Re: [asterisk-users] Asterisk DIAL() premature timeout on a PRI trunk to legacy PBX

[Vieri]

Thank you,
I did what you mentioned below.
It seems that I'm getting a hangupcause of 0 which I
believe is "not defined".
Is Alcatel the first party that is trying to
disconnect or is it Asterisk? (Not sure how to
interpret the debug info I'm posting below)

Whether it's Alcatel or Asterisk, what could be the
actual cause? (or where should I start looking?)

Thanks

INF-VOIP*CLI> pri debug span 1
Enabled debugging on span 1
-- Executing NoOp("SIP/4053-083189e8", "[ALCATEL
TEST] Start") in new stack
-- Executing Dial("SIP/4053-083189e8",
"Zap/g1/5900") in new stack
1 -- Making new call for cr 32781
-- Requested transfer capability: 0x00 - SPEECH
1 > Protocol Discriminator: Q.931 (8)  len=32
1 > Call Ref: len= 2 (reference 13/0xD) (Originator)
1 > Message type: SETUP (5)
1 > [04 03 80 90 a3]
1 > Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0 
Info transfer capability: Speech (0)
1 >  Ext: 1  Trans
mode/rate: 64kbps, circuit-mode (16)
1 >  Ext: 1  User
information layer 1: A-Law (35)
1 > [18 04 e9 81 83 81]
1 > Channel ID (len= 6) [ Ext: 1  IntID: Explicit, PRI
Spare: 0, Exclusive Dchan: 0
1 >ChanSel: Reserved
1 >   Ext: 1  DS1 Identifier: 1
1 >   Ext: 1  Coding: 0   Number
Specified   Channel Type: 3
1 >   Ext: 1  Channel: 1 ]
1 > [6c 06 21 80 34 30 35 33]
1 > Calling Number (len= 8) [ Ext: 0  TON: National
Number (2)  NPI: ISDN/Telephony Numbering Plan
(E.164/E.163) (1)
1 >   Presentation:
Presentation permitted, user number not screened (0)
'4053' ]
1 > [70 05 a1 35 39 30 30]
1 > Called Number (len= 7) [ Ext: 1  TON: National
Number (2)  NPI: ISDN/Telephony Numbering Plan
(E.164/E.163) (1) '5900' ]
1 > [a1]
1 > Sending Complete (len= 1)
-- Called g1/5900
1 < Protocol Discriminator: Q.931 (8)  len=10
1 < Call Ref: len= 2 (reference 13/0xD) (Terminator)
1 < Message type: CALL PROCEEDING (2)
1 < [18 03 a9 83 81]
1 < Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI
Spare: 0, Exclusive Dchan: 0
1  Protocol Discriminator: Q.931 (8)  len=9
1 > Call Ref: len= 2 (reference 13/0xD) (Originator)
1 > Message type: DISCONNECT (69)
1 > [08 02 81 90]
1 > Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU)
standard (0) 0: 0   Location: Private network serving
the local user (1)
1 >  Ext: 1  Cause: Normal Clearing
(16), class = Normal Event (1) ]
-- Hungup 'Zap/1-1'
  == Everyone is busy/congested at this time (1:1/0/0)
-- Executing NoOp("SIP/4053-083189e8", "[ALCATEL
TEST] hangupcause: 0") in new stack
-- Executing Hangup("SIP/4053-083189e8", "") in
new stack
  == Spawn extension (custom-TEST_ALCATEL, s, 4)
exited non-zero on 'SIP/4053-083189e8'
1 < Protocol Discriminator: Q.931 (8)  len=9
1 < Call Ref: len= 2 (reference 13/0xD) (Terminator)
1 < Message type: RELEASE (77)
1 < [08 02 81 90]
1 < Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU)
standard (0) 0: 0   Location: Private network serving
the local user (1)
1 <  Ext: 1  Cause: Normal Clearing
(16), class = Normal Event (1) ]
1 -- Processing IE 8 (cs0, Cause)
1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate
Null, peerstate Release Request
1 > Protocol Discriminator: Q.931 (8)  len=9
1 > Call Ref: len= 2 (reference 13/0xD) (Originator)
1 > Message type: RELEASE COMPLETE (90)
1 > [08 02 80 90]
1 > Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU)
standard (0) 0: 0   Location: User (0)
1 >  Ext: 1  Cause: Normal Clearing
(16), class = Normal Event (1) ]
1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate
Null, peerstate Null
1 NEW_HANGUP DEBUG: Destroying the call, ourstate
Null, peerstate Null
-- Channel 1/1, span 1 received AOC-E charging 0
units
INF-VOIP*CLI> quit

--- "Eric \"ManxPower\" Wieling" <[EMAIL PROTECTED]>
wrote:

> Looks like the Alcatel is sending back a busy. 
> Check the value of 
> HANGUPCAUSE with a Noop as the priority after the
> Dial.  You may also 
> want to do a pri debug span X to see the actual
> Q.931 ISDN messages that 
> are exchanged.
> 
> Vieri wrote:
> > An Asterisk extension calls an Alcatel extension
> via a
> > PRI link which rings 4 times for about 10-15
> seconds
> > and then drops.
> > So if the Alcatel user doesn't answer within 10-15
> > seconds the call is aborted.
> > (A timeout is *not* specified in the Asterisk Dial
> > command.)
> > It seems however tha

Re: [asterisk-users] DECT SIP phones

[Håkan Källberg]

On Thu, Sep 13, 2007 at 06:05:51PM -0600, Stephen Bosch wrote:
> I'm looking for a SIP DECT (cordless) phone for North American
> installations. I've heard only of the Siemens Gigaset S450/C450 phones.
> Apparently these aren't sold for use in NAm, even though they're
> supposed to be legal (in the United States, anyway).

Hello!

I would reccomend the Kirk DECT gateway. It is SIP capable
and avilable for N America.

We have a setup with the Skinny ( chan_sccp ) protocol and in Sweden,
but I wouldn't expect any problems in NA.

Our customer have used it for a while now.

Regards:Håkan


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[asterisk-users] VOIP Users Conference Friday September 14 @ 12:30PM EDT

[randulo]

Hello all,

Note: This is still an asterisk-based discussion, but we widened the
interests of the conference and also wish to avoid trademark abuse,
although we continue work closely with Digium,hence the change of
name.

Today, we'll be listening to the latest on Astricon which is fast
approaching. Our featured guest  is Daniel Berninger from FWD.
FreeWorldDialup.com has brought a lot of people into the asterisk
fold, including yours truly. Dan can explain what their latest
initiative is about.

http://VOIPUsersConference.org/join.php for more info.

The Talkshoe conference bridge has evolved to the point where you can
call it (whether via SIP or PSTN) and enter the conference code 22622#
and then 1# as a PIN if you do not wish to register. If you have a
PIN, callerID will be recognized if your PIN is your 10 digit phone
number. The text chat interface is also much improved if you use
Windows or Mac, but we will be on the IRC channel at freenode.net
anyway: #asterisk-users-conference

See you anon

randulo

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[asterisk-users] Asterisk voice quality tuning

[satish patel]

Dear all

  I have asterisk 1.4.11 on CentOS. I have SIP IP phone arround 100 
but i got Noice on voice call so what would be the resone and how to fine tune 
my voice quality on asterisk ?? what codec would be best for my asterisk





   
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Re: [asterisk-users] CallWithUs Service?

[John Meksavan]

Ira and Doug,

 Thanks for your inputs.  It seems like there are so many mixed reviews on 
every sip provider.  In the past, I have used Broadvoice, Vitelity, and 
Teliax.  All three have all the same problems- call quality and DTMF Tones.  
Some days, it would work perfectly fine, while on other days you wished you 
never went the sip route.


 There has to be some reasonable priced sip provider that would perform 
just as well as AT&T.  Does it exist?


-John



From: Ira <[EMAIL PROTECTED]>
Reply-To: Asterisk Users Mailing List - Non-Commercial 
Discussion
To: Asterisk Users Mailing List - Non-Commercial 
Discussion

Subject: Re: [asterisk-users] CallWithUs Service?
Date: Thu, 13 Sep 2007 14:44:08 -0700

At 12:32 PM 9/13/2007, you wrote:
>  I am thinking about selecting CALLWITHUS as my sip provider.  Has
> anybody ever used them?  How was the call quality?  DTMF Tones issues?

I use them as a a backup and they seem fine.

Ira


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Re: [asterisk-users] Fwd: Bad FCS error

[Russell Bryant]

Jan Prunk wrote:
> I am getting the following error in asterisk logs:
> 
> Sep 11 17:33:30 NOTICE[2187] chan_zap.c: PRI got event: HDLC Bad FCS (8)
> on Primary D-channel of span 1

Please contact Digium technical support about this issue.  This is something
that they are more than equipped to help you with.  Digium hardware comes with
free support, so don't hesitate to take advantage of it!  [EMAIL PROTECTED]

-- 
Russell Bryant
Software Engineer
Digium, Inc.

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Re: [asterisk-users] Include zaptel in kernel...

[Russell Bryant]

Robert La Ferla wrote:
> Is there any plan to include the zaptel drivers into the main Linux  
> kernel?  If not, there should be one.

If only it were that simple!

Yes, it has been discussed multiple times.  It is definitely something that we
see the benefit in doing and would like to do at some point in the future.
However, there is absolutely no time frame set for it right now.

-- 
Russell Bryant
Software Engineer
Digium, Inc.

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Re: [asterisk-users] DECT SIP phones

[Al lists]

Looks good!
i need to find a distributer to buy one.


On 9/13/07, Stephen Bosch <[EMAIL PROTECTED]> wrote:
>
> Anthony Francis wrote:
> > Aastra now makes a full SIP DECT system with cell style seamless hand
> > off from access point to access point.
> >
> > Caveat: This does not use standard wireless access points, you must
> > purchase their access points and handsets.
>
> That's okay, it's a DECT phone. It's not supposed to use standard
> wireless access points.
>
> I'll look into it.
>
> -Stephen-
>
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[asterisk-users] bug in 1.2.24

[Isaac Xiao]

Here is our dial plan. You need to avoid double recording as well when
you transfer the call to other extension.
exten => 7141,3,Set(CALLFILENAME=q${EXTEN}-${TIMESTAMP}-${UNIQUEID})
exten => 7141,4,Set(__FROM-EXT-QUEUES=ext-queues)
exten => 7141,5,MixMonitor(${CALLFILENAME}.gsm|b)
exten => 7141,6,Playback(custom/None)
exten => 7141,7,Queue(7141|t|||7200)

Here is the CLI log. 
  -- Executing Playback("Zap/9-1", "monitoring") in new stack
-- Playing 'monitoring' (language 'md')
-- Executing Playback("Zap/9-1", "press-1-to-msg") in new stack
-- Playing 'press-1-to-msg' (language 'md')
-- Executing Goto("Zap/9-1", "ext-queues|7141|1") in new stack
-- Goto (ext-queues,7141,1)
-- Executing NoOp("Zap/9-1", "do not answer call before entering
queue") in new stack
-- Executing SetCIDName("Zap/9-1", "CN") in new stack
-- Executing Set("Zap/9-1",
"CALLFILENAME=q7141-20070914-132445-1189740177.10324") in new stack
-- Executing Set("Zap/9-1", "__FROM-EXT-QUEUES=ext-queues") in new
stack
-- Executing MixMonitor("Zap/9-1",
"q7141-20070914-132445-1189740177.10324.gsm|b") in new stack
-- Executing Playback("Zap/9-1", "custom/None") in new stack
-- Executing Queue("Zap/9-1", "7141|t|||7200") in new stack

So Yes. As long as Zap/9-1 channel (customer's channel) not hangs up, it
will be always recorded.

Isaac


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Re: [asterisk-users] CallWithUs Service?

[Doug Crompton]

John,

 I have used callwithus for almost 9 months. I left Gizmo when they had a
doubling of their rates. I probably would have put up with the doubling
but the fact that you can set your callerid at callwithus (and not Gizmo)
was a big selling point. I have kept a minimum Verizon analog line for
local and 911 dialing and I wanted my announced callerid to be my verizon
number. With callwithus this is easy. I created a perl script to route the
calls wither local to verizon or all else to callwithus.

I also opted for the higher quality option. This is something like 1.5
rather than 1 cent/minute. This is not advertised but if you email them
they will bump it up for you. I was not happy, at times, with the call
quality at the 1 cent rate, although this may differ for you.

I connect via iax, but you can also use sip.

As far as I am concerned this is a no frills service that has worked for
me.

Doug

On Thu, 13 Sep 2007, John Meksavan wrote:

> Asterisk Users,
>
>   I am thinking about selecting CALLWITHUS as my sip provider.  Has anybody
> ever used them?  How was the call quality?  DTMF Tones issues?
>
>   Thanks in advance.
>
>
> -John
>
> _
> Gear up for Halo? 3 with free downloads and an exclusive offer.
> http://gethalo3gear.com?ocid=SeptemberWLHalo3_MSNHMTxt_1
>
>
>


"Those that sacrifice essential liberty to obtain a little temporary safety
 deserve neither liberty nor safety."  -- Ben Franklin (1759)


*  Doug Crompton   *
*  Richboro, PA 18954  *
*  215-431-6307*
*  *
* [EMAIL PROTECTED]*
* http://www.crompton.com  *




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Re: [asterisk-users] FreePBX (2.3) - Good? Bad? Ugly?

[shadowym]

FreePBX is a beautiful thing but nothing will prevent the inevitable train
wreck if your hardware is garbage.  Not saying yours is but..just sayin.

-Original Message-
From: Jay R. Ashworth [mailto:[EMAIL PROTECTED] 
Sent: Thursday, September 13, 2007 1:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] FreePBX (2.3) - Good? Bad? Ugly?

I'm about to (finally) do my first Asterisk install; SMB, 4 FXO, 4-6
stations, mostly IP (I'm looking at the Grandstream 201, to start), and
maybe X-lite on a couple of laptops via VPN.

We've got a 4xFXO box we bought off eBay, which unfortunately I can't
find to quote a model number off of, but I *think* it's a Grandstream
as well.

I've looked at several of the packages that turn Asterisk from a PBX
construction kit into an *actual* PBX, and so far FreePBX looks like
the one that matches my mental model of a small phone system best.

Anyone have any first hand experiences with it that they'd like to
share?

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink
[EMAIL PROTECTED]
Designer The Things I Think   RFC
2100
Ashworth & Associates http://baylink.pitas.com '87
e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647
1274




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Re: [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)

[shadowym]

Yes thank you for reminding me it is open source.  Thank you for reminding
me people can write their own code for it.

I'll get right on rewriting the entire sip code.  Should only take me a few
hours.  Including a couple hours to learn how to write c code.  How hard can
it be!

-Original Message-
From: Matthew Fredrickson [mailto:[EMAIL PROTECTED] 
Sent: Thursday, September 13, 2007 1:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)

shadowym wrote:
> Maybe his comments were taken out of context as they don't have the whole
> interview posted.  Why is he talking about queue games,  Biologicall and
> other extremely niche crap when there are huge holes in the basic offering
> (SLA and SCA)?

Considering it is an open source project, anybody that has access to the 
source code (i.e. everybody) can work on whatever they want to, whether 
it be SLA, SCA, or queue games for the more light hearted.

