[Asterisk-Users] Consol Phones for Asterisk

[Scott Wolfe]

What kind of phone would you suggest for a Multi-line operator phone? It
would have to be about 10-12 incoming lines. Most of time she would just
transfer them off but there are times when there will be about 8 people
parked (on hold). Thanks for your suggestions.

-Scott

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Re: [Asterisk-Users] MoH Problem

[Scott Wolfe]

Title: MoH Problem



I had the same problem. I had to reinstall 
mpg123

  - Original Message - 
  From: 
  Parker, Blake (MIS) 
  To: asterisk-users@lists.digium.com 
  
  Sent: Friday, March 25, 2005 7:01 
PM
  Subject: [Asterisk-Users] MoH 
  Problem
  
  When I put a call on hold I get the following out 
  put: 
      -- Started music on hold, class 
  'default', on SIP/beeforeCCM-8e33     -- Stopped music on hold on 
  SIP/beeforeCCM-8e33 
  And moh is never heard by the call on hold. 
  
  Any ideas? 
  Blake 
  
  

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[Asterisk-Users] Faxing through Broadvoice - HT286

[Scott Wolfe]

Has anyone had any luck using a fax machines with a 
HT286 through the broadvoice service? If so what is the class of fax machine you 
were using. I am getting nothing but communication errors. I was going to set 
this up and incoming and an outgoing machine. 
 
Thanks
 -Scott
 
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[Asterisk-Users] xlite regestration fails but calls to thru

[Scott Wolfe]

While on my network I can register ok with xlite 
but outside my firewall my Xlite says that regestraion has failed but I am still 
able to make calls through it. I have opened ports: 5060 udp/tcp and 1-2 
udp/tcp  is there another port Xlite needs for proper regestration? Is is 
this a network configuation error on Astrisks part? My Asterisk server is 
running a IP of 10.0.1.x and my Cisco firewall is passing the public IP address 
to it from the outside. 
 
Thanks for any advice.
 -Scott
 
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Re: [Asterisk-Users] xlite registration fails but calls to thru

[Scott Wolfe]

Title: Re: [Asterisk-Users] xlite registration fails but calls to thru



No. They are all there as shown in your image. 


  - Original Message - 
  From: 
  Robert Keller 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Saturday, April 02, 2005 4:12 
  PM
  Subject: Re: [Asterisk-Users] xlite 
  registration fails but calls to thru
  Make 
  sure the first three codecs are not grayed out.Robert Andrew Keller 
  Ferndale School District #502[EMAIL PROTECTED]360-383-9228 
  PH.360-383-9218 FAX"Paving the way for tomorrows genius."
  
  From: "Scott Wolfe" <[EMAIL PROTECTED]>Reply-To: 
  Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com>Date: 
  Sat, 2 Apr 2005 16:03:19 -0800To: 
  Subject: 
  [Asterisk-Users] xlite regestration fails but calls to 
  thruWhile 
  on my network I can register ok with xlite but outside my firewall my Xlite 
  says that regestraion has failed but I am still able to make calls through it. 
  I have opened ports: 5060 udp/tcp and 1-2 udp/tcp  is there 
  another port Xlite needs for proper regestration? Is is this a network 
  configuation error on Astrisks part? My Asterisk server is running a IP of 
  10.0.1.x and my Cisco firewall is passing the public IP address to it from the 
  outside. Thanks for 
  any advice. -Scott
  
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[Asterisk-Users] Concurrent calls: best provider?

[Scott Wolfe]

This brings up the question. What is the best service for concurrent calls?
In the case where I have a small business I might have 10-15 people needing
to call out and they could all be on at the same time.
 -Scott

- Original Message - 
From: "Kerry Garrison" <[EMAIL PROTECTED]>
To: "'Matt'" <[EMAIL PROTECTED]>; "'Asterisk Users Mailing List -
Non-Commercial Discussion'" 
Sent: Monday, April 04, 2005 11:48 AM
Subject: RE: [Asterisk-Users] broadvoice


> I called them about this and the vauge answer I got was that you get 2
> connections per account in order to allow the equivilant of a line with
call
> waiting. While there is no hard-wired limitation that I know of, it is
best
> not to abuse it so as to prevent them from enforcing one.
> -Kerry
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Matt
> Sent: Monday, April 04, 2005 11:30 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] broadvoice
>
> Hi,
> I'm currently routing my asterisk server out over broadvoice.. it seems I
> can do multiple outgoing and incoming calls does anyone know if
> broadvoice actually allows this or not?
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Re: [Asterisk-Users] Concurrent calls: best provider?

