[asterisk-users] TDM01B -1 FXO card not working.

[Jan du Toit]

Hi.

I'm trying to install and configure a TDM01B -1 FXO card.

I'm getting the following errors when starting up asterisk:
Jul 25 08:48:40 WARNING[1775]:
chan_zap.c:923 zt_open: Unable to specify channel 1: No such device
Jul 25 08:48:40 ERROR[1775]: chan_zap.c:6879 mkintf: Unable to open
channel 1: No such device
here = 0, tmp->channel = 1, channel = 1
Jul 25 08:48:40 ERROR[1775]: chan_zap.c:10311 setup_zap: Unable to
register channel '1'
Jul 25 08:48:40 WARNING[1775]: loader.c:414 __load_resource:
chan_zap.so: load_module failed, returning -1
Jul 25 08:48:40 WARNING[1775]: loader.c:554 load_modules: Loading
module chan_zap.so failed!

My /etc/zaptel.conf looks as follows:
fxsks=1
loadzone = us
defaultzone=us

The relevant stuff in my /etc/asterisk/zapata.conf looks as follows:
signalling=fxs_ks
echocancel=yes
echocancelwhenbridged=yes
echotraining=400
callerid=asreceived
group=1
context=sgs
channel => 1
(I have attached the whole zapata.conf file)

The card is properly installed in the box and the wctdm driver loads.
When I type in  modprobe wctdm the driver
loads and the light on the card goes on.

Typing in ztcf -vv gives the following:
Zaptel Configuration
==
Channel map:

Channel 01: FXS Kewlstart (Default) (Slaves: 01)

1 channels configured.

Typing in cat /proc/zaptel/1 produces
the following:
Span 1: WCTDM/0 "Wildcard TDM400P REV I
Board 1"

   1 WCTDM/0/0 FXSKS
   2 WCTDM/0/1
   3 WCTDM/0/2
   4 WCTDM/0/3

When I comment out the channel creation part of the zapata.conf
asterisk loads fine and the CLI command "Zap show status" produces the
following:
localhost*CLI> zap show status
Description  Alarms IRQ   
bpviol CRC4
Wildcard TDM400P REV I Board 1   OK 0 
0  0

But, when I type in "cat /dev/zap/1" I get the following:
cat: /dev/zap/1: No such device
Should it show sometging else.

The output given by "ztcf -vv" and "cat /proc/zaptel/1" seems fine to
me, but I'm not an expert. Is it fine?
Please, any help will be appreciated.

Thank you.

Jan.










;
; Zapata telephony interface
;
; Configuration file
;
; You need to restart Asterisk to re-configure the Zap channel
; CLI> reload chan_zap.so 
;   will reload the configuration file,
;   but not all configuration options are 
;   re-configured during a reload.


[trunkgroups]
;
; Trunk groups are used for NFAS or GR-303 connections.
;
; Group: Defines a trunk group.  
;group => ,[,...]
;
;trunkgroup  is the numerical trunk group to create
;dchannelis the zap channel which will have the 
;d-channel for the trunk.
;backup1 is an optional list of backup d-channels.
;
;trunkgroup => 1,24,48
;trunkgroup => 1,24
;
; Spanmap: Associates a span with a trunk group
;spanmap => ,[,]
;
;zapspan is the zap span number to associate
;trunkgroup  is the trunkgroup (specified above) for the mapping
;logicalspan is the logical span number within the trunk group to use.
;if unspecified, no logical span number is used.
;
;spanmap => 1,1,1
;spanmap => 2,1,2
;spanmap => 3,1,3
;spanmap => 4,1,4


[channels]
; Default language
;
;language=en
;
; Default context
;
context=sgs
;
; Switchtype:  Only used for PRI.
;
; national:   National ISDN 2 (default)
; dms100: Nortel DMS100
; 4ess:   AT&T 4ESS
; 5ess:   Lucent 5ESS
; euroisdn:   EuroISDN
; ni1:Old National ISDN 1
; qsig:   Q.SIG
;
switchtype=national
;
; Some switches (AT&T especially) require network specific facility IE
; supported values are currently 'none', 'sdn', 'megacom', 'accunet'
;
;nsf=none
;
; PRI Dialplan:  Only RARELY used for PRI.
;
; unknown:Unknown
; private:Private ISDN
; local:  Local ISDN
; national:   National ISDN
; international:  International ISDN
;
;pridialplan=national
;
; PRI Local Dialplan:  Only RARELY used for PRI (sets the calling number's 
numbering plan)
;
; unknown:Unknown
; private:Private ISDN
; local:  Local ISDN
; national:   National ISDN
; international:  International ISDN
;
;prilocaldialplan=national
;
; PRI callerid prefixes based on the given TON/NPI (dialplan)
; This is especially needed for euroisdn E1-PRIs
; 
; sample 1 for Germany 
;internationalprefix = 00
;nationalprefix = 0
;localprefix = 0711
;privateprefix = 07115678
;unknownprefix = 
;
; sample 2 for Germany 
;internationalprefix = +
;nationalprefix = +49
;localprefix = +49711
;privateprefix = +497115678
;unknownprefix = 
;
; PRI resetinterval: sets the time in seconds between restart of unused
; channels, defaults to 3600; minimum 60 seconds.  Some PBXs don't like
; channel restarts. so set the interval to a very long interval e.g. 1
; or 'never' to disable *entirely*.
;
;resetinterval = 3600 
;
; Overlap dialing mode (se

Re: [asterisk-users] X100P clone not working

[Frank Darner]

> > One last thing:
> >
> > Playback sounds now like MickeyMouse, much to slow
>
> One sanity check: try zttest . See that it gives values ov 100% or very
> close to that.

--- Results after 141 passes ---
Best: 99.499512 -- Worst: 83.691406 -- Average: 95.715467





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[asterisk-users] Asterisk/GPL and G.729 licensing

[Nick Hoffman]

Hi guys. I just stumbled upon 
http://www.voip-info.org/wiki/index.php?page=Asterisk+G.729+Licensing and 
read the section titled "Warning". I'm a bit confused now. Are you 
violating the GPL (or any other license) if you sell a computer with 
Asterisk and a G.729 license installed?

Cheers,
-- Nick
e: [EMAIL PROTECTED]
p: +61 7 5591 3588
f: +61 7 5591 6588

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[asterisk-users] kernel: Error! while loading wct4xxp module

[root linux]

Hi all,

I got this error message below after running the two
commands below: -

insmod zaptel
insmod wct4xxp t1e1override=0xFF
ztcfg -vvv

The error message is as below: -

Jul 25 13:23:00 test kernel: Zapata Telephony
Interface Registered on major 196
Jul 25 13:23:00 test kernel: Zaptel Version: 1.2.6
Echo Canceller: KB1
Jul 25 13:23:36 test kernel:  Error!
Jul 25 13:23:36 test kernel: TE4XXP: Version
Synchronization Error!
Jul 25 13:23:36 test last message repeated 388 times
Jul 25 13:23:36 test kernel: Found a Wildcard:
Wildcard TE205P

Below is my zaptel.conf config file: -

loadzone = us
defaultzone=us

span=1,0,0,ccs,hdb3
span=2,0,0,ccs,hdb3

bchan = 1-15, 17-31
dchan = 16
bchan = 32-46,48-62
dchan = 47


Regards,
rootlinux


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RE: [asterisk-users] Just bought a Polycom 501 - I feel likemyGXP-2000 was better...

[Douglas Garstang]

Not for our users. We held focus groups, and the Polycom's won in terms of 
ease-of-use over all the other phones investigated.

-Original Message- 
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 
Sent: Mon 7/24/2006 7:08 PM 
To: Asterisk Users Mailing List - Non-Commercial Discussion 
Cc: 
Subject: RE: [asterisk-users] Just bought a Polycom 501 - I feel 
likemyGXP-2000 was better...



On Mon, 24 Jul 2006, Mike wrote:
> As for the rest, it`s just personal opinion, but I still think the 
inerface
> of the GXP-2000 was a bit better...everything seemed to be where it 
should
> be.  At least now I`ll be able to appreciate the Polycom 501 and 
judge it in
> a functional state.

Yep the polycoms are not intuitive. That's been the reaction from
everyone who has used them so far. Ciscos Grandstreams Sipuras yes,
Polycoms no.

Nice speakerphone though. Shame about the configuration xml hell though.

-Dan
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Re: [asterisk-users] Voice with echo

[Martin Joseph]

On Jul 24, 2006, at 5:50 PM, Carlos Alberto Bernat Orozco wrote:


Hi group

I have my * box installed with a public IP address and I'm testing  
two extensions. I'm using SJphone for softphone. When I make the  
call from an extension to another, the voice sounds with echo.  
Besides sounds like creepy and it seems like a radio (for making a  
description)


Are you using a headset?  If not you are probably just hearing the  
speakers feedback through the mic.


Your configurations don't look like they are the problem to me.

Marty

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Re: [asterisk-users] Just bought a Polycom 501 - I feel likemyGXP-2000 was better...

[Dovid Bender]

Nice speakerphone though. Shame about the configuration xml hell though.


Polycom gave me my first lesson in XML. What a treat. Once you figure it out 
its a piece of cake. Real easy to provision multiple phones at once. 


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Re: [asterisk-users] Just bought a Polycom 501 - I feel like myGXP-2000 was better...

[Dovid Bender]

501 takes a little time to learn but much more 
worth it. My GXP-200 is collecting dust.

  - Original Message - 
  From: 
  Mike 
  To: 'Asterisk Users Mailing List - 
  Non-Commercial Discussion' 
  Sent: Monday, July 24, 2006 4:55 PM
  Subject: [asterisk-users] Just bought a 
  Polycom 501 - I feel like myGXP-2000 was better...
  
  Hello 
  everyone,
   
  I'm half writing 
  this to get some answers, and half-writing this to put in my 2 cents for 
  anybody who's looking to get his first VoIP phone.  Polycom's seem 
  to be highly regarded here, and after having bought a Polycom 501 as my 
  second phone (my first was the unloved Grandstream GXP-2000), I am left 
  wondering why.
   
  Never mind that 
  the setup of the Polycom was more complicated than the GXP-2000 by a few order 
  of magnitudes, that only matters the first time you do it.  But things 
  like 3-way conferences are harder to use (whatever happened to picking a line, 
  pressing CONF and picking another line) and in general, except for the 
  aesthetics of the phone, the GrandStream is an equal phone (for a much lesser 
  price) than the Polycom 501.
   
  My worst gripe 
  with this phone, is that I haven't managed to have it on the LAN without it 
  disappearing for a few seconds.  If I ping the phone (on the same LAN, on 
  the same underused hub actually) I get 5-6 responses, then timeoutsthen 
  another few responses, then timeouts again.  This translates into sound 
  being (badly) cut off when Im talking.  The same experiments yields good 
  results with the GXP-2000.  I understand ping might not be prioritized on 
  the Polycom, but this was done with no calls coming in or going out.  So 
  why is it disappearing?  Or is there a better test?
   
  If it helps, my 
  setup can be described as a Asterisk server (NO NAT) and a Polycom 501 (behind 
  a Nat).  The Grandstream handles that like a pro, I assumed the Polycom 
  would too, considering the reputation of Polycom and the price of the 
  phone.  
   
  I would be 
  grateful to anyone who clues me in on what I am doing wrong, because I am 95% 
  certain it's somehow my fault.  My gf would tell you it always is 
  :-)
   
   
  Mike
   
   
  
  

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Re: [asterisk-users] Polycom_acd_functions SIP trouble

[Dovid Bender]

I am sure you prob. know this but in your configs it shows secret commented 
out. Also it with a softphone if it dosent work then, then its your configs. 
Also did you remember to reload asterisk ?
- Original Message - 
From: "James Fromm" <[EMAIL PROTECTED]>

To: 
Sent: Monday, July 24, 2006 2:24 PM
Subject: [asterisk-users] Polycom_acd_functions SIP trouble


I'm trying to use the latest revision of Bweschke's branch from SVN for 
polycom_acd_functions.  Asterisk builds and runs without error but all SIP 
devices can't register when specifying a secret in sip.conf.  The Polycom 
601 I'm testing with and a copy of SJphone will not register. IAX from 
Idefisk works without error.


The error all SIP devices get is:

Jul 24 10:26:48 NOTICE[31524]: chan_sip.c:14203 handle_request_register: 
Registration from '' failed for 
'192.168.0.95' - Username/auth name mismatch


Commenting the definition of a secret in sip.conf for the device solves 
this.  Here's the config for one of the devices.


[1003]
type=friend
canreinvite=no
host=dynamic
username=1003
; secret=stuff
context=outbound
callerid="Jimmy" <1003>
[EMAIL PROTECTED]
nat=no

Why won't this revision accept the definition of a secret?  Am I missing 
something simple (stupid)?


Thanks,
Jay

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Re: [asterisk-users] Intercom feature on Polycom phones

[Dovid Bender]

I just went thru three days of scratching my head 
and finally got it done. If you need help with configs let me know. I also 
figured out how to have silent ringing where the ext. is called and the display 
show an incoming call but the phone dosent ring.
 

  - Original Message - 
  From: 
  Brian Vincent (C) 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Monday, July 24, 2006 1:19 PM
  Subject: RE: [asterisk-users] Intercom 
  feature on Polycom phones
  
  
  Yes.
   
  http://www.voip-info.org/wiki/view/Polycom+auto-answer+config
  
  ---Brian 
  VincentCopper Mountain Telecom[EMAIL PROTECTED] 
  -Original 
  Message-From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Stephen MurphySent: Monday, July 24, 
  2006 10:47 
  AMTo: asterisk-users@lists.digium.comSubject: [asterisk-users] Intercom 
  feature on Polycom phones
   
  I have the situation where my 
  client would like to ‘Intercom’ an extension similar to auto-answer. I have 
  polycom phones – can this be done?
   