Matthew Fredrickson

> 
>  
> 
> From: Al lists [mailto:[EMAIL PROTECTED] 
> Sent: Tuesday, September 11, 2007 8:28 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)
> 
>  
> 
> I liked the queue game concept!
> although it could be cruel!
> 
> 
> 
> On 9/11/07, Steve Totaro <[EMAIL PROTECTED]
>  > wrote:
> 
>
http://vonmag.com/editorial/web-exclusives/mark-spencer-digium-is-growing-up
> 
> Seems the Adtran relationship goes way back...
> 
> Thanks,
> Steve Totaro
> 
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> 
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-- 
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.




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Re: [asterisk-users] DECT SIP phones

[Stephen Bosch]

Anthony Francis wrote:
> Aastra now makes a full SIP DECT system with cell style seamless hand 
> off from access point to access point.
> 
> Caveat: This does not use standard wireless access points, you must 
> purchase their access points and handsets.

That's okay, it's a DECT phone. It's not supposed to use standard
wireless access points.

I'll look into it.

-Stephen-

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Re: [asterisk-users] DECT SIP phones

[Anthony Francis]

Al lists wrote:
> I'm using Linksys Wip300 and i'm not happy with it.
>
>
> On 9/13/07, *Dave Walker* <[EMAIL PROTECTED] 
> > wrote:
>
> On Thu, 2007-09-13 at 18:05 -0600, Stephen Bosch wrote:
> > Hi folks:
> >
> > I know it's come up a few times before, but I need some more detail.
> >
> > I'm looking for a SIP DECT (cordless) phone for North American
> > installations. I've heard only of the Siemens Gigaset S450/C450
> phones.
> > Apparently these aren't sold for use in NAm, even though they're
> > supposed to be legal (in the United States, anyway).
> >
> > On top of that, I understand they have some annoying issues anyway.
> >
>
> S450:
> A recent firmware (few days old) upgrade seems to have solved the
> issue
> of being able to transfer calls.  The handset still does not support
> 'Message Waiting Indicator", but does show missed calls.
>
> I am using this model, the audio IMO is superb and would recommend it.
>
> Failing that, there is the Aastra 480i-CT, (which is designed for
> the US
> market), but this includes a normal deskphone.  If this as good as
> the
> other Aastra products, then you can't go too far wrong.
>
> Kind Regards,
> Dave Walker
>
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Aastra now makes a full SIP DECT system with cell style seamless hand 
off from access point to access point.

Caveat: This does not use standard wireless access points, you must 
purchase their access points and handsets.

Anthony

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Re: [asterisk-users] DECT SIP phones

[Stephen Bosch]

Dave Walker wrote:
> On Thu, 2007-09-13 at 18:05 -0600, Stephen Bosch wrote:
>> Hi folks:
>>
>> I know it's come up a few times before, but I need some more detail.
>>
>> I'm looking for a SIP DECT (cordless) phone for North American
>> installations. I've heard only of the Siemens Gigaset S450/C450 phones.
>> Apparently these aren't sold for use in NAm, even though they're
>> supposed to be legal (in the United States, anyway).
>>
>> On top of that, I understand they have some annoying issues anyway.
>>
> 
> S450:
> A recent firmware (few days old) upgrade seems to have solved the issue
> of being able to transfer calls.  The handset still does not support
> 'Message Waiting Indicator", but does show missed calls.

Yeah, that message transfer issue was my primary concern. I'm glad it's
been corrected.

I could probably live without the MWI.

Where did you get yours, and where are you located?

Is anybody using these phones in North America?

> I am using this model, the audio IMO is superb and would recommend it.

Siemens phones (German phones in general ;) ) have a reputation for very
clear sound.

I'm also interested in the Openstage phones.

> Failing that, there is the Aastra 480i-CT, (which is designed for the US
> market), but this includes a normal deskphone.  If this as good as the
> other Aastra products, then you can't go too far wrong.

I'll be doing my first Aastra deployment shortly. Everybody I know who's
used them has been very pleased.

If the Aastra phone is true DECT then it should be possible to order
just the handsets for it.

-Stephen-

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Re: [asterisk-users] Agent Callback Login in 1.4

[Kevin P. Fleming]

Paul Hales wrote:

> I stand corrected - when I keyed AddQueueMember onto our in-house
> production server (1.2.23) I did not see that option. But on my test
> environment on my laptop (1.4.10) it's there looking back at me.

That is correct, we added it in 1.4 specifically because it was
necessary to allow AddQueueMember to be used to replace
AgentCallbackLogin :-)

-- 
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)

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[asterisk-users] Include zaptel in kernel...

[Robert La Ferla]

Is there any plan to include the zaptel drivers into the main Linux  
kernel?  If not, there should be one.


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Re: [asterisk-users] DECT SIP phones

[Al lists]

I'm using Linksys Wip300 and i'm not happy with it.


On 9/13/07, Dave Walker <[EMAIL PROTECTED]> wrote:
>
> On Thu, 2007-09-13 at 18:05 -0600, Stephen Bosch wrote:
> > Hi folks:
> >
> > I know it's come up a few times before, but I need some more detail.
> >
> > I'm looking for a SIP DECT (cordless) phone for North American
> > installations. I've heard only of the Siemens Gigaset S450/C450 phones.
> > Apparently these aren't sold for use in NAm, even though they're
> > supposed to be legal (in the United States, anyway).
> >
> > On top of that, I understand they have some annoying issues anyway.
> >
>
> S450:
> A recent firmware (few days old) upgrade seems to have solved the issue
> of being able to transfer calls.  The handset still does not support
> 'Message Waiting Indicator", but does show missed calls.
>
> I am using this model, the audio IMO is superb and would recommend it.
>
> Failing that, there is the Aastra 480i-CT, (which is designed for the US
> market), but this includes a normal deskphone.  If this as good as the
> other Aastra products, then you can't go too far wrong.
>
> Kind Regards,
> Dave Walker
>
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Re: [asterisk-users] DECT SIP phones

[Dave Walker]

On Thu, 2007-09-13 at 18:05 -0600, Stephen Bosch wrote:
> Hi folks:
> 
> I know it's come up a few times before, but I need some more detail.
> 
> I'm looking for a SIP DECT (cordless) phone for North American
> installations. I've heard only of the Siemens Gigaset S450/C450 phones.
> Apparently these aren't sold for use in NAm, even though they're
> supposed to be legal (in the United States, anyway).
> 
> On top of that, I understand they have some annoying issues anyway.
> 

S450:
A recent firmware (few days old) upgrade seems to have solved the issue
of being able to transfer calls.  The handset still does not support
'Message Waiting Indicator", but does show missed calls.

I am using this model, the audio IMO is superb and would recommend it.

Failing that, there is the Aastra 480i-CT, (which is designed for the US
market), but this includes a normal deskphone.  If this as good as the
other Aastra products, then you can't go too far wrong.

Kind Regards,
Dave Walker


signature.asc
Description: This is a digitally signed message part
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Re: [asterisk-users] Agent Callback Login in 1.4

[Paul Hales]

On Thu, 2007-09-13 at 12:01 -0700, Kevin P. Fleming wrote:
> Paul Hales wrote:
> 
> > It means that you end up reporting on SIP extensions, rather than agents
> > which doesn't work so well for call centres where people change desks or
> > have different staff using phones between shifts.
> 
> This has already been discussed multiple times on this list, and has not
> been true since revision 43316 on Sept. 20 of 2006 (i.e. since before
> Asterisk 1.4.0 was released).
> 
> AddQueueMember has a 'membername' argument that can be used to provide a
> logical member name for any interface you add as a queue member, and if
> provided this member name is used in the queue_log instead of the actual
> interface name.
> 

I stand corrected - when I keyed AddQueueMember onto our in-house
production server (1.2.23) I did not see that option. But on my test
environment on my laptop (1.4.10) it's there looking back at me.

Kind regards,

Paul Hales
AsteriskIT



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Re: [asterisk-users] DECT SIP phones

[Andrew Latham]

Google: Kirk Telecom

On 9/13/07, Stephen Bosch <[EMAIL PROTECTED]> wrote:
> Hi folks:
>
> I know it's come up a few times before, but I need some more detail.
>
> I'm looking for a SIP DECT (cordless) phone for North American
> installations. I've heard only of the Siemens Gigaset S450/C450 phones.
> Apparently these aren't sold for use in NAm, even though they're
> supposed to be legal (in the United States, anyway).
>
> On top of that, I understand they have some annoying issues anyway.
>
> Can anyone suggest a solid alternative DECT SIP phone that is available
> in North America?
>
> Cheers,
>
> -Stephen-
>
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-- 
/*
 Andrew Latham
 LATHAMA (lay-th-ham-eh)
 [EMAIL PROTECTED]
 [EMAIL PROTECTED]
*/

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Re: [asterisk-users] FreePBX (2.3) - Good? Bad? Ugly?

[Duncan Turnbull]

I am yet to use 2.3 but have 2.2 on 8 ubuntu based installations with Asterisk 
1.2.18 or greater

FreePbx is really useful as an interface to all the config files, stats etc, 
its also really great if your customers need some
control

The documentation has recently been updated and there is a lot of life in the 
project so I would recommend it

Just note, that like everything you still need to put some time into 
understanding what you are doing and how to get around the
systems

Cheers Duncan

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay R. Ashworth
Sent: Friday, 14 September 2007 8:57 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] FreePBX (2.3) - Good? Bad? Ugly?

On Thu, Sep 13, 2007 at 04:32:27PM -0400, Jay R. Ashworth wrote:
> I'm about to (finally) do my first Asterisk install; SMB, 4 FXO, 4-6
> stations, mostly IP (I'm looking at the Grandstream 201, to start), and
> maybe X-lite on a couple of laptops via VPN.
> 
> We've got a 4xFXO box we bought off eBay, which unfortunately I can't
> find to quote a model number off of, but I *think* it's a Grandstream
> as well.
> 
> I've looked at several of the packages that turn Asterisk from a PBX
> construction kit into an *actual* PBX, and so far FreePBX looks like
> the one that matches my mental model of a small phone system best.
> 
> Anyone have any first hand experiences with it that they'd like to
> share?

And I inadvertantly thread-jacked someone.  Sorry.  Fixed.

Cheers
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth & Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

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[asterisk-users] DECT SIP phones

[Stephen Bosch]

Hi folks:

I know it's come up a few times before, but I need some more detail.

I'm looking for a SIP DECT (cordless) phone for North American
installations. I've heard only of the Siemens Gigaset S450/C450 phones.
Apparently these aren't sold for use in NAm, even though they're
supposed to be legal (in the United States, anyway).

On top of that, I understand they have some annoying issues anyway.

Can anyone suggest a solid alternative DECT SIP phone that is available
in North America?

Cheers,

-Stephen-

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Re: [asterisk-users] Generating an old-fashioned dialtone

[Stephen Bosch]

Phil Reynolds wrote:
> On Wed, Sep 12, 2007 at 11:23:51AM -0600, Stephen Bosch wrote:
>> It's been years since I was in the UK. I can't remember what the modern
>> dial tone sounds like. When did it change?
> 
> The first version of it appeared in parts of Sutton Coldfield in 1976,
> but some places still had the old tone into the 1990s. The modern one is
> of a slightly higher pitch than the 1976 version. Much of Europe uses a
> similar tone. The "secondary" dial tone in France (that followed use of
> 19 when that was the International prefix) was quite similar too.

The German dialtone is a single frequency with no beat, which would
sound very different from the aforementioned tone only with a higher
pitch...

-Stephen-

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Re: [asterisk-users] Digium Appliance

[Kevin P. Fleming]

Noah Miller wrote:

> From what I've heard from Digium employees, you can treat it like a
> normal linux box (it's not locked in any way), so you should be able
> to do whatever you want with it.  I think the distro is very stripped
> down, though, so it may not have all the programs and utilities you
> might find in Debian or Fedora.

It runs uClinux; it has nothing except what is required for operation,
so it is nearly totally different from a traditional Linux distribution
running on an x86 platform.

As far as configuration editing, the Asterisk GUI has a tab that allows
editing any configuration file on the system, but of course there are
changes you could make that would break the GUI, so in general we tell
customers not to do that, since the box is intended to be a simple small
business PBX, not a general purpose Asterisk platform.

-- 
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)

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Re: [asterisk-users] Agent Callback Login in 1.4

[Anthony Francis]

Kevin P. Fleming wrote:
> Anthony Francis wrote:
>
>   
>> Right I am just saying that I can't use AEL in the DB.
>> My dialplan is in the DB, not the agents.
>> 
>
> It shouldn't be that hard to translate the AEL example into traditional
> dialplan language; in fact, Asterisk does that itself when you load the
> AEL into memory, so if you load it yourself and then do a 'dialplan
> show' you'll see the translated version, which you can then copy into
> your database.
>
>   

Ok, I don't ever use AEL. Therefore I would not have known that.
Thank you.

-- 
Thank you and have a wonderful day,

Anthony Francis
Rockynet VOIP
(303) 444-7052 opt 2
[EMAIL PROTECTED]


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Re: [asterisk-users] CallWithUs Service?

[Ira]

At 12:32 PM 9/13/2007, you wrote:
>  I am thinking about selecting CALLWITHUS as my sip provider.  Has 
> anybody ever used them?  How was the call quality?  DTMF Tones issues?

I use them as a a backup and they seem fine.

Ira 


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Re: [asterisk-users] TDM400P

[Anthony Francis]

exten => 1,1,Answer
exten =>1,2,Wait(1)

The wait in the dialplan before sending audio helps get rid of that problem.

Gustavo Gonzalez wrote:
>
> Hi all!  I have an issue with TDM400P FXO card. When a call enter into 
> my IVR and select the proper option, the person that ansswer the call 
> say your "thanks for contact us ..." but the caller cant hear this 
> words because a delay between asterisk and caller part or between 
> asterisk and the ATA device. What is the item on zapata.conf that can 
> affect this delays. Thanks for any help
>
>  
>
> Alejandro González
>
>  
>
> 
>
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-- 
Thank you and have a wonderful day,

Anthony Francis
Rockynet VOIP
(303) 444-7052 opt 2
[EMAIL PROTECTED]


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Re: [asterisk-users] Agent Callback Login in 1.4

[Kevin P. Fleming]

Anthony Francis wrote:

> Right I am just saying that I can't use AEL in the DB.
> My dialplan is in the DB, not the agents.

It shouldn't be that hard to translate the AEL example into traditional
dialplan language; in fact, Asterisk does that itself when you load the
AEL into memory, so if you load it yourself and then do a 'dialplan
show' you'll see the translated version, which you can then copy into
your database.

-- 
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)

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Re: [asterisk-users] Digium Appliance

[Noah Miller]

> Now I'm confused.   Laura from Digium said the AA50 was and I quote "not
> user editable when it comes to the config files".  Which is it?   Can I edit
> the config files, or not?
>
> Particularly of concern/interest to me is.. can I put my aastra phone config
> files on the flash and access them from the web?

>From what I've heard from Digium employees, you can treat it like a
normal linux box (it's not locked in any way), so you should be able
to do whatever you want with it.  I think the distro is very stripped
down, though, so it may not have all the programs and utilities you
might find in Debian or Fedora.