[Scott Wolfe]

So then in this case voip would not be the best solution?

- Original Message - 
From: "John Millican" <[EMAIL PROTECTED]>
To: 
Sent: Monday, April 04, 2005 2:41 PM
Subject: Re: [Asterisk-Users] Concurrent calls: best provider?


> On Monday April 04 2005 5:14 pm, Brian McSpadden  top posted:
> > I believe Kevin is meaning to buy a T-1 PRI from someone, like a phone
> > company of a CLEC.
> >
> > On Apr 4, 2005 3:40 PM, John Millican <[EMAIL PROTECTED]> wrote:
> > > On Monday April 04 2005 3:58 pm, Kevin Kiely wrote:
> > > > T1 PRI
> > > >
> > > >
> > > > This brings up the question. What is the best service for concurrent
> > > > calls?
> > > > In the case where I have a small business I might have 10-15 people
> > > > needing
> > > > to call out and they could all be on at the same time.
> > > >  -Scott
> > >
> > > Even with a T-1 you still need some one to provide termination that
will
> > > allow more than one call at a time on that account or multiple
accounts
> > > with the same or different providers.
> > > John M
>
> Well I canget T-1 from a local provider (since i live in BFE it is much
> cheaper than att, verizon,...) but they do not provide termination.  so
just
> wanted to clarify this for the op.
> john M
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Re: [Asterisk-Users] T.38 fax with SIP devices

[Scott Wolfe]

I have been on the same path although I am using a TDM400. No matter what I
did I could not get a fax to go through. Yesterday I moved the * server
outside my firewall and the rest of the network and now I am making more
progress. I blame it on old network hardware. I have two accounts, one with
Broadvoice and the other with LiveVoip. The Broadvoice fax is going through
with out any problems. LiveVoip faxing still fails. I would like to use
LiveVoip so I will keep at it.
-Scott



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

Sent: Thursday, April 07, 2005 7:52 AM
Subject: Re: [Asterisk-Users] T.38 fax with SIP devices


> Hello Mark,
>
> I have been working on a similar plan but am still looking for
> reasonable/tested hardware - can you tell me what devices you are
> using?
>
> Thanks,
>
> Jonathon
>
>
> On Apr 7, 2005 7:01 AM, Mark Dutton <[EMAIL PROTECTED]> wrote:
> >
> > Hi there
> >
> > I have a SIP ATA with a fax machine attached and a SIP FXO gateway to
the
> > PSTN. When I try to send faxes in either direction, I get nothing but
stony
> > silence. I have changed the gateway and the ATA to peer to peer mode to
test
> > them and they happily do the T.38 thing and faxes flow.
> >
> > It seems that they initially negotiate a G.729 codec, which is what I
want
> > and then when the receiving end detects the fax machine, it wants to
> > re-negotiate and use the t38fax codec. This is the working the Micronet
> > devices use at least.
> >
> > When I put the units into proxy mode and run them through Asterisk, they
> > fail at the negotiation stage.
> >
> > Now I have learned from my dealings with Asterisk and the newsgroup that
> > Asterisk does not do T.38. However, why should it not let devices do
T.38?
> > My debug messages from Asterisk don't show it saying no, but the
gateways
> > don't wont' setup the T.38 on Asterisk.
> >
> > I have chanded sip.conf to allow=all and there are no explicit rules in
the
> > registrations for the gateways.
> >
> > Does anyone have an idea here?
> >
> > For this venture to be truly usable, I have to be able to get FAX
working at
> > this basic level.
> >
> > Regards
> >
> > Mark Dutton
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> >
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> >
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[Asterisk-Users] FWD no longer doing IAX?

[Scott Wolfe]

Last night I signed up for a FWD account and was 
hoping to use iax to connect thier server. I have been unable to connect as of 
yet. I get a:
 
Registration of '64' rejected: Registration 
Refused.
 
I used the iax section of http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD to 
try and help me get this going.
 
iax2 show peers gives me.
Name/Username    
Host 
Mask 
Port  
Statusfwd-gw/64    65.39.205.121   (S)  
255.255.255.255  4569  
OK (76 ms)
 
I can login to my account via thier web page so it 
would seem that the account is set up. Here is the string I am using in 
iax.conf. 
register => 
64:[EMAIL PROTECTED]
 
I set off a message to thier tech support but I am 
just wondering if any of you were having any problems.
 