  Steve
  


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Re: [asterisk-users] G729 Softphone

[Dovid Bender]

I have the paid version of x-lite (the older style version). It lets me use 
g729. It was a setting some where.


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

Sent: Monday, July 24, 2006 12:47 PM
Subject: Re: [asterisk-users] G729 Softphone


I have the eyeBeam softphone but I don't see G729 in the list of  available 
codecs (BTW, this is the paid version not X-Lite). Any clues?


Thanks,
Daniel

On Jul 24, 2006, at 12:00 PM, Guillermo Salas M. wrote:


On Mon, 2006-07-24 at 11:41 -0400, Daniel Salama wrote:

Looking for a SIP or IAX softphone for a call center application that
can do G729 codec. Any recommendations? Ideally it would do screen
pops, meaning that it will understand the URL option of the Dial
command.



Give a try to Eyebeam at www.counterpath.com , it supports video and
voice with g729.

BOL Siphone is freeware that supports video/voice and uses de g723.1
codec you can download it at http://www.bol2000.com/download/sipphone/


Thanks,
Daniel

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--
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Calle 15 y Avenida 24 Esq
Edificio Barre #2 Primer Piso
Telefono : +593 5 262 8071
Celular  : +593 9 985 5138
e-mail   : [EMAIL PROTECTED]
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Linux User: 255902

Beat me, whip me, make me use Windows!

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Re: [asterisk-users] Operator in Voicemail

[Dovid Bender]

instead of a include for outs side callers how 
about making an ext. in your incoming context that goes to your vm ext. in your 
internal context i.e. Exten => 8000,1,Goto(VMExten,1,1)
 

  - Original Message - 
  From: 
  Anthony Davis 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Monday, July 24, 2006 12:20 
PM
  Subject: RE: [asterisk-users] Operator in 
  Voicemail
  
  
  I’m having the exact same problem here. I 
  originally thought it was a context problem. 
  However, to troubleshoot I tried placing 
  the following in every context (default, from-inside, from-outside, etc) in 
  extensions.conf with no luck:
    exten => 
  o,1,DIAL(SIP/100,100)
   
  Like Kevin, it works fine for our 
  internal users, just doesn’t work for callers coming from the 
  PSTN.
   
  Thanks,
  -AntD
  
  
  
  
  From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Kevin SavoySent: Monday, July 24, 2006 7:37 
  AMTo: 'Asterisk Users 
  Mailing List - Non-Commercial Discussion'Subject: [asterisk-users] Operator in 
  Voicemail
   
  I’ve 
  got an odd problem. I have set in Voicemail.conf operator=yes as a default. 
  This is so that when a caller is in the voicemail system they can press 0 and 
  be sent to the operator. This works fine when the caller is internal to the 
  system but NOT when the caller is calling in from the PSTN. Instead the caller 
  gets the message Press 1 to accept the recording. Pressing 0 again deletes the 
  message. How do I get this to work for outside callers calling 
  in??
   
  Thanks
   
   
   
  _
   
  Kevin 
  Savoy
  Business Unit 
  Telecom Analyst
  2218 4th Ave 
  W
  Williston, ND 58801
  Ph: 
  701-774-4023
  Fax: 
  701-774-2901
  http://www.novo1.com
  Novo 1 is a service mark of Novo 1, 
  Inc
   
  
  

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RE: [asterisk-users] Just bought a Polycom 501 - I feel like myGXP-2000 was better...

[Michael Graves]

On Mon, 24 Jul 2006 18:34:24 -0700, shadowym wrote:

>One of the problems with opinions on these forums IMHO is that they are
>mostly from technical types and not typical end users. 

Typical end users would have no business configuring a PBX such as Asterisk. A 
typical end user would leave that to a professional.

OTOH, those who are responsible for PBX systems often answer to "typical end 
users" in a fashion, and so very likely have faced a number of opinions about 
things...well founded and 
otherwise.

Michael



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Re: [asterisk-users] asterisk extra sounds: what for?

[Dovid Bender]

as it states. extra sounds that dont come with the reg. asterisk tar ball
- Original Message - 
From: "Giorgio Incantalupo" <[EMAIL PROTECTED]>

To: 
Sent: Monday, July 24, 2006 9:16 AM
Subject: [asterisk-users] asterisk extra sounds: what for?



Hi,
what are asterisk extra sounds (asterisk-sounds.x.x.x.tar.gz file) for??

TIA

Giorgio Incantalupo
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Re: [asterisk-users] Just bought a Polycom 501 - I feel like myGXP-2000 was better...

[C F]

Shadow, you mean to say that endusers like the GPX-2000 better than the Polycom?

On 7/24/06, shadowym <[EMAIL PROTECTED]> wrote:

One of the problems with opinions on these forums IMHO is that they are
mostly from technical types and not typical end users.

> -Original Message-
> From: C F [mailto:[EMAIL PROTECTED]
> Sent: Monday, July 24, 2006 2:12 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Just bought a Polycom 501 - I
> feel like myGXP-2000 was better...
>
> Feelings are for the ignorant.
> In any case, if you have trouble pinging your phone then you
> have something wrong on either your network, or you got a
> damaged phone.
> Here is my output from pinging a Polycom 501 while in a
> conversation with app_voicemail:
> Ping statistics for 192.168.1.246:
> Packets: Sent = 100, Received = 100, Lost = 0 (0% loss),
> Approximate round trip times in milli-seconds:
> Minimum = 1ms, Maximum = 2ms, Average = 1ms
>
> On 7/24/06, Mike <[EMAIL PROTECTED]> wrote:
> >
> >
> > Hello everyone,
> >
> > I'm half writing this to get some answers, and half-writing this to
> > put in my 2 cents for anybody who's looking to get his first VoIP
> > phone.  Polycom's seem to be highly regarded here, and after having
> > bought a Polycom 501 as my second phone (my first was the unloved
> > Grandstream GXP-2000), I am left wondering why.
> >
> > Never mind that the setup of the Polycom was more
> complicated than the
> > GXP-2000 by a few order of magnitudes, that only matters the first
> > time you do it.  But things like 3-way conferences are
> harder to use
> > (whatever happened to picking a line, pressing CONF and picking
> > another line) and in general, except for the aesthetics of the
> > phone, the GrandStream is an equal phone (for a much lesser
> price) than the Polycom 501.
> >
> > My worst gripe with this phone, is that I haven't managed
> to have it
> > on the LAN without it disappearing for a few seconds.  If I
> ping the
> > phone (on the same LAN, on the same underused hub actually)
> I get 5-6
> > responses, then timeoutsthen another few responses,
> then timeouts
> > again.  This translates into sound being (badly) cut off when Im
> > talking.  The same experiments yields good results with the
> GXP-2000.
> > I understand ping might not be prioritized on the Polycom, but this
> > was done with no calls coming in or going out.  So why is
> it disappearing?  Or is there a better test?
> >
> > If it helps, my setup can be described as a Asterisk server
> (NO NAT)
> > and a Polycom 501 (behind a Nat).  The Grandstream handles
> that like a
> > pro, I assumed the Polycom would too, considering the reputation of
> > Polycom and the price of the phone.
> >
> > I would be grateful to anyone who clues me in on what I am doing
> > wrong, because I am 95% certain it's somehow my fault.  My gf would
> > tell you it always is :-)
> >
> >
> > Mike
> >
> >
> > ___
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> >
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
>
>
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[asterisk-users] Unicall reload problem

[Jesus Mogollon]

I'm have a problem with Unicall not being able to recover from an Asterisk reload. When I try reloading, Unicall reports:Jul 24 22:50:44 ERROR[9252]: chan_unicall.c:3444 mkintf: Unable to open channel 1: Device or resource busy
here = 0, tmp->channel = 0, channel = 1Jul 24 22:50:44 ERROR[9252]: chan_unicall.c:4216 setup_unicall: Unable to register channel '1-15'Jul 24 22:50:44 WARNING[9252]: chan_unicall.c:4536 reload: Reload of chan_unicall.so is unsuccessful!
Why would this be the case?Jesus Mogollon
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Re: [asterisk-users] Goldmine CRM softphone + asterisk

[Dan Elder]

Anyone gotten the Goldmine (6.7 in our install) softphone to work with
asterisk?



What does your sip.conf look like?  Maybe md5 setting or something?
 



Here's my sip.conf, pretty brief


port = 5060   ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
disallow=all
;allow=ilbc
allow=gsm
allow=ulaw
allow=alaw
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
canreinvite=no
threewaycalling=yes
transfer=yes
progressinband=no
#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf

sip_additional.conf

[3000]
username=3000
type=friend
secret=
record_out=Adhoc
record_in=Adhoc
qualify=no
port=5060
nat=never
[EMAIL PROTECTED]
host=dynamic
dtmfmode=auto
context=from-internal
canreinvite=no
callerid=device <3000>

the server error messages are the things that have me most intrigued, 
like the GM client is doing something odd when trying to register... 
Frontrange say there is no debug capability inside the GM softphone.? Is 
there some way I can get * to tell me more than the error code?


Thx again!
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RE: [asterisk-users] Load balenced (ADSL) network connections, is it possible?

[James Harper]

> Is it possible to somehow have multiple NICs in the server each with a
> different IP address pointing to a different default gateway (router).
But
> then some how load balanced into a virtual network connection?
> 
> Any ideas or solutions would be appreciated - just in case I have gone
off
> at a wild tangent.

I have successfully tested a MultiLink PPPoE connection with DSL tails.
If your ISP supports this then that would be the best way to do it. If
they can supply a 512/512 DSL connection then just aggregate these
together until you have the required amount of bandwidth.

James
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RE: [asterisk-users] Just bought a Polycom 501 - I feel like myGXP-2000 was better...

[shadowym]

One of the problems with opinions on these forums IMHO is that they are
mostly from technical types and not typical end users. 

> -Original Message-
> From: C F [mailto:[EMAIL PROTECTED] 
> Sent: Monday, July 24, 2006 2:12 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Just bought a Polycom 501 - I 
> feel like myGXP-2000 was better...
> 
> Feelings are for the ignorant.
> In any case, if you have trouble pinging your phone then you 
> have something wrong on either your network, or you got a 
> damaged phone.
> Here is my output from pinging a Polycom 501 while in a 
> conversation with app_voicemail:
> Ping statistics for 192.168.1.246:
> Packets: Sent = 100, Received = 100, Lost = 0 (0% loss), 
> Approximate round trip times in milli-seconds:
> Minimum = 1ms, Maximum = 2ms, Average = 1ms
> 
> On 7/24/06, Mike <[EMAIL PROTECTED]> wrote:
> >
> >
> > Hello everyone,
> >
> > I'm half writing this to get some answers, and half-writing this to 
> > put in my 2 cents for anybody who's looking to get his first VoIP 
> > phone.  Polycom's seem to be highly regarded here, and after having 
> > bought a Polycom 501 as my second phone (my first was the unloved 
> > Grandstream GXP-2000), I am left wondering why.
> >
> > Never mind that the setup of the Polycom was more 
> complicated than the 
> > GXP-2000 by a few order of magnitudes, that only matters the first 
> > time you do it.  But things like 3-way conferences are 
> harder to use 
> > (whatever happened to picking a line, pressing CONF and picking 
> > another line) and in general, except for the aesthetics of the 
> > phone, the GrandStream is an equal phone (for a much lesser 
> price) than the Polycom 501.
> >
> > My worst gripe with this phone, is that I haven't managed 
> to have it 
> > on the LAN without it disappearing for a few seconds.  If I 
> ping the 
> > phone (on the same LAN, on the same underused hub actually) 
> I get 5-6 
> > responses, then timeoutsthen another few responses, 
> then timeouts 
> > again.  This translates into sound being (badly) cut off when Im 
> > talking.  The same experiments yields good results with the 
> GXP-2000.  
> > I understand ping might not be prioritized on the Polycom, but this 
> > was done with no calls coming in or going out.  So why is 
> it disappearing?  Or is there a better test?
> >
> > If it helps, my setup can be described as a Asterisk server 
> (NO NAT) 
> > and a Polycom 501 (behind a Nat).  The Grandstream handles 
> that like a 
> > pro, I assumed the Polycom would too, considering the reputation of 
> > Polycom and the price of the phone.
> >
> > I would be grateful to anyone who clues me in on what I am doing 
> > wrong, because I am 95% certain it's somehow my fault.  My gf would 
> > tell you it always is :-)
> >
> >
> > Mike
> >
> >
> > ___
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> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> 
> 
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RE: [asterisk-users] Operator Console(s)/Shared Call Appearances

[shadowym]

I believe they are 'looking' at some sort of implementation.  Supposedly not
until at least v2.6 which is after the next major release (v2.4) I believe.
In other words, it's a ways out!

This is perhaps the biggest obstacle I have to getting client buy in! 

> -Original Message-
> From: Peder @ NetworkOblivion [mailto:[EMAIL PROTECTED] 
> Sent: Monday, July 24, 2006 5:26 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Operator Console(s)/Shared Call 
> Appearances
> 
> Does anybody know if shared appearance / BLA is on the * 
> roadmap?  And if so, when it might appear?  I've seen people 
> asking for it for quite a while, but I've never seen anybody 
> say that it is "in process" or "on the roadmap".
> 
> 
> 
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Re: [asterisk-users] Operator Console(s)/Shared Call Appearances

[BJ Weschke]

On 7/24/06, Peder @ NetworkOblivion <[EMAIL PROTECTED]> wrote:

Does anybody know if shared appearance / BLA is on the * roadmap?  And
if so, when it might appear?  I've seen people asking for it for quite a
while, but I've never seen anybody say that it is "in process" or "on
the roadmap".



You need to subscribe to svn-commits. Mark has made some commits
recently that I believe is starting to lay the initial groundwork for
it.


--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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RE: [asterisk-users] Just bought a Polycom 501 - I feel likemy GXP-2000 was better...