- Noah

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Re: [asterisk-users] Agent Callback Login in 1.4

[Anthony Francis]

Kevin P. Fleming wrote:
> Anthony Francis wrote:
>
>   
>> So the most helpful thing would be a solid example of how to exactly 
>> duplicate the agent callback login behavior in a real-time friendly 
>> manner. The part I am missing is how we are to do authentication.
>> 
>
> Please define what you mean by Realtime friendly. Do you mean the agents
> you had defined using AgentCallbackLogin() were in a Realtime-managed
> database and not in agents.conf? If so, then the AEL example for
> replacing AgentCallbackLogin would need to use the existing dialplan
> applications/functions for querying the Realtime database to determine
> if the agent number being supplied is a valid agent, and extract what
> their password is.
>
>   
Right I am just saying that I can't use AEL in the DB.
My dialplan is in the DB, not the agents.

-- 
Thank you and have a wonderful day,

Anthony Francis
Rockynet VOIP
(303) 444-7052 opt 2
[EMAIL PROTECTED]


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Re: [asterisk-users] Voicemail in 1.4?

[Tilghman Lesher]

On Thursday 13 September 2007, Ken D'Ambrosio wrote:
> I got dragged away from Asterisk (somebody made me an offer I
> couldn't refuse for system administration), but I'm thinking about
> seeing if I can't get it deployed at my new employer.  Regardless,
> there are two things about older voicemail that used to annoy me:
>
> - Dial by name.  Has anyone made it so it can be first or last?

We're waiting on some unrelated translations to come through, but that
patch will wind up in trunk when the translations are ready.

> - Jump to voicemail; you used to have to actually dial the voicemail,
> whereas most voicemail systems allow you to go to your mailbox when
> you hear your voice prompt.  Any chance this has been rectified?

You can optionally specify an 'a' extension in the same context as
Voicemail, and when you press '*', it will go to that extension if it
exists.  Ditto for 'o' and '0'.

-- 
Tilghman

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Re: [asterisk-users] Voicemail in 1.4?

[Doug Lytle]

Ken D'Ambrosio wrote:
> - Dial by name.  Has anyone made it so it can be first or last?
>   
Yes

> - Jump to voicemail; you used to have to actually dial the voicemail,
> whereas most voicemail systems allow you to go to your mailbox when you
> hear your voice prompt.  Any chance this has been rectified?
>   

This can be programmed as well.

Doug


-- 
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."



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Re: [asterisk-users] Voicemail in 1.4?

[James FitzGibbon]

On 9/13/07, Ken D'Ambrosio <[EMAIL PROTECTED]> wrote:

> I got dragged away from Asterisk (somebody made me an offer I couldn't
> refuse for system administration), but I'm thinking about seeing if I
> can't get it deployed at my new employer.  Regardless, there are two
> things about older voicemail that used to annoy me:
>
> - Dial by name.  Has anyone made it so it can be first or last?


Yes, Directory() has a switch to make it search by first name.  You still
need to choose one or the other, or use the dialplan to ask the user whether
they want to search by first or last before calling Directory with or
without the switch, but it works.

- Jump to voicemail; you used to have to actually dial the voicemail,
> whereas most voicemail systems allow you to go to your mailbox when you
> hear your voice prompt.  Any chance this has been rectified?


Look at the 'a' extension, which will be executed if you hit '*' while
listening to the outgoing message.

-- 
j.
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[asterisk-users] Voicemail in 1.4?

[Ken D'Ambrosio]

I got dragged away from Asterisk (somebody made me an offer I couldn't
refuse for system administration), but I'm thinking about seeing if I
can't get it deployed at my new employer.  Regardless, there are two
things about older voicemail that used to annoy me:

- Dial by name.  Has anyone made it so it can be first or last?
- Jump to voicemail; you used to have to actually dial the voicemail,
whereas most voicemail systems allow you to go to your mailbox when you
hear your voice prompt.  Any chance this has been rectified?

Thanks,

-Ken


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[asterisk-users] TDM400P

[Gustavo Gonzalez]

Hi all!  I have an issue with TDM400P FXO card. When a call enter into my
IVR and select the proper option, the person that ansswer the call say your
"thanks for contact us ..." but the caller cant hear this words because a
delay between asterisk and caller part or between asterisk and the ATA
device. What is the item on zapata.conf that can affect this delays. Thanks
for any help 

 

Alejandro González



 

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Re: [asterisk-users] Agent Callback Login in 1.4

[Kevin P. Fleming]

Anthony Francis wrote:

> So the most helpful thing would be a solid example of how to exactly 
> duplicate the agent callback login behavior in a real-time friendly 
> manner. The part I am missing is how we are to do authentication.

Please define what you mean by Realtime friendly. Do you mean the agents
you had defined using AgentCallbackLogin() were in a Realtime-managed
database and not in agents.conf? If so, then the AEL example for
replacing AgentCallbackLogin would need to use the existing dialplan
applications/functions for querying the Realtime database to determine
if the agent number being supplied is a valid agent, and extract what
their password is.

-- 
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)

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[asterisk-users] FreePBX (2.3) - Good? Bad? Ugly?

[Jay R. Ashworth]

On Thu, Sep 13, 2007 at 04:32:27PM -0400, Jay R. Ashworth wrote:
> I'm about to (finally) do my first Asterisk install; SMB, 4 FXO, 4-6
> stations, mostly IP (I'm looking at the Grandstream 201, to start), and
> maybe X-lite on a couple of laptops via VPN.
> 
> We've got a 4xFXO box we bought off eBay, which unfortunately I can't
> find to quote a model number off of, but I *think* it's a Grandstream
> as well.
> 
> I've looked at several of the packages that turn Asterisk from a PBX
> construction kit into an *actual* PBX, and so far FreePBX looks like
> the one that matches my mental model of a small phone system best.
> 
> Anyone have any first hand experiences with it that they'd like to
> share?

And I inadvertantly thread-jacked someone.  Sorry.  Fixed.

Cheers
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth & Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

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[asterisk-users] FreePBX (2.3) - Good? Bad? Ugly?

[Jay R. Ashworth]

I'm about to (finally) do my first Asterisk install; SMB, 4 FXO, 4-6
stations, mostly IP (I'm looking at the Grandstream 201, to start), and
maybe X-lite on a couple of laptops via VPN.

We've got a 4xFXO box we bought off eBay, which unfortunately I can't
find to quote a model number off of, but I *think* it's a Grandstream
as well.

I've looked at several of the packages that turn Asterisk from a PBX
construction kit into an *actual* PBX, and so far FreePBX looks like
the one that matches my mental model of a small phone system best.

Anyone have any first hand experiences with it that they'd like to
share?

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth & Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

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Re: [asterisk-users] TE405P intermittent yellow alarm

[Matthew Fredrickson]

Richard van der Hoff wrote:
> Steve Totaro wrote:
>> Richard van der Hoff wrote:
>  >> [intermittent yellow alarm]
>>> At this point, I'd really like to know what a yellow alarm actually
>>> means. I've read that it indicates that that the other end of the E1 is
>>> in an alarm condition: however BT's terminating unit seems quite happy
>>> with no alarm conditions at all.
>>>
>> Check your cabling.  Replace it with new stuff.  Re-punch everything. 
>>
>> It is obviously somewhere in the line.  If the above does not fix it, 
>> maybe you can get a lucky and get a good tech out that will stick around 
>> to see the issue.
> 
> The only bit of cable I own here is the 2m length of cat-5 between the 
> te405P and BT's line terminating unit. And yes, I've replaced that about 
> 5 times now...
> 
> Thanks for your help, but again I'd like to ask: what does a yellow 
> alarm actually mean? From the driver source code I can see it is set 
> when the FRS0 register has bit 4 set - but that doesn't help a lot...

A yellow alarm means that the other end is seeing loss of signal 
(detected a red alarm from its perspective).  When it detects LOS, it 
transmits yellow alarm to notify the other end.

-- 
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.

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Re: [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)

[Matthew Fredrickson]

shadowym wrote:
> Maybe his comments were taken out of context as they don't have the whole
> interview posted.  Why is he talking about queue games,  Biologicall and
> other extremely niche crap when there are huge holes in the basic offering
> (SLA and SCA)?

Considering it is an open source project, anybody that has access to the 
source code (i.e. everybody) can work on whatever they want to, whether 
it be SLA, SCA, or queue games for the more light hearted.

Matthew Fredrickson

> 
>  
> 
> From: Al lists [mailto:[EMAIL PROTECTED] 
> Sent: Tuesday, September 11, 2007 8:28 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)
> 
>  
> 
> I liked the queue game concept!
> although it could be cruel!
> 
> 
> 
> On 9/11/07, Steve Totaro <[EMAIL PROTECTED]
>  > wrote:
> 
> http://vonmag.com/editorial/web-exclusives/mark-spencer-digium-is-growing-up
> 
> Seems the Adtran relationship goes way back...
> 
> Thanks,
> Steve Totaro
> 
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>  
> 
> 
> 
> 
> 
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-- 
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.

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Re: [asterisk-users] how to disable wctdm auto modprobe during boot

[Tzafrir Cohen]

On Thu, Sep 13, 2007 at 05:32:25PM +0200, gincantalupo wrote:
> Hi,
> I've installed Asterisk with a TDM400P on a Debian Etch distro.
> When I reboot the server I get zaptel and wctdm automatically loaded.
> I'd like to avoid this behaviour.

Why exactly would you like to disable this?
If this is done to affect the Zaptel cards registration order, then load
the module manually before it gets loaded automaticaly: add it to
/etc/modules .

> How can I disable this automatic "modprobe wctdm" during boot?

# This is probably the worng fix for your problem, but you asked for it:
echo 'blacklist wctdm' >/etc/modprobe.d/arbitrary_file_name

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] Asterisk cli

[Tzafrir Cohen]

On Thu, Sep 13, 2007 at 10:36:56AM -0500, Mark Michelson wrote:
> Rizwan Hisham wrote:
> > i connect remotely. I have tried both of these cases but no warnings 
> > or mesages still.
> 
> It could be that your logger.conf file doesn't know to send debug 
> messages to the cli. Make sure that the "console" line in logger.conf 
> includes "debug". Mine looks like:
> 
> console => notice,warning,error,debug

Don't do that.

Send debug logs to a different place.

Normally ou should not need to see verbose messages, and certainly nt
debug messages. The sane default verbosity level and debug level is 0.
This makes actual errors stand out.

Debug messages will just flood your console and make it non-functional.

-- 
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icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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[asterisk-users] TCP connection to AMI broken after 15 minutes

[Wai Wu]

 
Does anyone have this experience? My TCP connection the Asterisk Manager
Interface is chopped off after 15 minutes of operation.

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[asterisk-users] CallWithUs Service?

[John Meksavan]

Asterisk Users,

 I am thinking about selecting CALLWITHUS as my sip provider.  Has anybody 
ever used them?  How was the call quality?  DTMF Tones issues?


 Thanks in advance.


-John

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Re: [asterisk-users] Paging to external speaker like in airports etc...

[Lacy Moore - Aspendora]

On 9/13/07, Deepak Naidu <[EMAIL PROTECTED]> wrote:
>
> Hi, I have a production asterisk-1.2.8 system with FreePBX & PRI Digium
> card.
>
> I am looking for a paging system to an external speaker.  I can page to
> internal Polycom 501 VoIP.
>
> But, what hardware or system do I need to integrate with the asterisk to
> have this acheived.


You know what would be even better?  If we had a search engine that you
could type something into and it would produce a list of pages related to
this.

Oh wait, maybe that's what this does:
http://www.google.com/search?hl=en&q=Asterisk+paging

Google is a wonderful tool, learn to use it...

--
> Deepak
>
>
>  *Linux your Life,** Don't Window it [[]]*
>
>*{ All for the best }*
>
> --
> Yahoo! Answers - Get better answers from someone who knows. Try it 
> now.
>
>
>
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Somewhere I wish I wasn't
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Re: [asterisk-users] Agent Callback Login in 1.4

[Anthony Francis]

Kevin P. Fleming wrote:
> Paul Hales wrote:
>
>   
>> It means that you end up reporting on SIP extensions, rather than agents
>> which doesn't work so well for call centres where people change desks or
>> have different staff using phones between shifts.
>> 
>
> This has already been discussed multiple times on this list, and has not
> been true since revision 43316 on Sept. 20 of 2006 (i.e. since before
> Asterisk 1.4.0 was released).
>
> AddQueueMember has a 'membername' argument that can be used to provide a
> logical member name for any interface you add as a queue member, and if
> provided this member name is used in the queue_log instead of the actual
> interface name.
>
>   
So the most helpful thing would be a solid example of how to exactly 
duplicate the agent callback login behavior in a real-time friendly 
manner. The part I am missing is how we are to do authentication.

-- 
Thank you and have a wonderful day,

Anthony Francis
Rockynet VOIP
(303) 444-7052 opt 2
[EMAIL PROTECTED]


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Re: [asterisk-users] Agent Callback Login in 1.4

[Kevin P. Fleming]

Paul Hales wrote:

> It means that you end up reporting on SIP extensions, rather than agents
> which doesn't work so well for call centres where people change desks or
> have different staff using phones between shifts.

This has already been discussed multiple times on this list, and has not
been true since revision 43316 on Sept. 20 of 2006 (i.e. since before
Asterisk 1.4.0 was released).

AddQueueMember has a 'membername' argument that can be used to provide a
logical member name for any interface you add as a queue member, and if
provided this member name is used in the queue_log instead of the actual
interface name.

-- 
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)

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Re: [asterisk-users] Caller ID on Channelized T1 (E&M Wink)

[Eric "ManxPower" Wieling]

I've not seen an E&M/Wink that supported Caller*ID.  You can fake it by 
sending something like *CALLERID*DID and then on the far end break that 
out and set the callerid and goto the DID.

Willy Wouters wrote:
> Hi,
> 
> Normally my T1 implementations are PRI.
> However, I do have a customer who uses channelized T1 (24 channels).
> I have setup a 'test' environment, and have two T1 channels back-to-back 
> in my [*] box.
> Both are setup with signalling => em_w.
> Calls DO go back & forth, but I can not see the callerID being passed.
> 
> Any ideas?
> 
> WW
> 


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[asterisk-users] ZAP to invalid SIP device call looping

[Gustavo Cordeiro]

Hello,

  When I receive calls in one FXO port (TDM400 or A200, occurs in both) and 
it dial to one invalid SIP extension, the call never hangup.

  The call would have to be dropped, but it seems that "Starting simple 
switch on 'Zap/1-1'" and "Hungup 'Zap/1-1'" occurs almost at the same time.

  If the dial is made to a valid SIP extension, the call is proceeded and 
terminated without any problem.

  Anybody had the same situation?

  Thanks in advance!


