Re: [Asterisk-Users] where I can find some learning book about asterisk?
Hello, Take a look at http://www.signate.com You can also find various documentation resources at http://www.voip-info.org/tiki-index.php?page=Asterisk Regards Lamine - Original Message - From: "FCG ZHAO Zigang" <[EMAIL PROTECTED]> To: Sent: Friday, December 24, 2004 2:05 AM Subject: [Asterisk-Users] where I can find some learning book about asterisk? Hello , I learn handbook-draft.but I think I don't understand asterisk. where I can find some learning book about asterisk? thank u. B.R. John. -原始邮件- 发件人: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] 发送时间: 2004年12月24日 7:51 收件人: asterisk-users@lists.digium.com 主题: Asterisk-Users Digest, Vol 5, Issue 350 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than "Re: Contents of Asterisk-Users digest..." Today's Topics: 1. RE: rtp channels not through asterisk (Brian West) 2. Turning "*" Hangup off in queues ([EMAIL PROTECTED]) 3. Re: Voicemail email notification (Rich Adamson) 4. Can't Make Outgoing Call (Norman Zhang) 5. Re: Voicemail email notification (Dorn Hetzel) 6. Re: Asterisk in parallel with PSTN [OT] (Rich Adamson) 7. Re: rtp channels not through asterisk (Rich Adamson) 8. Re: Realtime sipbuddies table structure why? (Greg - Cirelle Enterprises) 9. RE: Polycom Buddies (Paul Hales) 10. Re: Queue - roundrobin member order (Adam Goryachev) 11. Re: Voicemail email notification (Rich Adamson) 12. Re: Can't Make Outgoing Call (Norman Zhang) 13. Re: Recommended IAX softphone. (Bruno Hertz) 14. Re: sip seeding vs registration (Greg - Cirelle Enterprises) 15. Asterisk 1.0.3 no RedHat zaptel script? (Jerry Geis) 16. Re: Recommended IAX softphone. (Erik Espinoza) -- Message: 1 Date: Thu, 23 Dec 2004 16:51:22 -0600 From: "Brian West" <[EMAIL PROTECTED]> Subject: RE: [Asterisk-Users] rtp channels not through asterisk To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="US-ASCII" canreinvite=yes Aterisk stays in the signaling path so unless you're running tcpdump or the like you'll never notice this. bkw > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of bijan > Sent: Thursday, December 23, 2004 4:46 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] rtp channels not through asterisk > > In wiki pages it is stated that The audio channels (RTP) may go directly > from phone to phone or may go through Asterisk's media bridge. > Currently with my settings, I notice that all rtp's are passing through my > asterisk. How could I achieve that they go directly from phone to phone? > I assume this way, my machine will have less load and therefore could > handle more calls. > > regards > Bijan Karimi > -- Message: 2 Date: Thu, 23 Dec 2004 19:16:19 -0600 (CST) From: [EMAIL PROTECTED] Subject: [Asterisk-Users] Turning "*" Hangup off in queues To: asterisk-users@lists.digium.com Message-ID: <[EMAIL PROTECTED]> Content-Type: TEXT/PLAIN; charset=US-ASCII Hi ! Can somebody tell me how to turn the "*" Hangup option utrned off in queues. I have not used any H option but still as an agent if I press "*" key the user gets disconnected. Somehow it is turned on by default. Can I turn this option off In my extensions.conf I have written : exten => 8000,3,Queue(supportq|t) plz help me inthis regard ... Thanks ! Usman. -- Message: 3 Date: Thu, 23 Dec 2004 16:51:34 -0600 From: Rich Adamson <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Voicemail email notification To: Asterisk Users Mailing List - Non-Commercial Discussion Message-ID: <[EMAIL PROTECTED]> Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1 > Are there any common silent failure modes for email > notification from the Voicemail module. I put the > email and pager email addresses in my entry in > voicemail.conf but no mail gets sent when I leave > a voicemail. No obvious error messages either, > unless I'm just not looking in the right place. > > Thanks for any clues :) Nop, that's it other then you have to have sendmail configured and running on the system (or have a substitute mail handler). Rich
[Asterisk-Users] where I can find some learning book about asterisk?
Hello ,
I learn handbook-draft.but I think I don't understand asterisk.
where I can find some learning book about asterisk?
thank u.
B.R.
John.