[asterisk]

On Mon, 24 Jul 2006, Mike wrote:

As for the rest, it`s just personal opinion, but I still think the inerface
of the GXP-2000 was a bit better...everything seemed to be where it should
be.  At least now I`ll be able to appreciate the Polycom 501 and judge it in
a functional state.


Yep the polycoms are not intuitive. That's been the reaction from 
everyone who has used them so far. Ciscos Grandstreams Sipuras yes, 
Polycoms no.


Nice speakerphone though. Shame about the configuration xml hell though.

-Dan
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[asterisk-users] Voice with echo

[Carlos Alberto Bernat Orozco]

Hi groupI have my * box installed with a public IP address and I'm testing two extensions. I'm using SJphone for softphone. When I make the call from an extension to another, the voice sounds with echo. Besides sounds like creepy and it seems like a radio (for making a description)
This is my sip.conf :Global Settings:  SIP Port:   5060  Bindaddress:    0.0.0.0  Videosupport:   No  AutoCreatePeer: No
  Allow unknown access:   Yes  Promsic. redir: No  SIP domain support: No  Call to non-local dom.: Yes  URI user is phone no:   No  Our auth realm  asterisk  Realm. auth:    No
  User Agent: Asterisk PBX  MWI checking interval:  10 secs  Reg. context:   (not set)  Caller ID:  asterisk  From: Domain:  Record SIP history: Off  Call Events:    Off
  IP ToS: 0x0  OSP Support:    No  SIP realtime:   DisabledGlobal Signalling Settings:---  Codecs: gsm,ulaw  Relax DTMF: No
  Compact SIP headers:    No  RTP Timeout:    60  RTP Hold Timeout:   0 (Disabled)  MWI NOTIFY mime type:   application/simple-message-summary  DNS SRV lookup: Yes  Pedantic SIP support:   No
  Reg. max duration:  3600 secs  Reg. default duration:  120 secs  Outbound reg. timeout:  20 secs  Outbound reg. attempts: 0  Notify ringing state:   YesDefault Settings:-
  Context:    default  Nat:    RFC3581  DTMF:   rfc2833  Qualify:    0  Use ClientCode: No  Progress inband:    Never  Language:   (Defaults to English)
  Musicclass: default  Voice Mail Extension:   asteriskAnd these are my extensions:;* extension de usuario 1 **exten => 2426098,1,dial(SIP/usuario1)
 exten => usuario1,1,goto(2426098,1) ; To be able to dial with text, "usuario1";* extension de usuario 2 **exten => 2418150,1,dial(SIP/usuario2) exten => usuario2,1,goto(2418150,1) ; To be able to dial with text, "usuario2"
This is an output for the conversation: --- (8 headers 0 lines)---Looking for 200.30.115.163 in default (domain )Transmitting (no NAT) to 
10.1.3.164:5060:SIP/2.0 404 Not FoundVia: SIP/2.0/UDP 10.1.3.164;rport;branch=z9hG4bK0a0103a4001044c56ac56b48061a;received=10.1.3.164
From: ;tag=124002584324To: ;tag=as162c41e3Call-ID: 
[EMAIL PROTECTED]CSeq: 222 OPTIONSUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYMax-Forwards: 70Contact: Accept: application/sdpContent-Length: 0I don't know if there is some problem with the codecs or on my configuration. Do I have to change some line?Thanks for any help.Carlos Bernat

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Re: [asterisk-users] Operator Console(s)/Shared Call Appearances

[Peder @ NetworkOblivion]

Does anybody know if shared appearance / BLA is on the * roadmap?  And 
if so, when it might appear?  I've seen people asking for it for quite a 
while, but I've never seen anybody say that it is "in process" or "on 
the roadmap".


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Re: [asterisk-users] NAT

[Nikolai Lusan]

On Mon, 2006-07-24 at 12:24 +0500, Atif Munir wrote:
> I am interested to configure my linux box for a server for my call
> center. What sort of NAT/IPTABLES I need to implement on my server?
> I have just masqurate the box and it is not workingbut i can have
> local calls ..

Won't work properly. If you bothered to read the docs regaurding VoIP
protocols 

Try looking at siproxd it will basically trans proxy it for you,
assuming you configure it properly.


-- 
Nikolai Lusan

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RE: [asterisk-users] Just bought a Polycom 501 - I feel likemy GXP-2000 was better...

[Mike]

Thanks Eric, you found it.  I just turned off the CDP setting and at first
glance, everything works.  Thanks.  No need to change hub (Im on a small
home network, which is why I can afford having a hub).

As for the rest, it`s just personal opinion, but I still think the inerface
of the GXP-2000 was a bit better...everything seemed to be where it should
be.  At least now I`ll be able to appreciate the Polycom 501 and judge it in
a functional state.

Mike

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
"ManxPower" Wieling
Sent: July 24, 2006 5:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Just bought a Polycom 501 - I feel likemy
GXP-2000 was better...

C F wrote:
> Feelings are for the ignorant.
> In any case, if you have trouble pinging your phone then you have 
> something wrong on either your network, or you got a damaged phone.
> Here is my output from pinging a Polycom 501 while in a conversation 
> with app_voicemail:
> Ping statistics for 192.168.1.246:
>Packets: Sent = 100, Received = 100, Lost = 0 (0% loss), 
> Approximate round trip times in milli-seconds:
>Minimum = 1ms, Maximum = 2ms, Average = 1ms

If he has something on his LAN that supports CDP, the phone is prolly trying
to get it's VLAN info via CDP.  Turn that off in the config file or by using
the interface on the actual phone.


--
Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga,
and Montgomery.
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Re: [asterisk-users] Goldmine CRM softphone + asterisk

[Steve Totaro]

Dan Elder wrote:

Anyone gotten the Goldmine (6.7 in our install) softphone to work with
asterisk? I've been pulling my hair out trying to figure out what the
problem is, the phone doesn't seem to register if I have a password on the
account. If I remove the password, it logs in & can make calls..but with the
pw, it doesn't work at all.. Here's a sip debug of the attempted
registration:

<-- SIP read from 192.168.1.4:5060:
REGISTER sip:192.168.1.253 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5060
To: 
From: ;tag=3b7780
CSeq: 1 REGISTER
Call-ID: [EMAIL PROTECTED]
Contact: 
Max-Forwards: 70
User-Agent: Desktop Phone Object, ver. 3.7.0.64
Expires: 3600
Content-Length: 0


--- (11 headers 0 lines)---
Transmitting (no NAT) to 192.168.1.4:5060:
SIP/2.0 503 Server error
Via: SIP/2.0/UDP 192.168.1.4:5060;received=192.168.1.4
From: ;tag=3b7780
To: ;tag=as794c4d62
Call-ID: [EMAIL PROTECTED]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: 
Content-Length: 0



Where can I find more info on the 503 server error message? Not finding much
w/google... If I put in a bogus password, the error shows bad auth, so I'm
assuming that when I enter the right pass, it's being authenticated, but not
actually registering.

Thanks in adavance for any pointers, been trying to get help from Goldmine
directly, but they're pretty useless & have been trying to fix this for
months now.

Thanks again


  


What does your sip.conf look like?  Maybe md5 setting or something?

Thanks,
Steve
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Re: [asterisk-users] create custom cdr's

[Carlos Chavez]

On Mon, 2006-07-24 at 18:16 -0400, William Piper wrote:
> 
> Thanks but I'd rather not do an AGI. 
> I know it is possible in extensions.conf, I've seen a little
> documentation about it on the web. 
>  
> Anyone have an idea on this?
>  
The only thing you can really do from the dialplan is to use the
CDR(userfield) function to add custom information, but I do not think
there is a way to put additional fields into the CDR database this way.

> 
-- 
Carlos Chavez Prats
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001


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Re: [asterisk-users] Clocking Multiple T1 Cards

[Steve Underwood]

Andrew Kohlsmith wrote:


On Monday 24 July 2006 12:11, Shaw Terwilliger wrote:
 


Thank you; this is the kind of information I was looking for.  The wiki
and other documents told me exactly what the configuration options did,
but I didn't know what kind of timing configuration was right for
multiple cards.
   



Essentially the timing is ONLY for the hardware on the card.  The Digium cards 
use a quad framer chip (maybe a dual for the TE210 but I don't think so) and 
it's a hardware limitation of the framer that all spans must share the same 
clock source.  Sangoma's cards use individual framers and don't have this 
limitation.  (essentially I think it was a cost/space tradeoff.)


Once the data is on the PCI bus, the clock source is irrelevant.  They're all 
close enough that it doesn't matter anymore.


This statement is very very wrong. The timing matters enormously. If the 
timing doesn't match, there will be frame slips, and things like modems 
will not work. The snag is, right now neither Asterisk or the cards it 
uses have the ability to lock their clocks together.


 Those framers want exact 
lock-step timing though, which is why your clocking settings are so very 
important, and why with telephony in general it is crucial to think about 
your clocking before throwing hardware at a solution.
 


Steve

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Re: [asterisk-users] create custom cdr's

[William Piper]

Thanks but I'd rather not do an AGI. 
I know it is possible in extensions.conf, I've seen a little documentation about it on the web. 
 
Anyone have an idea on this?
 
Thanks,
 
bp
 
On 7/24/06, Don <[EMAIL PROTECTED]> wrote:



I do it in AGI...don't use the asterisk CDR table for anything...make your own table and write whatever you want in it via AGI.


- Original Message - 
From: 
William Piper 
To: 
Asterisk Users Mailing List - Non-Commercial Discussion 
Sent: Monday, July 24, 2006 4:40 PM
Subject: [asterisk-users] create custom cdr's
 
List,
 
I'd like to create custom cdr columns in the database.
 
Here is what I'm trying to do:
exten => s,2,Set(CDR(ipaddress)=${ipaddress}) 

I'm able to get the ipaddress and set it to a variable but I'm not sure how to record the IP address to the CDR's. I know it has something to do with cdr_custom.conf but I'm not sure how to put it all together. 

 
FYI, I don't have cdr_custom.conf, is it something that was supposed to be installed with asterisk?
FYI2, I'm using ver 1.2.7.1
 
I know this is possible but I'd love it if someone could give me step by step directions on how to set it up or point me to a website that show this.
 
Thanks for the help,
 
bp
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Re: [asterisk-users] Just bought a Polycom 501 - I feel like my GXP-2000 was better...

[Jerry Jones]

Also on the poly, for the feature you are describing you be using  
join not conference


And on all pbx I have seen poly implements conference just like they  
historically  have



On Jul 24, 2006, at 4:50 PM, Carlos Chavez wrote:


On Mon, 2006-07-24 at 16:55 -0400, Mike wrote:


My worst gripe with this phone, is that I haven't managed to have it
on the LAN without it disappearing for a few seconds.  If I ping the
phone (on the same LAN, on the same underused hub actually) I get 5-6
responses, then timeoutsthen another few responses, then timeouts
again.  This translates into sound being (badly) cut off when Im
talking.  The same experiments yields good results with the GXP-2000.
I understand ping might not be prioritized on the Polycom, but this
was done with no calls coming in or going out.  So why is it
disappearing?  Or is there a better test?


This sounds like an issue with your hub.  Some phones are very
sensitive to cabling and cheap switches.  I would recommend that  
you try

another switch (specially if you are really using a hub) to test the
phone.  I have lots of Polycom and GXP-2000 phones and they all work
fine in the same network, only when we have used really cheap switches
do we run into any problems.




--
Carlos Chavez Prats
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001
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Re: [asterisk-users] Operator Console(s)/Shared Call Appearances

[Jerry Jones]

Asterisk does not yet support bridged calls

You can easily have a button labeled exec 1 ring on her phone at the  
same time it rings the execs phone, and have one light if he is on  
the phone


Also FOP works great

On Jul 23, 2006, at 3:42 PM, Mr. Jones wrote:


Thanks Sebastian -

You're right - I have limited experience in this area :)

I think the idea below is workable, except we actually want it to work
in the other direction - sort of.

Essentially we want the receptionist to screen the calls when she's
available. The executive should have option to answer the phone if its
after hours, or they know the receptionist isn't available (or perhaps
they recognize the caller ID and just want to take the call).

Can you think of how this might work? I suppose the executive could be
a member of his own queue?


What do you think about this idea;
1. Call comes in at one of the executive numbers.
2. Executive phone starts ringing for a predetermined time.
3. The callerid is changed to also reflect the name/number of called
executive, so that the receptionist knows for who the call was.
4. The call is dropped into a queue for the receptionist (queue  
because

multiple calls to the receptionist at the same time are possible).


This setup isn't all that hard, and doesn't require more than 4 sip
accounts / phones and one queue, with one agent. Furthermore, if your
company starts to grow, and more receptionists that have to answer  
the

phone are needed, it's quite easy, all you have to do is add a sip
account, one agent and add that agent to the existing queue. (About 2
minutes...)

--
Sebastian Berm
iPronto Communications
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Re: [asterisk-users] Just bought a Polycom 501 - I feel like my GXP-2000 was better...

[Carlos Chavez]

On Mon, 2006-07-24 at 16:55 -0400, Mike wrote:
>  
> My worst gripe with this phone, is that I haven't managed to have it
> on the LAN without it disappearing for a few seconds.  If I ping the
> phone (on the same LAN, on the same underused hub actually) I get 5-6
> responses, then timeoutsthen another few responses, then timeouts
> again.  This translates into sound being (badly) cut off when Im
> talking.  The same experiments yields good results with the GXP-2000.
> I understand ping might not be prioritized on the Polycom, but this
> was done with no calls coming in or going out.  So why is it
> disappearing?  Or is there a better test?
>  
This sounds like an issue with your hub.  Some phones are very
sensitive to cabling and cheap switches.  I would recommend that you try
another switch (specially if you are really using a hub) to test the
phone.  I have lots of Polycom and GXP-2000 phones and they all work
fine in the same network, only when we have used really cheap switches
do we run into any problems.

