Re: [asterisk-users] extension not found
Ben Schorr wrote: > > Is there some reason why I keep getting this same message from “cool > dude” over and over and over? And under different subject lines? > > I assumed that he was sending it to every post, trying to get a response. Sorta spammy in my opinion. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] extension not found
Is there some reason why I keep getting this same message from "cool
dude" over and over and over? And under different subject lines?
Ben M. Schorr
Chief Executive Officer
__
Roland Schorr & Tower
www.rolandschorr.com <http://www.rolandschorr.com/>
b...@rolandschorr.com
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of cool dude
Sent: Friday, February 12, 2010 21:24
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] extension not found
hi friend need ur help in dial plan, i want to allow exten 2000 to 2005
can make call outside and exten 2006 to 2010 can not make call outside.
heres my dial plan.
sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
context = others
[2000]
type=friend
context=outside
secret=1234
host=dynamic
[2001]
type=friend
context=outside
secret=1234
host=dynamic
[2002]
type=friend
context=outside
secret=1234
host=dynamic
[2003]
type=friend
contex=outside
secret=1234
host=dynamic
[2004]
type=friend
contex=outside
secret=1234
host=dynamic
[2005]
type=friend
contex=outside
secret=1234
host=dynamic
[2006]
type=friend
contex=internal
secret=1234
host=dynamic
[2007]
type=friend
contex=internal
secret=1234
host=dynamic
[2008]
type=friend
contex=internal
secret=1234
host=dynamic
[2009]
type=friend
contex=internal
secret=1234
host=dynamic
[2010]
type=friend
contex=internal
secret=1234
host=dynamic
## ##
vi /etc/asterisk/extensions.conf
[from-zaptel]
exten => s,1,wait(2)
exten => s,n,dial(sip/2000)
exten => s,n,dial(sip/2001)
exten => s,n,Playback(tt-weasels)
[others]
include => internal
include => outside
[internal]
exten => _20XX,1,Dial(SIP/${EXTEN})
exten => _20XX,n,VoiceMail(${ext...@others,u)
exten => _20XX,n,Hangup()
[outside]
exten => 2001,1,Dial(Zap/1-1/${EXTEN})
exten => 2001,n,Hangup
exten => 2002,1,Dial(Zap/1-1/${EXTEN})
exten => 2002,n,Hangup
exten => 2003,1,Dial(Zap/1-1/${EXTEN})
exten => 2003,n,Hangup
exten => 2004,1,Dial(Zap/1-1/${EXTEN})
exten => 2004,n,Hangup
exten => 2005,1,Dial(Zap/1-1/${EXTEN})
exten => 2005,n,Hangup
this is the log when i am calling from exten 2000 to outside
Connected to Asterisk 1.4.29 currently running on localhost (pid = 2243)
Verbosity is at least 3
[Feb 13 12:05:47] NOTICE[2482]: chan_sip.c:15124 handle_request_invite:
Call from '2002' to extension '9193696136' rejected because extension
not found.
Your Mail works best with the New Yahoo Optimized IE8. Get it NOW!
<http://in.rd.yahoo.com/tagline_ie8_new/*http:/downloads.yahoo.com/in/in
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[asterisk-users] extension not found
hi friend need ur help in dial plan, i want to allow exten 2000 to 2005 can
make call outside and exten 2006 to 2010 can not make call outside. heres my
dial plan.
sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
context = others
[2000]
type=friend
context=outside
secret=1234
host=dynamic
[2001]
type=friend
context=outside
secret=1234
host=dynamic
[2002]
type=friend
context=outside
secret=1234
host=dynamic
[2003]
type=friend
contex=outside
secret=1234
host=dynamic
[2004]
type=friend
contex=outside
secret=1234
host=dynamic
[2005]
type=friend
contex=outside
secret=1234
host=dynamic
[2006]
type=friend
contex=internal
secret=1234
host=dynamic
[2007]
type=friend
contex=internal
secret=1234
host=dynamic
[2008]
type=friend
contex=internal
secret=1234
host=dynamic
[2009]
type=friend
contex=internal
secret=1234
host=dynamic
[2010]
type=friend
contex=internal
secret=1234
host=dynamic
vi /etc/asterisk/extensions.conf
[from-zaptel]
exten => s,1,wait(2)
exten => s,n,dial(sip/2000)
exten => s,n,dial(sip/2001)
exten => s,n,Playback(tt-weasels)
[others]
include => internal
include => outside
[internal]
exten => _20XX,1,Dial(SIP/${EXTEN})
exten => _20XX,n,VoiceMail(${ext...@others,u)
exten => _20XX,n,Hangup()
[outside]
exten => 2001,1,Dial(Zap/1-1/${EXTEN})
exten => 2001,n,Hangup
exten => 2002,1,Dial(Zap/1-1/${EXTEN})
exten => 2002,n,Hangup
exten => 2003,1,Dial(Zap/1-1/${EXTEN})
exten => 2003,n,Hangup
exten => 2004,1,Dial(Zap/1-1/${EXTEN})
exten => 2004,n,Hangup
exten => 2005,1,Dial(Zap/1-1/${EXTEN})
exten => 2005,n,Hangup
this is the log when i am calling from exten 2000 to outside
Connected to Asterisk 1.4.29 currently running on localhost (pid = 2243)
Verbosity is at least 3
[Feb 13 12:05:47] NOTICE[2482]: chan_sip.c:15124 handle_request_invite: Call
from '2002' to extension '9193696136' rejected because extension not found.