-原始邮件-
发件人: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
[EMAIL PROTECTED]
发送时间: 2004年12月24日 7:51
收件人: asterisk-users@lists.digium.com
主题: Asterisk-Users Digest, Vol 5, Issue 350
Send Asterisk-Users mailing list submissions to
asterisk-users@lists.digium.com
To subscribe or unsubscribe via the World Wide Web, visit
http://lists.digium.com/mailman/listinfo/asterisk-users
or, via email, send a message with subject or body 'help' to
[EMAIL PROTECTED]
You can reach the person managing the list at
[EMAIL PROTECTED]
When replying, please edit your Subject line so it is more specific
than "Re: Contents of Asterisk-Users digest..."
Today's Topics:
1. RE: rtp channels not through asterisk (Brian West)
2. Turning "*" Hangup off in queues ([EMAIL PROTECTED])
3. Re: Voicemail email notification (Rich Adamson)
4. Can't Make Outgoing Call (Norman Zhang)
5. Re: Voicemail email notification (Dorn Hetzel)
6. Re: Asterisk in parallel with PSTN [OT] (Rich Adamson)
7. Re: rtp channels not through asterisk (Rich Adamson)
8. Re: Realtime sipbuddies table structure why?
(Greg - Cirelle Enterprises)
9. RE: Polycom Buddies (Paul Hales)
10. Re: Queue - roundrobin member order (Adam Goryachev)
11. Re: Voicemail email notification (Rich Adamson)
12. Re: Can't Make Outgoing Call (Norman Zhang)
13. Re: Recommended IAX softphone. (Bruno Hertz)
14. Re: sip seeding vs registration (Greg - Cirelle Enterprises)
15. Asterisk 1.0.3 no RedHat zaptel script? (Jerry Geis)
16. Re: Recommended IAX softphone. (Erik Espinoza)
--
Message: 1
Date: Thu, 23 Dec 2004 16:51:22 -0600
From: "Brian West" <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] rtp channels not through asterisk
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="US-ASCII"
canreinvite=yes
Aterisk stays in the signaling path so unless you're running tcpdump or the
like you'll never notice this.
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of bijan
> Sent: Thursday, December 23, 2004 4:46 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] rtp channels not through asterisk
>
> In wiki pages it is stated that The audio channels (RTP) may go directly
> from phone to phone or may go through Asterisk's media bridge.
> Currently with my settings, I notice that all rtp's are passing through my
> asterisk. How could I achieve that they go directly from phone to phone?
> I assume this way, my machine will have less load and therefore could
> handle more calls.
>
> regards
> Bijan Karimi
>
--
Message: 2
Date: Thu, 23 Dec 2004 19:16:19 -0600 (CST)
From: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Turning "*" Hangup off in queues
To: asterisk-users@lists.digium.com
Message-ID: <[EMAIL PROTECTED]>
Content-Type: TEXT/PLAIN; charset=US-ASCII
Hi !
Can somebody tell me how to turn the "*" Hangup option utrned off in
queues. I have not used any H option but still as an agent if I press "*"
key the user gets disconnected. Somehow it is turned on by
default. Can I turn this option off In my extensions.conf I have
written :
exten => 8000,3,Queue(supportq|t)
plz help me inthis regard ... Thanks !
Usman.
--
Message: 3
Date: Thu, 23 Dec 2004 16:51:34 -0600
From: Rich Adamson <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Voicemail email notification
To: Asterisk Users Mailing List - Non-Commercial Discussion
Message-ID: <[EMAIL PROTECTED]>
Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1
> Are there any common silent failure modes for email
> notification from the Voicemail module. I put the
> email and pager email addresses in my entry in
> voicemail.conf but no mail gets sent when I leave
> a voicemail. No obvious error messages either,
> unless I'm just not looking in the right place.
>
> Thanks for any clues :)
Nop, that's it other then you have to have sendmail configured
and running on the system (or have a substitute mail handler).
Rich
--
Message: 4
Date: Thu, 23 Dec 2004 14:58:04 -0800
From: Norman Zhang <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] Can't Make Outgoing Call
To: Asterisk Users Mailing List - Non-Commercial Discussion
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hi,
I can't get dial-out working. I'm trying to call 523936. Is there
something wrong with my setup here? Could someone please give me a few
pointers?
Regards,
Norman Zhang
[fwd-out]
exten => _8.,1,SetCallerID(${FWDUSERID})
exten => _8