> 
-- 
Carlos Chavez Prats
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001


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Re: [asterisk-users] Just bought a Polycom 501 - I feel like my GXP-2000 was better...

[Eric \"ManxPower\" Wieling]

C F wrote:

Feelings are for the ignorant.
In any case, if you have trouble pinging your phone then you have
something wrong on either your network, or you got a damaged phone.
Here is my output from pinging a Polycom 501 while in a conversation
with app_voicemail:
Ping statistics for 192.168.1.246:
   Packets: Sent = 100, Received = 100, Lost = 0 (0% loss),
Approximate round trip times in milli-seconds:
   Minimum = 1ms, Maximum = 2ms, Average = 1ms


If he has something on his LAN that supports CDP, the phone is prolly 
trying to get it's VLAN info via CDP.  Turn that off in the config file 
or by using the interface on the actual phone.



--
Now accepting new clients in Birmingham, Atlanta, Huntsville, 
Chattanooga, and Montgomery.

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Re: [asterisk-users] Urgent source code changes needed

[Joe Pukepail]

Why don't you detail what you are trying to accomplish on the list, perhaps someone will do it for free.  If it is a legitimate bug you could add an incident to the bug tracker. 
 
On 7/24/06, Bart Fisher <[EMAIL PROTECTED]> wrote:
I need someone to "patch" what I believe to be a simple change tochan_zap.c - I know if I attempt I'll screw it up :)
Whom would you approach for doing this? - My requests have received a'blank stare' from Free Lance sites and I'm running out of time on thisinstall.If you know someone or could handle this yourself, please contact me at
[EMAIL PROTECTED]ThanksBart___--Bandwidth and Colocation provided by Easynews.com
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Re: [asterisk-users] create custom cdr's

[Tristan]

It's even possible in the dialplan with files or database ;)


Regards,

Tristan

Don a écrit :
I do it in AGI...don't use the asterisk CDR table for anything...make 
your own table and write whatever you want in it via AGI.


- Original Message -
*From:* William Piper 
*To:* Asterisk Users Mailing List - Non-Commercial Discussion

*Sent:* Monday, July 24, 2006 4:40 PM
*Subject:* [asterisk-users] create custom cdr's

List,
 
I'd like to create custom cdr columns in the database.
 
Here is what I'm trying to do:

exten => s,2,Set(CDR(ipaddress)=${ipaddress})
 
I'm able to get the ipaddress and set it to a variable but I'm not

sure how to record the IP address to the CDR's. I know it has
something to do with cdr_custom.conf but I'm not sure how to put
it all together.
 
FYI, I don't have cdr_custom.conf, is it something that was

supposed to be installed with asterisk?
FYI2, I'm using ver 1.2.7.1 
 
I know this is possible but I'd love it if someone could give me

step by step directions on how to set it up or point me to a
website that show this.
 
Thanks for the help,
 
bp



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No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.1.394 / Virus Database: 268.10.4/396 - Release Date:
7/24/2006



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Re: [asterisk-users] Intercom feature on Polycom phones

[Alex Robar]

You could park them, and then use the intercom feature. The intercom feature in Polycoms requires you to set the SIP Header information in your call, and also use the XML config files to tell the Polycom how to answer a call with the header you will be sending. 
If you want all your phones to react the same way, do this in you sip.cfg file. Otherwise, do this in the cfg for the individual phone. Within the   tags, put this: In my extensions.conf I've just put this:exten => _*51XXX,1,SIPAddHeader(Alert-Info: Ring Answer)exten => _*51XXX,2,Dial(sip/${EXTEN:3:3})
exten => _*51XXX,3,HangupAnyone who needs to intercom dials via *51EXT, where ext is the three digit extension of the person they need to intercom.So, if your secretary were to park the person first, then intercom the called party with the parking space the person is in, this should be the solution to your issue.
AlexOn 7/24/06, C F <[EMAIL PROTECTED]> wrote:
And how exactly do  you plan on that person picking up line 2?  or isthat part of your question?On 7/24/06, Stephen Murphy <[EMAIL PROTECTED]> wrote:
> Basically the receptionist would like to have the ability to put a call on> hold and intercom an extension and annouce they have a call on say line 2>> -Original Message-> From: 
[EMAIL PROTECTED]> [mailto:[EMAIL PROTECTED]] On Behalf Of C F
> Sent: July 24, 2006 9:58 AM> To: Asterisk Users Mailing List - Non-Commercial Discussion> Subject: Re: [asterisk-users] Intercom feature on Polycom phones>> What do you mean by similar to auto answer?
>> On 7/24/06, Stephen Murphy <[EMAIL PROTECTED]> wrote:> >> >> >> >> > I have the situation where my client would like to 'Intercom' an extension
> > similar to auto-answer. I have polycom phones - can this be done?> >> >> >> > 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> >> >> >> ___> --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
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Re: [asterisk-users] ERROR 1045 (28000): Access denied for user

[Alex Robar]

This isn't even a question... It's just an error. Worse is that it appears to be an error for OpenSER, not Asterisk. Even so, your error looks to be a simple authentication error. Make sure the root user exists for your MySQL install and that you can login to MySQL via the command line with these credentials.
AlexOn 7/24/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]
> wrote:MySql password for root:Domain (realm) for the default user 'admin': 
localhost.localdomaincreating database openser ...ERROR 1045 (28000): Access denied for user 'root'@'localhost' (using password: Y ES)___--Bandwidth and Colocation provided by 
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Re: [asterisk-users] Just bought a Polycom 501 - I feel like my GXP-2000 was better...

[C F]

Feelings are for the ignorant.
In any case, if you have trouble pinging your phone then you have
something wrong on either your network, or you got a damaged phone.
Here is my output from pinging a Polycom 501 while in a conversation
with app_voicemail:
Ping statistics for 192.168.1.246:
   Packets: Sent = 100, Received = 100, Lost = 0 (0% loss),
Approximate round trip times in milli-seconds:
   Minimum = 1ms, Maximum = 2ms, Average = 1ms

On 7/24/06, Mike <[EMAIL PROTECTED]> wrote:



Hello everyone,

I'm half writing this to get some answers, and half-writing this to put in
my 2 cents for anybody who's looking to get his first VoIP phone.  Polycom's
seem to be highly regarded here, and after having bought a Polycom 501 as my
second phone (my first was the unloved Grandstream GXP-2000), I am left
wondering why.

Never mind that the setup of the Polycom was more complicated than the
GXP-2000 by a few order of magnitudes, that only matters the first time you
do it.  But things like 3-way conferences are harder to use (whatever
happened to picking a line, pressing CONF and picking another line) and
in general, except for the aesthetics of the phone, the GrandStream is an
equal phone (for a much lesser price) than the Polycom 501.

My worst gripe with this phone, is that I haven't managed to have it on the
LAN without it disappearing for a few seconds.  If I ping the phone (on the
same LAN, on the same underused hub actually) I get 5-6 responses, then
timeoutsthen another few responses, then timeouts again.  This
translates into sound being (badly) cut off when Im talking.  The same
experiments yields good results with the GXP-2000.  I understand ping might
not be prioritized on the Polycom, but this was done with no calls coming in
or going out.  So why is it disappearing?  Or is there a better test?

If it helps, my setup can be described as a Asterisk server (NO NAT) and a
Polycom 501 (behind a Nat).  The Grandstream handles that like a pro, I
assumed the Polycom would too, considering the reputation of Polycom and the
price of the phone.

I would be grateful to anyone who clues me in on what I am doing wrong,
because I am 95% certain it's somehow my fault.  My gf would tell you it
always is :-)


Mike


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Re: [asterisk-users] create custom cdr's

[Don]

I do it in AGI...don't use the asterisk CDR table 
for anything...make your own table and write whatever you want in it via 
AGI.

  - Original Message - 
  From: 
  William 
  Piper 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Monday, July 24, 2006 4:40 PM
  Subject: [asterisk-users] create custom 
  cdr's
  
  List,
   
  I'd like to create custom cdr columns in the database.
   
  Here is what I'm trying to do:
  exten => s,2,Set(CDR(ipaddress)=${ipaddress}) 
  
  I'm able to get the ipaddress and set it to a variable but 
  I'm not sure how to record the IP address to the CDR's. I know 
  it has something to do with cdr_custom.conf but I'm not sure how to put it all 
  together. 
   
  FYI, I don't have cdr_custom.conf, is it something that was supposed to 
  be installed with asterisk?
  FYI2, I'm using ver 1.2.7.1
   
  I know this is possible but I'd love it if someone could give me step by 
  step directions on how to set it up or point me to a website that show 
  this.
   
  Thanks for the help,
   
  bp
  
  

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  No virus found in this incoming message.Checked by AVG Free 
  Edition.Version: 7.1.394 / Virus Database: 268.10.4/396 - Release Date: 
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Re: [asterisk-users] Intercom feature on Polycom phones

[C F]

And how exactly do  you plan on that person picking up line 2?  or is
that part of your question?

On 7/24/06, Stephen Murphy <[EMAIL PROTECTED]> wrote:

Basically the receptionist would like to have the ability to put a call on
hold and intercom an extension and annouce they have a call on say line 2

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: July 24, 2006 9:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Intercom feature on Polycom phones

What do you mean by similar to auto answer?

On 7/24/06, Stephen Murphy <[EMAIL PROTECTED]> wrote:
>
>
>
>
> I have the situation where my client would like to 'Intercom' an extension
> similar to auto-answer. I have polycom phones - can this be done?
>
>
>
> Steve
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> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
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[asterisk-users] ERROR 1045 (28000): Access denied for user

[broadbandvoice]

MySql password for root:
Domain (realm) for the default user 'admin': localhost.localdomain

creating database openser ...
ERROR 1045 (28000): Access denied for user 'root'@'localhost' (using password: 
Y ES)
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[asterisk-users] sipbuddies realtime fields and latest documentation

[Thomas Winter]

Hi,

I used voip-info.org for setup my realtime users.
The mySQL table did not include for example the option call-limit.

Where I can find information whats the correct field name to adjust my mySQL 
table?

thanks

Thomas

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[asterisk-users] sms on wifi phones

[Jerry Geis]

I need help setting up SMS on a wifi phone?
Any done this?

I have a new DPH-540 from dlink. It says it does text messaging.
I tried sending a text message with SendMessage() and the CLI tells me 
the phone

does not support this.

I do the exact same thing with hitachi wip 5000 and that works.

The advertising says the dph-540 does text messaging so I presume we
are talking SMS now.
How do you setup SMS on a wifi phone.

THanks,

Jerry
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[asterisk-users] Just bought a Polycom 501 - I feel like my GXP-2000 was better...

[Mike]

Hello 
everyone,
 
I'm half writing 
this to get some answers, and half-writing this to put in my 2 cents for anybody 
who's looking to get his first VoIP phone.  Polycom's seem to 
be highly regarded here, and after having bought a Polycom 501 as my second 
phone (my first was the unloved Grandstream GXP-2000), I am left wondering 
why.
 
Never mind that the 
setup of the Polycom was more complicated than the GXP-2000 by a few order of 
magnitudes, that only matters the first time you do it.  But things like 
3-way conferences are harder to use (whatever happened to picking a line, 
pressing CONF and picking another line) and in general, except for the 
aesthetics of the phone, the GrandStream is an equal phone (for a much lesser 
price) than the Polycom 501.
 
My worst gripe with 
this phone, is that I haven't managed to have it on the LAN without it 
disappearing for a few seconds.  If I ping the phone (on the same LAN, on 
the same underused hub actually) I get 5-6 responses, then timeoutsthen 
another few responses, then timeouts again.  This translates into sound 
being (badly) cut off when Im talking.  The same experiments yields good 
results with the GXP-2000.  I understand ping might not be prioritized on 
the Polycom, but this was done with no calls coming in or going out.  So 
why is it disappearing?  Or is there a better test?
 
If it helps, my 
setup can be described as a Asterisk server (NO NAT) and a Polycom 501 (behind a 
Nat).  The Grandstream handles that like a pro, I assumed the Polycom would 
too, considering the reputation of Polycom and the price of the phone.  

 
I would be grateful 
to anyone who clues me in on what I am doing wrong, because I am 95% certain 
it's somehow my fault.  My gf would tell you it always is 
:-)
 
 
Mike
 
 
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[asterisk-users] RDNIS and IAX2

[Douglas Garstang]

I'll probably get blasted for this. I hope I'm wrong, and then a little 
blasting is ok. It appears that Asterisk may have let us down again as a 
'carrier grade' solution.

1. User A calls User B. The call is bridged.
2. User B wants to transfer User A to user C. When this happens, User B's phone 
sends a new call to Asterisk with RDNIS info contained in the SIP INVITE header.
3. However, user C isn't registered on the local system, so we do a DUNDi 
lookup to get an IAX2 path to the location of user C
4. We then connect to this DUNDi supplied IAX path so that we can dial user C, 
who is registered on a different Asterisk system.

It appears from this link that the Asterisk RDNIS implementation is completely 
broken. 
http://www.voip-info.org/wiki/view/RDNIS

Most importantly, RDNIS info is not passed along the IAX channel when the call 
is trunked to the Asterisk system where User C is registered. This makes it 
impossible to tell that this was a transferred call, and I'll spare the details 
right now as to why this breaks a whole lot of other things that we are trying 
to implement (like resetting the caller id to the original caller and so on).

Doug
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[asterisk-users] create custom cdr's

[William Piper]

List,
 
I'd like to create custom cdr columns in the database.
 
Here is what I'm trying to do:
exten => s,2,Set(CDR(ipaddress)=${ipaddress}) 

I'm able to get the ipaddress and set it to a variable but I'm not sure how to record the IP address to the CDR's. I know it has something to do with cdr_custom.conf but I'm not sure how to put it all together. 

 
FYI, I don't have cdr_custom.conf, is it something that was supposed to be installed with asterisk?
FYI2, I'm using ver 1.2.7.1
 
I know this is possible but I'd love it if someone could give me step by step directions on how to set it up or point me to a website that show this.
 
