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]
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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
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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'"
<asterisk-users@lists.digium.com>
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
<asterisk-users@lists.digium.com>
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
<asterisk-users@lists.digium.com>
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(${FWDUSERID})
exten => _8.,2,SetCIDName(${FWDUSERNAME})
exten => _8.,3,Dial(SIP/[EMAIL PROTECTED],70)
exten => _8.,4,Macro(fastbusy)
exten => _8.,5,Hangup
*CLI> -- Executing SetCallerID("SIP/101-e528", "533990") in new stack
-- Executing SetCIDName("SIP/101-e528", "Norman Zhang") in new stack
-- Executing Dial("SIP/101-e528", "SIP/[EMAIL PROTECTED]|70") in new stack
-- Called [EMAIL PROTECTED]
Dec 23 14:48:23 WARNING[1091111856]: chan_sip.c:683 retrans_pkt: Maximum
retries
exceeded on call [EMAIL PROTECTED] for
seqno 102
(Critical Request)
== No one is available to answer at this time
-- Executing Macro("SIP/101-e528", "fastbusy") in new stack
-- Executing Answer("SIP/101-e528", "") in new stack
-- Executing Wait("SIP/101-e528", "1") in new stack
-- Executing Playback("SIP/101-e528", "ss-noservice") in new stack
-- Playing 'ss-noservice' (language 'en')
-- Executing Wait("SIP/101-e528", "30") in new stack
------------------------------
Message: 5
Date: Thu, 23 Dec 2004 18:02:14 -0500
From: Dorn Hetzel <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Voicemail email notification
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=us-ascii
On Thu, Dec 23, 2004 at 04:51:34PM -0600, Rich Adamson wrote:
> > 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).
>
sendmail is running (well, actually, it's postfix, but it
responds to /usr/sbin/sendmail) ... still no mail gets
sent. is there any way to get * to log what happens when
it tries to call sendmail?
-Dorn
------------------------------
Message: 6
Date: Thu, 23 Dec 2004 16:53:22 -0600
From: Rich Adamson <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Asterisk in parallel with PSTN [OT]
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1
> > >I've got a configuration with PSTN line connected to FXO
> > >on TDM400P ringing through to a phone connected on a
> > >Sipura SPA-3000. The phone *does* ring before the
> > >caller-id is available. In fact, it shoes some
> > >alternate message like "waiting for caller id info"
> > >right after the first ring and then changes to
> > >the real caller-id after the 2nd ring.
> > >
> > >-Dorn
> >
> > I've always wondered if certain IP (regardless of proto) phones could do
> > the same? Basically initiate the call with fake callerid info and then
> > send an 'update' packet later to inform the phone of the new callerid?
> > Is this possible - even if it is only supported on certain phones?
> >
> > If this is possible, then we could modify * to allow the dialplan to
> > (optionally) start before callerid is received and then update the
> > ${CALLERID} variable(s) once the information is available. There are
> > situations where this is VERY desirable (obviously this only applies to
> > POTS though).
> >
> Seems like something similar must be going on in my setup,
> because * is clearly taking the inbound call from the
> TDM400P/FXO and ringing it through to the Sipura FXS port
> before the caller-id info is available.
The zapata.conf entry for the channel will need something like:
immediate=no
usecallerid=yes
If you have an analog phone on that same pstn line, you should notice
that * won't ring the internal sip phones until after the second pstn
ring. The CallerID is simply passed to the sip phone without any
special variables, etc.
------------------------------
Message: 7
Date: Thu, 23 Dec 2004 17:07:06 -0600
From: Rich Adamson <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] rtp channels not through asterisk
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1
> 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 rtps 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.
As bkw pointed out, use canreinvite=yes for each sip phone definition.
But, that will only work if the phones can reach each other directly
(the phones and/or asterisk can't be behind a nat/firewall box).
------------------------------
Message: 8
Date: Thu, 23 Dec 2004 18:11:18 -0500
From: Greg - Cirelle Enterprises <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Realtime sipbuddies table structure
why?????
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="us-ascii"; format=flowed
At 01:21 PM 12/23/04, you wrote:
>Greg - Cirelle Enterprises wrote:
>
>>Read it, makes no difference, it's broken :)
>>Also, it doesn't say why the table structure is the
>>way it is. just poor data modeling.
>
>God, I'm sure everyone on the list must be thinking, "Oh, why oh why
>didn't *Greg* write Asterisk instead of Mark; he seems so very much
>smarter. . . "
>
>B.
Don't claim to be smarter, just pointing out the obvious
greg
------------------------------
Message: 9
Date: Fri, 24 Dec 2004 10:15:17 +1100
From: "Paul Hales" <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] Polycom Buddies
To: "'Asterisk Users Mailing List'" <asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain;charset="us-ascii"
If anyone has a good guide to the buddy function, I would also love to read
it!
