date:20070125

I posed the same question to both our Cisco partner and direct to our Cisco
rep. Neither one could tell me what I would not be able to do with a
non-licenses IP phone. As you probably know the phone will work with out a
license, but that may not be acceptable in Cisco's eyes.

In the end we went with a different vendor anyways.

On 1/25/07, Pavel Jezek [EMAIL PROTECTED] wrote:

I think, ci$co phones can not be even purchased without licence...
btw, what is your reason, to buy ci$co phones, when known issues exist
with this phones, if working with anything other than callmanager? :-\
PJ

Peter Mitchell wrote:

I've got a question regarding Cisco IP Phones and licencing.

When using a third party PBX like asterisk is a licence required for
the Cisco phones ? Has anyone got anything in writing from Cisco to
clarify this ?

Eg can I just use CP-7961G or do I need CP-7961G-CH1 even though I'm
not using Cisco Callmanager ?

http://www.voip-info.org/wiki/view/Exposing+the+Cisco+Call+Manager+License+Scam
says no licence required.

Cisco site mentions All Cisco Unified IP phones require the purchase
of a phone technology license, regardless of call protocol being used.

http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet0900aecd802ff020.html

Cheers
Peter

--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.410 / Virus Database: 268.17.10/651 - Release Date:
24/01/2007

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

--
Bruce
Nortex Networks
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] 1.4 - SLA


There was talk last week that SLA in 1.4 was not working correctly and was
being rewritten for a 1.4.1 release.

On 1/25/07, Bill Gibbs [EMAIL PROTECTED] wrote:


 I have read that 1.4 has shared line appearances, which I assume will
work with Polycom phones.  Has anyone configured this and verified it
working?  I was going to start playing around with it but wanted to see if
anyone else has tackled it yet.



Bill

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users






--
Bruce
Nortex Networks
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] 1.4 - SLA

2007-01-25 Thread Steve Langstaff

On Fri, 12 Jan 2007 07:14:08 -0800 Kevin P. Fleming wrote:
 I won't waste your time, because the current SLA implementation is
 broken. We expect to have replaced it when Asterisk 1.4.1 is released,
 and there will be better documentation at that point as well.




From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs
Sent: 25 January 2007 14:54
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] 1.4 - SLA



I have read that 1.4 has shared line appearances, which I assume
will work with Polycom phones.  Has anyone configured this and verified
it working?  I was going to start playing around with it but wanted to
see if anyone else has tackled it yet.

 

Bill

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Do I need a CH1 licence for Cisco Phones ?

2007-01-25 Thread Cory Andrews

Technically, Cisco requires you to purchase both a Smartnet (To obtain a
CCO login for access to firmware), as well as a SIP/MGCP license token,
to utilize their phones with SIP firmware, regardless of platform.  

The CH1 nomenclature applies to Callmanager, the CCME nomenclature
applies to Callmanager Express.  

The appropriate license for SIP is SW-SM-UL-7960 if you are using a
Cisco 7960G phone.  The Cisco description for this license is SIP 
MGCP LICENSE FOR SINGLE 7960 IP PHONE

The article referenced in the WIKI referenced in this thread is not
entirely correct, in a few aspects.

A - Technically, a user needs a Smartnet for CCO access to firmware, as
well as a SIP/MGCP license to legitimately utilize Cisco's SIP firmware,
irregardless of platform.

B - The WIKI article also states that if you are using Callmanager and
you receive phones from a reseller that do not have a part # on the
label ending in CH1, that they are Spares and not Callmanager licensed
phones.  This is also technically incorrect.  In recent months I have
seen Cisco ship phones classified as CH1 (Callmanager) licensed, without
a CH1 part number on the box label.  Cisco tracks licensing by the
unique serial number of the phone, and I have seen them bulk register
spare phones as CH1 licensed phones, simply by updating their serial
number database accordingly and tagging serial numbers and licensed.  It
amounts to a virtual license, and likely allows Cisco to better manage
their inventory, as they can utilize phones originally produced as
spares, and easily convert them to CH1 licensed phones, just by updating
their serial number database accordingly.

Another common myth is that if you purchase used phones that were
originally sold as CH1 or CCME licensed units, that the license it
transferrable to the new owner of the phone.  According to Cisco, this
is not true, and a user is supposed to bear the cost of re-licensing.

It is quite confusing, and am neither supporting, nor critizing the
model, just relaying my experience.  I manage a business division that
is a Cisco premier partner with Unified CallManager Express
specialization and deal with licensing on a regular basis.


Cory Andrews

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pavel
Jezek
Sent: Thursday, January 25, 2007 8:12 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Do I need a CH1 licence for Cisco Phones ?

I think, ci$co phones can not be even purchased without licence...
btw, what is your reason, to buy ci$co phones, when known issues exist
with this phones, if working with anything other than callmanager?  :-\
PJ


Peter Mitchell wrote:

 I've got a question regarding Cisco IP Phones and licencing.
  
 When using a third party PBX like asterisk is a licence required for 
 the Cisco phones ? Has anyone got anything in writing from Cisco to 
 clarify this ?
  
 Eg can I just use CP-7961G or do I need CP-7961G-CH1 even though I'm 
 not using Cisco Callmanager ?

  


http://www.voip-info.org/wiki/view/Exposing+the+Cisco+Call+Manager+Licen
se+Scam 
 says no licence required.

  

 Cisco site mentions All Cisco Unified IP phones require the purchase 
 of a phone technology license, regardless of call protocol being used.

  


http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet0
900aecd802ff020.html 

  
 Cheers
 Peter


 --
 No virus found in this outgoing message.
 Checked by AVG Free Edition.
 Version: 7.1.410 / Virus Database: 268.17.10/651 - Release Date: 
 24/01/2007




 ___
 --Bandwidth and Colocation provided by Easynews.com --

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
   
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] TE110P and HDLC problems

2007-01-25 Thread Marc Patino Gómez


Hi!,

this issue makes me crazy. I read a lot of docs, also * mailling list 
and I try a lot of things  without success.


Any help will be appreciated. Here is the info:

Hardware:

Supermicro Server with motherboard X7DB8, chipset Intel 5000P, Xeon 5050
Digium TE110P

Software
-
Asterisk version 1.2.12.1
Zaptel version 1.2.8

/etc/zaptel.conf

loadzone=es
defaultzone=es
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31

The dammed errors:

Jan 25 14:11:52 NOTICE[4934]: chan_zap.c:8316 pri_dchannel: PRI got 
event: HDLC Bad FCS (8) on Primary D-channel of span 1
Jan 25 14:11:52 NOTICE[4934]: chan_zap.c:8316 pri_dchannel: PRI got 
event: HDLC Abort (6) on Primary D-channel of span 1
Jan 25 14:11:52 NOTICE[4934]: chan_zap.c:8316 pri_dchannel: PRI got 
event: HDLC Abort (6) on Primary D-channel of span 1

...

I tried the following without success:

- Disable Hyper Threading.
- Disable NIC's, RAID controller, usb, serial... to avoid shared IRQ, so 
TE110P has his own IRQ as shows lspci -vb.

- Also I tried with APIC and without APIC.
..


These HDLC errors appear when I physically loop the E1 interface in the 
Card and also appear, and more frequently, when I connect the E1 circuit 
(from the Telco) to the interface of the Card.



Thanks a lot

--


Marc Patino Gómez
Dpto. Sistemas

Claranet España. Servicios Internet
C/General Almirante 2-28, Torres Cerdá
08014 Barcelona
Tel. Información General: 902 884 633 
Tel. Soporte Técnico: 902 884 622

Fax: +34 93 445 19 20
www.claranet.es

Claranet Group: United Kingdom - Spain - France - Germany - Portugal - 
Netherlands - USA



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Queuing Problem with Asterisk

2007-01-25 Thread George C. Attopany

Hi David,

I removed the 1s as suggested, but it did not work.

Well, I have halted the beep/bleak/single-ring somehow, by taking down the 
inter-tie line between my asterisk and a PANASONIC TDA200.

What I have in place are:

ASTERISK SIDE:  ASTERISK PBX 1.2 + Adit600 Channel bank(which gives analog 
output and also takes PSTN lines into the Asterisk system) 

PANASONIC SIDE:  TDA200 Panasonic Hybrid PABX + PRI Card(this inter-ties to the 
Asterisk system)

Apart from the  bleak/single-ring on the  Queue extensions,  no visible problem 
exist with the inter-tie. 


Thanks.
  - Original Message - 
  From: David Gomillion 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Sent: Thursday, January 25, 2007 1:07 PM
  Subject: Re: [asterisk-users] Queuing Problem with Asterisk





  On 1/25/07, George C. Attopany [EMAIL PROTECTED] wrote:
Hi,

description of problem cut out for brevity



   
member = Zap/9-1
member = Zap/10-1
member = Zap/11-1
member = Zap/12-1
member = Zap/13-1 
member = Zap/14-1
member = Zap/15-1
member = Zap/16-1


  I don't think you want the -1 on the end of each line. Try:
  member = Zap/9
  member = Zap/10
  member = Zap/11
  member = Zap/12
  member = Zap/13
  member = Zap/14
  member = Zap/15
  member = Zap/16


  Hope that helps,

  David


--


  ___
  --Bandwidth and Colocation provided by Easynews.com --

  asterisk-users mailing list
  To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] RE: TDM2400 Hardware Echo Cancel

2007-01-25 Thread Giuffredi

Hi,

 

 

I had the same crackle problem with the same hardware.

Actually for me was a shared IRQ problem.

 

Now that I fixed it the situation is much better (maybe not perfect but
anyway good).

 

Maybe this IRQ are affecting more the Echo Module than the card?

 

Ciao

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Re: How to exit from console?

2007-01-25 Thread Tzafrir Cohen

On Thu, Jan 25, 2007 at 01:37:50PM +0100, Tomislav Parčina wrote:
 In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
   Try safe_asterisk , for an easy way to start asterisk in background, 
  
  a plain 'asterisk' is even better and safer.
  asterisk -U asterisk . is better. 
/etc/init.d/asterisk start
  is similar.
 
 Why is this better than safe_asterisk?

E.g: because you have a valid PID file of the controlling process. If
you actually want to kill it, you can.

And you don't need physical access to the system to get to the one and
only real console. OTOH, if you do have physical access, you have full
control of Asterisk, as you may inject custom dialplan.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] 1.4 - SLA

2007-01-25 Thread Bill Gibbs

Thanks, I had a notebook crash and must have missed that.  Appreciate the 
replies!  I will be patient.

Bill

-Original Message-
From: [EMAIL PROTECTED] on behalf of Steve Langstaff
Sent: Thu 1/25/2007 9:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] 1.4 - SLA

On Fri, 12 Jan 2007 07:14:08 -0800 Kevin P. Fleming wrote:
 I won't waste your time, because the current SLA implementation is
 broken. We expect to have replaced it when Asterisk 1.4.1 is released,
 and there will be better documentation at that point as well.