Thanks for the help,
 
bp
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Re: [asterisk-users] Polycom_acd_functions SIP trouble

[BJ Weschke]

On 7/24/06, Dean @ INKnBITs <[EMAIL PROTECTED]> wrote:

I've got the same problem, the only version I can find that works is 30432,
but the meetme conference does not compile in this version. A fix for the
newest version for username/auth name would be great!


- Original Message -
From: "James Fromm" <[EMAIL PROTECTED]>
To: 
Sent: Monday, July 24, 2006 7:24 PM
Subject: [asterisk-users] Polycom_acd_functions SIP trouble


> I'm trying to use the latest revision of Bweschke's branch from SVN for
> polycom_acd_functions.  Asterisk builds and runs without error but all
> SIP devices can't register when specifying a secret in sip.conf.  The
> Polycom 601 I'm testing with and a copy of SJphone will not register.
> IAX from Idefisk works without error.
>


It's going to be a bit guys before I can get in and figure out what's
going on with the later revisions of the branches. We're pretty tied
up with client work which comes first, but I'll send an email back out
as soon as we've had an opportunity to troubleshoot this and get a
resolution committed.

BJ

--
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Re: [asterisk-users] Transfers - No ringback or moh

[Nathan Bowyer]

On 7/24/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:

>
> Douglas Garstang wrote:
> >> -Original Message-
> >>
> >
> > When the extension on your desk is ringing, after you have
> pressed transfer key a second time(soft or hard key), does
> the original caller still hear music on hold, or ringback or nothing?
> >
> >
>
> Following your example, pressing transfer once, entering the
> extension
> (Caller C's) does not yield a second transfer option until C
> answers.
> Pressing the button anyway, does not get a response from the phone.
>
> When (B) selects the initial transfer, I have Cancel, Name, Blind.
>
> (A) hears hold music. During the transfer and Before (C) answers, the
> phone options are Cancel, Split.
>
> Once C answers, I have, Hold, Cancel, Transfer, More

We have SIP version 1.6.3. Polycom must have changed something...

Doug.


Not exactly.  There's an option in the Polycom config to disallow
"unattended" attended transfers.  In other words, you do not have the
option to press "Transfer" while you are getting in a RINGING progress
state.  Sounds Like one of you has that option enabled, the other has
it disabled.  I don't know exactly what the option is, but I've seen
it before.

Nathan
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RE: [asterisk-users] Transfers - No ringback or moh

[Douglas Garstang]

> -Original Message-
> From: Doug Lytle [mailto:[EMAIL PROTECTED]
> Sent: Monday, July 24, 2006 2:15 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Transfers - No ringback or moh
> 
> 
> Douglas Garstang wrote:
> >> -Original Message-
> >> 
> >   
> > When the extension on your desk is ringing, after you have 
> pressed transfer key a second time(soft or hard key), does 
> the original caller still hear music on hold, or ringback or nothing?
> >
> >   
> 
> Following your example, pressing transfer once, entering the 
> extension 
> (Caller C's) does not yield a second transfer option until C 
> answers.  
> Pressing the button anyway, does not get a response from the phone.
> 
> When (B) selects the initial transfer, I have Cancel, Name, Blind.
> 
> (A) hears hold music. During the transfer and Before (C) answers, the 
> phone options are Cancel, Split.
> 
> Once C answers, I have, Hold, Cancel, Transfer, More

We have SIP version 1.6.3. Polycom must have changed something...

Doug.
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Re: [asterisk-users] Transfers - No ringback or moh

[Doug Lytle]

Douglas Garstang wrote:

-Original Message-

  
When the extension on your desk is ringing, after you have pressed transfer key a second time(soft or hard key), does the original caller still hear music on hold, or ringback or nothing?


  


Following your example, pressing transfer once, entering the extension 
(Caller C's) does not yield a second transfer option until C answers.  
Pressing the button anyway, does not get a response from the phone.


When (B) selects the initial transfer, I have Cancel, Name, Blind.

(A) hears hold music. During the transfer and Before (C) answers, the 
phone options are Cancel, Split.


Once C answers, I have, Hold, Cancel, Transfer, More


Doug

--

Ben Franklin quote:

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deserve neither Liberty nor Safety."


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RE: [asterisk-users] Intercom feature on Polycom phones

[Stephen Murphy]

Basically the receptionist would like to have the ability to put a call on
hold and intercom an extension and annouce they have a call on say line 2

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: July 24, 2006 9:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Intercom feature on Polycom phones

What do you mean by similar to auto answer?

On 7/24/06, Stephen Murphy <[EMAIL PROTECTED]> wrote:
>
>
>
>
> I have the situation where my client would like to 'Intercom' an extension
> similar to auto-answer. I have polycom phones - can this be done?
>
>
>
> Steve
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>
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Re: [asterisk-users] Transfers - No ringback or moh

[Doug Lytle]

Douglas Garstang wrote:

People (Users that is) seem to think, pressing less is better.



Doug, as it turns out, the transfer button on the polycom and the transfer soft button, both behave in exactly the same way. 

  

What firmware?

I'm running Bootrom 3.1.3, sip.ld 1.5.2

Doug

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RE: [asterisk-users] Transfers - No ringback or moh

[Douglas Garstang]

> -Original Message-
> From: Doug Lytle [mailto:[EMAIL PROTECTED]
> Sent: Monday, July 24, 2006 1:35 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Transfers - No ringback or moh
> 
> 
> Douglas Garstang wrote:
> >> passes at all?
> >> 
> >
> > I don't think this is the same scenario. When you transfer 
> to an Asterisk extension, ie voicemail, your not going to get 
> a period of ring back as Asterisk will answer the call immediately.
> >
> >   
> 
> In this example, I'm dialing to another extension on my desk.

When the extension on your desk is ringing, after you have pressed transfer key 
a second time(soft or hard key), does the original caller still hear music on 
hold, or ringback or nothing?

Douglas.
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Re: [asterisk-users] Polycom_acd_functions SIP trouble

[Dean @ INKnBITs]

I've got the same problem, the only version I can find that works is 30432,
but the meetme conference does not compile in this version. A fix for the
newest version for username/auth name would be great!


- Original Message -
From: "James Fromm" <[EMAIL PROTECTED]>
To: 
Sent: Monday, July 24, 2006 7:24 PM
Subject: [asterisk-users] Polycom_acd_functions SIP trouble


> I'm trying to use the latest revision of Bweschke's branch from SVN for
> polycom_acd_functions.  Asterisk builds and runs without error but all
> SIP devices can't register when specifying a secret in sip.conf.  The
> Polycom 601 I'm testing with and a copy of SJphone will not register.
> IAX from Idefisk works without error.
>
> The error all SIP devices get is:
>
> Jul 24 10:26:48 NOTICE[31524]: chan_sip.c:14203 handle_request_register:
> Registration from '' failed for
> '192.168.0.95' - Username/auth name mismatch
>
> Commenting the definition of a secret in sip.conf for the device solves
> this.  Here's the config for one of the devices.
>
> [1003]
> type=friend
> canreinvite=no
> host=dynamic
> username=1003
> ; secret=stuff
> context=outbound
> callerid="Jimmy" <1003>
> [EMAIL PROTECTED]
> nat=no
>
> Why won't this revision accept the definition of a secret?  Am I missing
> something simple (stupid)?
>
> Thanks,
> Jay
>
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[asterisk-users] Urgent source code changes needed

[Bart Fisher]

I need someone to "patch" what I believe to be a simple change to 
chan_zap.c - I know if I attempt I'll screw it up :)


Whom would you approach for doing this? - My requests have received a 
'blank stare' from Free Lance sites and I'm running out of time on this 
install.


If you know someone or could handle this yourself, please contact me at 
[EMAIL PROTECTED]


Thanks

Bart


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RE: [asterisk-users] Operator in Voicemail

[Henk]

It is an Oh

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: maandag 24 juli 2006 21:33
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Operator in Voicemail

Are you sure this is saying "exten = 0" with a ZERO and not an Oh?
Looks like a lowercase Oh to me below.


Kevin Savoy wrote:
> This doesn't solve the problem. Still the same. Any other ideas?
> 
>  
> 
> 
> 
> This is what I am using:
> 
>  
> 
> exten = o,1,Answer()
> 
> exten = o,2,GoTo(default,3000,1)
> 
> exten = o,3,Hangup()
> 
>  
> 
> Hope this helps,
> 
>  
> 
> Henk
> 
>  
> 
> 
> 
> *From:* [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] *On Behalf Of *Anthony 
> Davis
> *Sent:* maandag 24 juli 2006 18:20
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* RE: [asterisk-users] Operator in Voicemail
> 
>  
> 
> I'm having the exact same problem here. I originally thought it was a 
> context problem.
> 
> However, to troubleshoot I tried placing the following in every context 
> (default, from-inside, from-outside, etc) in extensions.conf with no luck:
> 
>  / exten => o,1,DIAL(SIP/100,100)/
> 
>  
> 
> Like Kevin, it works fine for our internal users, just doesn't work for 
> callers coming from the PSTN.
> 
>  
> 
> Thanks,
> 
> -AntD
> 
> 
> 
> *From:* [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] *On Behalf Of *Kevin
Savoy
> *Sent:* Monday, July 24, 2006 7:37 AM
> *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
> *Subject:* [asterisk-users] Operator in Voicemail
> 
>  
> 
> I've got an odd problem. I have set in Voicemail.conf operator=yes as a 
> default. This is so that when a caller is in the voicemail system they 
> can press 0 and be sent to the operator. This works fine when the caller 
> is internal to the system but NOT when the caller is calling in from the 
> PSTN. Instead the caller gets the message Press 1 to accept the 
> recording. Pressing 0 again deletes the message. How do I get this to 
> work for outside callers calling in??

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Re: [asterisk-users] PAP2 TUI Configuration Menu

[Jamin W. Collins]

Nabeel Jafferali wrote:

I don't belive there is a way to turn it off, but you can prevent the IVR
menu being used to factory reset the device using the provisioning tools.


Would you happen to have any more specifics on this?  Perhaps an example 
of the reset option being disabled?  Can any of the other options, such 
as IP playback, be disabled?


--
Jamin W. Collins
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RE: [asterisk-users] Operator in Voicemail

[Kevin Savoy]

It is an o as in operator. That's what the manuals say. O extension is
operator.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Monday, July 24, 2006 2:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Operator in Voicemail

Are you sure this is saying "exten = 0" with a ZERO and not an Oh?
Looks like a lowercase Oh to me below.


Kevin Savoy wrote:
> This doesn't solve the problem. Still the same. Any other ideas?
> 
>  
> 
> 
> 
> This is what I am using:
> 
>  
> 
> exten = o,1,Answer()
> 
> exten = o,2,GoTo(default,3000,1)
> 
> exten = o,3,Hangup()
> 
>  
> 
> Hope this helps,
> 
>  
> 
> Henk
> 
>  
> 
> 
> 
> *From:* [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] *On Behalf Of *Anthony 
> Davis
> *Sent:* maandag 24 juli 2006 18:20
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* RE: [asterisk-users] Operator in Voicemail
> 
>  
> 
> I'm having the exact same problem here. I originally thought it was a 
> context problem.
> 
> However, to troubleshoot I tried placing the following in every context 
> (default, from-inside, from-outside, etc) in extensions.conf with no luck:
> 
>  / exten => o,1,DIAL(SIP/100,100)/
> 
>  
> 
> Like Kevin, it works fine for our internal users, just doesn't work for 
> callers coming from the PSTN.
> 
>  
> 
> Thanks,
> 
> -AntD
> 
> 
> 
> *From:* [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] *On Behalf Of *Kevin
Savoy
> *Sent:* Monday, July 24, 2006 7:37 AM
> *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
> *Subject:* [asterisk-users] Operator in Voicemail
> 
>  
> 
> I've got an odd problem. I have set in Voicemail.conf operator=yes as a 
> default. This is so that when a caller is in the voicemail system they 
> can press 0 and be sent to the operator. This works fine when the caller 
> is internal to the system but NOT when the caller is calling in from the 
> PSTN. Instead the caller gets the message Press 1 to accept the 
> recording. Pressing 0 again deletes the message. How do I get this to 
> work for outside callers calling in??

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Re: [asterisk-users] Transfers - No ringback or moh

[Doug Lytle]

Douglas Garstang wrote:

passes at all?



I don't think this is the same scenario. When you transfer to an Asterisk 
extension, ie voicemail, your not going to get a period of ring back as 
Asterisk will answer the call immediately.

  


In this example, I'm dialing to another extension on my desk.

Doug

--

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deserve neither Liberty nor Safety."


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Re: [asterisk-users] Operator in Voicemail

[Rich Adamson]

Are you sure this is saying "exten = 0" with a ZERO and not an Oh?
Looks like a lowercase Oh to me below.


Kevin Savoy wrote:

This doesn’t solve the problem. Still the same. Any other ideas?

 




This is what I am using:

 


exten = o,1,Answer()

exten = o,2,GoTo(default,3000,1)

exten = o,3,Hangup()

 


Hope this helps,

 


Henk

 




*From:* [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] *On Behalf Of *Anthony 
Davis

*Sent:* maandag 24 juli 2006 18:20
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* RE: [asterisk-users] Operator in Voicemail

 

I’m having the exact same problem here. I originally thought it was a 
context problem.


However, to troubleshoot I tried placing the following in every context 
(default, from-inside, from-outside, etc) in extensions.conf with no luck:


 / exten => o,1,DIAL(SIP/100,100)/

 

Like Kevin, it works fine for our internal users, just doesn’t work for 
callers coming from the PSTN.


