepts the SMS, but
doesnt send it -or- will send it after a LONG delay (couple of hours,
sometimes days).
Cheers,
Rene
respond
to it?
Rene Kluwen
Chimit
On 19:16, Sun 10 Jul 05, Peter Raaijmaker wrote:
(this time with subject)
Hello,
I?m trying to get Asterisk to accept incoming calls from budgetphone.nl.
When I dial my budgetphone nr on a PSTN KPN line it immediately gives a
busy
tone.
I tried X
Yups... at least via FWD it is still
working.
Rene Kluwen
Chimit
- Original Message -
From:
Liaan vd Merwe
To: asterisk-users@lists.digium.com
Sent: Friday, January 28, 2005 4:48
PM
Subject: [Asterisk-Users] Fwd and
Tollfree
Hallo all
do any of you
The first Grandstream phone that I ordered also had a defect.
After being in touch with their support test, I could just send the unit
back and I got a replacement phone.
Simple as that.
Rene Kluwen
Chimit
- Original Message -
From: Ronald Wiplinger [EMAIL PROTECTED]
To: Asterisk Users
disable the G729 stack in the Grandstream
*ponders* in case the problem is there.
Rene Kluwen
Chimit
- Original Message -
From: William Suffill [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, January 17, 2005 2:04
.
Either way, for me this solution works.
Rene Kluwen
Chimit
- Original Message -
From: Rene Kluwen [EMAIL PROTECTED]
To: William Suffill [EMAIL PROTECTED]; Asterisk Users Mailing
List - Non-Commercial Discussion asterisk-users@lists.digium.com
Sent: Monday, January 17, 2005 1:47 PM
Subject: Re
There's lots.
www.clickatell.com is one of them.
Google for sms gateway and you will find a bunch - especially in the
paid-add section.
Rene Kluwen
Chimit
- Original Message -
From: Brian C. Fertig [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk
Alternatively,
What I (we) do personally: In stead of having *
call my cellphone, it sends an MMS message with the message audio as
content.
Rene Kluwen
Chimit
- Original Message -
From:
Mike
Boger Jr
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Title: Passing PIN Numbers
This is a long shot, I am not sure if it will solve
your problem:
Did you try to change dtmfmode in
sip.conf?
Rene Kluwen
Chimit
- Original Message -
From:
Michael Di Martino
To: asterisk-users@lists.digium.com
Sent: Friday, January 14
make it work. I
changed the option back again because all other services (FWD, BRI, IAX2)
work like this and I don't want to break them.
Any suggestions about what I can change to make this work?
Cheers!
Rene Kluwen
Chimit
___
Asterisk-Users mailing
Andres,
Thanks for your answer, but as you can see in the output from show
translation in my original post my Asterisk DOES have G729 support.
Also the fact that softphones work but the Grandstream does not work
stumbles me.
Rene Kluwen
Chimit
- Original Message -
From: Andres [EMAIL
I am also interested.
Pls. contact me at [EMAIL PROTECTED]
Rene Kluwen
Chimit
- Original Message -
From: Nathan Goodwin [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, January 13, 2005 7:49 PM
Subject: Re
cards utilizing Winbond chips? Or do you guys think that
changing to capi would not help me in this matter?
Thanks in advance,
Rene Kluwen
Chimit
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and a SIP stack
built in. You put your SIM card in the box and it is able to dial out
for you.
Rene Kluwen
Chimit
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Result: Failed to resolve callee's address
harry gaillac wrote:
Hi all,
Anybody would be able to call my voicemail just for
test
sip:[EMAIL PROTECTED]
regards
harry
Le nouveau Yahoo! Messenger est arrivé ! Découvrez toutes les nouveautés pour
dialoguer
Hoi about having the calls forwarded by your phone
company?
Usually you can dial *21*number# or something and
your calls go to a remote party.
Same goes for delayed forwarding
*61*
Rene Kluwen
Chimit
- Original Message -
From:
Jim Dossey
To: Asterisk Users Mailing
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