[Asterisk-Users] Music On Hold Problem

Venu V vvenu at transdyne.co.in
Sun Nov 14 06:12:54 MST 2004 

Recently I have configured Music On Hold option in asterisk PBX. But I
am unable to listen to the audio properly and morever its getting breaks
for every 3 seconds. If any one know about this. Please help me

Thanks & Regards
V.Venu



-----Original Message-----
From: asterisk-users-request at lists.digium.com
[mailto:asterisk-users-request at lists.digium.com] 
Sent: Sunday, November 14, 2004 12:07 PM
To: asterisk-users at lists.digium.com
Subject: Asterisk-Users Digest, Vol 4, Issue 181

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Today's Topics:

   1. RE: SysMaster and GPL Violation (Brian)
   2. Re: getting callerid from spa3k to asterisk (Randy Bush)
   3. my asterisk drops connection when remote side puts	me on
      hold? (Steve Prior)
   4. Cisco ATA and G729 (kido noagbodji)
   5. Remote answer not detected (DB)
   6. Re: SysMaster and GPL Violation (Voip Business)
   7. RE: Cable for T1 connection: Crossover or	straightthrough?
      (Franceen Thompson)
   8. RE: Cisco ATA and G729 (Franceen Thompson)
   9. Queue/AgentCallbackLogin Problems (Franceen Thompson)


----------------------------------------------------------------------

Message: 1
Date: Sat, 13 Nov 2004 19:30:06 -0700
From: "Brian" <lists001 at brianchristie.com>
Subject: RE: [Asterisk-Users] SysMaster and GPL Violation
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
	<asterisk-users at lists.digium.com>
Message-ID: <20041114022609.KODQ347.fed1rmmtao02.cox.net at mica>
Content-Type: text/plain;	charset="us-ascii"

> > Are you saying that those of us that are using the product should
not be
> > allowed to voice our opinions about its licensing, development and
> > maintenance? That we should all just shut up and take whatever Mark
&
> > co. give us? If that's the case, then this is most definitely NOT an
> > open-source project at all.
> >
> -----Original Message-----
> From Brandon Patterson
> Sent: Saturday, November 13, 2004 7:15 PM
> Uh ok...So when will Asterisk be a licensed product? Will it take the
> form of a Redhat sort of platform... Fedora & with Redhat the pay me
money
> side of the house?
>
> Just a simple question: When can we expect to see Asterisk the
licensed
> as in paid for version ?
> 
> 
> Brandon

Right now.

As far as I know, you just need to contact Digium's sales department and
negotiate a licensing agreement with them.



------------------------------

Message: 2
Date: Sat, 13 Nov 2004 19:11:10 -0800
From: Randy Bush <randy at psg.com>
Subject: [Asterisk-Users] Re: getting callerid from spa3k to asterisk
To: splatters <asterisk-users at lists.digium.com>
Message-ID: <16790.52430.393126.59269 at ran.psg.com>
Content-Type: text/plain; charset=us-ascii

> if i have two sip contexts for my spa3k, on inbound and
> one outbound, e.g.
> 
>     [spa3k-in]
>     type=friend
>     host=dynamic
>     port=5061
>     auth=md5
>     secret=pfui
>     qualify=1000
>     canreinvite=yes
>     context=ext-in42
> 
>     [spa3k-out]
>     type=peer
>     auth=md5
>     secret=pfui
>     username=outpass
>     fromuser=outpass
>     host=spa3k.bogus.com
>     port=5061
>     nat=no
>     canreinvite=yes
>     context=ext-in42
> 
> and the spa3k's PSTN / Subscriber Information / User ID: = spack-in,
> 
> the incoming connection from spa3k to * is being routed to the
> spa3k-out context, not the spa3-in context.  see appended.
> 
> i suspect this is a bug in * 1.0.1.

i found the problem, or at least a work-around.

if i reverse the order of the above two sip contexts, the incoming
call is properly routed to the spa3k-in sip context as opposed to
the wrong one, spa3k-out.

my guess is that * is traversing a list and taking the first
context which has the ip address and port it wants without
checking the context name against the name which was received
over the wire.  so it depends on what order the contexts are
inserted in the list.

aiiiiiiiiiiiiiiiiiiiiii!

randy



------------------------------

Message: 3
Date: Sat, 13 Nov 2004 22:33:58 -0500
From: Steve Prior <sprior at geekster.com>
Subject: [Asterisk-Users] my asterisk drops connection when remote
	side puts	me on hold?
To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users at lists.digium.com>
Message-ID: <4196D226.7020208 at geekster.com>
Content-Type: text/plain; charset=us-ascii; format=flowed

I've got a TDM100P card with a fxo and fxs module in the US.  I'm
using kewlstart for all ports.  I've noticed that when I make
a call out from an analog phone out the POTS line that if after
talking to the party I called (in this case the phone company itself)
they put me on hold asterisk disconnects the call immediatly.