Regards,
Gustavo


- Asterisk 1.4.11


- Zaptel 1.4.5.1


- Call log:

  -- Starting simple switch on 'Zap/1-1'
  -- Executing [EMAIL PROTECTED]:1] Dial("Zap/1-1", "SIP/at001-a|30|r") in new 
stack
== Everyone is busy/congested at this time (1:0/0/1)
  -- Executing [EMAIL PROTECTED]:2] Hangup("Zap/1-1", "") in new stack
== Spawn extension (fxo-01, s, 2) exited non-zero on 'Zap/1-1'
  -- Hungup 'Zap/1-1'
  -- Starting simple switch on 'Zap/1-1'
  -- Executing [EMAIL PROTECTED]:1] Dial("Zap/1-1", "SIP/at001-a|30|r") in new 
stack
== Everyone is busy/congested at this time (1:0/0/1)
  -- Executing [EMAIL PROTECTED]:2] Hangup("Zap/1-1", "") in new stack
== Spawn extension (fxo-01, s, 2) exited non-zero on 'Zap/1-1'
  -- Hungup 'Zap/1-1'
  -- Starting simple switch on 'Zap/1-1'
  -- Executing [EMAIL PROTECTED]:1] Dial("Zap/1-1", "SIP/at001-a|30|r") in new 
stack
== Everyone is busy/congested at this time (1:0/0/1)
  -- Executing [EMAIL PROTECTED]:2] Hangup("Zap/1-1", "") in new stack
== Spawn extension (fxo-01, s, 2) exited non-zero on 'Zap/1-1'
  -- Hungup 'Zap/1-1'
  -- Starting simple switch on 'Zap/1-1'
  -- Executing [EMAIL PROTECTED]:1] Dial("Zap/1-1", "SIP/at001-a|30|r") in new 
stack
== Everyone is busy/congested at this time (1:0/0/1)
  -- Executing [EMAIL PROTECTED]:2] Hangup("Zap/1-1", "") in new stack
== Spawn extension (fxo-01, s, 2) exited non-zero on 'Zap/1-1'
  -- Hungup 'Zap/1-1'
  ...


- extensions.conf:

[fxo-01]
exten = s,1,dial(SIP/at001-a,30,r)
exten = s,n,hangup()


- zapata.conf:

[channels]
usecallerid = no
callwaiting = no
echocancel = yes
echotraining = yes
echocancelwhenbridged = yes
immediate = no
busydetect = yes
busycount = 3
callprogress = yes
progzone = br
faxdetect = no

context = fxo-01
signalling = fxs_ks
channel = 1

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[asterisk-users] Very fast playback

[Andre Courchesne]

Hi,

   It's my first attempt to run asterisk 1.4 (have been on 1.2 for a  
while) and I have a problem where playback and background are played  
very very fast. When I say fast is you get a few sounds that's it...

   Running kernel 2.6.20.4 and latest released asterisk packages  
(asterisk, libpri, zaptel).

   Any hints?

Andre COurchesne



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Re: [asterisk-users] Digium Appliance

[Matt]

Ok,
Now I'm confused.   Laura from Digium said the AA50 was and I quote "not
user editable when it comes to the config files".  Which is it?   Can I edit
the config files, or not?

Particularly of concern/interest to me is.. can I put my aastra phone config
files on the flash and access them from the web?

On 9/12/07, David Boyd <[EMAIL PROTECTED]> wrote:
>
> Hi Mat,
> i have been working with the aa50 for a couple of weeks now.  They are
> slick looking devices that still have a few bugs. I tried to use the
> device like an end user without previous knowledge of Asterisk or the
> asteriskGUI, and can say right off that a typical person will not be
> able to use the device by gui only. The interface does not create all
> entries required to configure either outbound routing or DID, outbound
> caller id for either sip or IAX looks to the fullname field in the
> users.conf file rather than CID entry They are working to correct
> the issues, however as of yet no known release date for firmware fixes.
> Having said that if you want to edit files via the gui by hand and make
> appropriate changes then the device seems to work ok.  Did have an issue
> where after reboot the system would register an IAX trunk with the
> provider but outbound calls would fail until you kicked the system to
> force a new registration.  A couple of times changes that were saved at
> the home page failed to commit to the flash card, replaced the flash and
> have not seen that issue again, but Little things that make me oogee
> about putting into a customer location right now.
>
> db
>
> On Wed, 2007-09-12 at 12:52 -0400, Matt wrote:
> > Hi,
> > Has anyone actually gotten their hands on an appliance yet?   If so,
> > how robust and working are they?  Any issues?
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[asterisk-users] problem

[Walter Willis]

I install asterisk 14.10.1 and update to asterisk 1.4.11 but the terminals
when call to extensions no work rining but not work voice, the error :


[Aug 15 19:42:56] WARNING[2171] chan_sip.c: Maximum retries exceeded on
transmission [EMAIL PROTECTED] for seqno 101 (Critical
Response)
[Aug 15 19:42:56] WARNING[2171] chan_sip.c: Maximum retries exceeded on
transmission [EMAIL PROTECTED] for seqno 101 (Critical
Response)
[Aug 15 19:42:57] WARNING[2171] chan_sip.c: Maximum retries exceeded on
transmission [EMAIL PROTECTED] for seqno 101 (Critical
Response)

my asterisk is over nat with forward the ports 5060 1-2 etc, it is
work fine but of the momento to other no work.

thanks. :)
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[asterisk-users] Caller ID on Channelized T1 (E&M Wink)

[Willy Wouters]

Hi,

Normally my T1 implementations are PRI.
However, I do have a customer who uses channelized T1 (24 channels).
I have setup a 'test' environment, and have two T1 channels back-to-back 
in my [*] box.
Both are setup with signalling => em_w.
Calls DO go back & forth, but I can not see the callerID being passed.

Any ideas?

WW

-- 
Willy Wouters, PhD
Asterisk Telephony
Web Applications
MAGU ENTERPRISES
Tel: 713-474-1534 


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Re: [asterisk-users] how to determine if a SIP extension has DND onoroff

[Vieri]

--- Steve Langstaff <[EMAIL PROTECTED]> wrote:

> Can you hook into the "qualify" code somehow? - that
> uses SIP OPTIONS.

I already knew of this wiki page:
http://www.voip-info.org/wiki/view/Asterisk+sip+qualify

So I did a "sip show peer" on the asterisk cli which I
am supposing is the same as the SIPPEER function.

When SIP softphone has DND turned OFF:

INF-VOIP*CLI> sip show peer 4053
INF-VOIP*CLI>

  * Name   : 4053
  Secret   : 
  MD5Secret: 
  Context  : from-internal
  Subscr.Cont. : 
  Language : es
  AMA flags: Unknown
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup: 1
  Pickupgroup  : 1
  Mailbox  : [EMAIL PROTECTED]
  VM Extension : asterisk
  LastMsgsSent : 0/0
  Call limit   : 0
  Dynamic  : Yes
  Callerid : "device" <4053>
  Expire   : 58
  Insecure : no
  Nat  : Always
  ACL  : No
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : No
  Trust RPID   : No
  Send RPID: No
  DTMFmode : rfc2833
  LastMsg  : 0
  ToHost   :
  Addr->IP : 10.215.147.240 Port 5060
  Defaddr->IP  : 0.0.0.0 Port 5060
  Def. Username: 4053
  SIP Options  : (none)
  Codecs   : 0xc (ulaw|alaw)
  Codec Order  : (ulaw,alaw)
  Status   : OK (127 ms)
  Useragent: SJphone/1.65.377a (SJ Labs)
  Reg. Contact : sip:[EMAIL PROTECTED]

When SIP softphone has DND turned ON:

INF-VOIP*CLI> sip show peer 4053
INF-VOIP*CLI>

  * Name   : 4053
  Secret   : 
  MD5Secret: 
  Context  : from-internal
  Subscr.Cont. : 
  Language : es
  AMA flags: Unknown
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup: 1
  Pickupgroup  : 1
  Mailbox  : [EMAIL PROTECTED]
  VM Extension : asterisk
  LastMsgsSent : 0/0
  Call limit   : 0
  Dynamic  : Yes
  Callerid : "device" <4053>
  Expire   : 45
  Insecure : no
  Nat  : Always
  ACL  : No
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : No
  Trust RPID   : No
  Send RPID: No
  DTMFmode : rfc2833
  LastMsg  : 0
  ToHost   :
  Addr->IP : 10.215.147.240 Port 5060
  Defaddr->IP  : 0.0.0.0 Port 5060
  Def. Username: 4053
  SIP Options  : (none)
  Codecs   : 0xc (ulaw|alaw)
  Codec Order  : (ulaw,alaw)
  Status   : OK (127 ms)
  Useragent: SJphone/1.65.377a (SJ Labs)
  Reg. Contact : sip:[EMAIL PROTECTED]
INF-VOIP*CLI>

I don't see any difference and "SIP Options  : (none)"
doesn't look "good".

(the SIP extension has qualify=yes)



  

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Re: [asterisk-users] Asterisk DIAL() premature timeout on a PRI trunk to legacy PBX

[Eric "ManxPower" Wieling]

Looks like the Alcatel is sending back a busy.  Check the value of 
HANGUPCAUSE with a Noop as the priority after the Dial.  You may also 
want to do a pri debug span X to see the actual Q.931 ISDN messages that 
are exchanged.

Vieri wrote:
> An Asterisk extension calls an Alcatel extension via a
> PRI link which rings 4 times for about 10-15 seconds
> and then drops.
> So if the Alcatel user doesn't answer within 10-15
> seconds the call is aborted.
> (A timeout is *not* specified in the Asterisk Dial
> command.)
> It seems however that either Asterisk or Alcatel drop
> the call prematurely (it's more likely to be on the
> Asterisk side).
> 
> What could I try?
> 
> The Asterisk log displays (* ext is 4053; Alcatel ext
> is 5900):
> 
> -- Executing
> Dial("SIP/4053-08311988","Zap/g1/5900||tTW") in new
> stack
> -- Requested transfer capability: 0x00 - SPEECH
> -- Called g1/5900
> -- Zap/2-1 is proceeding passing it to
> SIP/4053-08311988
> -- Zap/2-1 is ringing
> -- Zap/2-1 is busy
> -- Hungup 'Zap/2-1'
> == Everyone is busy/congested at this time (1:1/0/0)
> -- Executing Hangup("SIP/4053-08311988", "") in new
> stack
> == Spawn extension (from-internal, 5900, 4) exited
> non-zero on 'SIP/4053-08311988'
> -- Executing Macro("SIP/4053-08311988", "hangupcall")
> in new stack
> ...etc...
> 
> The Alcatel board is configured as:
> 
> Interface Type + PRA2
> CRC4 + YES
> Retransmission Timer : 100
> TEI Identity Check Timer : 100
> Polling Timer : 1000
> No. Of Retransmissions : 3
> Max Frame Size (Bytes) : 260
> Passive board + NO
> SS7 signaling + NO
> 
> (I also tried to increase the above "Timer" values but
> that did not change anything)
> 
> In Asterisk's /etc/zaptel.conf I have:
> 
> # TE120P (PRI):
> span=1,1,0,ccs,hdb3,crc4
> 
> bchan=1-15
> dchan=16
> bchan=17-31
> 
> What could be the problem here?
> 
> Thanks
> 
> 
> 
>   
> 
> Luggage? GPS? Comic books? 
> Check out fitting gifts for grads at Yahoo! Search
> http://search.yahoo.com/search?fr=oni_on_mail&p=graduation+gifts&cs=bz
> 
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[asterisk-users] Paging to external speaker like in airports etc...

[Deepak Naidu]

Hi, I have a production asterisk-1.2.8 system with FreePBX & PRI Digium card.

I am looking for a paging system to an external speaker.  I can page to 
internal Polycom 501 VoIP.

But, what hardware or system do I need to integrate with the asterisk to have 
this acheived.

--
Deepak



Linux your Life, Don't Window it [[]] 

   { All for the best }



   
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Re: [asterisk-users] Followme app_followme

[Kevin Kiely]

Easy now... I didn't see the closed posts after searching that's why I
looked elsewhere.  When I searched for the posts on the bug list they
weren't there and was able to see the resolution.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason Parker
Sent: Thursday, September 13, 2007 12:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Followme app_followme

Kevin Kiely wrote:
> When using app_followme, I am receiving the following warnings on the
> console.  We are calling the followme app with no options for additional
> voice announcements.  Is anyone else experiencing this issue with 1.4.11?
> 
> -- Executing [EMAIL PROTECTED]:1]
> FollowMe("SIP/101206006-b72223d8", "101206002") in new stack
> [Sep 13 12:10:37] WARNING[1447]: file.c:563 ast_openstream_full: File
> /var/spool/asterisk/followme.1189699837.464 does not exist in any format
> [Sep 13 12:10:37] WARNING[1447]: file.c:813 ast_streamfile: Unable to open
> /var/spool/asterisk/followme.1189699837.464 (format 0x4 (ulaw)): No such
> file or directory
> --  Playing 'followme/pls-hold-while-try'
> (language 'en')
> 

Kevin,
You have already posted 2 duplicate bugs reports which were closed and a
very clear answer was given as to why.  I honestly do not know how much more
clear I can make this.

Yes, it was a problem in 1.4.11.  However, this has ALREADY been fixed in
svn.
 It will be in the next release.

If you would like to have this fix, you can run the latest version of svn
branch 1.4.

-- 
Jason Parker
Digium

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No virus found in this incoming message.
Checked by AVG Free Edition. 
Version: 7.5.485 / Virus Database: 269.13.16/1004 - Release Date: 9/12/2007
5:22 PM
 


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Re: [asterisk-users] Direct dialing to correct extension from analog lines

[Lars Bensmann]

On Wed, Sep 12, 2007 at 06:40:46PM +0300, Tzafrir Cohen wrote:
> On Wed, Sep 12, 2007 at 04:55:39PM +0200, Lars Bensmann wrote:
> > Hi,
> > 
> > I have a problem with people that are calling from analog lines.
> > 
> > We have a block of numbers 12345 - 0 to -99. 
> 
> 00 to 99, right?

-0 is the main number which is quite common (at least in Germany).

-1 to -9 and -10 are not used as the old PBX used the first matching
extension. The first "real" extension is -11.

It seems there is no limit from the PSTN regarding the number of
extensions. I just tried -12345 and it works as well.

> > But people calling from analog lines are connected to our asterisk box
> > as soon as they finish dialing 12345. They don't get a chance to dial an
> > extension.
> 
> That is what overlapdial is for, right?
> 
> But you need a unidirectional overlap dial?

Thanks for pointing me in the right direction. I grepped the config
files earlier and it looked alright. But now I took a closer look and
overlapdial was enabled after the first two spans that are connected to
the PSTN.

Thanks,
Lars Bensmann

-- 
Copyri ah hell, just take it.
  -- Jeffrey Friedl ([EMAIL PROTECTED]) Active Perl

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[asterisk-users] Asterisk DIAL() premature timeout on a PRI trunk to legacy PBX

[Vieri]

An Asterisk extension calls an Alcatel extension via a
PRI link which rings 4 times for about 10-15 seconds
and then drops.
So if the Alcatel user doesn't answer within 10-15
seconds the call is aborted.
(A timeout is *not* specified in the Asterisk Dial
command.)
It seems however that either Asterisk or Alcatel drop
the call prematurely (it's more likely to be on the
Asterisk side).

What could I try?