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs
Sent: 25 January 2007 14:54
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] 1.4 - SLA

I have read that 1.4 has shared line appearances, which I assume
will work with Polycom phones.  Has anyone configured this and verified
it working?  I was going to start playing around with it but wanted to
see if anyone else has tackled it yet.

Bill

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] IAX softphone fails through PRI trunks with Hangup

2007-01-25 Thread Patrick W. Foster

I've a call center using IAX softphones provided by a third party.  
We've observed problems where the IAX phones seem unable to use our PRI 
trunks.  A sample anonymized call is provided below with the PRI debug 
calls embedded.  Any thoughts,
comments or suggestions would be welcome.  In anonymizing it, I preseved 
the format

and number of digits sent.

   -- Accepting AUTHENTICATED call from 192.168.1.164:
   requested format = alaw,
   requested prefs = (),
   actual format = ulaw,
   host prefs = (ulaw|alaw|gsm),
   priority = mine
   -- Executing Set(IAX2/4427-1, EMERGENCYROUTE=YES) in new stack
   -- Executing Macro(IAX2/4427-1, dialout-trunk|1|6167X||) in 
new stack

   -- Executing GotoIf(IAX2/4427-1, 1?3:2) in new stack
   -- Goto (macro-dialout-trunk,s,3)
   -- Executing Macro(IAX2/4427-1, user-callerid) in new stack
   -- Executing GotoIf(IAX2/4427-1, 0?report) in new stack
   -- Executing GotoIf(IAX2/4427-1, 0?start) in new stack
   -- Executing Set(IAX2/4427-1, REALCALLERIDNUM=4427) in new stack
   -- Executing NoOp(IAX2/4427-1, REALCALLERIDNUM is 4427) in new stack
   -- Executing Set(IAX2/4427-1, AMPUSER=4427) in new stack
   -- Executing Set(IAX2/4427-1, AMPUSERCIDNAME=USER18-IAX) in new 
stack

   -- Executing GotoIf(IAX2/4427-1, 0?report) in new stack
   -- Executing Set(IAX2/4427-1, CALLERID(all)=USER18-IAX 4427) 
in new stack
   -- Executing NoOp(IAX2/4427-1, Using CallerID USER18-IAX 
4427) in new stack

   -- Executing Macro(IAX2/4427-1, record-enable|4427|OUT) in new stack
   -- Executing GotoIf(IAX2/4427-1, 0  0?2:4) in new stack
   -- Goto (macro-record-enable,s,4)
   -- Executing AGI(IAX2/4427-1, 
recordingcheck|20070125-102531|1169738731.2435) in new stack

   -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
 recordingcheck|20070125-102531|1169738731.2435: Outbound recording not 
enabled

   -- AGI Script recordingcheck completed, returning 0
   -- Executing NoOp(IAX2/4427-1, No recording needed) in new stack
   -- Executing Macro(IAX2/4427-1, outbound-callerid|1) in new stack
   -- Executing GotoIf(IAX2/4427-1, 1?start) in new stack
   -- Goto (macro-outbound-callerid,s,3)
   -- Executing NoOp(IAX2/4427-1, REALCALLERIDNUM is 4427) in new stack
   -- Executing Set(IAX2/4427-1, USEROUTCID=8xx-6xx-) in new 
stack

   -- Executing Set(IAX2/4427-1, EMERGENCYCID=) in new stack
   -- Executing Set(IAX2/4427-1, TRUNKOUTCID=Business Name 
5xx-6xx-) in new stack

   -- Executing GotoIf(IAX2/4427-1, 0?trunkcid) in new stack
   -- Executing GotoIf(IAX2/4427-1, 1?trunkcid) in new stack
   -- Goto (macro-outbound-callerid,s,11)
   -- Executing GotoIf(IAX2/4427-1, 0?usercid) in new stack
   -- Executing Set(IAX2/4427-1, CALLERID(all)=Business Name 
5xx-6xx-) in new stack

   -- Executing GotoIf(IAX2/4427-1, 0?report) in new stack
   -- Executing Set(IAX2/4427-1, CALLERID(all)=8xx-6xx-) in 
new stack
   -- Executing NoOp(IAX2/4427-1, CallerID set to  8xx6xx) 
in new stack

   -- Executing Set(IAX2/4427-1, GROUP()=OUT_1) in new stack
   -- Executing GotoIf(IAX2/4427-1, 0?108) in new stack
   -- Executing Set(IAX2/4427-1, DIAL_NUMBER=6167X) in new stack
   -- Executing Set(IAX2/4427-1, DIAL_TRUNK=1) in new stack
   -- Executing AGI(IAX2/4427-1, fixlocalprefix) in new stack
   -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
   -- AGI Script fixlocalprefix completed, returning 0
   -- Executing Set(IAX2/4427-1, OUTNUM=6167X) in new stack
   -- Executing Set(IAX2/4427-1, custom=ZAP/g1) in new stack
   -- Executing GotoIf(IAX2/4427-1, 0?16) in new stack
   -- Executing Dial(IAX2/4427-1, ZAP/g1/6167X|150|r) in new stack
-- Making new call for cr 33745
   -- Requested transfer capability: 0x00 - SPEECH
 Protocol Discriminator: Q.931 (8)  len=46
 Call Ref: len= 2 (reference 977/0x3D1) (Originator)
 Message type: SETUP (5)
 [04 03 80 90 a2]
 Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer 
capability: Speech (0)
  Ext: 1  Trans mode/rate: 64kbps, 
circuit-mode (16)

  Ext: 1  User information layer 1: u-Law (34)
 [18 03 a9 83 84]
 Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, 
Exclusive Dchan: 0

ChanSel: Reserved
   Ext: 1  Coding: 0   Number Specified   Channel 
Type: 3

   Ext: 1  Channel: 4 ]
 [1e 02 80 83]
 Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard 
(0) 0: 0   Location: User (0)
   Ext: 1  Progress Description: Calling 
equipment is non-ISDN. (3) ]

 [6c 0c 21 81 38 30 30 36 39 35 39 38 39 37]
 Calling Number (len=14) [ Ext: 0  TON: National Number (2)  NPI: 
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
   Presentation: Presentation permitted, user 
number passed network screening (1) '8xx6xx' ]

 [70 0b 80 36 31 36 37 38 34 32 37 36 37]
 Called Number (len=13) [ Ext: 1  TON

[asterisk-users] dialplan and *

2007-01-25 Thread Giedrius Augys


Hi,
 I'm analyzing freepbx extensions. When creating ivr with freepbx, it
writes like this:
exten = ,1,Answer
exten = ,n,GotoIf($[${CONTEXT}=from-internal]?USERCID:SETCID)
exten = ,n(USERCID),Macro(user-callerid,)
exten = ,n(SETCID),Set(CALLERID(name)=${CALLERIDNAME})
exten =
,n,Set(MONITOR_FILENAME=/var/spool/asterisk/monitor/q${EXTEN}-${TIMESTAMP}-${UNIQUEID})
exten = ,n,Queue(|t|||300)
exten = *,1,Macro(agent-add,,)
exten = **,1,Macro(agent-del,,)

So my question is , what means these one/two asteriks (*,**
).Maybe it is like priority.?
Thanks
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Thomson ST2030S and BLF

2007-01-25 Thread Andrew Joakimsen


I know of the call pickup issues but what asterisk issue and what BLF issue?

On 1/25/07, Alberto Pastore [EMAIL PROTECTED] wrote:

Andrew Joakimsen ha scritto:
 Actually I noticed just three days ago there is a new release, and the
 releae notes seem to address

 Disable TrMail and Pickup keys
 Disable call progress indication
 ___
but it does not address poor guys' troubles with asterisk, blf and
call pickup...


BTW, anyone of you having problems also with RTTTL melodies?
My ST2030S phones seem to playback a RTTL melody at 1/5 its
original speed.

(I know this is not vital, but as soon as my users discover
the possibility of uploading rtttl ringtones, they begin
annoying me by asking how they work)
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] background() with m option

2007-01-25 Thread Jack Wei


Hi...

In my dialplan, I have the following:

exten = s,1,Background(${RECORDING}|m)
exten = s,n,Voicemail(${DID_NO})
exten = 0,1,Voicemail(${DID_NO})
exten = a,1,VoiceMailMain(${DID_NO})
exten = h,1,Hangup

In version 1.2, when I hit 0 during the playback, I will be directed 
to voicemail. But in verison 1.4, the call hangs up.


[Jan 24 16:05:37] DTMF[5754]: channel.c:2148 __ast_read: DTMF begin '0' 
received on SIP/5060-08c53e68
[Jan 24 16:05:37] DTMF[5754]: channel.c:2128 __ast_read: DTMF end '0' 
received on SIP/5060-08c53e68
== Spawn extension (play_recording, s, 1) exited non-zero on 
'SIP/5060-08c53e68'
-- Executing [EMAIL PROTECTED]:1] Hangup(SIP/5060-08c53e68, ) in new 
stack
== Spawn extension (play_recording, h, 1) exited non-zero on 
'SIP/5060-08c53e68'



Does anyone tell me why this is happening?

Thanks,

Jack
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk very slow when internet down

2007-01-25 Thread Steve Davies


On 1/25/07, Peter Mitchell [EMAIL PROTECTED] wrote:


Has anyone seen this issue with asterisk running like a dog when the
internet is down ?  Internal calls, incoming ISDN calls etc all seem to be
affected.  There is a local DNS server that is always available so I'm not
sure why asterisk is so unresponsive.


How fast does the DNS server respond to queries made by the asterisk
box? Perhaps set up a trace somewhere to discover what queries are
being made and what/when the responses come back.

Cheers,
Steve
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Asterisk 1.4 problem with ztdummy and MeetMe()

2007-01-25 Thread Stefan Wintermeyer


Hi,

when I build zaptel-1.2 and asterisk-1.2 I can modprobe ztdummy and  
start asterisk to be able to use MeetMe().


When I build zaptel-1.4 and asterisk-1.4 I can modprobe ztdummy and  
start asterisk but I am not able to use MeetMe().


What do I miss?

  Stefan

--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
  Asterisk? - http://www.das-asterisk-buch.de


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] NAT solutions

2007-01-25 Thread Brad Templeton

On Wed, Jan 24, 2007 at 11:09:21PM -0800, Yuan LIU wrote:
 From: Brad Templeton [EMAIL PROTECTED]
 
 On Mon, Jan 22, 2007 at 09:59:06AM +, Tim Panton wrote:
  In the meanwhile, use IAX, which understands about NAT pretty well.
  If you have multiple SIP phones on a LAN behind a NATing router, just
  put a small asterisk box on the LAN. It can manage your hairpin
  calls internally, save you bandwidth by trunking the IAX traffic
  to the central asterisk and avoid all the NAT hassle by using
  a single port (outgoing) and refreshing it often enough for the
  router to hold it open.
 