The INTERNET now has a personality. YOURS! See your Yahoo! Homepage.
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Re: [asterisk-users] Extension not found
Hi Michel,
Am Freitag, den 12.09.2008, 17:41 +0300 schrieb michel freiha:
> Dear All,
>
> I have the following scenario...When a customer dial 111 number a beep
> message will iplay in order to record and playback his voice...Else
> he'll be routed to another call flow as you can see in the context
> below:
>
>
> [a2billing]
> exten => _X.,1,Gotoif($[${EXTEN} = 111] ? custom-recordme,111,1)
> exten => _X.,2,DeadAGI,a2billing.php
> exten => _X.,3,Wait,2
> exten => _X.,4,Hangup
>
> But i have the following error when trying to dial 111:
>
> [Sep 12 14:16:32] WARNING[30978]: pbx.c:2483 __ast_pbx_run: Channel
> 'SIP/michofr-093833e0' sent into invalid extension '111' in context '
> custom-recordme', but no invalid handler
The above dialplan sends the call into context "custom-recordme" with
extension 111 and to priority 1, if the caller dials 111. For further
help we would need that context too.
Regards,
Karsten
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[asterisk-users] Extension not found
Dear All,
I have the following scenario...When a customer dial 111 number a beep
message will iplay in order to record and playback his voice...Else he'll be
routed to another call flow as you can see in the context below:
[a2billing]
exten => _X.,1,Gotoif($[${EXTEN} = 111] ? custom-recordme,111,1)
exten => _X.,2,DeadAGI,a2billing.php
exten => _X.,3,Wait,2
exten => _X.,4,Hangup
But i have the following error when trying to dial 111:
[Sep 12 14:16:32] WARNING[30978]: pbx.c:2483 __ast_pbx_run: Channel
'SIP/michofr-093833e0' sent into invalid extension '111' in context '
custom-recordme', but no invalid handler
Any help?
Regards
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Re: [asterisk-users] Extension not found
Dear Randulo, Thanks for your suggention. Now i am able to communicate between 2 computers. Regards, Baskar --- randulo <[EMAIL PROTECTED]> wrote: > On Mon, May 19, 2008 at 8:44 AM, bas karan > <[EMAIL PROTECTED]> wrote: > > [May 19 12:02:29] NOTICE[2559]: chan_sip.c:13879 > > handle_request_invite: Call from 'Phone3' to > extension > > '5' rejected because extension not found. > >-- Registered SIP 'Phone3' at 192.168.1.101 > port > > Extension.conf enteries are, > > exten => 3,1,Dial(SIP/Phone3,30,tr) > > exten => 4,1,Dial(SIP/Phone4,30,tr) > > exten => 5,1,Dial(SIP/Phone5,30,tr) > > Where is the [sip] context named in the phones > context= statement ? > > ___ > -- Bandwidth and Colocation Provided by > http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > Has your work life balance shifted? Find out - http://in.search.yahoo.com/search?&fr=na_onnetwork_mail_taglines&ei=UTF-8&rd=r1&p=work+life+balance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension not found
Thanks :-D change the context to default and everithing works fine. I assigned the sip context because that was the context on the example. Thanks :-) Nomar Alex Balashov wrote: Nomar Mora wrote: Alex Balashov wrote: Do you have dial plan routes for internal extension calls? Do you mean if I have configured the extension.conf? Yes, I config the extensions on the extension.conf file otherwise, no I have not. Thanks in Advance Nomar In the 'sip' context? -- 2008 Año del satélite Simón Bolívar ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension not found
Nomar Mora wrote: > Alex Balashov wrote: >> Do you have dial plan routes for internal extension calls? >> >> > Do you mean if I have configured the extension.conf? Yes, I config the > extensions on the extension.conf file > otherwise, no I have not. > > Thanks in Advance > Nomar > In the 'sip' context? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension not found
Alex Balashov wrote: > Do you have dial plan routes for internal extension calls? > > Do you mean if I have configured the extension.conf? Yes, I config the extensions on the extension.conf file otherwise, no I have not. Thanks in Advance Nomar -- 2008 Año del satélite Simón Bolívar ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension not found
Do you have dial plan routes for internal extension calls? Nomar Mora wrote: > Good day: > > I recently install asterisk-now. Setup a pair of SipXpert 160 phones and > all fine. Later I try to setup some Octtels VoIP Gateways SP4220. I > config the proxy setings like this: > > http://www.fundacitetachira.gob.ve/settings/Settings.png > > the sip.conf entrys are like the asterisk manual says > SipXpert 160 phone and the gateway sip config are like this: > > type=friend > host=xxx.yyy.zzz.aaa > dtmfmode=rfc2833 > mailbox=6050 > context=sip > callerid="Mukunda" <6001> > username=username > secret=secret > > I can make calls between the the sipxpert phones and make calls from the > sipxperts to the gateways but I can make calls from the gateways to any > extension. The log says: > > [May 22 16:54:49] NOTICE[2708] chan_sip.c: Call from '6004' to extension > '6000' rejected because extension not found. > > > > Any suggest? > > Thanks in advance > -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Extension not found