The Asterisk log displays (* ext is 4053; Alcatel ext
is 5900):

-- Executing
Dial("SIP/4053-08311988","Zap/g1/5900||tTW") in new
stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/5900
-- Zap/2-1 is proceeding passing it to
SIP/4053-08311988
-- Zap/2-1 is ringing
-- Zap/2-1 is busy
-- Hungup 'Zap/2-1'
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing Hangup("SIP/4053-08311988", "") in new
stack
== Spawn extension (from-internal, 5900, 4) exited
non-zero on 'SIP/4053-08311988'
-- Executing Macro("SIP/4053-08311988", "hangupcall")
in new stack
...etc...

The Alcatel board is configured as:

Interface Type + PRA2
CRC4 + YES
Retransmission Timer : 100
TEI Identity Check Timer : 100
Polling Timer : 1000
No. Of Retransmissions : 3
Max Frame Size (Bytes) : 260
Passive board + NO
SS7 signaling + NO

(I also tried to increase the above "Timer" values but
that did not change anything)

In Asterisk's /etc/zaptel.conf I have:

# TE120P (PRI):
span=1,1,0,ccs,hdb3,crc4

bchan=1-15
dchan=16
bchan=17-31

What could be the problem here?

Thanks



  

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Re: [asterisk-users] TE405P intermittent yellow alarm

[Richard Lyman]

a yellow alarm should mean there is a signal/encoding issue.
we used to see this on older gear when a single segment in the Rx leg 
would flip from SF to ESF.
which means the old gear wanted SF, but all the newer telco gear 
'defaults' to ESF nowdays, so there would be momentary yellow alarms as 
the gear in the middle flipped back to SF.

a red would be a loss of signal.

as asterisk views them...

yellow = remote alarm indication
red = loss of signal


Richard van der Hoff wrote:
> Steve Totaro wrote:
>   
>> Richard van der Hoff wrote:
>> 
>  >> [intermittent yellow alarm]
>   
>>> At this point, I'd really like to know what a yellow alarm actually
>>> means. I've read that it indicates that that the other end of the E1 is
>>> in an alarm condition: however BT's terminating unit seems quite happy
>>> with no alarm conditions at all.
>>>
>>>   
>> Check your cabling.  Replace it with new stuff.  Re-punch everything. 
>>
>> It is obviously somewhere in the line.  If the above does not fix it, 
>> maybe you can get a lucky and get a good tech out that will stick around 
>> to see the issue.
>> 
>
> The only bit of cable I own here is the 2m length of cat-5 between the 
> te405P and BT's line terminating unit. And yes, I've replaced that about 
> 5 times now...
>
> Thanks for your help, but again I'd like to ask: what does a yellow 
> alarm actually mean? From the driver source code I can see it is set 
> when the FRS0 register has bit 4 set - but that doesn't help a lot...
>
> Regards
>
> Richard
>
>   



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Re: [asterisk-users] Followme app_followme

[Jason Parker]

Kevin Kiely wrote:
> When using app_followme, I am receiving the following warnings on the
> console.  We are calling the followme app with no options for additional
> voice announcements.  Is anyone else experiencing this issue with 1.4.11?
> 
> -- Executing [EMAIL PROTECTED]:1]
> FollowMe("SIP/101206006-b72223d8", "101206002") in new stack
> [Sep 13 12:10:37] WARNING[1447]: file.c:563 ast_openstream_full: File
> /var/spool/asterisk/followme.1189699837.464 does not exist in any format
> [Sep 13 12:10:37] WARNING[1447]: file.c:813 ast_streamfile: Unable to open
> /var/spool/asterisk/followme.1189699837.464 (format 0x4 (ulaw)): No such
> file or directory
> --  Playing 'followme/pls-hold-while-try'
> (language 'en')
> 

Kevin,
You have already posted 2 duplicate bugs reports which were closed and a
very clear answer was given as to why.  I honestly do not know how much more
clear I can make this.

Yes, it was a problem in 1.4.11.  However, this has ALREADY been fixed in svn.
 It will be in the next release.

If you would like to have this fix, you can run the latest version of svn
branch 1.4.

-- 
Jason Parker
Digium

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[asterisk-users] Followme app_followme

[Kevin Kiely]

When using app_followme, I am receiving the following warnings on the
console.  We are calling the followme app with no options for additional
voice announcements.  Is anyone else experiencing this issue with 1.4.11?

-- Executing [EMAIL PROTECTED]:1]
FollowMe("SIP/101206006-b72223d8", "101206002") in new stack
[Sep 13 12:10:37] WARNING[1447]: file.c:563 ast_openstream_full: File
/var/spool/asterisk/followme.1189699837.464 does not exist in any format
[Sep 13 12:10:37] WARNING[1447]: file.c:813 ast_streamfile: Unable to open
/var/spool/asterisk/followme.1189699837.464 (format 0x4 (ulaw)): No such
file or directory
--  Playing 'followme/pls-hold-while-try'
(language 'en')


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Re: [asterisk-users] Looking for Asterisk Consultant in San Franicsco

[Bhrugu Mehta]

HI,
I have read your mail. I get ready for that but pls tell me what i do
at remote support.
thnks for sending mail.
Bhrugu mehta

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Re: [asterisk-users] DTMF error on asterisk

[Eric "ManxPower" Wieling]

HNAGUPCAUSE is more specific.  Cause 31 is "Normal, Unspecified" end of 
call.  Chances are it is a harmless message and is a telco caused issue.

See 
http://www.quintum.com/support/xplatform/network/Q931_Disconnect_Cause_Code_List.pdf

gincantalupo wrote:
> Hi satish,
> I get that error too (my Asterisk version is 1.2.x but should be the 
> same) when that Zap channel is not available and you are trying to use it.
> You should get a CHANUNAVAIL from Asterisk channel status.
> 
> Giorgio.
> 
> 
> satish patel wrote:
>> Dear all
>>
>>I have asterisk 1.4.11 on centos 4.x i have installed 2 
>> PRI on is asterisk and it is working fine but i got this DTMF error on 
>> asterisk CLI what is it ??
>>
>>
>> -- Zap/36-1 is ringing
>> -- Zap/36-1 answered SIP/5406-9fa59770
>> -- Channel 0/1, span 2 got hangup request, cause 31
>> [Sep 13 22:10:29] WARNING[7191]: app_dial.c:741 wait_for_answer: 
>> Unable to forward voice or dtmf

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Re: [asterisk-users] Asterisk cli

[Rizwan Hisham]

that was it. Thanx

On 9/13/07, Mark Michelson <[EMAIL PROTECTED]> wrote:
>
> Rizwan Hisham wrote:
> > i connect remotely. I have tried both of these cases but no warnings
> > or mesages still.
>
> It could be that your logger.conf file doesn't know to send debug
> messages to the cli. Make sure that the "console" line in logger.conf
> includes "debug". Mine looks like:
>
> console => notice,warning,error,debug
>
> Mark Michelson
>
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-- 
Best Regards
Rizwan Hisham
Software Engineer
Axvoice Inc.
www.axvoice.com
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Re: [asterisk-users] how to determine if a SIP extension has DND onoroff

[Steve Langstaff]

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of Vieri
> Sent: 13 September 2007 16:38
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] how to determine if a SIP 
> extension has DND onoroff
> 
> 
> --- Steve Langstaff <[EMAIL PROTECTED]> wrote:
> 
> > Sending the phone a SIP OPTIONS message *should* get you 
> the response 
> > code that the phone would respond with if you sent it an INVITE - I 
> > don't know how to do that from AGI though.
> 
> Do you know another way?
> I suppose it could be done by programming sockets and send 
> SIP protocol commands but I would need to do it
> *from* the server *to* the client (ie. the request would 
> originate on the * server).
> Is there a simple example as to how one may implement this, 
> say, in PHP or Perl? (I can create sockets but am unfamiliar 
> with SIP) 
> 
> Also, can this problem be handled the other way around? Can 
> Asterisk be configured somehow so that whenever someone tries 
> to call a particular extension and the latter yields a 
> 'response 486 "Do Not Disturb"' then the DND field in AstDB 
> for that extension is updated?
> This way the custom AGI script would only need to execute 
> "database show dnd"...
> 
> I need this particularly for queues that have the strict 
> option for joining and leaving.
> In this situation a custom cron script adds and removes 
> members dynamically from the queues. The problem I found is 
> that "strict" behavior works "as expected" when the agents, 
> even if added via AddQueueMember, are logged off or have 
> their softphone turned off but "fails" if they activate DND 
> (either by pressing the softphone DND button or dialing *78).
> So a "solution" I am thinking of implementing is to change 
> this custom cron script and make it "detect" if certain SIP 
> extensions have DND on or not (either with "database show 
> dnd" or any other reliable method). If it detects an 
> activated DND then it will execute a 
> RemoveQueueMember(queuenum, sipnum). If it detects that DND 
> is off again then it will run an AddQueueMember(queuenum, sipnum).
> 
> Help appreciated ;-)

Can you hook into the "qualify" code somehow? - that uses SIP OPTIONS.

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Re: [asterisk-users] DTMF error on asterisk

[gincantalupo]

Hi satish,
I get that error too (my Asterisk version is 1.2.x but should be the 
same) when that Zap channel is not available and you are trying to use it.
You should get a CHANUNAVAIL from Asterisk channel status.

Giorgio.


satish patel wrote:
> Dear all
>
>I have asterisk 1.4.11 on centos 4.x i have installed 2 
> PRI on is asterisk and it is working fine but i got this DTMF error on 
> asterisk CLI what is it ??
>
>
> -- Zap/36-1 is ringing
> -- Zap/36-1 answered SIP/5406-9fa59770
> -- Channel 0/1, span 2 got hangup request, cause 31
> [Sep 13 22:10:29] WARNING[7191]: app_dial.c:741 wait_for_answer: 
> Unable to forward voice or dtmf
> -- Hungup 'Zap/32-1'
>
>
>
>
>
>
> 
> Pinpoint customers 
> who
>  
> are looking for what you sell.
> 
>
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_
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FG&A srl - http://www.fgasoftware.com -
[EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu
Tel: 02997663.14, Fax: 0291390172  


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Re: [asterisk-users] how to determine if a SIP extension has DND on oroff

[Vieri]

--- Steve Langstaff <[EMAIL PROTECTED]> wrote:

> Sending the phone a SIP OPTIONS message *should* get
> you the response
> code that the phone would respond with if you sent
> it an INVITE - I
> don't know how to do that from AGI though.

Do you know another way?
I suppose it could be done by programming sockets and
send SIP protocol commands but I would need to do it
*from* the server *to* the client (ie. the request
would originate on the * server).
Is there a simple example as to how one may implement
this, say, in PHP or Perl? (I can create sockets but
am unfamiliar with SIP) 

Also, can this problem be handled the other way
around? Can Asterisk be configured somehow so that
whenever someone tries to call a particular extension
and the latter yields a 'response 486 "Do Not
Disturb"' then the DND field in AstDB for that
extension is updated?
This way the custom AGI script would only need to
execute "database show dnd"...

I need this particularly for queues that have the
strict option for joining and leaving.
In this situation a custom cron script adds and
removes members dynamically from the queues. The
problem I found is that "strict" behavior works "as
expected" when the agents, even if added via
AddQueueMember, are logged off or have their softphone
turned off but "fails" if they activate DND (either by
pressing the softphone DND button or dialing *78).
So a "solution" I am thinking of implementing is to
change this custom cron script and make it "detect" if
certain SIP extensions have DND on or not (either with
"database show dnd" or any other reliable method). If
it detects an activated DND then it will execute a
RemoveQueueMember(queuenum, sipnum). If it detects
that DND is off again then it will run an
AddQueueMember(queuenum, sipnum).

Help appreciated ;-)

> > -Original Message-
> > From: [EMAIL PROTECTED] 
> > [mailto:[EMAIL PROTECTED]
> On Behalf Of Vieri
> > Sent: 13 September 2007 14:30
> > To: asterisk-users@lists.digium.com
> > Subject: [asterisk-users] how to determine if a
> SIP extension 
> > has DND on oroff
> > 
> > I would like to determine through an AGI script if
> a specific 
> > SIP extension has DND on or off.
> > 
> > I know that if the SIP client dialed *78 or *79 it
> is usually 
> > enough to just do a:
> > 
> > database show dnd
> > 
> > to fetch the DND status from the database.
> > 
> > However, not all clients dial *78 or *79 (or
> whichever 
> > feature code is defined for DND).
> > 
> > Some softphones such as SJPhone have a DND button.
> > When pressed and someone tries to Dial() that
> extension, the 
> > Asterisk CLI shows something like this:
> > 
> > -- Called SIP/4053
> > -- Got SIP response 486 "Do Not Disturb" back
> from
> > 10.215.144.48
> > -- SIP/4053-08311988 is busy
> > 
> > So how could I get the "response code" *without*
> actually 
> > dialing from within an AGI script? (I don't want
> to establish 
> > a call, just want to see if the SIP client replies
> with a DND 
> > response code) Like a "ping" of some sort...
> > 
> > Vieri



   

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Re: [asterisk-users] Asterisk cli

[Mark Michelson]

Rizwan Hisham wrote:
> i connect remotely. I have tried both of these cases but no warnings 
> or mesages still.

It could be that your logger.conf file doesn't know to send debug 
messages to the cli. Make sure that the "console" line in logger.conf 
includes "debug". Mine looks like:

console => notice,warning,error,debug

Mark Michelson

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[asterisk-users] how to disable wctdm auto modprobe during boot

[gincantalupo]

Hi,
I've installed Asterisk with a TDM400P on a Debian Etch distro.
When I reboot the server I get zaptel and wctdm automatically loaded.
I'd like to avoid this behaviour.
How can I disable this automatic "modprobe wctdm" during boot?

Thanks

Giorgio


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Re: [asterisk-users] SMS in France - allways get "NAK"

[John Hughes]

John Hughes wrote:
> I'm trying to send an sms:
>
> smsq --motx-channel=CAPI/g1/0809101000 060739 "X"
>
> It seems to try to do something, but FT aren't happy:
>
> -- Attempting call on CAPI/g1/0809101000 for application SMS(0) (Retry 1)
>   == ISDN4#02: Setting up DTMF detector (PLCI=0x104, flag=1)
> [Sep 13 15:45:50] WARNING[23584]: pbx.c:5142 ast_pbx_outgoing_app2: 
> CAPI/ISDN4#02/0809101000-1 already has a call detail record??
>> Channel CAPI/ISDN4#02/0809101000-1 was answered.
>> Launching SMS(0) on CAPI/ISDN4#02/0809101000-1
> -- SMS RX 93 00 6D
> -- SMS TX 91 0D 01 01 0A 81 60 70 93 66 66 00 F1 01 58 5C
> -- SMS RX 96 0A 01 A0 00 70 90 31 51 54 45 80 24
> -- SMS TX 94 00 6C
>   == ISDN4#02: CAPI Hangingup for PLCI=0x104 in state 2
> [Sep 13 15:45:58] NOTICE[23584]: pbx_spool.c:371 attempt_thread: Call 
> completed to CAPI/g1/0809101000
>> ISDN4#02: CAPI INFO 0x3495: Call rejected
>
> As I understand it the "RX 96" is FT NAK'ing my "TX 91" "DELIVER".
>
> Any idea what I'm doing wrong?
>   
FT only accept the SMS if you use a valid sender address, adding the
--motx-callerid= option fixed it.