  Tim Panton
 
  www.mexuar.net
  www.westhawk.co.uk/
 
 IAX is a fine protocol as far as it goes, however this answer
 is really not a workable one.   There are only a few IAX phones,
 and they are not nearly as solid and full featured as the many
 SIP phones.   There are some IAX termination and origination
 providers, but there are far more SIP providers.
 ...
 IAX is great but SIP is also a reality, and putting
 Asterisk into the just works category is a really
 important milestone.  One I think that is intended
 to be improved a lot for 1.6.
 
 I have a really dumb question.  It appears that Yahoo, MSN, AIM, you name 
 them, they don't have a NAT problem, and some use SIP.  I don't think they 
 all stay in voice path, either.  What takes?

When you control both ends of the path, you can eliminate all NAT
problems.  Skype also deals almost perfectly with NAT (by using
other nodes as relays if necessary) as does IAX.   SIP was designed
without much attention to NAT and it's had to be added on later and
the different phones are all at different levels of implementation.

Some time ago, actually, the SIP and SDP groups devised the ICE
protocol for highly reliable NAT penetration, but it is still some
distance from wide adoption, and I don't know when anybody will code
up Asterisk adoption.

Larger services like you describe often solve NAT by relaying traffic
through their servers.   They use a trick, that if they suspect
an endpoint is behind NAT, they just ignore what they see in the
SDP, and send all traffic back to the source port/host that the
traffic comes from.  For RTP, they wait for packets to arrive at
the (external, routable) RTP port they provided, and send the
traffic back there instead of the often unroutable address in
the SDP.

Asterisk, if you set nat=yes, will do step 1 (SIP traffic back
to the source it came from, ignoring Contact header) but it does
not yet do the same for the RTP.   If it did, you would be unlikely
to get NAT trouble on phone to Asterisk calls, or calls hairpinned
through Asterisk.

But you don't want to hairpin unless absolutely necessary.  It costs
bandwidth and adds latency.  Latency no only makes calls annoying,
it increases the chance of echo.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] dialplan and *

2007-01-25 Thread Time Bandit


exten = ,n,Queue(|t|||300)
exten = *,1,Macro(agent-add,,)
exten = **,1,Macro(agent-del,,)

So my question is , what means these one/two asteriks (*,**
).Maybe it is like priority.?

It means that to login as an agent on the queue you have to dial
* and to logout you dial **.

hth
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] unable to create channel, in strange state, exited non-zero, etc.

2007-01-25 Thread Wayne Jensen


I'm having various issues that may or may not be related to each other (I'm
pretty sure they are).  We've had this system for a year now (quad T1 card,
right now we have 1 T1 coming in, 2 going out to channel banks) and we've
had intermittent ghost calls--it appears that what is happening is a call
is made, the number being called is disconnected/busy/answering
machine/whatever, we hang up and the phone starts ringing.  Answer the phone
and it's that same call still not hung up.

That problem hadn't happened very often so we didn't worry too much about
it.  It was just a little annoying.  We had pretty low traffic on the system
though and were mostly still using our old phone system.

On Monday I switched so that all of the phones are going through this
Asterisk system and Tuesday morning we started having major problems.  Calls
were being dropped in the middle of the call.  At times, everyone who was
trying to make a call would get a fast busy and Unable to create channel
of type 'Zap' showed up in the logs.

Yesterday we had various Red and Yellow alarms.  Here are a few lines from
the logs:

Yesterday:
Jan 24 10:01:47 WARNING[12435] chan_zap.c: Detected alarm on channel 24: Red
Alarm
Jan 24 10:02:11 NOTICE[25813] app_dial.c: Unable to create channel of type
'Zap' (cause 34 - Circuit/channel congestion)

Today:
Jan 25 08:39:23 WARNING[6863] chan_zap.c: zt hook failed: Device or resource
busy
Jan 25 08:40:32 WARNING[932] chan_zap.c: Ring/Off-hook in strange state 6 on
channel 2


From console:

 == Spawn extension (phones-agent, 1510272, 2) exited non-zero on
'Zap/95-1'
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] On-hold calls dropped when new call comes in

2007-01-25 Thread Lars George


Hi,

We have a very basic setup of Asterisk 1.2 with a 4 inbound line Digium card. The phones are 
Grandstream GXP-2000 with the latest stable firmware.


When we get calls and put them on hold and then get a new external call coming in, it drops the 
person on hold. They just get disconnected.


What could be wrong?

Note: we have changed the phones firmware a few times but that made no difference. We are even now 
connected to different external telephone lines and it stayed the same. So somehow I think it is our 
Asterisk config.


Regards,
Lars
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] SVN trunk synchro failure

2007-01-25 Thread Administrator TOOTAI


Hi,

does anyone have some informations on when the SVN repository of 
digium.com will be synchronized again? Since few days we are sticked 
with trunk #51363.


--
Daniel
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Digium Forums

2007-01-25 Thread Dovid B

- Original Message - 
From: Paul [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Wednesday, January 24, 2007 4:23 PM
Subject: Re: [asterisk-users] Digium Forums

Dovid B wrote:

Hi List,
Does anyone know where I can get support for the digium forums ? my
user ID and pass just stoped working as of yesterday. The forums say
to go to asterisk.org for any password issues. I am able to log in
there with out any issues. For some reason when I try to log in to the
forums it wont accept it. Anyone have an ideas ?

A forum user reported that his user ID got changed. It looks like maybe
they merged in the AsteriskNow user/passwords. I know that when I login
with the user/password I created to download AsteriskNow, it works for
forum access.

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

I can log in to asterisk and asterisknow with out a problem. I just cant use 
the forums.
Ding Dong. Any Digium people here ? 

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] setting up AMD

2007-01-25 Thread Asterisk

On Wed, 2007-01-24 at 12:20 -0800, Michael Collins wrote:
  
 
  
 

 __
 From:[EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Peter
 Halliday
 Sent: Wednesday, January 24, 2007 11:56 AM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] setting up AMD
 
 
  
 
 I'm trying get this working.  I've looked through the list, and can't
 see how to get AMD to print out more.  I have it call and say Hello
 like I normally would.  I've tried to say more and less doesn't seem
 to matter.  After I hangup it does recognize hangup.  Here's logging
 during an attempt where I make outbound call and answer, but then
 hangup after 1-2 seconds: 
 
 Jan 24 17:01:37 VERBOSE[31455] logger.c: -- AMD:
 SIP/sip.broadvoice.com-098c4aa8 6079362172 (null) (Fmt: 4)
 Jan 24 17:01:37 VERBOSE[31455] logger.c: -- AMD: initialSilence
 [8000] greeting [1500] afterGreetingSilence [300] totalAnalysisTime
 [5000] minimumWordLength [120] betweenWordsSilence [50]
 maximumNumberOfWords [5] silenceThreshold [256] 
 Jan 24 17:01:44 DEBUG[31437] chan_sip.c: Auto destroying call
 '[EMAIL PROTECTED]'
 Jan 24 17:01:45 DEBUG[31447] manager.c: Manager received command
 'Command'
 Jan 24 17:01:45 DEBUG[31447] manager.c: Manager received command
 'Command'
 Jan 24 17:01:45 DEBUG[31447] manager.c: Manager received command
 'Command' 
 Jan 24 17:01:56 VERBOSE[31455] logger.c: -- AMD: HANGUP
 Jan 24 17:01:56 DEBUG[31455] app_amd.c: Got hangup
 
 The amd.conf:
 [amd]
 initial_silence= 3500
 greeting   = 1500 
 after_greeting_silence = 300
 total_analysis_time= 5000
 min_word_length= 120
 between_words_silence  = 50
 maximum_number_of_words= 5
 silence_threshold  = 256
 
 In extensions.conf
 [outboundmsg1]
 exten = s,1,NoCDR
 exten = s,n,AMD



 exten = s,n,GotoIf($[${AMDSTATUS}=AMD_PERSON]?humn:mach)



*
exten = s,n,GotoIf($[${AMDSTATUS}=HUMAN]?humn:mach)
*

 exten = s,n(mach),WaitForSilence(2500)
 exten = s,n,Playback(outboundmsgs/msg1) 
 exten = s,n,Hangup
 exten = s,n(humn),WaitForSilence(500)
 exten = s,n,Playback(outboundmsgs/msg1)
 exten = s,n,Hangup
 
 
 
 Peter,
 
 It looks like your initial silence setting might be having trouble.
  The amd.conf file has a value of 3500 but the log file is showing
 8000.  Try changing the amd.conf to something like 3000 and issue a
 “reload” at the CLI. Make another test call and see if the trace still
 shows 8000 for the initial silence.  I think having an initial silence
 value that is longer than the total analysis time might be causing the
 undesired behavior.
 
  
 
 Let us know what happens when you try to modify the initial silence
 value.  
 
  
 
 -MC
 
 
 ___
 --Bandwidth and Colocation provided by Easynews.com --
 
 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users



Try to replace the AMD_PERSON with HUMAN as depicted above between 






The AMD_STATUS that works for me is not person but human.



dave




___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Cannot xfer parked callers


Here's how it's currently working:

1) Call comes in
2) Operator parks call (700)
3) Operator picks up call on another phone (701)
4) Operator tries to transfer to a different phone (we use #0) but the 
transfer doesn't work.


We can transfer initial callers all we want and it works fine.  Once a 
call is parked, however, we can no longer transfer the caller.


Any ideas?

Thanks,
Jay
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] low audio (sometimes)

2007-01-25 Thread Jerry Geis


Hi all,

I am using asterisk 1.2.14 release on a 3GIG box/1 GIG RAM with a 
TDM2400E card.

For the most part my echo problems are gone (I have not noticed any issue).
The problem I have is SOMETIMES I get really low audio.
This typically happens when a call is coming in to the TDM2400E card
and going back out the same card to my cell phone.

My card is alone on its interrupt.

What might I look at or try to fix this issue?

Thanks

Jerry
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Cannot xfer parked callers


Jay,

there is a bug in Mantis regarding this, a change was made to allow native
bridging of parked calls. The change has been reverted in a more recent SVN
version of 1.2. See http://bugs.digium.com/view.php?id=8804

On 1/25/07, Jay Moore [EMAIL PROTECTED] wrote:


Here's how it's currently working:

1) Call comes in
2) Operator parks call (700)
3) Operator picks up call on another phone (701)
4) Operator tries to transfer to a different phone (we use #0) but the
transfer doesn't work.

We can transfer initial callers all we want and it works fine.  Once a
call is parked, however, we can no longer transfer the caller.

Any ideas?

Thanks,
Jay
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users





--
Bruce
Nortex Networks
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Adding 4 more POTS lines

2007-01-25 Thread Jim Freeze


Hello

I have a working * server with a TDM card and 4 FXO ports.
We have 4 lines now and need to add 2 more lines (and possibly two more
later).