I've looked around the web pages, but can't figure out what might be
causing this and how to fix it - can anyone give me a clue?

Thanks
Steve Prior



------------------------------

Message: 4
Date: Sun, 14 Nov 2004 03:36:55 -0000
From: "kido noagbodji" <kido at cafe.tg>
Subject: [Asterisk-Users] Cisco ATA and G729
To: <asterisk-users at lists.digium.com>
Message-ID: <043a01c4c9fb$30e932a0$7c40f850 at DNTKIDO>
Content-Type: text/plain; charset="iso-8859-1"

Hi all,

I am new to asterisk. I was able, but not without pain to install it on
a FreeBSD box. I set up a cisco ATA 186 and the SJlabs softphone to work
with the PBX.
Three remarks:
* On the SJphone, i use the GSM and the G711 (ulaw and alaw) codec. In
the h323.conf file i enabled those codec. Everything works great!!!
* However, when i set my Cisco ATA to G711, i can't hear any sound
unless I press at least two or three keys(any random keys). I am using
the demo context of extension.conf file. Can that be due to a fast start
problem? Anyone knows how to checkthe faststartcapabilities of an ATA
186?
* Also when i set my ATA codec to g729 and in asterisk i allow=g729, i
get a very low weird sound. What is that due to? I am guessing that i
don't have the codec installed on the system. Is there an open source
g729 codec available for FreeBSD?

Any help will be very much appreciated,

Thanks.

Kido
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Message: 5
Date: Sat, 13 Nov 2004 22:45:13 -0500
From: DB <DB at M-and-D.com>
Subject: [Asterisk-Users] Remote answer not detected
To: asterisk-users at lists.digium.com
Message-ID: <4196D4C9.1060600 at M-and-D.com>
Content-Type: text/plain; charset=us-ascii; format=flowed

> 
> Il dom, 2004-11-14 alle 00:13, DB ha scritto:
>> Here's my a section of my simple extensions.conf
> <snip>
>> exten => s,5,Dial(Zap/4/2326932|15)
>> exten => s,6,Voicemail,u100
> <snip>
>> It works, but when the call is routed out on ZAP/4 (at priority 5), 
>> Asterisk seems to not realize the call is answered. After 15 seconds
it 
>> proceeds to voicemail interrupting the call. Can anyone help?
> 
> eh, perhaps with some details about your zap...
> ie what card?
> zaptel.conf?
> zapata.conf?
> 
> matteo, still without divinatory powers


Hi - thanks for the reply - here's that info:

card is TDM22B

zaptel.conf:
==================
fxoks=1-2 # Make sure that the FXS(green) modules are closest
fxsks=3-4 # This is for the FXO module(s) becaus
defaultzone=us
loadzone=us
==================

zapata.conf:
===================
[trunkgroups]
[channels]
switchtype=national
signalling=fxo_ls
rxwink=300              ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
busydetect=yes
callprogress=yes
progzone=us

signalling=fxo_ks
echocancel=yes ; You can set this to 32, 64, or 128, tweak to your
needs.
echocancelwhenbridged=yes
echotraining=400 ; Asterisk trains to the beginning of the call, number 
is in milliseconds
callerid=asreceived
group=1
context=MD_line1 ; Points to the default context of your extensions.conf
channel => 1

signalling=fxo_ks
echocancel=yes ; You can set this to 32, 64, or 128, tweak to your
needs.
echocancelwhenbridged=yes
echotraining=400 ; Asterisk trains to the beginning of the call, number 
is in milliseconds
callerid=asreceived
group=2
context=MD_line2 ; Points to the default context of your extensions.conf
channel => 2

signalling=fxs_ks
group=3
context=incoming_9141252
channel=> 3 ; Again if you only have one FXO module remove the '-4'

signalling=fxs_ks
group=4
context=incoming_3493729
channel=> 4 ; Again if you only have one FXO module remove the '-4'
================================

DB





------------------------------

Message: 6
Date: Sat, 13 Nov 2004 23:03:49 -0600
From: Voip Business <voipbusiness at gmail.com>
Subject: Re: [Asterisk-Users] SysMaster and GPL Violation
To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users at lists.digium.com>
Message-ID: <f8a0e7510411132103ed30018 at mail.gmail.com>
Content-Type: text/plain; charset=US-ASCII

Guys, in fact we will give an Applause to Sysmaster guys that are
doing a great job in their products (world wide sales), this guys are
doing money , and in this point is ,, Mark (and/or Digium is not
receiving  money for that).