-- SMS RX 93 00 6D
-- SMS TX 91 38 01 04 0A 81 60 58 70 11 11 00 F1 32 D3 B2 9B 0C AA CF 41 61 
36 1B 94 7F D7 E5 A0 F6...
-- SMS RX 95 09 01 00 70 90 31 51 95 25 80 A5
-- SMS TX 94 00 6C




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Re: [asterisk-users] No Sound on Zap Channels

[Jon Weisman]

I have a feeling the dchannel is bad. I'll investigate further and post my 
findings.

-Jon


- Original Message - 
From: "Atis" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Thursday, September 13, 2007 4:12 AM
Subject: Re: [asterisk-users] No Sound on Zap Channels


> On 9/13/07, Hoai-Anh Ngo-Vi <[EMAIL PROTECTED]> wrote:
>> Have you answered the channel?
>
> Voicemail doesn't require Answer(). It does that itself, as you
> usually get to voicemail after Dial(). It would be silly to require to
> do Answer after each Dial and then send to voicemail.
>
> Regards,
> Atis
>
>>
>> Von: [EMAIL PROTECTED]
>> [mailto:[EMAIL PROTECTED] Im Auftrag
>> von Jon Weisman
>
>> I've got a strange issue here. When I make a SIP call to say my voicemail
>> app, I hear audio just fine. However when I dial from PSTN into my 
>> Asterisk
>> box, I see that its playing the voice files, but I hear nothing, then the
>> call drops. I'm running Fedora Core 6, and Asterisk 1.2.24. CLI output
>> below. T-1 is PRI, showing normal, dchannel is up as well. Any help is
>> greatly appreciated.
>>
>>
>>
>>
>>
>>
>>
>>
>> Thanks,
>>
>>
>> Jon
>>
>>
>>
>>
>>
>>
>>
>>
>>  -- Accepting call from '2125551212' to '6465551212' on channel 0/23, 
>> span 4
>>  -- Executing VoiceMail("Zap/95-1", "u100") in new stack
>>  -- Playing 'vm-theperson' (language 'en')
>>  -- Playing 'digits/1' (language 'en')
>>  -- Playing 'digits/0' (language 'en')
>>  -- Playing 'digits/0' (language 'en')
>>  -- Playing 'vm-isunavail' (language 'en')
>>  -- Playing 'vm-intro' (language 'en')
>>  -- Channel 0/23, span 4 got hangup request, cause 34
>>== Spawn extension (default, 6465551212, 1) exited non-zero on 
>> 'Zap/95-1'
>>  -- Hungup 'Zap/95-1'
>> ___
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>
>
> -- 
> Atis Lezdins,
> IT Responsible of BEST Riga,
> [EMAIL PROTECTED]
> ICQ: 142239285
> Skype: atis.lezdins
> Cell Phone: +371 28806004 [Tele2, Latvia]
> Work phone: +1 800 7502835 [Toll free, USA]
> ?BEST? -> www.BEST.eu.org
>
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Re: [asterisk-users] TE405P intermittent yellow alarm

[Don Pobanz]

On Thursday, September 13, 2007 4:58 AM Richard van der Hoff said
> Thanks for your help, but again I'd like to ask: what does a yellow 
> alarm actually mean? From the driver source code I can see it is set 
> when the FRS0 register has bit 4 set - but that doesn't help a lot...
> 

All of my experience has been with T1s, not E1s but I assume the alarms
mean the same even though they are transmitted differently. 

Suppose that there are three pieces of equipment 'A', 'B', and 'C' and
the signal from 'A' to 'B' has been interrupted (designated by the 'X'
in the diagram) so that 'B' is not seeing an incoming signal. 'B' will
be in red alarm, and 'B' will transmit back to 'A' a yellow alarm
indicator. When 'A' see the yellow alarm indicator, 'A' will go into
yellow alarm. 

Just to complete the picture, if 'B' is not the end point of the circuit
'B' will then send to 'C' a blue alarm indicator. As you can see, a
signal not making it between two pieces of equipment can cause 3 (or
more) alarms. 


Yellow   red blue 
 alarm   alarm   alarm
|-| |-| |-|
| |---X>| |>| | 
|  A  | |  B  | |  C  | 
| |<| |<| | 
|-| |-| |-|

I hope this helps. 

Don Pobanz

> Regards
> 
> Richard

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Re: [asterisk-users] Conference bridge.

[Alex Balashov]

On Thu, 13 Sep 2007, Paul Hales wrote:

> On Wed, 2007-09-12 at 16:44 -0400, Alex Balashov wrote:
>> Any recommendations for an affordable SIP conference bridge unit?  I mean
>> one that isn't crappy;  something where the duplex and cancellation
>> functions that are traditionally built into such devices actually work.
>
> Do you want something cheap or something that works?

   True, true.

--
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Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671

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[asterisk-users] DTMF error on asterisk

[satish patel]

Dear all

   I have asterisk 1.4.11 on centos 4.x i have installed 2 PRI on 
is asterisk and it is working fine but i got this DTMF error on asterisk CLI 
what is it ?? 


-- Zap/36-1 is ringing
-- Zap/36-1 answered SIP/5406-9fa59770
-- Channel 0/1, span 2 got hangup request, cause 31
[Sep 13 22:10:29] WARNING[7191]: app_dial.c:741 wait_for_answer: Unable to 
forward voice or dtmf
-- Hungup 'Zap/32-1'







   
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Re: [asterisk-users] Fwd: Bad FCS error

[Eric "ManxPower" Wieling]

This error message means there was corrupted data received from the 
card.  This can be caused by many things.  The T-1/E-1 could be having 
errors, there could be a bad SmartJack (the telco box on your wall), it 
can also be cause by lost or delayed interrupts.

As you can see below your T-1/E-1 card is sharing an IRQ with your 
Ethernet card.  This is a Very Bad Thing.  Digium cards and drivers have 
gotten much better at sharing interrupts, but I think it is still a bad 
idea to have two devices that generate many interrupts on the same IRQ. 
  Try moving the card to a different slot.

Jan Prunk wrote:
> -- Forwarded message --
> From: Jan Prunk <[EMAIL PROTECTED]>
> Date: Sep 13, 2007 3:35 PM
> Subject: Bad FCS error
> To: asterisk-users@lists.digium.com
> 
> Hello !
> 
> I am getting the following error in asterisk logs:
> 
> Sep 11 17:33:30 NOTICE[2187] chan_zap.c: PRI got event: HDLC Bad FCS (8) on
> Primary D-channel of span 1
> Sep 11 21:45:08 NOTICE[2187] chan_zap.c: PRI got event: HDLC Bad FCS (8) on
> Primary D-channel of span 1
> Sep 11 23:57:26 NOTICE[2187] chan_zap.c: PRI got event: HDLC Bad FCS (8) on
> Primary D-channel of span 1
> Sep 12 01:59:01 NOTICE[2187] chan_zap.c: PRI got event: HDLC Bad FCS (8) on
> Primary D-channel of span 1
> Sep 12 03:52:31 NOTICE[2187] chan_zap.c: PRI got event: HDLC Bad FCS (8) on
> Primary D-channel of span 1
> Sep 12 04:32:41 NOTICE[2187] chan_zap.c: PRI got event: HDLC Abort (6) on
> Primary D-channel of span 1
> 
> The result is that when someone phones, the call gets disrupted, and
> sometimes it breaks.
> I wonder if this was fixed in the latest releases of asterisk.
> I am using asterisk version 1.2.13 on Debian sarge system
> And I am using zaptel module (if its related) 1.2.11
> 
> cat /proc/interrupts
>CPU0   CPU1
>   0: 1327389665  0IO-APIC-edge  timer
>   1:240  0IO-APIC-edge  i8042
>   9:  0  0   IO-APIC-level  acpi
> 169: 1361816047  0   IO-APIC-level  wcte11xp, eth0
> 177: 648682  0   IO-APIC-level  3ware Storage Controller
> 201:  0  0 PCI-MSI  pciehp
> 209:  0  0 PCI-MSI  pciehp
> NMI:  1  0
> LOC: 1327446761 1327446897
> ERR:  0
> MIS:  0
> 


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[asterisk-users] SMS in France - allways get "NAK"

[John Hughes]

I'm trying to send an sms:

smsq --motx-channel=CAPI/g1/0809101000 060739 "X"

It seems to try to do something, but FT aren't happy:

-- Attempting call on CAPI/g1/0809101000 for application SMS(0) (Retry 1)
  == ISDN4#02: Setting up DTMF detector (PLCI=0x104, flag=1)
[Sep 13 15:45:50] WARNING[23584]: pbx.c:5142 ast_pbx_outgoing_app2: 
CAPI/ISDN4#02/0809101000-1 already has a call detail record??
   > Channel CAPI/ISDN4#02/0809101000-1 was answered.
   > Launching SMS(0) on CAPI/ISDN4#02/0809101000-1
-- SMS RX 93 00 6D
-- SMS TX 91 0D 01 01 0A 81 60 70 93 66 66 00 F1 01 58 5C
-- SMS RX 96 0A 01 A0 00 70 90 31 51 54 45 80 24
-- SMS TX 94 00 6C
  == ISDN4#02: CAPI Hangingup for PLCI=0x104 in state 2
[Sep 13 15:45:58] NOTICE[23584]: pbx_spool.c:371 attempt_thread: Call completed 
to CAPI/g1/0809101000
   > ISDN4#02: CAPI INFO 0x3495: Call rejected

As I understand it the "RX 96" is FT NAK'ing my "TX 91" "DELIVER".

Any idea what I'm doing wrong?

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[asterisk-users] Fwd: Bad FCS error

[Jan Prunk]

-- Forwarded message --
From: Jan Prunk <[EMAIL PROTECTED]>
Date: Sep 13, 2007 3:35 PM
Subject: Bad FCS error
To: asterisk-users@lists.digium.com

Hello !

I am getting the following error in asterisk logs:

Sep 11 17:33:30 NOTICE[2187] chan_zap.c: PRI got event: HDLC Bad FCS (8) on
Primary D-channel of span 1
Sep 11 21:45:08 NOTICE[2187] chan_zap.c: PRI got event: HDLC Bad FCS (8) on
Primary D-channel of span 1
Sep 11 23:57:26 NOTICE[2187] chan_zap.c: PRI got event: HDLC Bad FCS (8) on
Primary D-channel of span 1
Sep 12 01:59:01 NOTICE[2187] chan_zap.c: PRI got event: HDLC Bad FCS (8) on
Primary D-channel of span 1
Sep 12 03:52:31 NOTICE[2187] chan_zap.c: PRI got event: HDLC Bad FCS (8) on
Primary D-channel of span 1
Sep 12 04:32:41 NOTICE[2187] chan_zap.c: PRI got event: HDLC Abort (6) on
Primary D-channel of span 1

The result is that when someone phones, the call gets disrupted, and
sometimes it breaks.
I wonder if this was fixed in the latest releases of asterisk.
I am using asterisk version 1.2.13 on Debian sarge system
And I am using zaptel module (if its related) 1.2.11

cat /proc/interrupts
   CPU0   CPU1
  0: 1327389665  0IO-APIC-edge  timer
  1:240  0IO-APIC-edge  i8042
  9:  0  0   IO-APIC-level  acpi
169: 1361816047  0   IO-APIC-level  wcte11xp, eth0
177: 648682  0   IO-APIC-level  3ware Storage Controller
201:  0  0 PCI-MSI  pciehp
209:  0  0 PCI-MSI  pciehp
NMI:  1  0
LOC: 1327446761 1327446897
ERR:  0
MIS:  0

Thank you for answering !

Regards,
Jan Prunk
-- 
+--+
| Jan PrunkGPG key: 00E80E86   |
| E-mail: [EMAIL PROTECTED] Fingerprint: 77C5 156E 29A4 EB6C 1C4A   |
| http://blog.prunk.be  5EBA 414A 29F5 00E8 0E86   |
+--+

-- 
+--+
| Jan PrunkGPG key: 00E80E86   |
| E-mail: [EMAIL PROTECTED] Fingerprint: 77C5 156E 29A4 EB6C 1C4A   |
| http://blog.prunk.be  5EBA 414A 29F5 00E8 0E86   |
+--+
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Re: [asterisk-users] how to determine if a SIP extension has DND on oroff

[Steve Langstaff]

Sending the phone a SIP OPTIONS message *should* get you the response
code that the phone would respond with if you sent it an INVITE - I
don't know how to do that from AGI though. 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of Vieri
> Sent: 13 September 2007 14:30
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] how to determine if a SIP extension 
> has DND on oroff
> 
> I would like to determine through an AGI script if a specific 
> SIP extension has DND on or off.
> 
> I know that if the SIP client dialed *78 or *79 it is usually 
> enough to just do a:
> 
> database show dnd
> 
> to fetch the DND status from the database.
> 
> However, not all clients dial *78 or *79 (or whichever 
> feature code is defined for DND).
> 
> Some softphones such as SJPhone have a DND button.
> When pressed and someone tries to Dial() that extension, the 
> Asterisk CLI shows something like this:
> 
> -- Called SIP/4053
> -- Got SIP response 486 "Do Not Disturb" back from
> 10.215.144.48
> -- SIP/4053-08311988 is busy
> 
> So how could I get the "response code" *without* actually 
> dialing from within an AGI script? (I don't want to establish 
> a call, just want to see if the SIP client replies with a DND 
> response code) Like a "ping" of some sort...
> 
> Vieri
> 
> 
> 
>   
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[asterisk-users] how to determine if a SIP extension has DND on or off

[Vieri]

I would like to determine through an AGI script if a
specific SIP extension has DND on or off.

I know that if the SIP client dialed *78 or *79 it is
usually enough to just do a:

database show dnd

to fetch the DND status from the database.

However, not all clients dial *78 or *79 (or whichever
feature code is defined for DND).

Some softphones such as SJPhone have a DND button.
When pressed and someone tries to Dial() that
extension, the Asterisk CLI shows something like this:

-- Called SIP/4053
-- Got SIP response 486 "Do Not Disturb" back from
10.215.144.48
-- SIP/4053-08311988 is busy

So how could I get the "response code" *without*
actually dialing from within an AGI script? (I don't
want to establish a call, just want to see if the SIP
client replies with a DND response code)
Like a "ping" of some sort...

Vieri



  

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Re: [asterisk-users] Different Networks

[Mike Hammett]

"That server traceroutes out that interface."

Yes, I can lynx to google.com.