I'm wondering the best upgrade path for this situation.

The simplest I can invision is adding another TDM400 card with
4 FXO ports, and use 2 now and the remaining 2 later.

Are there success stories with using 2 TDM cards?
Any info will be appreciated.

Thanks
--
Jim Freeze
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] IMAP Voicemail Storage


I am doing some testing with 1.4 and the imap storage and a exchange 2003
server. I have not had any positive results so far using the notes on the
wiki or the docs in the release. My current settings are

imapserver=server
imapport=143
imapfolder=Voicemail
;imapflags=novalidate-cert
expungeonhangup=no

[default]
1114 =1114,Bruce Reeves,[EMAIL PROTECTED]
,,attach=yes|imapuser=breeves|imappasswd=secret

I have also tried specifying a global account with
authuser=pbx
authpassword=secret

In either case I get a

[Jan 25 13:37:43] ERROR[3564]: app_voicemail.c:8417 mm_log: IMAP Error:
Login aborted
[Jan 25 13:37:43] ERROR[3564]: app_voicemail.c:4647 init_mailstream: Can't
connect to imap server {server:143/imap/user=breeves}INBOX
[Jan 25 13:37:43] ERROR[3564]: app_voicemail.c:2529 inboxcount: IMAP
mailstream is NULL

Can someone with this working or who sees the problem point me in the right
direction to get this working.

--
Bruce
Nortex Networks
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Cannot xfer parked callers


Bruce,

I'm running 1.2.14.  I am not willing to switch to 1.4 yet due to the 
stability issue.  From what I read on the page you linked, I could not 
find what version had the supposed fix.  I also can't seem to find a 
later 1.2 version of Asterisk (if one exists).


Any suggestions?

Thanks,
Jay

Bruce Reeves wrote:

Jay,

there is a bug in Mantis regarding this, a change was made to allow native
bridging of parked calls. The change has been reverted in a more recent SVN
version of 1.2. See http://bugs.digium.com/view.php?id=8804

On 1/25/07, Jay Moore [EMAIL PROTECTED] wrote:


Here's how it's currently working:

1) Call comes in
2) Operator parks call (700)
3) Operator picks up call on another phone (701)
4) Operator tries to transfer to a different phone (we use #0) but the
transfer doesn't work.

We can transfer initial callers all we want and it works fine.  Once a
call is parked, however, we can no longer transfer the caller.

Any ideas?

Thanks,
Jay
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users








___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Adding 4 more POTS lines


Jim,

I have 2 TDM400s in my * box (as well as a T1 card).  I use all 8 ports, 
and aside from some minor echoing during peak periods, it's running 
smooth as ice.


Jay

Jim Freeze wrote:

Hello

I have a working * server with a TDM card and 4 FXO ports.
We have 4 lines now and need to add 2 more lines (and possibly two more
later).

I'm wondering the best upgrade path for this situation.

The simplest I can invision is adding another TDM400 card with
4 FXO ports, and use 2 now and the remaining 2 later.

Are there success stories with using 2 TDM cards?
Any info will be appreciated.

Thanks




___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] IMAP Voicemail Storage

On Thursday 25 January 2007 3:27 pm, Bruce Reeves wrote:
 I am doing some testing with 1.4 and the imap storage and a exchange 2003
 server. I have not had any positive results so far using the notes on the
 wiki or the docs in the release. My current settings are

I've done some work with the IMAP voicemail storage and Courier-IMAP, and have 
had it working.

It does seem like it just cannot get to your IMAP server; have you tried the 
imaptools test program (the name escapes me, it's the only binary produced by 
the imaptools package), giving it the same IMAP connection string as what 
Asterisk reports?

My development box is offline at the moment or I could give you specific 
details.

-A.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] TE110P and HDLC problems

2007-01-25 Thread Matthew Fredrickson

There was a recent driver fix that *might* help you.  It's not in an  
official 1.x.x release yet, but if you check out 1.2 from svn, you  
should get the latest version of the driver with the fix.


Matthew Fredrickson

On Jan 25, 2007, at 9:15 AM, Marc Patino Gómez wrote:


Hi!,

this issue makes me crazy. I read a lot of docs, also * mailling list  
and I try a lot of things  without success.


Any help will be appreciated. Here is the info:

Hardware:

Supermicro Server with motherboard X7DB8, chipset Intel 5000P, Xeon  
5050

Digium TE110P

Software
-
Asterisk version 1.2.12.1
Zaptel version 1.2.8

/etc/zaptel.conf

loadzone=es
defaultzone=es
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31

The dammed errors:

Jan 25 14:11:52 NOTICE[4934]: chan_zap.c:8316 pri_dchannel: PRI got  
event: HDLC Bad FCS (8) on Primary D-channel of span 1
Jan 25 14:11:52 NOTICE[4934]: chan_zap.c:8316 pri_dchannel: PRI got  
event: HDLC Abort (6) on Primary D-channel of span 1
Jan 25 14:11:52 NOTICE[4934]: chan_zap.c:8316 pri_dchannel: PRI got  
event: HDLC Abort (6) on Primary D-channel of span 1

...

I tried the following without success:

- Disable Hyper Threading.
- Disable NIC's, RAID controller, usb, serial... to avoid shared IRQ,  
so TE110P has his own IRQ as shows lspci -vb.

- Also I tried with APIC and without APIC.
..


These HDLC errors appear when I physically loop the E1 interface in  
the Card and also appear, and more frequently, when I connect the E1  
circuit (from the Telco) to the interface of the Card.



Thanks a lot

--  
--- 
-


Marc Patino Gómez
Dpto. Sistemas

Claranet España. Servicios Internet
C/General Almirante 2-28, Torres Cerdá
08014 Barcelona
Tel. Información General: 902 884 633 Tel. Soporte Técnico: 902 884 622
Fax: +34 93 445 19 20
www.claranet.es

Claranet Group: United Kingdom - Spain - France - Germany - Portugal -  
Netherlands - USA


--- 
-


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Cannot xfer parked callers


Jay,

The proble is in both 1.2.14 and 1.4 the fix mentioned in the bug was added
to the svn revisions of both versions. If you are not wanting to switch from
1.2.14 to 1.2 svn the you can edit the features.c file and add the lines
mentioned in the notes back to the file, then make and make install.



On 1/25/07, Jay Moore [EMAIL PROTECTED] wrote:


Bruce,

I'm running 1.2.14.  I am not willing to switch to 1.4 yet due to the
stability issue.  From what I read on the page you linked, I could not
find what version had the supposed fix.  I also can't seem to find a
later 1.2 version of Asterisk (if one exists).

Any suggestions?

Thanks,
Jay

Bruce Reeves wrote:
 Jay,

 there is a bug in Mantis regarding this, a change was made to allow
native
 bridging of parked calls. The change has been reverted in a more recent
SVN
 version of 1.2. See http://bugs.digium.com/view.php?id=8804

 On 1/25/07, Jay Moore [EMAIL PROTECTED] wrote:

 Here's how it's currently working:

 1) Call comes in
 2) Operator parks call (700)
 3) Operator picks up call on another phone (701)
 4) Operator tries to transfer to a different phone (we use #0) but the
 transfer doesn't work.

 We can transfer initial callers all we want and it works fine.  Once a
 call is parked, however, we can no longer transfer the caller.

 Any ideas?

 Thanks,
 Jay
 ___
 --Bandwidth and Colocation provided by Easynews.com --

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users





 

 ___
 --Bandwidth and Colocation provided by Easynews.com --

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users





--
Bruce
Nortex Networks
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] RE: TDM2400 Hardware Echo Cancel

2007-01-25 Thread Nick Whitaker

I'm having the same crackle/static issue.  Seems more noticeable on
outbound calls over inbound ones.  I'm running a TDM2400 with 7 FXO
lines currently in use.  Card is on it's own IRQ, Athlon 3200 processor,
Nvidia chipset.  It's somewhat intermittent - much like my zttest
results - I'm not sure if they are related or not.
 
-Nick


 
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] setting up AMD

2007-01-25 Thread Peter Halliday


Where can I get the latest copy of this file.  I thought google found
ithere, but it doesn't compile correctly on 1.2.14.  And the copy on
voip-info.org that I found initially appears to be old.  It's not in
the 1.2tree.



On 1/25/07, Asterisk [EMAIL PROTECTED] wrote:


On Wed, 2007-01-24 at 12:20 -0800, Michael Collins wrote:





 __
 From:[EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Peter
 Halliday
 Sent: Wednesday, January 24, 2007 11:56 AM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] setting up AMD




 I'm trying get this working.  I've looked through the list, and can't
 see how to get AMD to print out more.  I have it call and say Hello
 like I normally would.  I've tried to say more and less doesn't seem
 to matter.  After I hangup it does recognize hangup.  Here's logging
 during an attempt where I make outbound call and answer, but then
 hangup after 1-2 seconds:

 Jan 24 17:01:37 VERBOSE[31455] logger.c: -- AMD:
 SIP/sip.broadvoice.com-098c4aa8 6079362172 (null) (Fmt: 4)
 Jan 24 17:01:37 VERBOSE[31455] logger.c: -- AMD: initialSilence
 [8000] greeting [1500] afterGreetingSilence [300] totalAnalysisTime
 [5000] minimumWordLength [120] betweenWordsSilence [50]
 maximumNumberOfWords [5] silenceThreshold [256]
 Jan 24 17:01:44 DEBUG[31437] chan_sip.c: Auto destroying call
 '[EMAIL PROTECTED]'
 Jan 24 17:01:45 DEBUG[31447] manager.c: Manager received command
 'Command'
 Jan 24 17:01:45 DEBUG[31447] manager.c: Manager received command
 'Command'
 Jan 24 17:01:45 DEBUG[31447] manager.c: Manager received command
 'Command'
 Jan 24 17:01:56 VERBOSE[31455] logger.c: -- AMD: HANGUP
 Jan 24 17:01:56 DEBUG[31455] app_amd.c: Got hangup

 The amd.conf:
 [amd]
 initial_silence= 3500
 greeting   = 1500
 after_greeting_silence = 300
 total_analysis_time= 5000
 min_word_length= 120
 between_words_silence  = 50
 maximum_number_of_words= 5
 silence_threshold  = 256

 In extensions.conf
 [outboundmsg1]
 exten = s,1,NoCDR
 exten = s,n,AMD



 exten = s,n,GotoIf($[${AMDSTATUS}=AMD_PERSON]?humn:mach)



*
exten = s,n,GotoIf($[${AMDSTATUS}=HUMAN]?humn:mach)
*

 exten = s,n(mach),WaitForSilence(2500)
 exten = s,n,Playback(outboundmsgs/msg1)
 exten = s,n,Hangup
 exten = s,n(humn),WaitForSilence(500)
 exten = s,n,Playback(outboundmsgs/msg1)
 exten = s,n,Hangup



 Peter,

 It looks like your initial silence setting might be having trouble.
  The amd.conf file has a value of 3500 but the log file is showing
 8000.  Try changing the amd.conf to something like 3000 and issue a
 reload at the CLI. Make another test call and see if the trace still
 shows 8000 for the initial silence.  I think having an initial silence
 value that is longer than the total analysis time might be causing the
 undesired behavior.



