You need to tell asterisk to stay in the media path. You must add 't'
to the Dial options:
exten => 21,1,Dial(${phone1},20,trwL(30:24:6))
Or set canreinvite=no in your sip peer definition.
Julian J. M.
> but even adding it and commenting out "automon => *
_disturb]
exten => s,1,Wait(2)
exten => s,2,Answer
exten => s,3,Playback(pls-try-call-later)
exten => s,4,Voicemail(u${ARG1})
exten => s,5,Hangup
Julian J. M.
On Sun, 27 Feb 2005 19:58:32 +0100, Riphagen, Ferdy
<[EMAIL PROTECTED]> wrote:
> Hello All,
>
> I have a
.
.
Julian J. M.
> SIP debug shows that phone registers with public IP address of the site,
> while calls somehow go to local address.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/li
=yes
.
.
Julian J. M.
On Sun, 27 Feb 2005 11:04:25 +0100, Michiel van Baak
<[EMAIL PROTECTED]> wrote:
> On 20:52, Sun 27 Feb 05, Rudolf Ladyzhenskii wrote:
> > Thanks for suggestion.
> >
> > Unfortunately did not work.
> > What does this option do anyway
Dial(SIP/whatever,30,m)
instead of 'r'
http://www.voip-info.org/wiki-Asterisk+cmd+dial
Julian
On Fri, 25 Feb 2005 17:18:59 +0500, Muhammad Muzzamil Luqman
<[EMAIL PROTECTED]> wrote:
>
> I want to setup a senario in which the callers hears to some music file
> while the phone is ringing and a
You could add
exten => 1,2,Goto(context,2,2)
But I don't know what will happen when, after 5 secs, dial SIP/2 is
executed again...
Julian
On Fri, 25 Feb 2005 12:56:14 +0100, Elmar Haneke <[EMAIL PROTECTED]> wrote:
> > exten => s,1,Dial(Local/[EMAIL PROTECTED]&Local/[EMAIL
> > PROTECTED]&Local
Are you running apache as root or as the asterisk user? If not, maybe
it's a permissions problem...
Julian J. M.
On Fri, 25 Feb 2005 03:39:30 -0600, Martin Keding
<[EMAIL PROTECTED]> wrote:
> I install WebVmail today on a Fedora 2 box. I got the cgi script running etc
> an
Hello,
I've just uploaded a patch to amportal project at sourceforge, to
support Zap Extensions...
http://sourceforge.net/tracker/index.php?func=detail&aid=1146433&group_id=121515&atid=690574
I'd appreciate some feedback ;)
Greetings
Julian J. M.
On Sun, 20 F
What phone are you using when calling? Does it have silence supression
on? Try disablig it... It could be a timing issue.
Julian.
On Mon, 21 Feb 2005 12:27:54 -0600, Anton Krall
<[EMAIL PROTECTED]> wrote:
> Brian:
>
> Found the MOH random answer on the wiki, you were right... All the basic
> st
http://www.voip-info.org/wiki-Asterisk+config+extensions.conf+sorting
On Tue, 22 Feb 2005 09:07:43 +0200, Altus Snyman <[EMAIL PROTECTED]> wrote:
> Good day all
> I registered at a few sip server in different countries
> Now I want to route outgoing calls for that country threw that sip
> server
You can't change the callerid on an outgoing PSTN call (at least on
analog lines).
To modifiy the callerid on incoming calls, you could do something like
this (not tested):
[incoming-line1]
exten => s,1,setCidName("Line1: " . ${CALLERID})
exten => s,2,Goto(Incoming,s,1)
[incoming]
exten => s,1,.
Check your soundcard controls... maybe it's recording "what you hear"
or PCM, thus sending it again to the other party.
Julianjm.
On Mon, 21 Feb 2005 09:47:55 +0200, Nic le Roux <[EMAIL PROTECTED]> wrote:
>
> Good Morning,
>
> I have a weird situation,
> I'm testing with Xlite as SIP phon
Why?
When you register with another provider, the use or not of MD5 Auth is
up to him/her...
When other clients are registering with you, you may require MD5 auth
(Auth=MD5)... If you don't want to have a cleartext password, you can
use md5secret=. instead of secret=
Greetings
Julian.
On
On Fri, 04 Feb 2005 00:38:35 -0600, Eric Wieling <[EMAIL PROTECTED]> wrote:
> [EMAIL PROTECTED] wrote:
>
> >>Anybody with an idea how will I set me Asterisk to send call straight to my
> >>extension with out playing demo-Congrat MENU.
> >
> > Comething like this
> >
> > exten => s,1,Answer
> > ext
Why are you connecting a phone to the PSTN?
On Fri, 4 Feb 2005 10:15:52 -0600, [EMAIL PROTECTED]
<[EMAIL PROTECTED]> wrote:
>
>
>
> Hi asterisk GURUs I have an issue with a call forwarding setup, I have the
> next zapata conf:
>
>
>
> -zapata.conf-
>
>
On Mon, 31 Jan 2005 06:12:53 -0600, Rich Adamson <[EMAIL PROTECTED]> wrote:
> If you incorrectly connect a fxs module to a pstn line, you will
> likely blow the module making it useless.
Wow, i think it shoud be more fool-proof ;) Lucky I tried first with
an analog phone in my TDM11B...
Julian.
Dont you need allow=ulaw before the register=> line?
On Sat, 29 Jan 2005 13:13:28 -0700, Joseph <[EMAIL PROTECTED]> wrote:
> When I try to call out to FWD over IAX2 I get:
> Call rejected by 65.39.205.121: Unable to negotiate codec
>
> I'm using asterisk-1.0.5 (the same settings works fine with
Check sample.call in the asterisk tarball... edit the file, move it to
/var/spool/asterisk/outgoing and it'll dial and connect de callee with
the extension of your choice...
Greetings
On Mon, 24 Jan 2005 02:57:13 -0600, Roger Hanson <[EMAIL PROTECTED]> wrote:
> I'm trying to get a script working
You can set the default sip context to a nonexistant, and set the
correct one in the peer definition... Although i guess there must be a
better solution ;)
JulianJM
On Tue, 11 Jan 2005 11:35:36 +0100, Florian Effenberger <[EMAIL PROTECTED]>
wrote:
> Hello there,
>
> I am playing around with As
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