-Scott
 
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[Asterisk-Users] I dont want to hear the FXS port ring - TDM400?

[Scott Wolfe]

Is there a way to have an FXS 
port not ring but just pick up? Here is what I am doing.
 
I have Mitel 200SX plugged into 
one of FXS ports on my TDM400 so that my Mitel users can make calls out via 
VoIP. Currently when I dial that Mitel extension from the Mitel, it rings the 
port on TDM400 and then answers, then drops to a dial tone so I can continue to 
dial. Preferably I would like not to hear a ring but just give me back the dial 
tone.  
 
Here is my code in the 
extensions.conf and all works great except I don’t want to hear the FXS port 
ring.  

 
exten => 
s,1,Wait(0)
exten => 
s,n,Answer
exten => 
s,n,DigitTimeout,5 
; Set Digit Timeout to 5 seconds
exten => 
s,n,ResponseTimeout,10 
; Set Response Timeout to 10 seconds
exten => 
s,n,DISA,no-password|local
 
 
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[Asterisk-Users] TE11OP -> Mitel 200Sx??

[Scott Wolfe]

Hello all. I just received a 
TE110P and am trying to hook it to my Mitel 200SX has anyone successfully done 
this? My configuration is as follows.
 
Asterisk -> TE110P 
->Kentrox (csu/dsu) -> Mitel T1 Card. 
 
All I get is a blinking yellow on 
my TE110P card and an alarm on my Mitel. T1 card. 
 
Any advice would be great. 
 
Zaptel.conf
span=1,0,1,d4,ami
e&m=1-23 
dchan=24 
 
Zapata.conf 
signalling=em_w
switchtype=dms100
echocancel=yes
echocancelwhenbridged=yes
echotraining=400 
callerid=asreceived
group=1
context=default
channel => 1-23 
 
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Re: [Asterisk-Users] TE11OP -> Mitel 200Sx??

[Scott Wolfe]

 
 
Thanks,
  This is what I have now, but my Mitel PBX 
and Asterisk Box are unable to communicate via the T1 connection. Asterisk loads 
ok but I get error lights (blinking orange) on my TE110P and on my Mitel T1 
card. Hu
 
 
-Scott
 
/etc/zaptel.conf
loadzone = usdefaultzone=usspan=1,0,0,d4,amibchan=1-23
dchan=24
 
/etc/asterisk/zapata.conf
[trunkgroups]
 
[channels]context=defaultswitchtype=dms100rxwink=300usecallerid=nohidecallerid=nocallwaiting=nousecallingpres=yescallwaitingcallerid=nothreewaycalling=yestransfer=yescanpark=yescancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=yesrxgain=0.0 
;into the pstn twords the 
telcotxgain=0.0callgroup=1pickupgroup=1immediate=yes
 
signalling=pri_cpegroup=1context=default 
emdigitwait=500channel => 1-23 ; Set this to 
1-15,17-31 for E1
 
 


  - Original Message - 
  From: 
  Michael D 
  Schelin 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Friday, April 22, 2005 4:48 
PM
  Subject: Re: [Asterisk-Users] TE11OP 
  -> Mitel 200Sx??
  Hello Henrye&m=1-23 
  should be bchan=1-23you have it set for 
  analogalsosignaling=pri_cpeHenry Devito wrote:
  


Don't you need one of these directives so the 
PRI knows which is master and which is slave?
 

pri_cpe: PRI signaling, CPE side 
pri_net: PRI signaling, Network side 
 
Henry

  - 
  Original Message - 
  From: 
  Scott 
  Wolfe 
  To: 
  Asterisk-Users@lists.digium.com 
  
  Sent: 
  Friday, April 22, 2005 11:01 AM
  Subject: 
  [Asterisk-Users] TE11OP -> Mitel 200Sx??
  
  
  Hello all. I just received 
  a TE110P and am trying to hook it to my Mitel 200SX has anyone 
  successfully done this? My configuration is as follows.
  
  Asterisk -> TE110P 
  ->Kentrox (csu/dsu) -> Mitel T1 Card. 
  
  All I get is a blinking 
  yellow on my TE110P card and an alarm on my Mitel. T1 card. 
  
  Any advice would be great. 
  