 Let us know what happens when you try to modify the initial silence
 value.



 -MC


 ___
 --Bandwidth and Colocation provided by Easynews.com --

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users



Try to replace the AMD_PERSON with HUMAN as depicted above between






The AMD_STATUS that works for me is not person but human.



dave




___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Adding 4 more POTS lines

2007-01-25 Thread Leif Neland


Jim Freeze wrote:

Hello

I have a working * server with a TDM card and 4 FXO ports.
We have 4 lines now and need to add 2 more lines (and possibly two
more later).

I'm wondering the best upgrade path for this situation.

The simplest I can invision is adding another TDM400 card with
4 FXO ports, and use 2 now and the remaining 2 later.

Are there success stories with using 2 TDM cards?
Any info will be appreciated.


I have no experience with the TDM cards, but costwise it is not the best 
solution, in my opinion.


A TDM04B (4FXO) cost around $378 at voiplink.com, while a  Grandstream 
GXW-4104 (also 4FXO) cost $250 at voxilla.com., and the 8FXO version costs 
$400, almost the same as the 4FXO card.


Having the pstn-ip conversion outside the server reduces the load and makes 
an easier install.


I'm using the GXW-4104 , and besides it has trouble detecting danish 
callerid (a standard not used anywhere else in the world...), i have no 
complaints against it.


Imho, ymmw etc.

Leif


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Cannot xfer parked callers


Ah, I misread.  I'll probably do that and hopefully it'll fix the issue.

Thanks!
Jay

Bruce Reeves wrote:

Jay,

The proble is in both 1.2.14 and 1.4 the fix mentioned in the bug was added
to the svn revisions of both versions. If you are not wanting to switch 
from

1.2.14 to 1.2 svn the you can edit the features.c file and add the lines
mentioned in the notes back to the file, then make and make install.



On 1/25/07, Jay Moore [EMAIL PROTECTED] wrote:


Bruce,

I'm running 1.2.14.  I am not willing to switch to 1.4 yet due to the
stability issue.  From what I read on the page you linked, I could not
find what version had the supposed fix.  I also can't seem to find a
later 1.2 version of Asterisk (if one exists).

Any suggestions?

Thanks,
Jay

Bruce Reeves wrote:
 Jay,

 there is a bug in Mantis regarding this, a change was made to allow
native
 bridging of parked calls. The change has been reverted in a more recent
SVN
 version of 1.2. See http://bugs.digium.com/view.php?id=8804

 On 1/25/07, Jay Moore [EMAIL PROTECTED] wrote:

 Here's how it's currently working:

 1) Call comes in
 2) Operator parks call (700)
 3) Operator picks up call on another phone (701)
 4) Operator tries to transfer to a different phone (we use #0) but the
 transfer doesn't work.

 We can transfer initial callers all we want and it works fine.  Once a
 call is parked, however, we can no longer transfer the caller.

 Any ideas?

 Thanks,
 Jay
 ___
 --Bandwidth and Colocation provided by Easynews.com --

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users





 



 ___
 --Bandwidth and Colocation provided by Easynews.com --

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users








___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Adding 4 more POTS lines

2007-01-25 Thread Jim Freeze


On 1/25/07, Jay Moore [EMAIL PROTECTED] wrote:


Jim,

I have 2 TDM400s in my * box (as well as a T1 card).  I use all 8 ports,
and aside from some minor echoing during peak periods, it's running
smooth as ice.



Hi Jay. Thanks for the info.  Digium logged onto my box early on and
fixed some echo problems with a code change and recompile.

Do you have any theories on the cause of the echo only for peak periods?

Also, I suppose there is no problem leaving 2 FXO ports unused for a time.

Jim



--
Jim Freeze
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Adding 4 more POTS lines

2007-01-25 Thread Jim Freeze


Hi Leif

On 1/25/07, Leif Neland [EMAIL PROTECTED] wrote:


I have no experience with the TDM cards, but costwise it is not the best
solution, in my opinion.

A TDM04B (4FXO) cost around $378 at voiplink.com, while a  Grandstream
GXW-4104 (also 4FXO) cost $250 at voxilla.com., and the 8FXO version costs
$400, almost the same as the 4FXO card.



I suppose that is my alternative - remove the 4FXO card and add an 8FXO
card.
But I'm not seeing the prices you list. The Digium TDM2402B is listed at
$837.00.
Am I missing something?

 http://www.voiplink.com/Digium_TDM2402B_p/digium-tdm2402b.htm




--
Jim Freeze
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] IMAP Voicemail Storage

2007-01-25 Thread David Gomillion


Since you've done some work with Courier and Asterisk's IMAP voicemail, is
there a place you documented your findings? I'm interested in merging the
two. Is there any way to do it without having to ask all of my users for
their passwords?


On 1/25/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote:


On Thursday 25 January 2007 3:27 pm, Bruce Reeves wrote:
 I am doing some testing with 1.4 and the imap storage and a exchange
2003
 server. I have not had any positive results so far using the notes on
the
 wiki or the docs in the release. My current settings are

I've done some work with the IMAP voicemail storage and Courier-IMAP, and
have
had it working.

It does seem like it just cannot get to your IMAP server; have you tried
the
imaptools test program (the name escapes me, it's the only binary produced
by
the imaptools package), giving it the same IMAP connection string as what
Asterisk reports?

My development box is offline at the moment or I could give you specific
details.

-A.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] IMAP Voicemail Storage


I tried testing with mtest in the imap toolkit, I think that is what you
meant and it connects. The connection string
{server:143/imap/user=breeves}INBOX prompts for a password then
connects. I will keep digging the answer has to be around.

On 1/25/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote:


On Thursday 25 January 2007 3:27 pm, Bruce Reeves wrote:
 I am doing some testing with 1.4 and the imap storage and a exchange
2003
 server. I have not had any positive results so far using the notes on
the
 wiki or the docs in the release. My current settings are

I've done some work with the IMAP voicemail storage and Courier-IMAP, and
have
had it working.

It does seem like it just cannot get to your IMAP server; have you tried
the
imaptools test program (the name escapes me, it's the only binary produced
by
the imaptools package), giving it the same IMAP connection string as what
Asterisk reports?

My development box is offline at the moment or I could give you specific
details.

-A.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users





--
Bruce
Nortex Networks
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] IMAP Voicemail Storage

On Thursday 25 January 2007 4:48 pm, David Gomillion wrote:
 Since you've done some work with Courier and Asterisk's IMAP voicemail, is
 there a place you documented your findings? I'm interested in merging the
 two. Is there any way to do it without having to ask all of my users for
 their passwords?

There really weren't any findings; I wrote a small patch which corrected how 
the IMAP connection string was built, but other than that it just worked.

As far as not asking all your users for their passwords -- I'm not sure what 
you mean -- Asterisk needs to know the voicemail passwords, and those are 
stored in voicemail.conf.  I'm not using IMAP server passwords at all.

-A.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] IMAP Voicemail Storage

On Thursday 25 January 2007 5:01 pm, Bruce Reeves wrote:
 I tried testing with mtest in the imap toolkit, I think that is what you
 meant and it connects. The connection string
 {server:143/imap/user=breeves}INBOX prompts for a password then
 connects. I will keep digging the answer has to be around.

I'm using server authentication -- i.e. one login for the entire server, not 
per-user authentication... That's a critical difference I think.  Did you set 
up the IMAP user password in voicemail.conf? (I did not have to do that).

-A.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] IMAP Voicemail Storage


Andrew,

I tried both, did you set the server authentication with authuser=user and
authpassword=password in the general section?

On 1/25/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote:


On Thursday 25 January 2007 5:01 pm, Bruce Reeves wrote:
 I tried testing with mtest in the imap toolkit, I think that is what you
 meant and it connects. The connection string
 {server:143/imap/user=breeves}INBOX prompts for a password then
 connects. I will keep digging the answer has to be around.

I'm using server authentication -- i.e. one login for the entire server,
not
per-user authentication... That's a critical difference I think.  Did you
set
up the IMAP user password in voicemail.conf? (I did not have to do that).

-A.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users





--
Bruce
Nortex Networks
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Adding 4 more POTS lines

2007-01-25 Thread Leif Neland


Jim Freeze wrote:

Hi Leif

On 1/25/07, Leif Neland [EMAIL PROTECTED] wrote:


I have no experience with the TDM cards, but costwise it is not the
best solution, in my opinion.

A TDM04B (4FXO) cost around $378 at voiplink.com, while a
Grandstream GXW-4104 (also 4FXO) cost $250 at voxilla.com., and the
8FXO version costs $400, almost the same as the 4FXO card.





I suppose that is my alternative - remove the 4FXO card and add an
8FXO card.
But I'm not seeing the prices you list. The Digium TDM2402B is
listed at $837.00.
Am I missing something?

 http://www.voiplink.com/Digium_TDM2402B_p/digium-tdm2402b.htm



You misunderstand me.
A Grandstream GXW-4108 with 8FXO cost $400, almost the same as a TDM404B 
fully populated 4FXO card.


Leif

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] IAX softphone fails through PRI trunks with Hangup

2007-01-25 Thread Tim Panton



On 25 Jan 2007, at 16:48, Patrick W. Foster wrote:

I've a call center using IAX softphones provided by a third party.   
We've observed problems where the IAX phones seem unable to use our  
PRI trunks.  A sample anonymized call is provided below with the  
PRI debug calls embedded.  Any thoughts,
comments or suggestions would be welcome.  In anonymizing it, I  
preseved the format

and number of digits sent.


Do you have an IAX trace (either etherreal or IAX2 debug ) of a  
failed call ?


We had a similar problem in an early version of our  IAX softphone.
When I put the state-machine together I didn't expect the ringing  
message _ever_

come after a call is answered.

But it can, if you have

exten = s,1,Answer()
exten = s,2,Playback(your-call-may-be-recorded-blah-blah)
exten = s,3,Dial(Zap/g1/004416128824245) ; this line can generate a  
ringing message


Tim.



Tim Panton

www.mexuar.net
www.westhawk.co.uk/



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] setting up AMD

2007-01-25 Thread Matt Florell


From the VICIDIAL SCRATCH_INSTALL doc:


- cd asterisk-1.2.14/apps
- wget http://www.eflo.net/files/app_amd2.c
- mv app_amd2.c app_amd.c
- vi Makefile
 replace this line(line 32):
  app_mixmonitor.so app_stack.so
 with this line:
  app_mixmonitor.so app_stack.so app_amd.so
- wget http://www.eflo.net/files/amd2.conf
- mv amd2.conf /etc/asterisk/amd.conf

It works with Asterisk 1.2.14 just fine.