But Back to basics of Open source you can sale , modify , distribute
and so on , so on so on,, BUT give the credits of the developer (in
this case that is the only thing thery are not respecting)

Now ,, Guys this is an oportunity to have a back benefit of that ,,
because why I will pay a 150K usd for a NORFA (sysmaster new system)if
for much less I can have an Asterisk up and running,

In my point of view technically Asterisk (as it is right now) is GREAT
lets take advance of that ,, instead of 100's of brains trying to make
asterisk more and more and more features.... GUYS FOR GOD SAINT lets
do it NEAT (GUI, Administration etc)

Why dont Asterisk comunity opens a group for Asterisk Simplification

This is the Opinion of a Non guru fellow ben available for a monthly
donation for that proyect (what we need is to be Several like me to
pay good programers and develop a GUI) and off course with a
"licencing" that for all the donators and contributors the GUI has no
Cost (including Digium) but for others will have a  cost.


AGAIN , this is only my point of view and I respect every coments about
this :)

Regards


Humberto


On Sat, 13 Nov 2004 19:30:06 -0700, Brian <lists001 at brianchristie.com>
wrote:
> 
> 
> > > Are you saying that those of us that are using the product should
not be
> > > allowed to voice our opinions about its licensing, development and
> > > maintenance? That we should all just shut up and take whatever
Mark &
> > > co. give us? If that's the case, then this is most definitely NOT
an
> > > open-source project at all.
> > >
> > -----Original Message-----
> > From Brandon Patterson
> > Sent: Saturday, November 13, 2004 7:15 PM
> > Uh ok...So when will Asterisk be a licensed product? Will it take
the
> > form of a Redhat sort of platform... Fedora & with Redhat the pay me
money
> > side of the house?
> >
> > Just a simple question: When can we expect to see Asterisk the
licensed
> > as in paid for version ?
> >
> >
> > Brandon
> 
> Right now.
> 
> As far as I know, you just need to contact Digium's sales department
and
> negotiate a licensing agreement with them.
> 
> 
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at 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
>


------------------------------

Message: 7
Date: Sat, 13 Nov 2004 23:13:37 -0700
From: "Franceen Thompson" <franceen at thompson.cc>
Subject: RE: [Asterisk-Users] Cable for T1 connection: Crossover or
	straightthrough?
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
	<asterisk-users at lists.digium.com>
Message-ID: <000001c4ca11$1449eb60$6601a8c0 at hickey>
Content-Type: text/plain;	charset="Windows-1252"

You've got a 50/50 shot.

Try the crossover.

http://www.cisco.com/en/US/products/hw/routers/ps233/products_tech_note0
9186a00800a3f09.shtml#topic2

It would be more helpful for you to send your /etc/zaptel.conf file and
/etc/asterisk/Zapata.conf file.

You should have something like the following for your zaptel.conf file:

#zaptel.conf
span=1,1,0,esf,b8zs
loadzone = us
defaultzone=us

also do an

%asterisk -r and send the info from the CLI that is show when you try to
dial something.  It's pretty intuitive output.

Tim.






> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> bounces at lists.digium.com] On Behalf Of Andrew Kohlsmith
> Sent: Saturday, November 13, 2004 6:09 PM
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] Cable for T1 connection: Crossover or
> straightthrough?
> 
> On November 13, 2004 12:11 pm, Malcolm Bader wrote:
> > This is my second asterisk server but the first one with a T100P
card.
> > I connected it to the phone company(SBC) jack but have only a busy
> > signal when calling the T1's number and nothing in the asterisk log
> > files to indicate a connection.
> > Do I need to use a crossover cable?
> 
> Is the T100P's light Green or flashing red?
> 
> Green means it sees the other side, and the other side sees it.  i.e.
the
> cabling is fine.  Orange (well it's trying to be yellow) means that
the
> other
> side can't see the T100P, but the T100P is seeing the other side.
> 
> -A.
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at 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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------------------------------

Message: 8
Date: Sat, 13 Nov 2004 23:17:37 -0700
From: "Franceen Thompson" <franceen at thompson.cc>
Subject: RE: [Asterisk-Users] Cisco ATA and G729
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
	<asterisk-users at lists.digium.com>
Message-ID: <000101c4ca11$a3c2bd80$6601a8c0 at hickey>
Content-Type: text/plain; charset="windows-1252"

I'm not sure about the G711 codec on the ATA, but I know you need to
purchase the g729 from digium.  
 