  
  Zaptel.conf
  span=1,0,1,d4,ami
  e&m=1-23 
  dchan=24 
  
  Zapata.conf 
  signalling=em_w
  switchtype=dms100
  echocancel=yes
  echocancelwhenbridged=yes
  echotraining=400 
  callerid=asreceived
  group=1
  context=default
  channel => 1-23 
  
  
  
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Re: [Asterisk-Users] TE11OP -> Mitel 200Sx??

[Scott Wolfe]

The Switch is since 1995 and I get a SX-200 Digital G1005 ENH 672P / F25.0 
09-FEB1994 when I look up the software on the switch board so if I am 
reading what your telling me then I have to do D4/AMI. So does my zaptel 
look correct? Maybe my cableing is off.
Thanks,
 -Scott
- Original Message - 
From: "Henry Devito" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>; "Asterisk Users Mailing List - 
Non-Commercial Discussion" 
Sent: Friday, April 22, 2005 8:34 PM
Subject: Re: [Asterisk-Users] TE11OP -> Mitel 200Sx??


Of course there are exceptions to the rules.  I see now on a couple 
software releases where they do allow PRI with D4/AMI and PRI with 
esf/b8zs.  It's been a year or so since I messed with trunking on a 200, 
I've mostly been installing and maintaining the SX2000's and 3300's.

Henry

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

Sent: Friday, April 22, 2005 9:13 PM
Subject: RE: [Asterisk-Users] TE11OP -> Mitel 200Sx??


I have done the same thing with an sx200 and a pri circuit
My sx200 can only do ami d4 and e&m channels
Here's parts of my config that takes the pri and converts it to e&m with
ANI & DNIS
zaptel.conf
# t1 connected to the PRI circuit
span=1,1,0,exf,b8zs
# t1 connected to SX200
# the t1 card on my sx200 did d4 ami and I supplied ANI and DNIS through
the dial plan
span=2,0,0,d4,ami
bchan=1-23
dchan=24
e&m=25-47
-
zapata.conf
[channels]
echocancel=yes
echocancelwhenbridged=yes
echotraining=no
rxgain=0.0
txgain=0.0
useincomingcalleridonzaptransfer=yes
restrictcid=no
context=default
usecallingpres=yes
usercallerid=yes
hidecallerid=no
callerid="Company Name"<8005551212>
signalling=pri_cpe
switchtype=dms100
group=1
channel => 1-23
group=2
signalling=em_w
emdigitwait=500
channel => 24-47
# I needed the emdigitwait=500 to wait long enough for the SX200 to dial
out it's digits
--
extensions.conf
# our PRI circiut gave us the last 4 digits of the dialed number and this
is how I passed
#   *ANI*DNIS*  to the SX200 for it to decode
# the first group were individual numbers that mapped to faxes and modems
exten => 1234,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
exten => ,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
exten => ,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
# this set mapped our did 5000 - 5199 to the SX200
exten => _5[0-1]XX,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
The reset of the dial plan took what ever I set up in the sx200 ARS to 
dial
out and
sent out put Zap/G1

Hope this helps

--
From: Henry Devito[SMTP:[EMAIL PROTECTED]
Reply To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, April 22, 2005 8:56 PM
To: Scott Wolfe; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] TE11OP -> Mitel 200Sx??
<><>
I was wrong.  I just looked in my Mitel I&M's.  What level software are 
you
on in the SX200?  Up until a certain level 200's could only do D4/AMI 
T1's,
they could not do PRI's.  If it is a newer switch within the past 3 years
or an older switch with later software than you can do PRI, but the
signaling and framing must be ESF/B8ZS.

Henry
 - Original Message -
 From: Scott Wolfe
 To: [EMAIL PROTECTED] ; Asterisk Users Mailing List - Non-Commercial
Discussion
 Sent: Friday, April 22, 2005 7:04 PM
 Subject: Re: [Asterisk-Users] TE11OP -> Mitel 200Sx??