 


Thanks,

-AntD



*From:* [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] *On Behalf Of *Kevin Savoy

*Sent:* Monday, July 24, 2006 7:37 AM
*To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
*Subject:* [asterisk-users] Operator in Voicemail

 

I’ve got an odd problem. I have set in Voicemail.conf operator=yes as a 
default. This is so that when a caller is in the voicemail system they 
can press 0 and be sent to the operator. This works fine when the caller 
is internal to the system but NOT when the caller is calling in from the 
PSTN. Instead the caller gets the message Press 1 to accept the 
recording. Pressing 0 again deletes the message. How do I get this to 
work for outside callers calling in??


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RE: [asterisk-users] Transfers - No ringback or moh

[Douglas Garstang]

> -Original Message-
> From: Doug Lytle [mailto:[EMAIL PROTECTED]
> Sent: Monday, July 24, 2006 12:56 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Transfers - No ringback or moh
> 
> 
> Douglas Garstang wrote:
> >> -Original Message-
> >> From: Doug Lytle [mailto:[EMAIL PROTECTED]
> >> Sent: Monday, July 24, 2006 12:14 PM
> >> To: Asterisk Users Mailing List - Non-Commercial Discussion
> >> Subject: Re: [asterisk-users] Transfers - No ringback or moh
> >>
> >>
> >> 
> > Doug,
> >
> > The transfer soft button can do both attended and non 
> attended transfers. If user B presses the transfer soft 
> button before user C picks up, it's an unattended transfer. 
> If user B presses the transfer soft button after user C has 
> answered, then it's an attended transfer.
> >
> >   
> I'm not able reproduct this.  If I don't select Blind before entering 
> the extension or number when transferring to user C, it only offers a 
> cancel or split.  Once user C answers, then I'm offered the Transfer 
> soft button.
> 
> > Don't see what the Polycom digit map has to do with it. 
> >   
> 
> Not wanting to press yet another button.
> 
> People (Users that is) seem to think, pressing less is better.

Doug, as it turns out, the transfer button on the polycom and the transfer soft 
button, both behave in exactly the same way. 

If user B wants to transfer user A to user C, ATTENDED, user B simply presses 
the transfer button and waits for user C to pick up before pressing the 
transfer button again. 

If user B wants to transfer user A to user C, UNATTENDED, user B simply presses 
the transfer button and presses it again before user C picks up.

In any case, when the phone of user C is ringing, user A does not hear a ring.

Doug.
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RE: [asterisk-users] Transfers - No ringback or moh

[Douglas Garstang]

> -Original Message-
> From: Doug Lytle [mailto:[EMAIL PROTECTED]
> Sent: Monday, July 24, 2006 11:48 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Transfers - No ringback or moh
> 
> 
> Douglas Garstang wrote:
> >> -Original Message-
> >> 
> > And that's normal, for user A to just hear dead air?
> >
> >   
> 
> 
> I have a Polycom IP501 sitting on my desk (Test phone):
> 
> I call it with my Avaya phone
> pick up the ringing extension
> press transfer button (I hear hold music on the Avaya)
> I dial the voice mail extension on the Asterisk
> I press the transfer button again.
> Hold music stops and I hear Comedian Mail.
> 
> So, is the dead air that you hear, the silence that you would 
> get when 
> the hold music stops and both parties have been bridged, or 
> no audio is 
> passes at all?

I don't think this is the same scenario. When you transfer to an Asterisk 
extension, ie voicemail, your not going to get a period of ring back as 
Asterisk will answer the call immediately.

Douglas.
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Re: [asterisk-users] Asterisk Realtime Macros

[Ron Wellsted]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Jon Scottorn wrote:
> Hi,
> 
>I am trying to get asterisk Realtime to work.  I have a fresh
> installed 1.2.10 setup on a debian system.  I have taken the defaul
> setup and put it into the mysql database. 
> I have setup two extensions 101 and 102. 
> 
> If I setup the extension like such:
> 
>exten => 101,1,Dial(SIP/101)
>exten => 102,1,Dial)SIP/102)
> 
> I can dial back and forth between the two phones.
> 
> When I switch it to use the stdexten macro and change the extension like
> such
> 
>exten => 101,1,Macro(stdexten,101,sip/101)
>exten => 102,1,Macro(stdexten,102,sip/102)
> 
> I can not dial each extension and this is what reports on asterisk cli:
> 
> -- SIP Seeding peer from astdb: '102' at [EMAIL PROTECTED]:5060 for 3600
> -- SIP Seeding peer from astdb: '101' at [EMAIL PROTECTED]:1093 for 3600
> -- Executing Macro("SIP/101-081a7f90", "stdexten,102,sip/102")
> Jul 24 10:36:37 WARNING[23358]: app_macro.c:149 macro_exec: No such
> context 'macro-stdexten,102,sip/102' for macro 'stdexten,102,sip/102'
>   == Auto fallthrough, channel 'SIP/101-081a7f90' status is 'UNKNOWN'
> -- SIP Seeding peer from astdb: '101' at [EMAIL PROTECTED]:1093 for 3600
> 
> My question is what has to be in the mysql extenstions_table to get the
> macro to work?
> 
> Here is what is in my extensions_table:
> 
> mysql> select * from extensions_table;
> ++-+---+--+---+--+
> | id | context | exten | priority | app   | appdata  |
> ++-+---+--+---+--+
> |  1 | default | 101   |1 | Macro | stdexten,101,sip/101 |
> |  2 | default | 102   |1 | Macro | stdexten,102,sip/102 |
> 
> Thanks in advance for any help.
> 
> */Jon Scottorn/*
> /Systems Administrator/
> /The Possibility Forge, Inc./
> /http://www.possibilityforge.com/
> /435.635.0591 x.1004/

You must use "|" as the separator instead of "," in the realtime engine
so stdexten,101,sip/101 should be stdexten|101|sip/101 etc.

- --
Ron Wellsted
[EMAIL PROTECTED] http://www.wellsted.org.uk
N 52.567623, W 2.137621 Linux Counter No. 202120
FWD:519961
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[asterisk-users] Goldmine CRM softphone + asterisk

[Dan Elder]

Anyone gotten the Goldmine (6.7 in our install) softphone to work with
asterisk? I've been pulling my hair out trying to figure out what the
problem is, the phone doesn't seem to register if I have a password on the
account. If I remove the password, it logs in & can make calls..but with the
pw, it doesn't work at all.. Here's a sip debug of the attempted
registration:

<-- SIP read from 192.168.1.4:5060:
REGISTER sip:192.168.1.253 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5060
To: 
From: ;tag=3b7780
CSeq: 1 REGISTER
Call-ID: [EMAIL PROTECTED]
Contact: 
Max-Forwards: 70
User-Agent: Desktop Phone Object, ver. 3.7.0.64
Expires: 3600
Content-Length: 0


--- (11 headers 0 lines)---
Transmitting (no NAT) to 192.168.1.4:5060:
SIP/2.0 503 Server error
Via: SIP/2.0/UDP 192.168.1.4:5060;received=192.168.1.4
From: ;tag=3b7780
To: ;tag=as794c4d62
Call-ID: [EMAIL PROTECTED]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: 
Content-Length: 0



Where can I find more info on the 503 server error message? Not finding much
w/google... If I put in a bogus password, the error shows bad auth, so I'm
assuming that when I enter the right pass, it's being authenticated, but not
actually registering.

Thanks in adavance for any pointers, been trying to get help from Goldmine
directly, but they're pretty useless & have been trying to fix this for
months now.

Thanks again

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Re: [asterisk-users] Asterisk Realtime Macros

[Benchev]

On Monday 24 July 2006 19:47, Jon Scottorn wrote:
> If I setup the extension like such:
>
>exten => 101,1,Dial(SIP/101)
>exten => 102,1,Dial)SIP/102)
>
> I can dial back and forth between the two phones.
>
> When I switch it to use the stdexten macro and change the extension like
> such
>
>exten => 101,1,Macro(stdexten,101,sip/101)
>exten => 102,1,Macro(stdexten,102,sip/102)
>
> I can not dial each extension and this is what reports on asterisk cli:
>

> My question is what has to be in the mysql extenstions_table to get the
> macro to work?
>
> Here is what is in my extensions_table:
>
> mysql> select * from extensions_table;
> ++-+---+--+---+--+
>
> | id | context | exten | priority | app   | appdata  |
>
> ++-+---+--+---+--+
>
> |  1 | default | 101   |1 | Macro | stdexten,101,sip/101 |
> |  2 | default | 102   |1 | Macro | stdexten,102,sip/102 |
>
You should substitute commas with a "pipe";
 i.e. your "select" should look:
|  1 | default | 101   |1 | Macro | stdexten|101|sip/101 |
|  2 | default | 102   |1 | Macro | stdexten|102|sip/102 |
On the wiki is complete mess with the commas ...

Benchev
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Re: [asterisk-users] Circuit/channel Congestion

[Doug Lytle]

Lincoln Zuljewic Silva wrote:
It's a propretary hardware that is Telephony Server for a predictive 
dialer. We would like to use asterisk as pbx of that hardware instead 
of another pbx like avaya ou nortel...


Then I'm sorry, I can't help you.

Doug

--

Ben Franklin quote:

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


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Re: [asterisk-users] Transfers - No ringback or moh

[Doug Lytle]

Douglas Garstang wrote:

-Original Message-
From: Doug Lytle [mailto:[EMAIL PROTECTED]
Sent: Monday, July 24, 2006 12:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Transfers - No ringback or moh




Doug,

The transfer soft button can do both attended and non attended transfers. If 
user B presses the transfer soft button before user C picks up, it's an 
unattended transfer. If user B presses the transfer soft button after user C 
has answered, then it's an attended transfer.

  
I'm not able reproduct this.  If I don't select Blind before entering 
the extension or number when transferring to user C, it only offers a 
cancel or split.  Once user C answers, then I'm offered the Transfer 
soft button.


Don't see what the Polycom digit map has to do with it. 
  


Not wanting to press yet another button.

People (Users that is) seem to think, pressing less is better.

Doug


--

Ben Franklin quote:

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deserve neither Liberty nor Safety."


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RE: [asterisk-users] Operator in Voicemail

[Kevin Savoy]

This doesn’t solve the problem. Still the same. Any
other ideas?

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Henk
Sent: Monday, July 24, 2006 12:25
PM
To: 'Asterisk Users Mailing List -
 Non-Commercial Discussion'
Subject: RE: [asterisk-users]
Operator in Voicemail



 

This is what I am using:

 

exten = o,1,Answer()

exten = o,2,GoTo(default,3000,1)

exten = o,3,Hangup()

 

Hope this helps,

 

Henk

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony Davis
Sent: maandag 24 juli 2006 18:20
To: Asterisk Users Mailing List -
 Non-Commercial Discussion
Subject: RE: [asterisk-users]
Operator in Voicemail



 

I’m having the exact same problem here. I originally
thought it was a context problem. 

However, to troubleshoot I tried placing the following in
every context (default, from-inside, from-outside, etc) in extensions.conf with
no luck:

  exten =>
o,1,DIAL(SIP/100,100)

 

Like Kevin, it works fine for our internal users, just
doesn’t work for callers coming from the PSTN.

 

Thanks,

-AntD









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin Savoy
Sent: Monday, July 24, 2006 7:37
AM
To: 'Asterisk Users Mailing List -
 Non-Commercial Discussion'
Subject: [asterisk-users] Operator
in Voicemail



 

I’ve
got an odd problem. I have set in Voicemail.conf operator=yes as a default.
This is so that when a caller is in the voicemail system they can press 0 and
be sent to the operator. This works fine when the caller is internal to the
system but NOT when the caller is calling in from the PSTN. Instead the caller
gets the message Press 1 to accept the recording. Pressing 0 again deletes the
message. How do I get this to work for outside callers calling in??

 

Thanks

 

 

 

_

 

Kevin Savoy

Business Unit Telecom
Analyst

2218 4th Ave W

Williston, ND 58801

Ph: 701-774-4023

Fax: 701-774-2901

http://www.novo1.com

Novo 1 is a service mark of Novo 1, Inc

 






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RE: [asterisk-users] Transfers - No ringback or moh

[Douglas Garstang]

> -Original Message-
> From: Doug Lytle [mailto:[EMAIL PROTECTED]
> Sent: Monday, July 24, 2006 12:14 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Transfers - No ringback or moh
> 
> 
> Douglas Garstang wrote:
> > I don't know... all I know is that when user C starts to 
> ring, and user B has dropped from the call, the music on hold 
> stops for user A, until user C answers. I would have expected 
> User A to hear ringing at this point.
> >
> >   
> 
> Then they need to do a blind transfer.
> 
> Transfer button,
> 
> Blind (Soft button)
> Extensions or phone number to transfer too
> 
> Depending on your digit map for the Polycom, you may have to press 
> something after that.
> 
> My digit map matches against 4 digit extensions that we use 
> internally, 
> 7 and 10 digit number starting with a 9 are also matched.

Doug,

The transfer soft button can do both attended and non attended transfers. If 
user B presses the transfer soft button before user C picks up, it's an 
unattended transfer. If user B presses the transfer soft button after user C 
has answered, then it's an attended transfer.

Doing a blind/unattended transfer isn't going to make any difference, as the 
ringback (or lack of it) to user A, occurs as soon as user B presses the 
transfer soft button a second time.

Don't see what the Polycom digit map has to do with it. 

Doug.


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[asterisk-users] Zap DMTF detect error

[Woodoo People .pGa!]

Hi!

Today, as the linux is runnig 136 days ago, with asterisk running 50days ago
both * and zaptel is 1.0.10

all the pbx worked well, but they called me at the morning, because the IVR does
not detect any DMTF code. (DMTF detect is not worked via sip trunk and 
dtmfmode=inband
with worked with dmtfmode=rfc2833).

Any ideas about that?

(yes, i know, i have to upgrade :-)

-- 
WoodOO-[P]an[G]alaktikan[A]gent-People <][> http://shadow.pganet.com
[EMAIL PROTECTED]@RedHat.users
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Re: [asterisk-users] Re: Overriding # at the end

[Delca]

Hi Jay, thanks for the help, it was really useful :)
I realized that my extensions.conf was a mess and i re-do it and
modified the ATA dial plan and now it's more structured and scalable.