Braxis*CLI> iax2 reload
  == Parsing '/etc/asterisk/iax.conf': Found
   > doing dnsmgr_lookup for '208.100.1.33'
   > doing dnsmgr_lookup for '208.100.1.33'
  == Parsing '/etc/asterisk/users.conf': Found
  == Loaded firmware 'iaxy.bin'
  == Parsing '/etc/asterisk/iaxprov.conf': Found
-- Loaded provisioning template 'default'
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: 
REGREQ
   Timestamp: 00017ms  SCall: 2  DCall: 0 [208.100.1.33:4569]
   USERNAME: rwestics
   REFRESH : 60

Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: 
REGREQ
   Timestamp: 00017ms  SCall: 3  DCall: 0 [208.100.1.33:4569]
   USERNAME: ottos
   REFRESH : 60

Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK
   Timestamp: 00017ms  SCall: 9  DCall: 2 [208.100.1.33:4569]
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: 
REGAUTH
   Timestamp: 9ms  SCall: 9  DCall: 2 [208.100.1.33:4569]
   AUTHMETHODS : 3
   CHALLENGE   : 29638146
   USERNAME: rwestics

Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: 
REGREQ
   Timestamp: 00035ms  SCall: 2  DCall: 9 [208.100.1.33:4569]
   USERNAME: rwestics
   REFRESH : 60
   MD5 RESULT  : 1c113a5aaa20100f2c864544b892fea3

Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK
   Timestamp: 00017ms  SCall: 00013  DCall: 3 [208.100.1.33:4569]
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: 
REGAUTH
   Timestamp: 00010ms  SCall: 00013  DCall: 3 [208.100.1.33:4569]
   AUTHMETHODS : 3
   CHALLENGE   : 43340858
   USERNAME: ottos

Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: 
REGREQ
   Timestamp: 00036ms  SCall: 3  DCall: 00013 [208.100.1.33:4569]
   USERNAME: ottos
   REFRESH : 60
   MD5 RESULT  : 4d18ad3b06bc96496f59655367093ecf

Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK
   Timestamp: 00035ms  SCall: 9  DCall: 2 [208.100.1.33:4569]
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: 
REGACK
   Timestamp: 00036ms  SCall: 9  DCall: 2 [208.100.1.33:4569]
   USERNAME: rwestics
   DATE TIME   : 2007-09-13  07:00:54
   REFRESH : 60
   APPARENT ADDRES : IPV4 24.14.116.22:4569
   CALLING NUMBER  : 8159092441
   CALLING NAME: West and Associates

-- Registered IAX2 to '208.100.1.33', who sees us as 24.14.116.22:4569 
with no messages waiting

Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK
   Timestamp: 00036ms  SCall: 2  DCall: 9 [208.100.1.33:4569]
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK
   Timestamp: 00036ms  SCall: 00013  DCall: 3 [208.100.1.33:4569]
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: 
REGACK
   Timestamp: 00040ms  SCall: 00013  DCall: 3 [208.100.1.33:4569]
   USERNAME: ottos
   DATE TIME   : 2007-09-13  07:00:54
   REFRESH : 60
   APPARENT ADDRES : IPV4 24.14.116.22:4569
   CALLING NUMBER  : 8157582715
   CALLING NAME: Ottos Nightclub

-- Registered IAX2 to '208.100.1.33', who sees us as 24.14.116.22:4569 
with no messages waiting

Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK
   Timestamp: 00040ms  SCall: 3  DCall: 00013 [208.100.1.33:4569]
Braxis*CLI> iax2 no debug
IAX2 Debugging Disabled
The 'iax2 no debug' command is deprecated and will be removed in a future 
release. Please use 'iax2 set debug off' instead.
Braxis*CLI> iax2 show peers
Name/UsernameHost Mask Port  Status
rwestics (Unspecified)   (D)  255.255.255.255  0 
Unmonitored
rwest1/rwest1(Unspecified)   (D)  255.255.255.255  0 
Unmonitored
224/224  (Unspecified)   (D)  255.255.255.255  0 
Unmonitored
ics/ottos(Unspecified)   (D)  255.255.255.255  0 UNKNOWN
4 iax2 peers [0 online, 1 offline, 3 unmonitored]



That's what happens after I do a iax2 show peers.  So apparently calls are 
coming in, but showing the peers isn't bringing up any IP addresses.  I can 
also make outbound calls.

So...  apparently Asterisk is working except for the servers aren't showing 
up in the peer list.




-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com


- Original Message - 
From: "Erik Anderson" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Wednesday, September 12, 2007 12:44 PM
Subject: Re: [asterisk-users] Different Networks


> On 9/7/07, Mike Hammett <[EMAIL PROTECTED]> wrote:
>> If it has nothing to do with Asterisk, then why does every other device 
>> work
>> as its supposed to?
>
> You never answered as to whether 

Re: [asterisk-users] Asterisk cli

[Rizwan Hisham]

i connect remotely. I have tried both of these cases but no warnings or
mesages still.

On 9/13/07, Bhrugu Mehta <[EMAIL PROTECTED]> wrote:
>
> Hi,
>
> 1. If you are connecting to remotly with asterisk server you have to use
> asterisk -vvvrc
> 2. if your asterisk server is your pc then you have to use  asterisk
> -c
>
> ok
> enjoy
>
> Bhrugu mehta
>
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-- 
Best Regards
Rizwan Hisham
Software Engineer
Axvoice Inc.
www.axvoice.com
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Re: [asterisk-users] Asterisk cli

[Bhrugu Mehta]

Hi,

1. If you are connecting to remotly with asterisk server you have to use
asterisk -vvvrc
2. if your asterisk server is your pc then you have to use  asterisk
-c

ok
enjoy

Bhrugu mehta

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Re: [asterisk-users] ztdummy problem in fedora7, kernel 2.6.22.5-76.fc7

[Patrick]

On Thu, 2007-09-13 at 15:03 +0500, saqib butt wrote:
> hi there
> 
> i am facing problem in installing the ztdummy module in fedora7,
> 2.6.22.5-76.fc7 is the version of the kernel. here are some logs for
> your kind consideration, 
> i have tried varios solution from voip-info.org and many more, but in
> vain.
[snip]

You need to apply the patch from:
http://bugs.digium.com/file_download.php?file_id=15160&type=bug
to your zaptel source and rebuild zaptel. You can read more here:
http://bugs.digium.com/view.php?id=10314

This should fix your issue and even give you high resolution timers.

Regards,
Patrick


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[asterisk-users] [phpAGI] generate a call from a SIP device to Asterisk

[nik600]

Hi

i need to generate a call from a SIP hardware device to Asterisk.
The device isn't registered with a sip account to Asterisk.

What i've done, is to do this (using phpAGI):

.
$asm->Originate(SIP/[EMAIL PROTECTED],2000,"default","1");
.

And on the extension 2000 in the context "default"

exten => 2000,1,ChanSpy(|g(100))
exten => 2000,2,Hangup

Is it correct ?

or shall i do something else?


-- 
/*/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/nikstresser

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Re: [asterisk-users] Asterisk cli

[Dave Walker]

On Thu, 2007-09-13 at 15:19 +0500, Rizwan Hisham wrote:
> Hi all,
> I dont see any notices or warning anymore on my asterisk cli. but
> these notices and warnings appear in the asterisk log. I have tried
> settings the verbosity and debug levels (core set verbos and core set
> debug) but still they dont appear on cli. Maybe i have accidentally
> turned them. Is that possible? if yes then how do i turn them on? 


Have you tried connecting to asterisk with:
$ asterisk -rvvv

Kind Regards,
Dave Walker


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Re: [asterisk-users] ztdummy problem in fedora7, kernel 2.6.22.5-76.fc7

[Dave Walker]

On Thu, 2007-09-13 at 15:03 +0500, saqib butt wrote:
> hi there
> 
> i am facing problem in installing the ztdummy module in fedora7,
> 2.6.22.5-76.fc7 is the version of the kernel. here are some logs for
> your kind consideration, 
> i have tried varios solution from voip-info.org and many more, but in
> vain.
> Message from "/var/log/messages"
> 
> Sep 13 14:27:14 localhost kernel: Zapata Telephony Interface
> Registered on major
> Sep 13 14:27:14 localhost kernel: Zaptel Version: 1.4.5.1
> Sep 13 14:27:14 localhost kernel: Zaptel Echo Canceller: MG2
> Sep 13 14:27:14 localhost kernel: ztdummy: Unknown symbol rtc_register



Hi Saqib

Strangely it would seem that your kernel does not have a Real Time Clock
(RTC) compiled in.  ztdummy can function fine without it, but
automatically assumes all 2.6.* kernel's to have it.

Therefore you can either recompile your kernel adding RTC support, or
modify ztdummy source.

Try commenting out the instances of "#define USE_RTC"  (add "//" before
it) in zaptel.c.  Then:
make clean
make
make install

Let me know how you get on.

Kind Regards,
Dave Walker


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[asterisk-users] Asterisk cli

[Rizwan Hisham]

Hi all,
I dont see any notices or warning anymore on my asterisk cli. but these
notices and warnings appear in the asterisk log. I have tried settings the
verbosity and debug levels (core set verbos and core set debug) but still
they dont appear on cli. Maybe i have accidentally turned them. Is that
possible? if yes then how do i turn them on?

-- 
Best Regards
Rizwan Hisham
Software Engineer
Axvoice Inc.
www.axvoice.com
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[asterisk-users] ztdummy problem in fedora7, kernel 2.6.22.5-76.fc7

[saqib butt]

hi there

i am facing problem in installing the ztdummy module in fedora7,
2.6.22.5-76.fc7 is the version of the kernel. here are some logs for your
kind consideration,
i have tried varios solution from voip-info.org and many more, but in vain.


Message from "/var/log/messages"

Sep 13 14:27:14 localhost kernel: Zapata Telephony Interface Registered on
major
Sep 13 14:27:14 localhost kernel: Zaptel Version: 1.4.5.1
Sep 13 14:27:14 localhost kernel: Zaptel Echo Canceller: MG2
Sep 13 14:27:14 localhost kernel: ztdummy: Unknown symbol rtc_register
Sep 13 14:27:14 localhost kernel: ztdummy: Unknown symbol rtc_unregister
Sep 13 14:27:14 localhost kernel: ztdummy: Unknown symbol rtc_control

Message from "modprobe ztdummy" at console,

FATAL: Error inserting ztdummy (/lib/modules/2.6.22.5-76.fc7/misc/ztdummy.ko):
Unknown symbol in module, or unknown parameter (see dmesg)

Message from "dmesg"

Zapata Telephony Interface Registered on major 196
Zaptel Version: 1.4.5.1
Zaptel Echo Canceller: MG2
ztdummy: Unknown symbol rtc_register
ztdummy: Unknown symbol rtc_unregister
ztdummy: Unknown symbol rtc_control
ztdummy: Unknown symbol rtc_register
ztdummy: Unknown symbol rtc_unregister
ztdummy: Unknown symbol rtc_control
ztdummy: Unknown symbol rtc_register
ztdummy: Unknown symbol rtc_unregister
ztdummy: Unknown symbol rtc_control



-- 
Regards,

Saqib Butt.
Cell: (0092)3024068471.
E-mail: [EMAIL PROTECTED]
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Re: [asterisk-users] TE405P intermittent yellow alarm

[Richard van der Hoff]

Steve Totaro wrote:
> Richard van der Hoff wrote:
 >> [intermittent yellow alarm]
>> At this point, I'd really like to know what a yellow alarm actually
>> means. I've read that it indicates that that the other end of the E1 is
>> in an alarm condition: however BT's terminating unit seems quite happy
>> with no alarm conditions at all.
>>
> Check your cabling.  Replace it with new stuff.  Re-punch everything. 
> 
> It is obviously somewhere in the line.  If the above does not fix it, 
> maybe you can get a lucky and get a good tech out that will stick around 
> to see the issue.

The only bit of cable I own here is the 2m length of cat-5 between the 
te405P and BT's line terminating unit. And yes, I've replaced that about 
5 times now...

Thanks for your help, but again I'd like to ask: what does a yellow 
alarm actually mean? From the driver source code I can see it is set 
when the FRS0 register has bit 4 set - but that doesn't help a lot...

Regards

Richard

-- 
Richard van der Hoff <[EMAIL PROTECTED]>
Project Manager
Tel: +44 (0) 845 666 7778
http://www.mxtelecom.com

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Re: [asterisk-users] Flash IDE

[Tim Panton]

On 11 Sep 2007, at 12:32, Gordon Henderson wrote:

> On Tue, 11 Sep 2007, Juan Sandro wrote:
>
>>
>> Hi
>>
>> We have a number offices accommodating 4-6 people each hence it is  
>> very
>> important for PBX to be fanless and silent. We have been looking  
>> at using
>> IDE flash disks also called DOM. The performance tests we have  
>> done so far
>> satisfy our requirements, however we are concerned with DOM  
>> durability.
>>
>> We have installed debian and vanilla asterisk on 1GB DOM. All  
>> seems to work
>> fine at the moment however will DOM last? How long it will last?  
>> Is anyone
>> able to share similar experience? Any other information/tips?
>
> You could read the archives from a week or 2 ago under the heading:
>Build your own "appliance"
>
> I use these deices, but I unload them entirely into RAM.
>
> I have seen devices (eary mikrotik routers?) with them as live (and  
> ext3
> no less!) filesystems, but I would be very concerend about their  
> lifespan.
>
> One thing to note and this might well shaft you is that they use  
> POI mode
> rather than DMA (or at least the ones I'm using do) so they will  
> really
> crowbar the bus & cpu when doing transfers to/from them, however  
> with only
> 4-6 people and not doing much like writing voicemail, etc. you may not
> notice it.
>
> If you're sticking a "normal" disctibution on it, I'd suggest  
> dumping the
> DOM and getting a laptop type IDE/SATA drive and using that  
> instead. It's
> not silent, but will be very quiet.
>
> Gordon
>

As an extra option, I have a couple of nslu2's running asterisk 1.4  
very happily.
I'd guess 6 users would be about the limit for these boxes as they  
only have 32Mb
of RAM, but they are small, quiet and cheap. One is running from a  
USB harddrive
and the other from a pen drive. In theory you might be able to  
squeeze Asterisk onto
the builtin flash, but I haven't bothered.

Tim.

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[asterisk-users] fax machine detection for outgoing call on DIVA card

[lemmel]

   Hello,
I need to detect both fax and answer machine, and it should be valuable that 
the detection will be run by the Diva card itself. So :
- I read Diva Documentation, and I found that the Diva could send some 
specific DTMF, if I had "[..] enabled [this functionnality] by the 
application for a designated controller through a manufacturer request 
command 9 [...]", but I didn't figure how to activate it ; has someone an 
idea ?
- I read my capi.conf, and found the faxdetect parameter, which "enable 
faxdetection and redirection to EXTEN fax for incoming and/or outgoing 
calls", but I didn't succeed to perform that[1] ; has someone an idea about 
it ?

[1] I altered my capi.conf file, and put the fax entension just after the 
Dial, and called a fax machine, but nothing happens. The capi log is in the 
attachment.