 Thanks,
   This is what I have now, but my Mitel PBX and Asterisk Box are unable
to communicate via the T1 connection. Asterisk loads ok but I get error
lights (blinking orange) on my TE110P and on my Mitel T1 card. Hu
 -Scott
 /etc/zaptel.conf
 loadzone = us
 defaultzone=us
 span=1,0,0,d4,ami
 bchan=1-23
 dchan=24
 /etc/asterisk/zapata.conf
 [trunkgroups]
 [channels]
 context=default
 switchtype=dms100
 rxwink=300
 usecallerid=no
 hidecallerid=no
 callwaiting=no
 usecallingpres=yes
 callwaitingcallerid=no
 threewaycalling=yes
 transfer=yes
 canpark=yes
 cancallforward=yes
 callreturn=yes
 echocancel=yes
 echocancelwhenbridged=yes
 rxgain=0.0 ;into the pstn twords the telco
 txgain=0.0
 callgroup=1
 pickupgroup=1
 immediate=yes
 signalling=pri_cpe
 group=1
 context=default
 emdigitwait=500
 channel => 1-23 ; Set this to 1-15,17-31 for E1

   - Original Message -
   From: Michael D Schelin
   To: Asterisk Users Mailing List - Non-Commercial Discussion
   Sent: Friday, April 22, 2005 4:48 PM
   Subject: Re: [Asterisk-Users] TE11OP -> Mitel 200Sx??
   Hello Henry
   e&m=1-23 should be bchan=1-23
   you have it set for analog
   also
   signaling=pri_cpe
   Henry Devito wrote:
 Don't you need one of these directives so the PRI knows whic

Re: [Asterisk-Users] TE11OP -> Mitel 200Sx??

[Scott Wolfe]

Thanks Henry,
 -Scott
- Original Message - 
From: "Henry Devito" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Saturday, April 23, 2005 11:05 PM
Subject: Re: [Asterisk-Users] TE11OP -> Mitel 200Sx??


I am trying to locate the manual for that level software.  If it's not here 
at home it is at my office and I will look everything up in the morning.
----- Original Message - 
From: "Scott Wolfe" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
; <[EMAIL PROTECTED]>
Sent: Saturday, April 23, 2005 9:00 PM
Subject: Re: [Asterisk-Users] TE11OP -> Mitel 200Sx??


The Switch is since 1995 and I get a SX-200 Digital G1005 ENH 672P / 
F25.0 09-FEB1994 when I look up the software on the switch board so if I 
am reading what your telling me then I have to do D4/AMI. So does my 
zaptel look correct? Maybe my cableing is off.
Thanks,
 -Scott
- Original Message - 
From: "Henry Devito" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>; "Asterisk Users Mailing List - 
Non-Commercial Discussion" 
Sent: Friday, April 22, 2005 8:34 PM
Subject: Re: [Asterisk-Users] TE11OP -> Mitel 200Sx??


Of course there are exceptions to the rules.  I see now on a couple 
software releases where they do allow PRI with D4/AMI and PRI with 
esf/b8zs.  It's been a year or so since I messed with trunking on a 200, 
I've mostly been installing and maintaining the SX2000's and 3300's.

Henry

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

Sent: Friday, April 22, 2005 9:13 PM
Subject: RE: [Asterisk-Users] TE11OP -> Mitel 200Sx??


I have done the same thing with an sx200 and a pri circuit
My sx200 can only do ami d4 and e&m channels
Here's parts of my config that takes the pri and converts it to e&m 
with
ANI & DNIS

zaptel.conf
# t1 connected to the PRI circuit
span=1,1,0,exf,b8zs
# t1 connected to SX200
# the t1 card on my sx200 did d4 ami and I supplied ANI and DNIS 
through
the dial plan

span=2,0,0,d4,ami
bchan=1-23
dchan=24
e&m=25-47
-
zapata.conf
[channels]
echocancel=yes
echocancelwhenbridged=yes
echotraining=no
rxgain=0.0
txgain=0.0
useincomingcalleridonzaptransfer=yes
restrictcid=no
context=default
usecallingpres=yes
usercallerid=yes
hidecallerid=no
callerid="Company Name"<8005551212>
signalling=pri_cpe
switchtype=dms100
group=1
channel => 1-23
group=2
signalling=em_w
emdigitwait=500
channel => 24-47
# I needed the emdigitwait=500 to wait long enough for the SX200 to 
dial
out it's digits

--
extensions.conf
# our PRI circiut gave us the last 4 digits of the dialed number and 
this
is how I passed
#   *ANI*DNIS*  to the SX200 for it to decode

# the first group were individual numbers that mapped to faxes and 
modems

exten => 1234,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
exten => ,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
exten => ,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
# this set mapped our did 5000 - 5199 to the SX200
exten => _5[0-1]XX,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
The reset of the dial plan took what ever I set up in the sx200 ARS to 
dial
out and
sent out put Zap/G1

Hope this helps

--
From: Henry Devito[SMTP:[EMAIL PROTECTED]
Reply To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, April 22, 2005 8:56 PM
To: Scott Wolfe; Asterisk Users Mailing List - Non-Commercial 
Discussion
Subject: Re: [Asterisk-Users] TE11OP -> Mitel 200Sx??