Thanks!
Santiago

On 7/21/06, Jay Milk <[EMAIL PROTECTED]> wrote:

Delca wrote:
> Fixed, i'm the kind of guy who ask and later find the solution :$ it
> is a Linksys PAP-2 ATA setting in Regional -> Control Timer Values ->
> Interdigit Long Timer (this is in advanced mode).
>
> Sorry :)
> Santiago
>
> On 7/20/06, Delca <[EMAIL PROTECTED]> wrote:
>> Hi, I'm using a Linksys PAP-2 to test my next PBX with asterisk.
>> The problem  I'm haivng is that when I dial the extension, I've to end
>> it with # and then it starts calling is there any way to override that
>> # so with just dialing the 3-digit extension I'll be able to call?
>> This actually works great with a voipjet configuration I already have
>> .. when I dial an US number (i.e.: 12245684486 ) it starts dialing
>> that number.
>> But if I do the same with an extension, I just have to wait until i
>> press #. I just want to dial 123 :(
>>
>>
>> Cheers!
>> Santiago
That's probably not the best solution.  You may want to look at
dial-plans here.  For example, I have this one:

(*xxS0|011x.|1xxS0|2xxS0|6xxxS0|7xxS0|8.|911S0)

*xx are services and are immediately called
011x. allows for international numbers
1xxS0 makes sure 1+10 digit US numbers are called instantly
2xxSO makes sure extensions (three digits, all beginning with "2") are
dialed instantly
etc...
don't forget 911S0 -- this dials 911 immediately as well.

Check the sipura website for config info on dial plans.
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Re: [asterisk-users] Circuit/channel Congestion

[Lincoln Zuljewic Silva]

It's a propretary hardware that is Telephony Server for a predictive 
dialer. We would like to use asterisk as pbx of that hardware instead of 
another pbx like avaya ou nortel...


Doug Lytle wrote:


Lincoln Zuljewic Silva wrote:

Yes, I saw that...I just don't know what do to anymore. The framing 
and protocol between machines are the same and the pri signaling is 
ok in both machines.



Not that this is it, but you have:

signalling=pri_net

It should probably be pri_cpe, net being the PBX.

What PBX is it?

Ours it a Avaya Definity G3R.

Doug



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[asterisk-users] Polycom_acd_functions SIP trouble

[James Fromm]

I'm trying to use the latest revision of Bweschke's branch from SVN for 
polycom_acd_functions.  Asterisk builds and runs without error but all 
SIP devices can't register when specifying a secret in sip.conf.  The 
Polycom 601 I'm testing with and a copy of SJphone will not register. 
IAX from Idefisk works without error.


The error all SIP devices get is:

Jul 24 10:26:48 NOTICE[31524]: chan_sip.c:14203 handle_request_register: 
Registration from '' failed for 
'192.168.0.95' - Username/auth name mismatch


Commenting the definition of a secret in sip.conf for the device solves 
this.  Here's the config for one of the devices.


[1003]
type=friend
canreinvite=no
host=dynamic
username=1003
;   secret=stuff
context=outbound
callerid="Jimmy" <1003>
[EMAIL PROTECTED]
nat=no

Why won't this revision accept the definition of a secret?  Am I missing 
something simple (stupid)?


Thanks,
Jay

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Re: [asterisk-users] Circuit/channel Congestion

[Doug Lytle]

Lincoln Zuljewic Silva wrote:
Yes, I saw that...I just don't know what do to anymore. The framing 
and protocol between machines are the same and the pri signaling is ok 
in both machines.


Not that this is it, but you have:

signalling=pri_net

It should probably be pri_cpe, net being the PBX.

What PBX is it?

Ours it a Avaya Definity G3R.

Doug

--

Ben Franklin quote:

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


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Re: [asterisk-users] Transfers - No ringback or moh

[Doug Lytle]

Douglas Garstang wrote:

I don't know... all I know is that when user C starts to ring, and user B has 
dropped from the call, the music on hold stops for user A, until user C 
answers. I would have expected User A to hear ringing at this point.

  


Then they need to do a blind transfer.

Transfer button,

Blind (Soft button)
Extensions or phone number to transfer too

Depending on your digit map for the Polycom, you may have to press 
something after that.


My digit map matches against 4 digit extensions that we use internally, 
7 and 10 digit number starting with a 9 are also matched.


Doug

--

Ben Franklin quote:

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


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Re: [asterisk-users] Asterisk, IAXModem and Hylafax

[Lee Howard]

JR Richardson wrote:


Is there any of you Hylafax guru's out there who may have a few
minutes to help me get up to speed with the Hylafax setup and
configuration conventions. 



You'll do well to look for help in HylaFAX forums/communities/archives 
rather than in an Asterisk one.


That said, the best advice I can give you is to learn how your MTA works 
in delivering mail and how you can go about delivering certain e-mails 
(however you wish to indicate them) to a processing script (MDA).  You 
can write that processing script to call on sendfax, or you can use 
faxmail there, instead (if faxmail suits you).


All of this has been done hundreds upon hundreds of times by people all 
over the world with sendmail, postfix, and exim, so expect there to be 
information available for your situation to at least get you started.


But again, looking for HylaFAX mail-to-fax information in an Asterisk 
forum is probably not going to be an extremely effective approach.


Lee.

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Re: [asterisk-users] Circuit/channel Congestion

[Lincoln Zuljewic Silva]

Yes, I saw that...I just don't know what do to anymore. The framing and 
protocol between machines are the same and the pri signaling is ok in 
both machines.


Doug Lytle wrote:


Lincoln Zuljewic Silva wrote:


Its another pbx.



It looks like your setup between the PBX and the Asterisk box is not 
correct.  Your span line shows:


My pri show span 1:
Primary D-channel: 16
Status: Provisioned, Down, Active

Notice that it shows Provisioned, *DOWN*, Active.

If setup correctly, it should show up.

Doug



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RE: [asterisk-users] Transfers - No ringback or moh

[Douglas Garstang]

> -Original Message-
> From: Doug Lytle [mailto:[EMAIL PROTECTED]
> Sent: Monday, July 24, 2006 11:48 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Transfers - No ringback or moh
> 
> 
> Douglas Garstang wrote:
> >> -Original Message-
> >> 
> > And that's normal, for user A to just hear dead air?
> >
> >   
> 
> 
> I have a Polycom IP501 sitting on my desk (Test phone):
> 
> I call it with my Avaya phone
> pick up the ringing extension
> press transfer button (I hear hold music on the Avaya)
> I dial the voice mail extension on the Asterisk
> I press the transfer button again.
> Hold music stops and I hear Comedian Mail.
> 
> So, is the dead air that you hear, the silence that you would 
> get when 
> the hold music stops and both parties have been bridged, or 
> no audio is 
> passes at all?

I don't know... all I know is that when user C starts to ring, and user B has 
dropped from the call, the music on hold stops for user A, until user C 
answers. I would have expected User A to hear ringing at this point.

Doug.
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Re: [asterisk-users] Circuit/channel Congestion

[Doug Lytle]

Lincoln Zuljewic Silva wrote:

Its another pbx.


It looks like your setup between the PBX and the Asterisk box is not 
correct.  Your span line shows:


My pri show span 1:
Primary D-channel: 16
Status: Provisioned, Down, Active

Notice that it shows Provisioned, *DOWN*, Active.

If setup correctly, it should show up.

Doug

--

Ben Franklin quote:

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


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Re: [asterisk-users] Transfers - No ringback or moh

[Doug Lytle]

Douglas Garstang wrote:

-Original Message-


And that's normal, for user A to just hear dead air?

  



I have a Polycom IP501 sitting on my desk (Test phone):

   I call it with my Avaya phone
   pick up the ringing extension
   press transfer button (I hear hold music on the Avaya)
   I dial the voice mail extension on the Asterisk
   I press the transfer button again.
   Hold music stops and I hear Comedian Mail.

So, is the dead air that you hear, the silence that you would get when 
the hold music stops and both parties have been bridged, or no audio is 
passes at all?


Doug

--

Ben Franklin quote:

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


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Re: [asterisk-users] G729 Softphone

[Alyed Tzompa]

		As far as I there is no free softphone that can handle G729 codec. So you will need a licenced one.
Have a call center working with eyebeam from counterpath (previously
known as Xten) for about a year with no problems. Don't know if it
supports 
the URL option, but I'm pretty sure it will. Anyway you can ask this directly to their tech support. Alyed
		
		
		
Return-Path: <[EMAIL PROTECTED]> Mon Jul 24 10:09:29 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP;   Mon, 24 Jul 2006 10:09:29 -0700Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])	by lists.digium.com (Postfix) with ESMTP id 41C65C202;	Mon, 24 Jul 2006 09:49:46 -0700 (MST)
		
		Daniel Salama a écrit :> Looking for a SIP or IAX softphone for a call center application that > can do G729 codec. Any recommendations? Ideally it would do screen pops, > meaning that it will understand the URL option of the Dial command.Of course I'm a little biased, but I think MozPhone is well suited to call center application: it does natively support URL option of Dial or Queue command. It does not support G729 though, but speex will give you nice quality / low bandwith.Have a look at http://moziax.mozdev.org/ and please send feedback / comment / questions to MozPhone's mailing list at: http://moziax.mozdev.org/list.htmlThanks,Jean-Denis___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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[asterisk-users] Lots of Asterisk child processes

[Mike Clark]

We are experiencing a sporadic issue across several systems using 1.2.x 
releases of Asterisk where there are lots of child processes  spawned 
off the main Asterisk process that appear to be just sitting around 
doing nothing. They don't accumulate any CPU time, so don't know if they 
are deadlocked on something. Eventually, the systems will hang when a 
number of these processes accumulate. Anyone else experienced this or is 
there any specific known issue that may cause this?


Thanks,

Mike Clark
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RE: [asterisk-users] Operator in Voicemail

[T. Shaw]

Hmm this works for me. I'm using 1.2.7.1, but doing VOIP only. No PSTN lines 
coming in.


exten => o,1,Playback(walks-into-bar-mail)

Currently i have a "place holder" for when my customer gets a real 
receptionist, then i'll substitute the Playback with a Dial application.  
this is placed in the context of [incoming], after a caller hits the IVR


Terrelle



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

Subject: RE: [asterisk-users] Operator in Voicemail
Date: Mon, 24 Jul 2006 09:20:03 -0700

I'm having the exact same problem here. I originally thought it was a
context problem.

However, to troubleshoot I tried placing the following in every context
(default, from-inside, from-outside, etc) in extensions.conf with no
luck:

  exten => o,1,DIAL(SIP/100,100)



Like Kevin, it works fine for our internal users, just doesn't work for
callers coming from the PSTN.



Thanks,

-AntD

  _

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin
Savoy
Sent: Monday, July 24, 2006 7:37 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Operator in Voicemail



I've got an odd problem. I have set in Voicemail.conf operator=yes as a
default. This is so that when a caller is in the voicemail system they
can press 0 and be sent to the operator. This works fine when the caller
is internal to the system but NOT when the caller is calling in from the
PSTN. Instead the caller gets the message Press 1 to accept the
recording. Pressing 0 again deletes the message. How do I get this to
work for outside callers calling in??



Thanks







_



Kevin Savoy

Business Unit Telecom Analyst

2218 4th Ave W

Williston, ND 58801

Ph: 701-774-4023

Fax: 701-774-2901

http://www.novo1.com 

Novo 1 is a service mark of Novo 1, Inc



<< image001.gif >>




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Re: [asterisk-users] Circuit/channel Congestion

[Lincoln Zuljewic Silva]

Its another pbx.

Doug Lytle wrote:


Lincoln Zuljewic Silva wrote:


grep 4509 extensions.conf
exten => 4509,1,Dial(Zap/g1/4509)



Doug Lytle wrote:


Lincoln Zuljewic Silva wrote:


Where is your configuration for extension 4509?




I still don't know how you are hooking up extension 4509.  Is this 
hanging off of a channel bank, is it a SIP phone or on another PBX?


Doug



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RE: [asterisk-users] Operator in Voicemail

[Henk]

This is what I am using:

 

exten = o,1,Answer()

exten = o,2,GoTo(default,3000,1)

exten = o,3,Hangup()

 

Hope this helps,

 

Henk

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony Davis
Sent: maandag 24 juli 2006 18:20
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users]
Operator in Voicemail



 

I’m having the exact same problem
here. I originally thought it was a context problem. 

However, to troubleshoot I tried placing
the following in every context (default, from-inside, from-outside, etc) in
extensions.conf with no luck:

  exten
=> o,1,DIAL(SIP/100,100)

 

Like Kevin, it works fine for our internal
users, just doesn’t work for callers coming from the PSTN.

 

Thanks,

-AntD









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin Savoy
Sent: Monday, July 24, 2006 7:37
AM
To: 'Asterisk
 Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Operator
in Voicemail



 

I’ve got an odd problem. I have set in Voicemail.conf
operator=yes as a default. This is so that when a caller is in the voicemail
system they can press 0 and be sent to the operator. This works fine when the
caller is internal to the system but NOT when the caller is calling in from the
PSTN. Instead the caller gets the message Press 1 to accept the recording.
Pressing 0 again deletes the message. How do I get this to work for outside
callers calling in??

 

Thanks

 

 

 

_

 

Kevin Savoy

Business Unit
Telecom Analyst

2218
  4th Ave W

Williston, ND 58801

Ph: 701-774-4023

Fax: 701-774-2901

http://www.novo1.com

Novo 1 is a service mark of Novo 1, Inc

 






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RE: [asterisk-users] Intercom feature on Polycom phones

[Brian Vincent \(C\)]

Yes.