Regards,
PaulH
-----Original Message-----
From: Nihal [mailto:[EMAIL PROTECTED]
Sent: Friday, 24 December 2004 6:11 AM
To: Asterisk Users Mailing List
Subject: [Asterisk-Users] Polycom Buddies
I've got two Polycom 500's that I'm playing with, and I want view the status
of either phone, (busy/on the phone/etc.) from the other.
I've got this cute little 'Buddies' button, and I can add contacts to that.
But the status doesnt actually update.
Do I need to setup realtime for asterisk? Can anyone point me to some
documentation or give me some hints?
Thanks,
Nihal
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------------------------------
Message: 10
Date: Fri, 24 Dec 2004 10:21:26 +1100
From: Adam Goryachev <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Queue - roundrobin member order
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain
On Fri, 2004-12-24 at 04:52, Matthew Boehm wrote:
> Whoever was listed first in the list always got the call first. This isn't
> what I was expecting RR to do. I was expecting call #1 to goto agent 1. if
> call 2 comes in and 1 is still on phone it goes to 2. if 1 is not on phone
> it still goes to 2. and then 3, 4 etc..until it loops back around.
>
> but what did happen was that 1 got all the calls. 2 never got calls unless 1
> was on the phone. and 3 never got calls unless both 1 and 2 where on.
>
> i changed it to random so our CSR girls will have something to do. :)
Why not use rrmemory ? Since what you 'expected' from roundrobin is
exactly what rrmemory says it does?
Regards,
Adam
------------------------------
Message: 11
Date: Thu, 23 Dec 2004 17:19:49 -0600
From: Rich Adamson <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Voicemail email notification
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
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).
> >
> sendmail is running (well, actually, it's postfix, but it
> responds to /usr/sbin/sendmail) ... still no mail gets
> sent. is there any way to get * to log what happens when
> it tries to call sendmail?
You should see asterisk logging the email in /var/log/asterisk/messages
like this:
Dec 19 09:07:58 DEBUG[31885]: Sent mail to [EMAIL PROTECTED] with co
mmand '/usr/sbin/sendmail -t'
And, the mail handler (in my case, sendmail) in /var/log/maillog.
Might have to play around with the asterisk debug level to get the
entries (since it says "DEBUG" in the above). Check /etc/asterisk/logger.
Rich
------------------------------
Message: 12
Date: Thu, 23 Dec 2004 15:26:58 -0800
From: Norman Zhang <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Can't Make Outgoing Call
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
> 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?
> [fwd-out]
> exten => _8.,1,SetCallerID(${FWDUSERID})
> exten => _8.,2,SetCIDName(${FWDUSERNAME})
> exten => _8.,3,Dial(SIP/[EMAIL PROTECTED],70)
I found out that I need to replace
exten => _8.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED],70)
May I ask why? context fwd is defined in sip.conf as follows
[fwd]
type=friend
secret=mysecret
username=533990
fromuser=533990
fromdomain=fwd.pulver.com
host=fwd.pulver.com
dtmfmode=inband
nat=yes
canreinvite=no
Regards,
Norman Zhang
------------------------------
Message: 13
Date: Fri, 24 Dec 2004 00:27:28 +0100
From: Bruno Hertz <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Recommended IAX softphone.
To: asterisk-users@lists.digium.com
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain
On Thu, 2004-12-23 at 16:36 -0600, Michael Van Donselaar wrote:
> iaxComm is Open Source, and currently runs on Win32 and i386Linux platforms.
> Earlier versions run on Mac OSX, but I don't have hardware to compile it, and
> have not had any recent reports.
Thanks Michael
I've tried it and it seemed a reasonable choice to me, with it's codec
support, clean gui plus being open source. I guess I'll go for it
then ...
Regards, Bruno.
------------------------------
Message: 14
Date: Thu, 23 Dec 2004 18:43:35 -0500
From: Greg - Cirelle Enterprises <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] sip seeding vs registration
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="us-ascii"; format=flowed
At 03:43 PM 12/23/04, you wrote:
>Oh, I see. This is the realtime connected problem.
>Can't say too much constructive about that without info, I'm not a fan of it.
>
>We need a debug trace of the registration process (SIP trace and *
>messages) to debug why it failed,
>not just a one-line message, and anything after that is useless, as you
>point out.
>
>However, I don't think it has anything do to with loading (or not) all
>your modules, unless you're running out of memory.
The module elimination was to try and rule out memory issues as the machine
is limited to 512MB RAM.
When utilizing the app_realtime:
The CLI interface is consistently issuing these messages.
it has been a slow day with no real phone activity.