<><>
I was wrong.  I just looked in my Mitel I&M's.  What level software are 
you
on in the SX200?  Up until a certain level 200's could only do D4/AMI 
T1's,
they could not do PRI's.  If it is a newer switch within the past 3 
years
or an older switch with later software than you can do PRI, but the
signaling and framing must be ESF/B8ZS.

Henry
 - Original Message -
 From: Scott Wolfe
 To: [EMAIL PROTECTED] ; Asterisk Users Mailing List - Non-Commercial
Discussion
 Sent: Friday, April 22, 2005 7:04 PM
 Subject: Re: [Asterisk-Users] TE11OP -> Mitel 200Sx??

 Thanks,
   This is what I have now, but my Mitel PBX and Asterisk Box are 
unable
to communicate via the T1 connection. Asterisk loads ok but I get error
lights (blinking orange) on my TE110P and on my Mitel T1 card. Hu

 -Scott
 /etc/zaptel.conf
 loadzone = us
 defaultzone=us
 span=1,0,0,d4,ami
 bchan=1-23
 dchan=24
 /etc/asterisk/zapata.conf
 [trunkgroups]
 [channels]
 context=default
 switchtype=dms100
 rxwink=300
 usecallerid=no
 hidecallerid=no
 callwaiting=no
 usecallingpres=yes
 callwaitingcallerid=no
 threewaycalling=yes
 transfer=y

Re: [Asterisk-Users] VoiPSupply Dot Com: Epilogue

[Scott Wolfe]

Maybe I should my pictures in with me and supermodels. :-)

Cheers,
   -Scott

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

Sent: Friday, May 27, 2005 12:26 PM
Subject: RE: [Asterisk-Users] VoiPSupply Dot Com: Epilogue


LOL - You mean he actually 'met' Newt Gingrich?  How dare you not extend
him credit!!!
I mean seriously... For such a distinguished individual...

Hey, not only have I met the heads of several multi-billion dollar
corps, I have gotten absolutely blasted drunk with them.
So I should get credit, a 40% discount, and your daughters phone number,
right???  LOL

Seriously, though.  I think it is a sigh of relief that this hopefully
will be all over and off the list.
I for one have seen enough positive comments to know that your company
is a quality player.
The fact that you have followed up with the community and been so
forthright also says a lot.
Mistakes happen.  Sometimes people get inconvenienced.  The quality
companies address the issue and fix it as best they can.
I don't think we can ask for much more than that.  Keep up the good work
and keep that pricing aggressive...  8)

Cheers,
Wiley Siler
Who has been drunk with "important" people





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Re: [Asterisk-Users] TE11OP -> Mitel 200Sx??

[Scott Wolfe]

Does anyone know the pinout to make a cable so that My Asterisk can talk to
my Mitel 200SX?


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

Sent: Sunday, April 24, 2005 1:47 PM
Subject: Re: [Asterisk-Users] TE11OP -> Mitel 200Sx??