 

http://www.voip-info.org/wiki/view/Polycom+auto-answer+config



---
Brian Vincent
Copper Mountain Telecom
[EMAIL PROTECTED] 



-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stephen Murphy
Sent: Monday, July
 24, 2006 10:47 AM
To:
asterisk-users@lists.digium.com
Subject: [asterisk-users] Intercom
feature on Polycom phones

 

I have the situation where my client
would like to ‘Intercom’ an extension similar to auto-answer. I
have polycom phones – can this be done?

 

Steve



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[asterisk-users] core dumps when phpagi script ends?

[Dan Kirshner]

I've set up my IVR system as a PHP script using phpagi.php.  The typical 
way for users to exit the system is by hanging up.  This seems to work 
fine -- Asterisk promptly ends the call/connection -- but a side effect 
seems to be that it always leaves a core. file in 
/var/lib/asterisk/agi-bin.  Anything simple I should be doing to avoid this?


Thanks.

--Dan
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Re: [asterisk-users] Intercom feature on Polycom phones

[C F]

What do you mean by similar to auto answer?

On 7/24/06, Stephen Murphy <[EMAIL PROTECTED]> wrote:





I have the situation where my client would like to 'Intercom' an extension
similar to auto-answer. I have polycom phones – can this be done?



Steve
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RE: [asterisk-users] Transfers - No ringback or moh

[Douglas Garstang]

> -Original Message-
> From: Doug Lytle [mailto:[EMAIL PROTECTED]
> Sent: Monday, July 24, 2006 9:34 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Transfers - No ringback or moh
> 
> 
> Douglas Garstang wrote:
> > 1. User A dials User B.
> > 2. User A and User B are connected.
> > 3. User B hits the transfer soft key. User A gets music on hold.
> > 4. User B dials user C. User C's phone rings, and user A 
> continues to 
> > hear music on hold.
> > 5. When User B presses the transfer soft key again to complete the 
> > transfer, the music on hold for User A stops.
> 
> Because user B just did an attended transfer.

And that's normal, for user A to just hear dead air?

Doug.
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Re: [asterisk-users] IP CDR

[William Piper]

On 7/24/06, Khaled Chehab <[EMAIL PROTECTED]> wrote:




 Hi 
 Please how can I get the user 
register ip address and put it at cdr ,its too important
 Thanks
 
 
Check out this page:
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sippeer
 
I believe that some variation of that command and the "Set(CDR(cdrfield)=whatever)" will do it for you.
Perhaps there is an easier way, but I don't know it.
 
Good luck,
 
bp
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Re: [asterisk-users] G729 Softphone

[Jean-Denis Girard]

Daniel Salama a écrit :
Looking for a SIP or IAX softphone for a call center application that 
can do G729 codec. Any recommendations? Ideally it would do screen pops, 
meaning that it will understand the URL option of the Dial command.


Of course I'm a little biased, but I think MozPhone is well suited to 
call center application: it does natively support URL option of Dial or 
Queue command. It does not support G729 though, but speex will give you 
nice quality / low bandwith.
Have a look at http://moziax.mozdev.org/ and please send feedback / 
comment / questions to MozPhone's mailing list at: 
http://moziax.mozdev.org/list.html



Thanks,
Jean-Denis
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[asterisk-users] Asterisk Realtime Macros

[Jon Scottorn]

Hi,

   I am trying to get asterisk Realtime to work.  I have a fresh installed 1.2.10 setup on a debian system.  I have taken the defaul setup and put it into the mysql database.  
I have setup two extensions 101 and 102.  

If I setup the extension like such:

   exten => 101,1,Dial(SIP/101)
   exten => 102,1,Dial)SIP/102)

I can dial back and forth between the two phones.

When I switch it to use the stdexten macro and change the extension like such

   exten => 101,1,Macro(stdexten,101,sip/101)
   exten => 102,1,Macro(stdexten,102,sip/102)

I can not dial each extension and this is what reports on asterisk cli:

-- SIP Seeding peer from astdb: '102' at [EMAIL PROTECTED]:5060 for 3600
    -- SIP Seeding peer from astdb: '101' at [EMAIL PROTECTED]:1093 for 3600
    -- Executing Macro("SIP/101-081a7f90", "stdexten,102,sip/102")
Jul 24 10:36:37 WARNING[23358]: app_macro.c:149 macro_exec: No such context 'macro-stdexten,102,sip/102' for macro 'stdexten,102,sip/102'
  == Auto fallthrough, channel 'SIP/101-081a7f90' status is 'UNKNOWN'
    -- SIP Seeding peer from astdb: '101' at [EMAIL PROTECTED]:1093 for 3600

My question is what has to be in the mysql extenstions_table to get the macro to work?

Here is what is in my extensions_table:

mysql> select * from extensions_table;
++-+---+--+---+--+
| id | context | exten | priority | app   | appdata  |
++-+---+--+---+--+
|  1 | default | 101   |    1 | Macro | stdexten,101,sip/101 | 
|  2 | default | 102   |    1 | Macro | stdexten,102,sip/102 |

Thanks in advance for any help.




Jon Scottorn
Systems Administrator
The Possibility Forge, Inc.
http://www.possibilityforge.com
435.635.0591 x.1004





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Re: [asterisk-users] G729 Softphone

[Daniel Salama]

I have the eyeBeam softphone but I don't see G729 in the list of  
available codecs (BTW, this is the paid version not X-Lite). Any clues?


Thanks,
Daniel

On Jul 24, 2006, at 12:00 PM, Guillermo Salas M. wrote:


On Mon, 2006-07-24 at 11:41 -0400, Daniel Salama wrote:

Looking for a SIP or IAX softphone for a call center application that
can do G729 codec. Any recommendations? Ideally it would do screen
pops, meaning that it will understand the URL option of the Dial
command.



Give a try to Eyebeam at www.counterpath.com , it supports video and
voice with g729.

BOL Siphone is freeware that supports video/voice and uses de g723.1
codec you can download it at http://www.bol2000.com/download/sipphone/


Thanks,
Daniel

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--
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esq
Edificio Barre #2 Primer Piso
Telefono : +593 5 262 8071
Celular  : +593 9 985 5138
e-mail   : [EMAIL PROTECTED]
www  : http://www.manta.telconet.net
   http://www.telcocarrier.net

Linux User: 255902

Beat me, whip me, make me use Windows!

Please avoid sending me Word or PowerPoint attachments.
See http://www.fsf.org/philosophy/no-word-attachments.html

Please avoid the Top Posting, see
http://es.wikipedia.org/wiki/Top-posting

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[asterisk-users] Intercom feature on Polycom phones

[Stephen Murphy]

I have the situation where my client would like to
‘Intercom’ an extension similar to auto-answer. I have polycom
phones – can this be done?

 

Steve






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Re: [asterisk-users] Cyberdata paging speakers - anyone use them?

[Christopher Snell]

For our stores, it would be nicer to have some kind of device that
automatically mutes our music before playing input from the Asterisk
pager.  We already have a store full of speakers, no reason to
duplicate them.  Has anybody heard of such a thing?

On 7/21/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:

POE sip speaker
http://www.cyberdata.net/voip/voip-speaker.html

Anyone use these?
How well do they work?




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Re: [asterisk-users] Clocking Multiple T1 Cards

[Andrew Kohlsmith]

On Monday 24 July 2006 12:11, Shaw Terwilliger wrote:
> Thank you; this is the kind of information I was looking for.  The wiki
> and other documents told me exactly what the configuration options did,
> but I didn't know what kind of timing configuration was right for
> multiple cards.

Essentially the timing is ONLY for the hardware on the card.  The Digium cards 
use a quad framer chip (maybe a dual for the TE210 but I don't think so) and 
it's a hardware limitation of the framer that all spans must share the same 
clock source.  Sangoma's cards use individual framers and don't have this 
limitation.  (essentially I think it was a cost/space tradeoff.)

Once the data is on the PCI bus, the clock source is irrelevant.  They're all 
close enough that it doesn't matter anymore.  Those framers want exact 
lock-step timing though, which is why your clocking settings are so very 
important, and why with telephony in general it is crucial to think about 
your clocking before throwing hardware at a solution.

-A.
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Re: [asterisk-users] Circuit/channel Congestion

[Lincoln Zuljewic Silva]

In my line, i need to use "0" because I send the clock to the other 
machine...


Thomas Laurids Pedersen wrote:


I have the same card, but in my zaptel.conf I have the following line
span=1,1,0,hdb3,crc4

as you can see from the status your line is down.

BR Thomas



  
Lincoln Zuljewic  
Silva 
<[EMAIL PROTECTED]  To 
.com> Asterisk Users Mailing List -   
Sent by:  Non-Commercial Discussion   
asterisk-users-bo
[EMAIL PROTECTED]  cc 
m.com 
  Subject 
  [asterisk-users] Circuit/channel
07/24/2006 04:58  Congestion  
PM
  
  
Please respond to 
 Asterisk Users   
 Mailing List -   
 Non-Commercial   
   Discussion 
<[EMAIL PROTECTED] 
ists.digium.com>  
  
  





Hello all. I have a Digium TE110P board and when I do a 'dial 4509' I got:

Jul 24 11:44:33 NOTICE[8180]: app_dial.c:1049 dial_exec_full: Unable to
create channel of type 'Zap' (cause 34 - Circuit/channel congestion)


My zaptel.conf:
span=1,0,0,cas,hdb3,crc4
bchan=1-15,17-31
dchan=16

My zapata.conf:
[channels]
context=demo
priindication=outofband
pridialplan=local
prilocaldialplan=local
overlapdial=yes
immediate=no
callprogress=yes
busydetect=no
switchtype=euroisdn
signalling=pri_net

group=1
callgroup=1
pickupgroup=1
channel => 1-15,17-31

My /proc/zaptel/1
Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" HDB3//CRC4

  1 WCT1/0/1 Clear (In use)
  2 WCT1/0/2 Clear (In use)
  3 WCT1/0/3 Clear (In use)
  4 WCT1/0/4 Clear (In use)
  5 WCT1/0/5 Clear (In use)
  6 WCT1/0/6 Clear (In use)
  7 WCT1/0/7 Clear (In use)
  8 WCT1/0/8 Clear (In use)
  9 WCT1/0/9 Clear (In use)
 10 WCT1/0/10 Clear (In use)
 11 WCT1/0/11 Clear (In use)
 12 WCT1/0/12 Clear (In use)
 13 WCT1/0/13 Clear (In use)
 14 WCT1/0/14 Clear (In use)
 15 WCT1/0/15 Clear (In use)
 16 WCT1/0/16 HDLCFCS (In use)
 17 WCT1/0/17 Clear (In use)
 18 WCT1/0/18 Clear (In use)
 19 WCT1/0/19 Clear (In use)
 20 WCT1/0/20 Clear (In use)
 21 WCT1/0/21 Clear (In use)
 22 WCT1/0/22 Clear (In use)
 23 WCT1/0/23 Clear (In use)
 24 WCT1/0/24 Clear (In use)
 25 WCT1/0/25 Clear (In use)
 26 WCT1/0/26 Clear (In use)
 27 WCT1/0/27 Clear (In use)
 28 WCT1/0/28 Clear (In use)
 29 WCT1/0/29 Clear (In use)
 30 WCT1/0/30 Clear (In use)
 31 WCT1/0/31 Clear (In use)

My pri show span 1:
Primary D-channel: 16
Status: Provisioned, Down, Active
Switchtype: EuroISDN
Type: Network
Window Length: 0/7
Sentrej: 0
SolicitFbit: 0
Retrans: 0
Busy: 0
Overlap Dial: -1
T200 Timer: 1000
T203 Timer: 1
T305 Timer: 3
T308 Timer: 4000
T313 Timer: 4000
N200 Counter: 3

My zap show channels:
  Chan Extension  Context Language   MusicOnHold
pseudodemo
 1demo
 2demo
 3demo
 4demo
 5demo
 6demo
 7demo
 8demo
 9demo
10demo
11demo
12demo
13demo
14demo
15demo
17demo
18demo
19demo
20demo
21demo
22demo
23demo
24demo
25demo
26demo
27demo
28demo
29demo
30demo
31demo

Thanks a lot!
Lincoln
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Re: [asterisk-users] Circuit/channel Congestion

[Doug Lytle]

Lincoln Zuljewic Silva wrote:

grep 4509 extensions.conf
exten => 4509,1,Dial(Zap/g1/4509)



Doug Lytle wrote:


Lincoln Zuljewic Silva wrote:


Where is your configuration for extension 4509?


I still don't know how you are hooking up extension 4509.  Is this 
hanging off of a channel bank, is it a SIP phone or on another PBX?


Doug

--

Ben Franklin quote:

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


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RE: [asterisk-users] Operator in Voicemail

[Anthony Davis]

I’m having the exact same problem here. I originally
thought it was a context problem. 

However, to troubleshoot I tried placing the following in
every context (default, from-inside, from-outside, etc) in extensions.conf with
no luck:

  exten =>
o,1,DIAL(SIP/100,100)

 

Like Kevin, it works fine for our internal users, just doesn’t
work for callers coming from the PSTN.

 

Thanks,

-AntD









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin Savoy
Sent: Monday, July 24, 2006 7:37
AM
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: [asterisk-users] Operator
in Voicemail



 

I’ve
got an odd problem. I have set in Voicemail.conf operator=yes as a default.
This is so that when a caller is in the voicemail system they can press 0 and
be sent to the operator. This works fine when the caller is internal to the
system but NOT when the caller is calling in from the PSTN. Instead the caller
gets the message Press 1 to accept the recording. Pressing 0 again deletes the
message. How do I get this to work for outside callers calling in??

 

Thanks

 

 

 

_

 

Kevin Savoy

Business Unit Telecom
Analyst

2218 4th Ave
  W

Williston, ND 58801

Ph: 701-774-4023

Fax: 701-774-2901

http://www.novo1.com

Novo 1 is a service mark of Novo 1, Inc

 






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