MATT---


On 1/25/07, Peter Halliday [EMAIL PROTECTED] wrote:

Where can I get the latest copy of this file.  I thought google found
ithere, but it doesn't compile correctly on 1.2.14.  And the copy on
voip-info.org that I found initially appears to be old.  It's not in the 1.2
tree.




On 1/25/07, Asterisk [EMAIL PROTECTED] wrote:
 On Wed, 2007-01-24 at 12:20 -0800, Michael Collins wrote:
 
 
 
 
 
 
__
  From:[EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On
Behalf Of Peter
  Halliday
  Sent: Wednesday, January 24, 2007 11:56 AM
  To: asterisk-users@lists.digium.com
  Subject: [asterisk-users] setting up AMD
 
 
 
 
  I'm trying get this working.  I've looked through the list, and can't
  see how to get AMD to print out more.  I have it call and say Hello
  like I normally would.  I've tried to say more and less doesn't seem
  to matter.  After I hangup it does recognize hangup.  Here's logging
  during an attempt where I make outbound call and answer, but then
  hangup after 1-2 seconds:
 
  Jan 24 17:01:37 VERBOSE[31455] logger.c: -- AMD:
  SIP/sip.broadvoice.com-098c4aa8 6079362172 (null) (Fmt: 4)
  Jan 24 17:01:37 VERBOSE[31455] logger.c: -- AMD: initialSilence
  [8000] greeting [1500] afterGreetingSilence [300] totalAnalysisTime
  [5000] minimumWordLength [120] betweenWordsSilence [50]
  maximumNumberOfWords [5] silenceThreshold [256]
  Jan 24 17:01:44 DEBUG[31437] chan_sip.c: Auto destroying call
  '[EMAIL PROTECTED]'
  Jan 24 17:01:45 DEBUG[31447] manager.c: Manager received command
  'Command'
  Jan 24 17:01:45 DEBUG[31447] manager.c: Manager received command
  'Command'
  Jan 24 17:01:45 DEBUG[31447] manager.c: Manager received command
  'Command'
  Jan 24 17:01:56 VERBOSE[31455] logger.c: -- AMD: HANGUP
  Jan 24 17:01:56 DEBUG[31455] app_amd.c: Got hangup
 
  The amd.conf:
  [amd]
  initial_silence= 3500
  greeting   = 1500
  after_greeting_silence = 300
  total_analysis_time= 5000
  min_word_length= 120
  between_words_silence  = 50
  maximum_number_of_words= 5
  silence_threshold  = 256
 
  In extensions.conf
  [outboundmsg1]
  exten = s,1,NoCDR
  exten = s,n,AMD



  exten =
s,n,GotoIf($[${AMDSTATUS}=AMD_PERSON]?humn:mach)



 *
 exten = s,n,GotoIf($[${AMDSTATUS}=HUMAN]?humn:mach)
 *

  exten = s,n(mach),WaitForSilence(2500)
  exten = s,n,Playback(outboundmsgs/msg1)
  exten = s,n,Hangup
  exten = s,n(humn),WaitForSilence(500)
  exten = s,n,Playback(outboundmsgs/msg1)
  exten = s,n,Hangup
 
 
 
  Peter,
 
  It looks like your initial silence setting might be having trouble.
   The amd.conf file has a value of 3500 but the log file is showing
  8000.  Try changing the amd.conf to something like 3000 and issue a
  reload at the CLI. Make another test call and see if the trace still
  shows 8000 for the initial silence.  I think having an initial silence
  value that is longer than the total analysis time might be causing the
  undesired behavior.
 
 
 
  Let us know what happens when you try to modify the initial silence
  value.
 
 
 
  -MC
 
 
  ___
  --Bandwidth and Colocation provided by Easynews.com --
 
  asterisk-users mailing list
  To UNSUBSCRIBE or update options visit:
 
http://lists.digium.com/mailman/listinfo/asterisk-users



 Try to replace the AMD_PERSON with HUMAN as depicted above between

 

 


 The AMD_STATUS that works for me is not person but human.



 dave




 ___
 --Bandwidth and Colocation provided by Easynews.com --

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:

http://lists.digium.com/mailman/listinfo/asterisk-users



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:

http://lists.digium.com/mailman/listinfo/asterisk-users




___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Adding 4 more POTS lines

2007-01-25 Thread cb


On Jan 25, 2007, at 5:38 PM, Leif Neland wrote:

A Grandstream GXW-4108 with 8FXO cost $400, almost the same as a  
TDM404B fully populated 4FXO card.


I'm currently testing a GXW-4108... my verdict is still out. I've had  
some problems, some minor, some major.


In the minor department, it does not always reboot when instructed to  
via the web interface. I think I've tracked it to the reboot button  
on a regular screen is ignored, but the reboot from the post update  
screen goes thru. This is likely a minor bug in the firmware.


Next into the minor-ish... the documentation isn't great. It it  
written assuming you know a lot more about this stuff than I did when  
I started. The more I've played and learned, when I go back and  
reread parts of the docs, they then make more sense to me.


Heading into the not so minor, but not really major... the logging  
sucks. It only supports a syslog server, which isn't a huge deal, but  
having a web interface to read the logs would have been nice. But,  
the logging doesn't seem to give much info (even in Debug mode), and  
seems to randomly stop working entirely. Sometimes it will start  
again when you power cycle the unit (not just a software reboot, but  
physically turn it off and back on), other times it needs to be  
defaulted to factory settings to get the logging going again, which  
is totally unacceptable.


Also in the not so minor category, there doesn't appear to be any  
easy way of backing up the config files. When it polls the tftp  
server on boot, it does look for a config file, but since there  
doesn't appear to be any way to save one out of the unit, and no  
documentation or otherwise (that I've found) to create one from  
scratch... it makes it very difficult to save settings and then  
easily restore them.


And then into the potentially major catagory... I've run into a  
problem that I *think* I've tracked to the unit doesn't recognize the  
dial-tone issued by my PSTN provider (Verizon). It works inbound and  
outbound just fine at my house, where it is connected to a LinkSys  
PAP that interfaces with Verizon's VoiceWing service. But when I move  
it to a real POTS line, it works inbound, but outbound single stage  
dialing stalls. This is a problem that I only just identified last  
night, and have been working on it today and as I said I *think* it  
may be that it isn't accepting the dial-tone. There is an option to  
ignore the dial-tone and not wait, but I haven't tested that yet (at  
3am I gave up at the office I was connecting it to and brought it  
back to my house where it promptly started working again... I'm  
hoping to retest on POTS tonight or tomorrow).



All of the above are probably fixable via a firmware update. I'm  
currently running the latest that Grandstream has on their web site,  
but I have not yet contacted them to see if they have a newer beta  
version available that hasn't been publicly posted. My guess is, all  
the issues will be worked out in due time.


With the only show stopper appearing to be the dial-tone issue (or  
whatever is causing it to fail on the POTS lines), it may be a good  
buy if you can either verify that it works with your PSTN provider  
first, or have the ability to return it if it doesn't (in my case, I  
could probably return it to the dealer, but A: it has been over a  
month since I bought it, and B: I'm not done playing with it to see  
what might or might not be wrong, and since for me price is the  
single most important factor, I'm willing to keep at this one to see  
if I can get it all working correctly.)


-chris
www.mythtech.net


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] setting up AMD

2007-01-25 Thread Peter Halliday


That's the same code as I have.  It's identical.  Are you using it over a
SIP channel?

Peter

On 1/25/07, Matt Florell [EMAIL PROTECTED] wrote:


From the VICIDIAL SCRATCH_INSTALL doc:

- cd asterisk-1.2.14/apps
- wget http://www.eflo.net/files/app_amd2.c
- mv app_amd2.c app_amd.c
- vi Makefile
  replace this line(line 32):
   app_mixmonitor.so app_stack.so
  with this line:
   app_mixmonitor.so app_stack.so app_amd.so
- wget http://www.eflo.net/files/amd2.conf
- mv amd2.conf /etc/asterisk/amd.conf

It works with Asterisk 1.2.14 just fine.


MATT---


On 1/25/07, Peter Halliday [EMAIL PROTECTED] wrote:
 Where can I get the latest copy of this file.  I thought google found
 ithere, but it doesn't compile correctly on 1.2.14.  And the copy on
 voip-info.org that I found initially appears to be old.  It's not in the
1.2
 tree.




 On 1/25/07, Asterisk [EMAIL PROTECTED] wrote:
  On Wed, 2007-01-24 at 12:20 -0800, Michael Collins wrote:
  
  
  
  
  
  
 __
   From:[EMAIL PROTECTED]
   [mailto:[EMAIL PROTECTED] On
 Behalf Of Peter
   Halliday
   Sent: Wednesday, January 24, 2007 11:56 AM
   To: asterisk-users@lists.digium.com
   Subject: [asterisk-users] setting up AMD
  
  
  
  
   I'm trying get this working.  I've looked through the list, and
can't
   see how to get AMD to print out more.  I have it call and say Hello
   like I normally would.  I've tried to say more and less doesn't seem
   to matter.  After I hangup it does recognize hangup.  Here's logging
   during an attempt where I make outbound call and answer, but then
   hangup after 1-2 seconds:
  
   Jan 24 17:01:37 VERBOSE[31455] logger.c: -- AMD:
   SIP/sip.broadvoice.com-098c4aa8 6079362172 (null) (Fmt: 4)
   Jan 24 17:01:37 VERBOSE[31455] logger.c: -- AMD: initialSilence
   [8000] greeting [1500] afterGreetingSilence [300] totalAnalysisTime
   [5000] minimumWordLength [120] betweenWordsSilence [50]
   maximumNumberOfWords [5] silenceThreshold [256]
   Jan 24 17:01:44 DEBUG[31437] chan_sip.c: Auto destroying call
   '[EMAIL PROTECTED]'
   Jan 24 17:01:45 DEBUG[31447] manager.c: Manager received command
   'Command'
   Jan 24 17:01:45 DEBUG[31447] manager.c: Manager received command
   'Command'
   Jan 24 17:01:45 DEBUG[31447] manager.c: Manager received command
   'Command'
   Jan 24 17:01:56 VERBOSE[31455] logger.c: -- AMD: HANGUP
   Jan 24 17:01:56 DEBUG[31455] app_amd.c: Got hangup
  
   The amd.conf:
   [amd]
   initial_silence= 3500
   greeting   = 1500
   after_greeting_silence = 300
   total_analysis_time= 5000
   min_word_length= 120
   between_words_silence  = 50
   maximum_number_of_words= 5
   silence_threshold  = 256
  
   In extensions.conf
   [outboundmsg1]
   exten = s,1,NoCDR
   exten = s,n,AMD
 
 
 
   exten =
 s,n,GotoIf($[${AMDSTATUS}=AMD_PERSON]?humn:mach)
 
 
 
  *
  exten = s,n,GotoIf($[${AMDSTATUS}=HUMAN]?humn:mach)
  *
 
   exten = s,n(mach),WaitForSilence(2500)
   exten = s,n,Playback(outboundmsgs/msg1)
   exten = s,n,Hangup
   exten = s,n(humn),WaitForSilence(500)
   exten = s,n,Playback(outboundmsgs/msg1)
   exten = s,n,Hangup
  
  
  
   Peter,
  
   It looks like your initial silence setting might be having trouble.
The amd.conf file has a value of 3500 but the log file is showing
   8000.  Try changing the amd.conf to something like 3000 and issue a
   reload at the CLI. Make another test call and see if the trace
still
   shows 8000 for the initial silence.  I think having an initial
silence
   value that is longer than the total analysis time might be causing
the
   undesired behavior.
  
