[Asterisk-Users] where I can find some learning book about
asterisk?
Mamadou Lamine KA
lamineka at chaka.sn
Fri Dec 24 03:07:06 MST 2004
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" <Zigang.ZHAO at alcatel-sbell.com.cn>
To: <asterisk-users at lists.digium.com>
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: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com]代表
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发送时间: 2004年12月24日 7:51
收件人: asterisk-users at 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 (usman at user.iphonica.net)
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" <brian at bkw.org>
Subject: RE: [Asterisk-Users] rtp channels not through asterisk
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
<asterisk-users at lists.digium.com>
Message-ID: <auto-000005445809 at cgp1.tulsaconnect.com>
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: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> bounces at lists.digium.com] On Behalf Of bijan
> Sent: Thursday, December 23, 2004 4:46 PM
> To: asterisk-users at 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: usman at user.iphonica.net
Subject: [Asterisk-Users] Turning "*" Hangup off in queues
To: asterisk-users at lists.digium.com
Message-ID: <Pine.LNX.4.44.0412231912470.14849-100000 at news.icns.com>
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 <radamson at routers.com>
Subject: Re: [Asterisk-Users] Voicemail email notification
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <Chameleon.1103842356.adar0 at vegas>
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 <norman.zhang at rd.arkonnetworks.com>
Subject: [Asterisk-Users] Can't Make Outgoing Call
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <41CB4D7C.9040007 at rd.arkonnetworks.com>
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/${EXTEN}@fwd,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/8523936 at fwd|70") in new stack
-- Called 8523936 at fwd
Dec 23 14:48:23 WARNING[1091111856]: chan_sip.c:683 retrans_pkt: Maximum
retries
exceeded on call 4e566ea1004c2bb015f3bd8e2b98db61 at fwd.pulver.com 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 <asterisk-users at dorn.hetzel.org>
Subject: Re: [Asterisk-Users] Voicemail email notification
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <20041223230214.GA21192 at lilah.hetzel.org>
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 <radamson at routers.com>
Subject: Re: [Asterisk-Users] Asterisk in parallel with PSTN [OT]
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <Chameleon.1103843111.adar0 at vegas>
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 <radamson at routers.com>
Subject: Re: [Asterisk-Users] rtp channels not through asterisk
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <Chameleon.1103843336.adar0 at vegas>
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 <gcirino at cirelle.com>
Subject: Re: [Asterisk-Users] Realtime sipbuddies table structure
why?????
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <5.1.0.14.0.20041223181038.00a71d90 at pop3.cedata.com>
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" <paulh at adairs.com.au>
Subject: RE: [Asterisk-Users] Polycom Buddies
To: "'Asterisk Users Mailing List'" <asterisk-users at lists.digium.com>
Message-ID: <20041223231141.1066CD4003 at mailbox.adairs.com.au>
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:nihal at claim.md]
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 <mailinglists at websitemanagers.com.au>
Subject: Re: [Asterisk-Users] Queue - roundrobin member order
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <1103844086.9857.7.camel at workhorse>
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 <radamson at routers.com>
Subject: Re: [Asterisk-Users] Voicemail email notification
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <Chameleon.1103844175.adar0 at vegas>
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 2011234567 at yourcarrier.com 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 <norman.zhang at rd.arkonnetworks.com>
Subject: Re: [Asterisk-Users] Can't Make Outgoing Call
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <41CB5442.4040103 at rd.arkonnetworks.com>
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(${FWDUSERNAME})
> exten => _8.,3,Dial(SIP/${EXTEN}@fwd,70)
I found out that I need to replace
exten => _8.,3,Dial(SIP/${EXTEN:1}@fwd.pulver.com,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 <brrhtz at yahoo.de>
Subject: Re: [Asterisk-Users] Recommended IAX softphone.