P.S. : it is the second mail, for I forgot the attachment, and in the archive 
web page 
(http://lists.digium.com/pipermail/asterisk-users/2007-September/thread.html), 
my mail seems to have the wrong id (as I had answer a message instead to make 
a new one) for it belongs to a thread.
*CLI> -- Executing [EMAIL PROTECTED]:1] NoOp("SIP/22-08503e28", " a 
compose le <120> depuis le poste <22> de type < - 0>") in new stack
-- Executing [EMAIL PROTECTED]:2] Dial("SIP/22-08503e28", 
"CAPI/contr2/120/b") in new stack
   > data = contr2/120/b format=8
   > parsed dialstring: 'contr2' 'NULL' '120' 'b'
   > capi request controller = 2
  == contr2#02: setting format alaw - 0x8 (alaw)
   > parsed dialstring: 'contr2' 'NULL' '120' 'b'
   > capi: peerlink -1 allocated, peer is unlinked
   > CAPI devicestate requested for contr2#02/120
  == contr2#02: Call CAPI/contr2#02/120-2 with B3  (pres=0x00, ton=0x00)
CONNECT_REQ ID=002 #0x060c LEN=0048
  Controller/PLCI/NCCI= 0x2
  CIPValue= 0x1
  CalledPartyNumber   = <80>120
  CallingPartyNumber  = <00 80>22
  CalledPartySubaddress   = default
  CallingPartySubaddress  = default
  BProtocol
   B1protocol = 0x1
   B2protocol = 0x1
   B3protocol = 0x0
   B1configuration= default
   B2configuration= default
   B3configuration= default
   GlobalConfiguration= default
  BC  = default
  LLC = default
  HLC = default
  AdditionalInfo
   BChannelinformation= <00 00>
   Keypadfacility = default
   Useruserdata   = default
   Facilitydataarray  = default
   SendingComplete= default

-- Called contr2/120/b
CONNECT_CONF ID=002 #0x060c LEN=0014
  Controller/PLCI/NCCI= 0x302
  Info= 0x0

-- contr2#02: received CONNECT_CONF PLCI = 0x302
INFO_IND ID=002 #0x0687 LEN=0017
  Controller/PLCI/NCCI= 0x302
  InfoNumber  = 0x1e
  InfoElement = <81 88>

INFO_RESP ID=002 #0x0687 LEN=0012
  Controller/PLCI/NCCI= 0x302

-- contr2#02: info element PI 81 88
   > contr2#02: In-band information available
CONNECT_B3_REQ ID=002 #0x060d LEN=0013
  Controller/PLCI/NCCI= 0x302
  NCPI= default

-- contr2#02: sent CONNECT_B3_REQ PLCI=0x302
-- CAPI queue frame:[ TYPE: Control (4) SUBCLASS: Unknown control '14' (14) 
] [contr2#02]
INFO_IND ID=002 #0x0688 LEN=0016
  Controller/PLCI/NCCI= 0x302
  InfoNumber  = 0x18
  InfoElement = <8a>

INFO_RESP ID=002 #0x0688 LEN=0012
  Controller/PLCI/NCCI= 0x302

-- contr2#02: info element CHANNEL IDENTIFICATION 8a
INFO_IND ID=002 #0x0689 LEN=0015
  Controller/PLCI/NCCI= 0x302
  InfoNumber  = 0x800d
  InfoElement = default

INFO_RESP ID=002 #0x0689 LEN=0012
  Controller/PLCI/NCCI= 0x302

-- contr2#02: info element SETUP ACK
CONNECT_B3_CONF ID=002 #0x060d LEN=0014
  Controller/PLCI/NCCI= 0x120302
  Info= 0x0

CONNECT_B3_ACTIVE_IND ID=002 #0x068b LEN=0013
  Controller/PLCI/NCCI= 0x120302
  NCPI= default

CONNECT_B3_ACTIVE_RESP ID=002 #0x068b LEN=0012
  Controller/PLCI/NCCI= 0x120302

-- CAPI/contr2#02/120-2 is making progress passing it to SIP/22-08503e28
INFO_IND ID=002 #0x068c LEN=0015
  Controller/PLCI/NCCI= 0x302
  InfoNumber  = 0x8002
  InfoElement = default

INFO_RESP ID=002 #0x068c LEN=0012
  Controller/PLCI/NCCI= 0x302

-- contr2#02: info element CALL PROCEEDING
-- CAPI queue frame:[ TYPE: Control (4) SUBCLASS: Unknown control '15' (15) 
] [contr2#02]
-- CAPI/contr2#02/120-2 is proceeding passing it to SIP/

Re: [asterisk-users] online active call watching

[Tim H. Panton]

You can also monitor active calls with SNMP. This should be a lower load
than manager (I've no stats to prove it). It is also easier to secure as
the asterisk MIB is read only :-)

I'm in the process of writing a proof of concept that monitors to Asterisk
over snmp.

Tim

- Original Message -
From: "Mojo with Horan & Company, LLC" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Wednesday, September 12, 2007 6:09:40 PM (GMT) Europe/London
Subject: Re: [asterisk-users] online active call watching

Dinesh Nair wrote:
> On Mon, 10 Sep 2007 13:43:46 -0800, Mojo with Horan & Company, LLC wrote:
>
>   
>> Though still in the proof-of-concept stage, my project "AstSee" from 
>> http://www.astsee.com/ might be fun to play with if you're using 
>> linux/XWindows.  There are screenshots there.
>> 
>
> that may be so, but without source, there's no way we can test it on
> freebsd. i'll stick with fop for the timebeing, thank you. 
>
>   
Ok, so source is available now.  Do your worst:  Innovate!


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Re: [asterisk-users] res_snmp

[Tim H. Panton]

Yes, res_snmp seems to be sensitive to the specific version of net_snmp.
I wrote some notes on this - see 
http://www.voip-info.org/wiki/view/Asterisk+monitoring

Basically I ended up installing netsnmp from source, and things started working.

I'm currently writing a little demo program which lets you see calls via snmp
but I'm a bit stuck on the graphical representation.

Which snmp tools will you use to monitor asterisk ?

Tim.

- Original Message -
From: "yonoko molomo" <[EMAIL PROTECTED]>
To: asterisk-users@lists.digium.com
Sent: Wednesday, September 12, 2007 4:51:11 PM (GMT) Europe/London
Subject: [asterisk-users] res_snmp

Hi,

I have problems compiling asterisk 1.4.11 with res_snmp.
I do 'make menuselect', and I see that this resource module depends on netsnmp.
I am using centOS 4.5.
I do:
> yum install net-snmp net-snmp-devel net-snmp-utils net-snmp-libs
I don't know if i am missing something.

I go to the source directory and I do:
./configure

but still does not work:
> ...
> checking for curses.h... (cached) yes
> checking for net-snmp-config... /usr/bin/net-snmp-config
> checking for snmp_register_callback in -lnetsnmp... no
> ...

When i run 'make menuselect' I cannot get rid of the XXX, and i can't
select res_snmp.

I also tried to install net-snmp-perl package but it does not help. i
have no clue how to continue.

any ideas?
thanks

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Re: [asterisk-users] No Sound on Zap Channels

[Atis]

On 9/13/07, Hoai-Anh Ngo-Vi <[EMAIL PROTECTED]> wrote:
> Have you answered the channel?

Voicemail doesn't require Answer(). It does that itself, as you
usually get to voicemail after Dial(). It would be silly to require to
do Answer after each Dial and then send to voicemail.

Regards,
Atis

>
> Von: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Im Auftrag
> von Jon Weisman

> I've got a strange issue here. When I make a SIP call to say my voicemail
> app, I hear audio just fine. However when I dial from PSTN into my Asterisk
> box, I see that its playing the voice files, but I hear nothing, then the
> call drops. I'm running Fedora Core 6, and Asterisk 1.2.24. CLI output
> below. T-1 is PRI, showing normal, dchannel is up as well. Any help is
> greatly appreciated.
>
>
>
>
>
>
>
>
> Thanks,
>
>
> Jon
>
>
>
>
>
>
>
>
>  -- Accepting call from '2125551212' to '6465551212' on channel 0/23, span 4
>  -- Executing VoiceMail("Zap/95-1", "u100") in new stack
>  -- Playing 'vm-theperson' (language 'en')
>  -- Playing 'digits/1' (language 'en')
>  -- Playing 'digits/0' (language 'en')
>  -- Playing 'digits/0' (language 'en')
>  -- Playing 'vm-isunavail' (language 'en')
>  -- Playing 'vm-intro' (language 'en')
>  -- Channel 0/23, span 4 got hangup request, cause 34
>== Spawn extension (default, 6465551212, 1) exited non-zero on 'Zap/95-1'
>  -- Hungup 'Zap/95-1'
> ___
>
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-- 
Atis Lezdins,
IT Responsible of BEST Riga,
[EMAIL PROTECTED]
ICQ: 142239285
Skype: atis.lezdins
Cell Phone: +371 28806004 [Tele2, Latvia]
Work phone: +1 800 7502835 [Toll free, USA]
?BEST? -> www.BEST.eu.org

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Re: [asterisk-users] Zap channels: no sound with certain call paths

[Tzafrir Cohen]

On Wed, Sep 12, 2007 at 09:08:23PM -0400, Christian Weeks wrote:
> Hi,
> A most peculiar and vexing problem for you all. I hope I have been
> verbose enough without being a firehose ;)
> 
> The set up:
> I have a channel bank, using the r1t1 rhino driver with a rhino T1 card
> (the channel bank itself is a very legacy piece of equipment)- this
> supplies FXS for all the house phones. Also, a Wildcard TDM400P, using
> the wctdm module with 1 FXO module, this supplies FXO to the upstream
> telco (a single line).
> 
> The problem:
> Lately, and without any configuration changes, incoming calls that route
> through the Wildcard (from the telco) to the channel bank (well, a phone
> connected to the channel bank) have no voice in either direction.
> Obviously, this is rather frustrating. The same configuration has worked
> quite reliably for the past year or so, so I am reasonably confident
> that the problem isn't directly configuration related (though I have,
> since this started occuring tried various configs).
> 
> The version where this started to occur (intermittently) was
> asterisk/zaptel in debian etch (the 1.2 branch). I have since upgraded
> to zaptel/asterisk from debian sid (the 1.4 branch) and the problems
> have gotten marginally worse.
> 
> Stuff I have tried:
> 1. Zap->Zap (calling one channel bank extn from another) works fine.
> 2. Zap->anywhere (calling out from CB to telco through wildcard, or to
> SIP provider, or IAX provider) works fine.
> 3. telco->Zap (calling in from telco to CB line) fails: no voice.
> 4. SIP/IAX->Zap (calling in from a SIP client to CB line) works.
> 
> Diagnostics examined:
> 1. ztmonitor  -v shows expected signals, from the asterisk
> perspective. But e.g. in scenario 3 above, there is no received voice
> from the zap line. Which is consistent with the dialled CB line not
> being properly connected somehow.
> 
> Oddities noticed:
> 1. Sometimes, when picking up a CB line, there is no dialtone. Only
> resolution has been to reset the computer.
> 2. There are several odd messages in the log files:
> (/var/log/syslog)
> [..snip..]
> Sep 12 17:52:04 phone kernel: Got pulse digit 36 on R1T1/0/3??
> (note: lots of these, at least one per CB line, whenever we restart or
> reprobe the module)

This means many close on-hook/of-hook events. Close enough to create 36
pulse dials. This is from zaptel.ko .

> [..snip..]
> Sep 12 17:53:29 phone asterisk[2638]: rc_avpair_new: unknown attribute
> 1490026597
> (lots of these too, there seems to be a correlation between these
> messages and no voice routings)
> (/var/log/asterisk/messages (I have verbosity up nice and high))

Make sure you have debug enabled and logged if you have strange things
in chan_zap and want to full understand them.

What version of Asterisk is it?

> [Sep 12 20:35:20] WARNING[3174] chan_zap.c: Ring/Off-hook in strange
> state 6 on channel 25
> (I've had this since I set the environment up. No one seems to be able
> to give a sane answer as to why).
> 
> Finally, here's an interesting oddity. I can get the voice to come up,
> in certain circumstances, by doing the following:
> 1. Dial in from telco using cellphone.
> 2. Answer with CB Zap line. No voice.
> 3. Hang up the CB Zap line.
> 4. Re-open any Zap CB line, execute a dial that uses telco line.
> 5. The telco line picks up (to execute the dial); voice is now connected
> to the still waiting original call.
> 
> Here's the log file:
> [Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Starting simple switch
> on 'Zap/25-1'
> [Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Executing
> [EMAIL PROTECTED]:1] Goto("Zap/25-1", "incoming-home|s|1") in new stack
> [Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Goto
> (incoming-home,s,1)
> [Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Executing
> [EMAIL PROTECTED]:1] NoOp("Zap/25-1", """ ") in new stack
> [Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Executing
> [EMAIL PROTECTED]:2] Set("Zap/25-1", "TRANSFER_CONTEXT=transfer") in new
> stack
> [Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Executing
> [EMAIL PROTECTED]:3] GotoIfTime("Zap/25-1", "9:00-20:00|*|*|*?s-DAY|1")
> in new stack
> [Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Goto
> (incoming-home,s-DAY,1)
> [Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Executing
> [EMAIL PROTECTED]:1] Dial("Zap/25-1",
> "Zap/1&Zap/3&Zap/2&Zap/10&Zap/5&Zap/6&SIP/cpw...)
> [Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Called 1
> [Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Called 3
> [Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Called 2
> [Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Called 10
> [Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Called 5
> [Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Called 6
> [Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Called 
> [Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Zap/1-1 is ringing
> [Sep 12 18:23:51] VERBOSE[3051] logger.c: -- Zap/3-1 is ringing
> [Sep 12 18:23:51] VERBOSE[3051] logge

Re: [asterisk-users] No Sound on Zap Channels

[Hoai-Anh Ngo-Vi]

Have you answered the channel?

 

  _  

Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Jon Weisman
Gesendet: Donnerstag, 13. September 2007 05:28
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: [asterisk-users] No Sound on Zap Channels

 

All,

 

I've got a strange issue here. When I make a SIP call to say my voicemail
app, I hear audio just fine. However when I dial from PSTN into my Asterisk
box, I see that its playing the voice files, but I hear nothing, then the
call drops. I'm running Fedora Core 6, and Asterisk 1.2.24. CLI output
below. T-1 is PRI, showing normal, dchannel is up as well. Any help is
greatly appreciated.

 

 

Thanks,

Jon

 

 

 -- Accepting call from '2125551212' to '6465551212' on channel 0/23, span 4
-- Executing VoiceMail("Zap/95-1", "u100") in new stack
-- Playing 'vm-theperson' (language 'en')
-- Playing 'digits/1' (language 'en')
-- Playing 'digits/0' (language 'en')
-- Playing 'digits/0' (language 'en')
-- Playing 'vm-isunavail' (language 'en')
-- Playing 'vm-intro' (language 'en')
-- Channel 0/23, span 4 got hangup request, cause 34
  == Spawn extension (default, 6465551212, 1) exited non-zero on 'Zap/95-1'
-- Hungup 'Zap/95-1'

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[asterisk-users] Support of simple E1 CAS signaling (MFCR-like)

[Lukas Rypl]

I have a legacy device, which has multiple analog POTS lines and two E1
trunks with CAS signaling in 16th timeslot. It is in fact MFCR2
signaling without register signaling. Only the following link states are
signaled:

 TxRx
ABCD  ABCD
Idle1001  1001
Seized  0001
Seize Ack 1101
Answered  0101

ClearBack 1101

ClearFwd1001

Blocked 1101  1101


 I tried to connect it via Digium TE205p card to the Asterisk but I not
sure about the proper signaling. I started with MFCR2 and unicall
library, which has the same basic signaling, but it stops after the
first R2 signal is transmitted from Asterisk because it does not receive
any response (this is expected behavior). My first idea about simple
"disabling" register signaling in MFCR2 library was pretty naive,
because it is really complex piece of code.

 The question is: is there any signaling implemented in Asterisk, which
will work with this type of CAS signaling?

 Thanks for any help.

 Lukas

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