  
  
   Let us know what happens when you try to modify the initial silence
   value.
  
  
  
   -MC
  
  
   ___
   --Bandwidth and Colocation provided by Easynews.com --
  
   asterisk-users mailing list
   To UNSUBSCRIBE or update options visit:
  
 http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 
  Try to replace the AMD_PERSON with HUMAN as depicted above between
 
  
 
  
 
 
  The AMD_STATUS that works for me is not person but human.
 
 
 
  dave
 
 
 
 
  ___
  --Bandwidth and Colocation provided by Easynews.com --
 
  asterisk-users mailing list
  To UNSUBSCRIBE or update options visit:
 
 http://lists.digium.com/mailman/listinfo/asterisk-users
 


 ___
 --Bandwidth and Colocation provided by Easynews.com --

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:

 http://lists.digium.com/mailman/listinfo/asterisk-users



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] IMAP Voicemail Storage

2007-01-25 Thread David Gomillion


On 1/25/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote:


On Thursday 25 January 2007 4:48 pm, David Gomillion wrote:
 Since you've done some work with Courier and Asterisk's IMAP voicemail,
is
 there a place you documented your findings? I'm interested in merging
the
 two. Is there any way to do it without having to ask all of my users for
 their passwords?

There really weren't any findings; I wrote a small patch which corrected
how
the IMAP connection string was built, but other than that it just worked.

As far as not asking all your users for their passwords -- I'm not sure
what
you mean -- Asterisk needs to know the voicemail passwords, and those are
stored in voicemail.conf.  I'm not using IMAP server passwords at all.




I mean that I would like to have a system in place so that Asterisk, as a
privileged service, can gain access to Courier's IMAP storage. Having to
keep track of all of our users' passwords in the Asterisk configuration is
going to provide a ridiculous amount of administration, as we force them to
change their passwords often in our single-sign on environment.


-A.

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Do I need a CH1 licence for Cisco Phones ?

2007-01-25 Thread Peter Mitchell

Cory,

I know the 7940 and 7960 had a SIP licence you could buy.  It was simple,
buy the phone and then buy the SIP licence if you want to use it for
asterisk.

79X1 phones now come bundled with licences - and I can't find a separate SIP
licence like the old 79x0 models.

Whats the non callmanager - SIP licence number for 79X1 ?  

I've only found CM CME licence codes SW-CCM-UL-7941 SW-CCM-UL-7961
SW-CCME-UL-7961 (these codes may only be for our distributor)

Cheers
Peter.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Cory Andrews
Sent: Friday, 26 January 2007 1:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Do I need a CH1 licence for Cisco Phones ?

Technically, Cisco requires you to purchase both a Smartnet (To obtain a
CCO login for access to firmware), as well as a SIP/MGCP license token,
to utilize their phones with SIP firmware, regardless of platform.  

The CH1 nomenclature applies to Callmanager, the CCME nomenclature
applies to Callmanager Express.  

The appropriate license for SIP is SW-SM-UL-7960 if you are using a
Cisco 7960G phone.  The Cisco description for this license is SIP 
MGCP LICENSE FOR SINGLE 7960 IP PHONE

The article referenced in the WIKI referenced in this thread is not
entirely correct, in a few aspects.

A - Technically, a user needs a Smartnet for CCO access to firmware, as
well as a SIP/MGCP license to legitimately utilize Cisco's SIP firmware,
irregardless of platform.

B - The WIKI article also states that if you are using Callmanager and
you receive phones from a reseller that do not have a part # on the
label ending in CH1, that they are Spares and not Callmanager licensed
phones.  This is also technically incorrect.  In recent months I have
seen Cisco ship phones classified as CH1 (Callmanager) licensed, without
a CH1 part number on the box label.  Cisco tracks licensing by the
unique serial number of the phone, and I have seen them bulk register
spare phones as CH1 licensed phones, simply by updating their serial
number database accordingly and tagging serial numbers and licensed.  It
amounts to a virtual license, and likely allows Cisco to better manage
their inventory, as they can utilize phones originally produced as
spares, and easily convert them to CH1 licensed phones, just by updating
their serial number database accordingly.

Another common myth is that if you purchase used phones that were
originally sold as CH1 or CCME licensed units, that the license it
transferrable to the new owner of the phone.  According to Cisco, this
is not true, and a user is supposed to bear the cost of re-licensing.

It is quite confusing, and am neither supporting, nor critizing the
model, just relaying my experience.  I manage a business division that
is a Cisco premier partner with Unified CallManager Express
specialization and deal with licensing on a regular basis.


Cory Andrews

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pavel
Jezek
Sent: Thursday, January 25, 2007 8:12 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Do I need a CH1 licence for Cisco Phones ?

I think, ci$co phones can not be even purchased without licence...
btw, what is your reason, to buy ci$co phones, when known issues exist
with this phones, if working with anything other than callmanager?  :-\
PJ


Peter Mitchell wrote:

 I've got a question regarding Cisco IP Phones and licencing.
  
 When using a third party PBX like asterisk is a licence required for 
 the Cisco phones ? Has anyone got anything in writing from Cisco to 
 clarify this ?
  
 Eg can I just use CP-7961G or do I need CP-7961G-CH1 even though I'm 
 not using Cisco Callmanager ?

  


http://www.voip-info.org/wiki/view/Exposing+the+Cisco+Call+Manager+Licen
se+Scam 
 says no licence required.

  

 Cisco site mentions All Cisco Unified IP phones require the purchase 
 of a phone technology license, regardless of call protocol being used.

  


http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet0
900aecd802ff020.html 

  
 Cheers
 Peter


 --
 No virus found in this outgoing message.
 Checked by AVG Free Edition.
 Version: 7.1.410 / Virus Database: 268.17.10/651 - Release Date: 
 24/01/2007




 ___
 --Bandwidth and Colocation provided by Easynews.com --

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
   
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To

[asterisk-users] Failing to compile chan_capi

2007-01-25 Thread Cosmin Prund

I've got a brand new Eicon Diva Server BRI card and I want to configure 
it with Asterisk. I managed to get asterisk and zaptel to compile and 
install, I've compiled and installed the drivers for the Diva card and 
now I need to compile and install the chan_driver for chan_capi. 
Unfortunately this fails miserably. I get the following messages:


I'm using: Kernel 2.6.16.37.4, zaptel-1.4.0 and asterisk-1.4.0., 
chan_capi 0.7.1


//--

[EMAIL PROTECTED] chan_capi-0.7.1]# make
./create_config.sh /usr/src/asterisk-1.4.0/include
Checking Asterisk version... 1.4.0
* found stringfield in ast_channel
* found 'indicate' with data
config.h complete.
gcc -pipe -fPIC -Wall -Wmissing-prototypes -Wmissing-declarations -g  
-I/usr/src/asterisk-1.4.0/include -D_REENTRANT -D_GNU_SOURCE  -O6 
-march=i686  -Wno-missing-prototypes -Wno-missing-declarations 
-DCRYPTO   -c -o chan_capi.o chan_capi.c

In file included from chan_capi.c:82:
chan_capi.h:41:2: warning: #warning If you dont update your libcapi20, 
some fax features are not available
chan_capi.c:146: warning: type defaults to `int' in declaration of 
`STANDARD_LOCAL_USER'

chan_capi.c:146: warning: data definition has no type or storage class
chan_capi.c:147: warning: type defaults to `int' in declaration of 
`LOCAL_USER_DECL'

chan_capi.c:147: warning: data definition has no type or storage class
chan_capi.c: In function `capi_new':
chan_capi.c:2069: error: too few arguments to function `ast_channel_alloc'
chan_capi.c:2083: error: structure has no member named `type'
chan_capi.c: In function `pbx_capicommand_exec':
chan_capi.c:4613: warning: implicit declaration of function `LOCAL_USER_ADD'
chan_capi.c:4628: warning: implicit declaration of function 
`LOCAL_USER_REMOVE'

chan_capi.c: At top level:
chan_capi.c:5275: error: unknown field `send_digit' specified in initializer
chan_capi.c:5275: warning: initialization from incompatible pointer type
make: *** [chan_capi.o] Error 1

//--

Since the configuration method is a bit too much for me, here's part of 
chan_capi Makefile. I think I've been blind as I haven't found the 
documentation for WHAT needs to go WHERE in this Makefile...


.PHONY: openpbx

INSTALL_PREFIX=/usr/lib/asterisk

ASTERISK_HEADER_DIR=/usr/src/asterisk-1.4.0/include

MODULES_DIR=/usr/lib/asterisk/modules

CONFIG_DIR=/etc/asterisk


//--

If anyone has any idea what I'm doing wrong, please help me,
Thanks,
Cosmin Prund
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] NTL Hangup

2007-01-25 Thread Leo Ann Boon


Kyle Gordon wrote:

Hi all,

I'm currently on an NTL PSTN line, using Asterisk 1.2 and a X101P 
cheapo card.


The problem lies with detecting when the far end has hung up. It fails 
to detect it, and will only cleardown when the silence timeout has 
been reached. Now, I've seen the thread at 
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg32337.html, 
to which nothing has come of it. That was almost 2 years ago, so I was 
wondering if there's been any progress?

2 things:
a. You need to show us your zaptel.conf and zapata.conf.
b. Do you know the tone plan used by ntl? I guess it should be the UK 
standard.


Leo




___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] SPA3K to SPA3K DTMF issue

2007-01-25 Thread Doug Crompton

As per my (numerous) prior statements on this subject Asterisk WILL
NOT properly work with the spa-3000 DTMF in rfc2833. Use INBAND when
dealing with Asterisk on both the FXO/FXS ports of the spa3k if you are
dealing with Asterisk. This is a setting in BOTH sip.conf and spa3k pstn
and line 1 tabs.

Symptoms are no DTMF after call completion (voicemail
from outside to fxo) and IVR attempts from FXS attached analog phones.