HYPERLINK
"http://www.digium.com/index.php?menu=asterisk_g729"http://www.digium.co
m/index.php?menu=asterisk_g729
 
pretty inexpensive at $10 each.  That's for "concurrent" connections to
the server. 
 
 
Tim.
 
 
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of kido
noagbodji
Sent: Saturday, November 13, 2004 8:37 PM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Cisco ATA and G729
 
Hi all,
 
I am new to asterisk. I was able, but not without pain to install it on
a FreeBSD box. I set up a cisco ATA 186 and the SJlabs softphone to work
with the PBX.
Three remarks:
* On the SJphone, i use the GSM and the G711 (ulaw and alaw) codec. In
the h323.conf file i enabled those codec. Everything works great!!!
* However, when i set my Cisco ATA to G711, i can't hear any sound
unless I press at least two or three keys(any random keys). I am using
the demo context of extension.conf file. Can that be due to a fast start
problem? Anyone knows how to checkthe faststartcapabilities of an ATA
186?
* Also when i set my ATA codec to g729 and in asterisk i allow=g729, i
get a very low weird sound. What is that due to? I am guessing that i
don't have the codec installed on the system. Is there an open source
g729 codec available for FreeBSD?
 
Any help will be very much appreciated,
 
Thanks.
 
Kido

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Message: 9
Date: Sat, 13 Nov 2004 23:36:59 -0700
From: "Franceen Thompson" <franceen at thompson.cc>
Subject: [Asterisk-Users] Queue/AgentCallbackLogin Problems
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
	<asterisk-users at lists.digium.com>
Message-ID: <000501c4ca14$5c80fb50$6601a8c0 at hickey>
Content-Type: text/plain; charset="windows-1252"

I am having a few problems with my queue.  I am using the
AgentCallbackLogin feature.  When the call comes to the user, it does
not "announce" the call to the agent.  It waits until you enter the "#".
After you hit #. It will play the queue-support announcement to the
agent and tell them to press # if they want the call.  
 
The agent will have to hit # multiple times if they want the call to
come through.
 
Any suggestions?
 
Tim.
 
 
Extenstions.conf
 
exten => 999,1,AgentCallbackLogin(${CALLERIDNUM}|@default)
exten => 999,2,Hangup
 
exten => 1,1,Playback(welcome)
exten => 1,2,SetVar(QUEUE_PRIO=10)
exten => 1,3,Queue(cssupport|t||queue-support|120)
 
Queues.Conf
 
[cssupport]
music = random
announce = queue-support
strategy = rrmemory
context = exitqueue
timeout = 45
retry = 10
wrapuptime=60
maxlen = 2
announce-frequency = 60
announce-holdtime = yes
announce-round-seconds = 10
queue-youarenext = queue-youarenext             
queue-thereare  = queue-thereare                
queue-callswaiting = queue-callswaiting        
queue-holdtime = queue-holdtime                 
queue-minutes = queue-minutes                  
queue-seconds = queue-seconds                   
queue-thankyou = queue-thankyou                 
queue-lessthan = queue-less-than                
joinempty = no
leavewhenempty = yes
member => Agent/309
member => Agent/311
 
CLI>
  -- Executing Playback("Zap/1-1", "welcome") in new stack
    -- Playing 'welcome' (language 'en')
    -- Executing SetVar("Zap/1-1", "QUEUE_PRIO=10") in new stack
    -- Executing Queue("Zap/1-1", "cssupport|t||queue-support|120") in
new stack
    -- Started music on hold, class 'random', on Zap/1-1
    -- Stopped music on hold on Zap/1-1
    -- Playing 'queue-youarenext' (language 'en')
    -- Told Zap/1-1 in cssupport their queue position (which was 1)
    -- Playing 'queue-thankyou' (language 'en')
    -- Started music on hold, class 'random', on Zap/1-1
Nov 13 23:28:26 NOTICE[4125]: app_queue.c:749 wait_for_answer: No one is
answering queue 'cssupport'
Nov 13 23:28:36 NOTICE[4125]: app_queue.c:749 wait_for_answer: No one is
answering queue 'cssupport'
Nov 13 23:28:47 NOTICE[4125]: app_queue.c:749 wait_for_answer: No one is
answering queue 'cssupport'
Nov 13 23:28:57 NOTICE[4125]: app_queue.c:749 wait_for_answer: No one is
answering queue 'cssupport'
Nov 13 23:29:08 NOTICE[4125]: app_queue.c:749 wait_for_answer: No one is
answering queue 'cssupport'

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