"
Sent: Sunday, July 24, 2005 11:30 PM
Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202
Sorry to be a pain... but
I restarted my * and now when I launch * get this:
== Parsing '/etc/asterisk/zapata.conf': Found
Jul 24 18:52:45 WARNING[6817]: chan_zap.c:932
Message -
From: "Mark Edwards" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Sunday, July 24, 2005 10:13 PM
Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202
OK Angus
just start here
mv extensions.c
July 24, 2005 10:06 PM
Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202
Ok your extensions.conf doesn't mention anything about an
extension/number equal to 202 or 200. You must know that the name of a
SIP and IAX2 peer is only an "address", you will have to assign a
nu
ook in a certain .h file?
Angus
- Original Message -
From: "dbruce" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Sunday, July 24, 2005 10:10 PM
Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202
The
uot;
Sent: Sunday, July 24, 2005 10:06 PM
Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202
Ok your extensions.conf doesn't mention anything about an extension/number
equal to 202 or 200. You must know that the name of a SIP and IAX2 peer is
only an "ad
en => s,3,Background(submenuopts) ; "Thanks for calling the sales
> > department. Press 1 for steve, 2 for..."
> > ;exten => 1,1,Goto(default,steve,1)
> > ;exten => 2,1,Goto(default,mark,2)
> >
> > [default]
> > ;
> > ; By default we include the demo. In a prod
t with a
> single
> ; digit that is fairly large (like 6 or 7) so that you have plenty of room
> to
> ; overlap extensions and menu options without conflict. You can alias them
> with
> ; names, too and use global variables
>
> ;exten => 6245,hint,SIP/Grandstream1&
t;
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Sunday, July 24, 2005 2:50 PM
Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202
> I think the 777 may be a bit of a Red Herring. I dialed 777 as a test. I
> can't dial 202 from 200 if I actu
eed to edit meetme.conf to enable this
room)
;
;exten => 8600,1,Meetme(1234)
;
; Or playing an announcement to the called party, as soon it answers
;
;exten = 8700,1,Dial(${MARK},30,A(/path/to/my/announcemsg))
;
; For more information on applications, just type "show applications" at
your
manner... such
as:
INVITE
407 Proxy Authentication Required
ACK
INVITE
404 Not Found
ACK
The idea was to provide a clue... not to provide a complete working
dialplan
and phone configuration. Providing new users with "the complete package"
is
a dis-service to them. They will only learn fr
ercial Discussion"
Sent: Sunday, July 24, 2005 12:53 PM
Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202
> Derek: you reply is uncorrect. If Angus has the extension 777 in his
> dialplan/extensions.conf which will dial 202. The name of the peer has
> absolutel
m.com
<mailto:asterisk-users@lists.digium.com>
*Sent:* Sunday, July 24, 2005 11:51 AM
*Subject:* [Asterisk-Users] Why can't sip/200 call sip/202
I have 2 sip accounts setup - 200 and 202. If I do sip show peers I
get:
sip show peers
Name/usernam
need to dial 202
on extension 200, not 777.
Regards,
Derek
- Original Message -
From:
Angus
Comber
To: asterisk-users@lists.digium.com
Sent: Sunday, July 24, 2005 11:51
AM
Subject: [Asterisk-Users] Why can't
sip/200 call sip/202
I have 2 sip acc
> I have 2 sip accounts setup - 200 and 202. If I do sip show peers I get:
>
> sip show peers
> Name/usernameHostDyn Nat ACL Mask Port Status
> 202/202 192.168.0.6 D 255.255.255.255 5060
> Unmonitored
> 201/201 (Unspecified)
I have 2 sip accounts setup - 200 and 202. If
I do sip show peers I get:
sip show peersName/username
Host Dyn Nat
ACL Mask
Port
Status202/202
192.168.0.6
D 255.255.255.255
5060
Unmonitored201/201
(Unspecified)
D
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