Using INFO negates use of dtmf control functions on your fxs/fxo ports -
transfer etc. - Take your pick of what is more important to you.

There should really be a wiki on this! It gets asked often.

I might qualify that this is an issue with 1.2.x (and probably earlier) -
not sure if any fixes make this work or work better in 1.4. Fault
(apparently) lies with both sipura(linksys) and digium.

Since in this case you are connecting the spa3k's thru Asterisk this would
apply. I have not tried connecting two spa3k's directly together via
network to see if they play together in this regard.

Doug


On Wed, 24 Jan 2007, Mark Coccimiglio wrote:

 My experience has been to be consistant.  The only time I have had
 problems with DTMF is when I am not using the same DTMF encoding
 technique on all hardware.  Your choices are: INFO, RFC2833 or
 INBAND.  Some equipment also has an AUTO option but I would not
 recomend it.  Stick with INFO or rfc2833 and be consistant across the
 enterprise.

 Mark C
 IS Manager
 http://www.psh-inc.com

 [EMAIL PROTECTED] wrote:

  Hi all,
 
  Has anyone faced an issue when sending DTMF from FXS of one SPA3K to
  FXO of another SPA3K through asterisk?
 
  Im not able to send it properly. Wanna be sure if its an issue faced
  by all..
 
  If you have a fix for it, pls guide me.
 
  Thanks
 
  Dan
 
 
 
 
 ___
 --Bandwidth and Colocation provided by Easynews.com --
 
 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
 
 



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  *



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] DeStar 0.2.2 released!

2007-01-25 Thread Santiago José Ruano Rincón

Hi,

someone has made me realize that a more detailed description is needed
for those who don't know about DeStar, so:

DeStar is a Web-based management and configuration tool for the Asterisk
PBX. 

DeStar's main features include:

* Hosted PBX and virtual PBX features, which allow you to have several
PBXs on a single machine. 
* Extensions can be managed for SIP, IAX, Zap, and more. 
* Auto-attendants are supported. 
* Trunks can be managed for SIP, IAX, Zap, ZapPRI, and more. 
* Dialout patterns (i.e. local, national, mobile-phones) can be used. 
* Asternic Flash Operator Panel is integrated. 
* Many application applets are included for voice mail, meeting rooms,
and more. 
* It is extensible through a pluguin-based architecture.

Best regards,

Santiago Ruano Rincón
http://destar.berlios.de



signature.asc
Description: Esta parte del mensaje está firmada	digitalmente
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] dacs support on Digium T1 equipment.

2007-01-25 Thread Shane Spencer


Heya everybody.

I have been peering into the code for zaptel for a while now, I am
keenly interested in the dacs support, being able to apparently
redirect certain spans to other spans.  Not sure if this has to be on
the same T1 interface or can be used between T1 interfaces on the same
board or possible two different cards.  Any information on the
functionality of this would be greatly appreciated!  I don' t have the
equipment any more to test it out.

I have been wanting to bridge T1 devices together outside of the
dial plan for a long time. However this time I need to be able to
monitor the audio data and call information as well.  I am fine
programming something that can talk to the zaptel drivers, but I need
to know if channels placed into a dacs configuration can be monitored
at all.

If I do what needs to be done with just using a simple dialplan I have
echo concerns, I am wondering how much of a concern echo will be
between two spans placed into a dacs configuration.  I knew if there
is echo I can do nothing about it, asterisk needs to be in the middle
to help adjust that.  If there is a low chance of echo it would save
me quite a bit of money by not requiring echo cancellation capable T1
boards.

Shane R. Spencer
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] setting up AMD

2007-01-25 Thread Matt Florell


I tested it over a SIP channel and an IAX channel and it did work, but
I have not used it in production that way. I only use Zap channels(T1
PRI) In prodution at the locations that I use AMD at.

MATT---

On 1/25/07, Peter Halliday [EMAIL PROTECTED] wrote:

That's the same code as I have.  It's identical.  Are you using it over a
SIP channel?

Peter


On 1/25/07, Matt Florell  [EMAIL PROTECTED] wrote:
 From the VICIDIAL SCRATCH_INSTALL doc:

 - cd asterisk-1.2.14/apps
 - wget http://www.eflo.net/files/app_amd2.c
 - mv app_amd2.c app_amd.c
 - vi Makefile
   replace this line(line 32):
app_mixmonitor.so app_stack.so
   with this line:
app_mixmonitor.so app_stack.so app_amd.so
 - wget http://www.eflo.net/files/amd2.conf
 - mv amd2.conf /etc/asterisk/amd.conf

 It works with Asterisk 1.2.14 just fine.


 MATT---


 On 1/25/07, Peter Halliday [EMAIL PROTECTED] wrote:
  Where can I get the latest copy of this file.  I thought google found
  ithere, but it doesn't compile correctly on 1.2.14.  And the copy on
  voip-info.org that I found initially appears to be old.  It's not in the
1.2
  tree.
 
 
 
 
  On 1/25/07, Asterisk [EMAIL PROTECTED] wrote:
   On Wed, 2007-01-24 at 12:20 -0800, Michael Collins wrote:
   
   
   
   
   
   
 
__
From:[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
  Behalf Of Peter
Halliday
Sent: Wednesday, January 24, 2007 11:56 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] setting up AMD
   
   
   
   
I'm trying get this working.  I've looked through the list, and
can't
see how to get AMD to print out more.  I have it call and say Hello
like I normally would.  I've tried to say more and less doesn't seem
to matter.  After I hangup it does recognize hangup.  Here's logging
during an attempt where I make outbound call and answer, but then
hangup after 1-2 seconds:
   
Jan 24 17:01:37 VERBOSE[31455] logger.c: -- AMD:
SIP/sip.broadvoice.com-098c4aa8 6079362172 (null) (Fmt: 4)
Jan 24 17:01:37 VERBOSE[31455] logger.c: -- AMD: initialSilence
[8000] greeting [1500] afterGreetingSilence [300] totalAnalysisTime
[5000] minimumWordLength [120] betweenWordsSilence [50]
maximumNumberOfWords [5] silenceThreshold [256]
Jan 24 17:01:44 DEBUG[31437] chan_sip.c: Auto destroying call
   
'[EMAIL PROTECTED] '
Jan 24 17:01:45 DEBUG[31447] manager.c: Manager received command
'Command'
Jan 24 17:01:45 DEBUG[31447] manager.c: Manager received command
'Command'
Jan 24 17:01:45 DEBUG[31447] manager.c: Manager received command
'Command'
Jan 24 17:01:56 VERBOSE[31455] logger.c: -- AMD: HANGUP
Jan 24 17:01:56 DEBUG[31455] app_amd.c: Got hangup
   
The amd.conf:
[amd]
initial_silence= 3500
greeting   = 1500
after_greeting_silence = 300
total_analysis_time= 5000
min_word_length= 120
between_words_silence  = 50
maximum_number_of_words= 5
silence_threshold  = 256
   
In extensions.conf
[outboundmsg1]
exten = s,1,NoCDR
exten = s,n,AMD
  
  
  
exten =
  s,n,GotoIf($[${AMDSTATUS}=AMD_PERSON]?humn:mach)
  
  
  
   *
   exten = s,n,GotoIf($[${AMDSTATUS}=HUMAN]?humn:mach)
   *
  
exten = s,n(mach),WaitForSilence(2500)
exten = s,n,Playback(outboundmsgs/msg1)
exten = s,n,Hangup
exten = s,n(humn),WaitForSilence(500)
exten = s,n,Playback(outboundmsgs/msg1)
exten = s,n,Hangup
   
   
   
Peter,
   
It looks like your initial silence setting might be having trouble.
 The amd.conf file has a value of 3500 but the log file is showing
8000.  Try changing the amd.conf to something like 3000 and issue a
reload at the CLI. Make another test call and see if the trace
still
shows 8000 for the initial silence.  I think having an initial
silence
value that is longer than the total analysis time might be causing
the
undesired behavior.
   
   
   
Let us know what happens when you try to modify the initial silence
value.
   
   
   
-MC
   
   
___
--Bandwidth and Colocation provided by Easynews.com --
   
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   
 
http://lists.digium.com/mailman/listinfo/asterisk-users
  
  
  
   Try to replace the AMD_PERSON with HUMAN as depicted above between
  
   
  
   
  
  
   The AMD_STATUS that works for me is not person but human.
  
  
  
   dave
  
  
  
  
   ___
   --Bandwidth and Colocation provided by Easynews.com --
  
   asterisk-users mailing list
   To UNSUBSCRIBE or update options visit:
  
 
http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] TC400B Transcoder Card Shipping

2007-01-25 Thread Andres

I just saw the TC400B transcoder card at the IT Expo in Fort 
Lauderdale.  The Digium representative confirmed it was shipping.  Does 
anybody have one of this and can give us some feedback?

Thanks,

--
Andres
Technical Support
http://www.telesip.net

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] IMAP Voicemail Storage

On Thursday 25 January 2007 6:30 pm, David Gomillion wrote:
 I mean that I would like to have a system in place so that Asterisk, as a
 privileged service, can gain access to Courier's IMAP storage. Having to
 keep track of all of our users' passwords in the Asterisk configuration is
 going to provide a ridiculous amount of administration, as we force them to
 change their passwords often in our single-sign on environment.

How do they log on to check their voicemail?  Is your SSO system entirely 
numeric?

-A.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] background() with m option

2007-01-25 Thread Franz Wu

I have same problem and no mailing list response. I suggest we go for 
reporting bug.



- Original Message - 
From: Jack Wei [EMAIL PROTECTED]

To: asterisk-users@lists.digium.com
Sent: Friday, January 26, 2007 1:16 AM
Subject: [asterisk-users] background() with m option



Hi...

In my dialplan, I have the following:

exten = s,1,Background(${RECORDING}|m)
exten = s,n,Voicemail(${DID_NO})
exten = 0,1,Voicemail(${DID_NO})
exten = a,1,VoiceMailMain(${DID_NO})
exten = h,1,Hangup

In version 1.2, when I hit 0 during the playback, I will be directed to 
voicemail. But in verison 1.4, the call hangs up.


[Jan 24 16:05:37] DTMF[5754]: channel.c:2148 __ast_read: DTMF begin '0' 
received on SIP/5060-08c53e68
[Jan 24 16:05:37] DTMF[5754]: channel.c:2128 __ast_read: DTMF end '0' 
received on SIP/5060-08c53e68
== Spawn extension (play_recording, s, 1) exited non-zero on 
'SIP/5060-08c53e68'
-- Executing [EMAIL PROTECTED]:1] Hangup(SIP/5060-08c53e68, ) in new 
stack
== Spawn extension (play_recording, h, 1) exited non-zero on 
'SIP/5060-08c53e68'



Does anyone tell me why this is happening?

Thanks,

Jack
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users 


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] IMAP Voicemail Storage