-- SIP Seeding '40853' at [EMAIL PROTECTED]:5060 for 3600
-- Saved useragent "Sipura/SPA2000-2.0.10(e)" for peer 40853
-- SIP Seeding '40853' at [EMAIL PROTECTED]:5060 for 3600
-- SIP Seeding '40854' at [EMAIL PROTECTED]:5061 for 3600
-- SIP Seeding '40854' at [EMAIL PROTECTED]:5061 for 3600
-- Saved useragent "Sipura/SPA2000-2.0.10(e)" for peer 40854
-- SIP Seeding '40854' at [EMAIL PROTECTED]:5061 for 3600
-- SIP Seeding '52221' at [EMAIL PROTECTED]:5060 for 3600
-- SIP Seeding '52221' at [EMAIL PROTECTED]:5060 for 3600
Dec 23 16:22:44 NOTICE[12551]: chan_sip.c:7742 handle_request: Registration
from '<sip:[EMAIL PROTECTED]>' failed for '192.168.70.26'
-- SIP Seeding '52221' at [EMAIL PROTECTED]:5060 for 3600
-- Saved useragent "Grandstream BT100 1.0.5.20" for peer 52221
-- SIP Seeding '52221' at [EMAIL PROTECTED]:5060 for 3600
-- SIP Seeding '1002' at [EMAIL PROTECTED]:5060 for 1800
-- SIP Seeding '1002' at [EMAIL PROTECTED]:5060 for 1800
-- Saved useragent "X-Lite release 1103m" for peer 1002
-- SIP Seeding '1002' at [EMAIL PROTECTED]:5060 for 1800
-- SIP Seeding '40852' at [EMAIL PROTECTED]:5060 for 3600
-- SIP Seeding '40852' at [EMAIL PROTECTED]:5060 for 3600
Dec 23 16:48:47 NOTICE[12551]: chan_sip.c:7742 handle_request: Registration
from '<sip:[EMAIL PROTECTED]>' failed for '192.168.70.25'
-- SIP Seeding '40852' at [EMAIL PROTECTED]:5060 for 3600
-- Saved useragent "Grandstream BT100 1.0.5.20" for peer 40852
-- SIP Seeding '40852' at [EMAIL PROTECTED]:5060 for 3600
-- SIP Seeding '1002' at [EMAIL PROTECTED]:5060 for 1800
-- SIP Seeding '1002' at [EMAIL PROTECTED]:5060 for 1800
-- Saved useragent "X-Lite release 1103m" for peer 1002
The /var/log/asterisk/messages file gives
Dec 23 12:24:00 NOTICE[12551]: Registration from '<sip:[EMAIL PROTECTED]>'
failed for '192.168.70.26'
Dec 23 12:50:05 NOTICE[12551]: Registration from '<sip:[EMAIL PROTECTED]>'
failed for '192.168.70.25'
Dec 23 13:23:41 NOTICE[12551]: Registration from '<sip:[EMAIL PROTECTED]>'
failed for '192.168.70.26'
Dec 23 14:23:22 NOTICE[12551]: Registration from '<sip:[EMAIL PROTECTED]>'
failed for '192.168.70.26'
Dec 23 14:49:26 NOTICE[12551]: Registration from '<sip:[EMAIL PROTECTED]>'
failed for '192.168.70.25'
Dec 23 15:23:03 NOTICE[12551]: Registration from '<sip:[EMAIL PROTECTED]>'
failed for '192.168.70.26'
Dec 23 16:22:44 NOTICE[12551]: Registration from '<sip:[EMAIL PROTECTED]>'
failed for '192.168.70.26'
Dec 23 16:48:47 NOTICE[12551]: Registration from '<sip:[EMAIL PROTECTED]>'
failed for '192.168.70.25'
Dec 23 17:22:25 NOTICE[12551]: Registration from '<sip:[EMAIL PROTECTED]>'
failed for '192.168.70.26'
D
Restoring the system to using *.conf files eliminates all of this output
and calls
going directly to voicemail
Unfortunately, I don't have the exact channel cannot be created or ?
messages as there were
non today and are usually seen in the CLI.
Unless I'm mistaken, these general messages indicate a registration
failure, do they not?
When a call comes in and goes directly to voicemail while the extension is
sitting idle
waiting for a call, not busy or off the hook, I think is an issue.
g
------------------------------
Message: 15
Date: Thu, 23 Dec 2004 18:50:54 -0500
From: Jerry Geis <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] Asterisk 1.0.3 no RedHat zaptel script?
To: asterisk-users@lists.digium.com
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=us-ascii; format=flowed
Did you do a "make config" in the zaptel source directory?
THat works for me.
Jerry
------------------------------
Message: 16
Date: Thu, 23 Dec 2004 15:50:57 -0800
From: Erik Espinoza <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Recommended IAX softphone.
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=US-ASCII
I'd recommend Firefly by Virbiage. It's free and works on third party
networks with sip/iax2 support.
On Fri, 24 Dec 2004 00:27:28 +0100, Bruno Hertz <[EMAIL PROTECTED]> wrote:
> On Thu, 2004-12-23 at 16:36 -0600, Michael Van Donselaar wrote:
>
> > iaxComm is Open Source, and currently runs on Win32 and i386Linux platforms.
> > Earlier versions run on Mac OSX, but I don't have hardware to compile it,
> > and
> > have not had any recent reports.
>
> Thanks Michael
>
> I've tried it and it seemed a reasonable choice to me, with it's codec
> support, clean gui plus being open source. I guess I'll go for it
> then ...
>
> Regards, Bruno.
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
------------------------------
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