Good day: I recently install asterisk-now. Setup a pair of SipXpert 160 phones and all fine. Later I try to setup some Octtels VoIP Gateways SP4220. I config the proxy setings like this: http://www.fundacitetachira.gob.ve/settings/Settings.png the sip.conf entrys are like the asterisk manual says SipXpert 160 phone and the gateway sip config are like this: type=friend host=xxx.yyy.zzz.aaa dtmfmode=rfc2833 mailbox=6050 context=sip callerid="Mukunda" <6001> username=username secret=secret I can make calls between the the sipxpert phones and make calls from the sipxperts to the gateways but I can make calls from the gateways to any extension. The log says: [May 22 16:54:49] NOTICE[2708] chan_sip.c: Call from '6004' to extension '6000' rejected because extension not found. Any suggest? Thanks in advance -- 2008 Año del satélite Simón Bolívar ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension not found
On Mon, May 19, 2008 at 11:36 AM, bas karan <[EMAIL PROTECTED]> wrote: > I am new to this concept, Could you explain me little > bit extra please? You will need to put extensions in contexts. The context is a fundamental concept of the dialplan. All extensions are inside a context. In your sip.conf file you have given the phones a context you call "sip". That means that when a call enters it appears in the sip context (under [sip] in extensions.conf) The error message clearly tells you that the extension has not been found, meaning the call is not in the correct context. You should probably read this book: http://tfot.leifmadsen.com to get the basic concepts down or maybe this one: http://snurl.com/dasbuch Best, /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension not found
Dear Randulo, Thanks for your replay. I am new to this concept, Could you explain me little bit extra please? Thanks & Regards, Baskar --- randulo <[EMAIL PROTECTED]> wrote: > On Mon, May 19, 2008 at 8:44 AM, bas karan > <[EMAIL PROTECTED]> wrote: > > [May 19 12:02:29] NOTICE[2559]: chan_sip.c:13879 > > handle_request_invite: Call from 'Phone3' to > extension > > '5' rejected because extension not found. > >-- Registered SIP 'Phone3' at 192.168.1.101 > port > > Extension.conf enteries are, > > exten => 3,1,Dial(SIP/Phone3,30,tr) > > exten => 4,1,Dial(SIP/Phone4,30,tr) > > exten => 5,1,Dial(SIP/Phone5,30,tr) > > Where is the [sip] context named in the phones > context= statement ? > > ___ > -- Bandwidth and Colocation Provided by > http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > Meet people who discuss and share your passions. Go to http://in.promos.yahoo.com/groups/bestofyahoo/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension not found
On Mon, May 19, 2008 at 8:44 AM, bas karan <[EMAIL PROTECTED]> wrote: > [May 19 12:02:29] NOTICE[2559]: chan_sip.c:13879 > handle_request_invite: Call from 'Phone3' to extension > '5' rejected because extension not found. >-- Registered SIP 'Phone3' at 192.168.1.101 port > Extension.conf enteries are, > exten => 3,1,Dial(SIP/Phone3,30,tr) > exten => 4,1,Dial(SIP/Phone4,30,tr) > exten => 5,1,Dial(SIP/Phone5,30,tr) Where is the [sip] context named in the phones context= statement ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Extension not found
Dear Friends, This is Baskar from Chennai, trying to configure asterisk. Now I planned to start with communication between 2 systems using soft phones. When I tried to call the other computer I am getting the following error message on asterisk terminal, Connected to Asterisk 1.4.18 currently running on asterisker (pid = 2478) Verbosity is at least 3 [May 19 12:02:29] NOTICE[2559]: chan_sip.c:13879 handle_request_invite: Call from 'Phone3' to extension '5' rejected because extension not found. -- Registered SIP 'Phone3' at 192.168.1.101 port 5060 expires 70 asterisker*CLI> SIP.conf Entries are as follows: [Phone3] type = friend secret=Phone3 host = dynamic defaultip = 192.168.1.101 dtmfmode = rfc2833 context = sip callerid = "Phone3" <3> [Phone4] type = friend secret=Phone4 host = dynamic defaultip = 127.0.0.1 dtmfmode = rfc2833 context = sip callerid = "Phone4" <4> [Phone5] type = friend secret=Phone5 host = dynamic defaultip = 192.168.1.51 dtmfmode = rfc2833 context = sip callerid = "Phone5" <5> Extension.conf enteries are, exten => 3,1,Dial(SIP/Phone3,30,tr) exten => 4,1,Dial(SIP/Phone4,30,tr) exten => 5,1,Dial(SIP/Phone5,30,tr) Please help me to fix this issue. Thank in advance. Regards, Baskar Bollywood, fun, friendship, sports and more. You name it, we have it on http://in.promos.yahoo.com/groups/bestofyahoo/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