To: asterisk-users at lists.digium.com
Message-ID: <1103844448.4059.55.camel at caruso.quasi.local>
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 <gcirino at cirelle.com>
Subject: Re: [Asterisk-Users] sip seeding vs registration
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <5.1.0.14.0.20041223181729.00a80da0 at pop3.cedata.com>
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 40853 at 192.168.70.24:5060 for 3600
-- Saved useragent "Sipura/SPA2000-2.0.10(e)" for peer 40853
-- SIP Seeding '40853' at 40853 at 192.168.70.24:5060 for 3600
-- SIP Seeding '40854' at 40854 at 192.168.70.24:5061 for 3600
-- SIP Seeding '40854' at 40854 at 192.168.70.24:5061 for 3600
-- Saved useragent "Sipura/SPA2000-2.0.10(e)" for peer 40854
-- SIP Seeding '40854' at 40854 at 192.168.70.24:5061 for 3600
-- SIP Seeding '52221' at 52221 at 192.168.70.26:5060 for 3600
-- SIP Seeding '52221' at 52221 at 192.168.70.26:5060 for 3600
Dec 23 16:22:44 NOTICE[12551]: chan_sip.c:7742 handle_request: Registration
from '<sip:52221 at 192.168.70.2>' failed for '192.168.70.26'
-- SIP Seeding '52221' at 52221 at 192.168.70.26:5060 for 3600
-- Saved useragent "Grandstream BT100 1.0.5.20" for peer 52221
-- SIP Seeding '52221' at 52221 at 192.168.70.26:5060 for 3600
-- SIP Seeding '1002' at 1002 at 192.168.70.251:5060 for 1800
-- SIP Seeding '1002' at 1002 at 192.168.70.251:5060 for 1800
-- Saved useragent "X-Lite release 1103m" for peer 1002
-- SIP Seeding '1002' at 1002 at 192.168.70.251:5060 for 1800
-- SIP Seeding '40852' at 40852 at 192.168.70.25:5060 for 3600
-- SIP Seeding '40852' at 40852 at 192.168.70.25:5060 for 3600
Dec 23 16:48:47 NOTICE[12551]: chan_sip.c:7742 handle_request: Registration
from '<sip:40852 at 192.168.70.2>' failed for '192.168.70.25'
-- SIP Seeding '40852' at 40852 at 192.168.70.25:5060 for 3600
-- Saved useragent "Grandstream BT100 1.0.5.20" for peer 40852
-- SIP Seeding '40852' at 40852 at 192.168.70.25:5060 for 3600
-- SIP Seeding '1002' at 1002 at 192.168.70.251:5060 for 1800
-- SIP Seeding '1002' at 1002 at 192.168.70.251:5060 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:52221 at 192.168.70.2>'
failed for '192.168.70.26'
Dec 23 12:50:05 NOTICE[12551]: Registration from '<sip:40852 at 192.168.70.2>'
failed for '192.168.70.25'
Dec 23 13:23:41 NOTICE[12551]: Registration from '<sip:52221 at 192.168.70.2>'
failed for '192.168.70.26'
Dec 23 14:23:22 NOTICE[12551]: Registration from '<sip:52221 at 192.168.70.2>'
failed for '192.168.70.26'
Dec 23 14:49:26 NOTICE[12551]: Registration from '<sip:40852 at 192.168.70.2>'
failed for '192.168.70.25'
Dec 23 15:23:03 NOTICE[12551]: Registration from '<sip:52221 at 192.168.70.2>'
failed for '192.168.70.26'
Dec 23 16:22:44 NOTICE[12551]: Registration from '<sip:52221 at 192.168.70.2>'
failed for '192.168.70.26'
Dec 23 16:48:47 NOTICE[12551]: Registration from '<sip:40852 at 192.168.70.2>'
failed for '192.168.70.25'
Dec 23 17:22:25 NOTICE[12551]: Registration from '<sip:52221 at 192.168.70.2>'
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 <geisj at pagestation.com>
Subject: [Asterisk-Users] Asterisk 1.0.3 no RedHat zaptel script?
To: asterisk-users at lists.digium.com
Message-ID: <41CB59DE.2010106 at pagestation.com>
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 <erik.espinoza at gmail.com>
Subject: Re: [Asterisk-Users] Recommended IAX softphone.
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <b86db13f04122315506f5edd49 at mail.gmail.com>
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 <brrhtz at yahoo.de> 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.
>
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