> According to the Mitel manuals that version of SX-200D can only use a
> regular 24 channel T1.  It can not use a PRI interface.  You are going to
> have to configure * to use a standard T1 not a PRI D4/AMI is the correct
> signaling.
> - Original Message - 
> From: "Scott Wolfe" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Sent: Sunday, April 24, 2005 11:09 AM
> Subject: Re: [Asterisk-Users] TE11OP -> Mitel 200Sx??
>
>
> > Thanks Henry,
> >  -Scott
> >
> > - Original Message - 
> > From: "Henry Devito" <[EMAIL PROTECTED]>
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> > 
> > Sent: Saturday, April 23, 2005 11:05 PM
> > Subject: Re: [Asterisk-Users] TE11OP -> Mitel 200Sx??
> >
> >
> >>I am trying to locate the manual for that level software.  If it's not
> >>here at home it is at my office and I will look everything up in the
> >>morning.
> >> - Original Message - 
> >> From: "Scott Wolfe" <[EMAIL PROTECTED]>
> >> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> >> ; <[EMAIL PROTECTED]>
> >> Sent: Saturday, April 23, 2005 9:00 PM
> >> Subject: Re: [Asterisk-Users] TE11OP -> Mitel 200Sx??
> >>
> >>
> >>> The Switch is since 1995 and I get a SX-200 Digital G1005 ENH 672P /
> >>> F25.0 09-FEB1994 when I look up the software on the switch board so if
I
> >>> am reading what your telling me then I have to do D4/AMI. So does my
> >>> zaptel look correct? Maybe my cableing is off.
> >>> Thanks,
> >>>  -Scott
> >>> - Original Message - 
> >>> From: "Henry Devito" <[EMAIL PROTECTED]>
> >>> To: <[EMAIL PROTECTED]>; "Asterisk Users Mailing List -
> >>> Non-Commercial Discussion" 
> >>> Sent: Friday, April 22, 2005 8:34 PM
> >>> Subject: Re: [Asterisk-Users] TE11OP -> Mitel 200Sx??
> >>>
> >>>
> >>>> Of course there are exceptions to the rules.  I see now on a couple
> >>>> software releases where they do allow PRI with D4/AMI and PRI with
> >>>> esf/b8zs.  It's been a year or so since I messed with trunking on a
> >>>> 200, I've mostly been installing and maintaining the SX2000's and
> >>>> 3300's.
> >>>>
> >>>> Henry
> >>>>
> >>>>
> >>>>
> >>>> - Original Message - 
> >>>> From: "Dennis Walker" <[EMAIL PROTECTED]>
> >>>> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> >>>> 
> >>>> Sent: Friday, April 22, 2005 9:13 PM
> >>>> Subject: RE: [Asterisk-Users] TE11OP -> Mitel 200Sx??
> >>>>
> >>>>
> >>>>>I have done the same thing with an sx200 and a pri circuit
> >>>>>
> >>>>> My sx200 can only do ami d4 and e&m channels
> >>>>>
> >>>>> Here's parts of my config that takes the pri and converts it to e&m
> >>>>> with
> >>>>> ANI & DNIS
> >>>>>
> >>>>> zaptel.conf
> >>>>>
> >>>>> # t1 connected to the PRI circuit
> >>>>> span=1,1,0,exf,b8zs
> >>>>>
> >>>>> # t1 connected to SX200
> >>>>> # the t1 card on my sx200 did d4 ami and I supplied ANI and DNIS
> >>>>> through
> >>>>> the dial plan
> >>>>>
> >>>>> span=2,0,0,d4,ami
> >>>>>
> >>>>> bchan=1-23
> >>>>> dchan=24
> >>>>> e&m=25-47
> >>>>> -
> >>>>> zapata.conf
> >>>>>
> >>>>> [channels]
> >>>>>
> >>>>> echocancel=yes
> >>>>> echocancelwhenbridged=yes
> >>>>> echotr

[Asterisk-Users] Legacy PBX -> * -> Voip Calls problems

[Scott Wolfe]

If have installed a TE110P and have connected it to my Mitel 200SX. I can 
dial to the Mitel via the T1 connection but when I dial from the Mitel to try 
and go out the Asterisk box via Voip it fails. I can see the calls getting to 
the Asterisk box from the Mitel but it just loops though its Zap channels then 
fails. Do I have spilt incoming and out going channels on a T1?
 
Thanks,
  
-Scott
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Re: [Asterisk-Users] Livevoip 800 Choppy Audio

[Scott Wolfe]

Wiley,
  Long time no chat. I just got asterisk going with my Legacy Mitel PBX.
YEA!!! I have been following your emails on Livevoip and am wondering if not
them then who? I am still looking for something that will allow me several
concurrent connections for a small business setting.

Take care and good weekend.

-Scott


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

Sent: Friday, June 03, 2005 3:36 PM
Subject: RE: [Asterisk-Users] Livevoip 800 Choppy Audio


Go to dslreports.com and look in the forum for LiveVoip.

Or alternately you can search this list with google via the
site:lists.digium.com parameter.

I spent two months working through problems with LiveVOip.

I highly recommend against them.

Cheers,
Wiley


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John
Kington
Sent: Friday, June 03, 2005 3:08 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Livevoip 800 Choppy Audio

I just signed up with livevoip for 800 DID and have very choppy audio.
>From PSTN to my asterisk is ok but asterisk to PSTN is terrible. I am
using IAX and was assigned to server iax01.nyc.*. I do not believe it is
a bandwidth problem on my end and I have no problems using iax with
gafachi. I opened a ticket with livevoip but no response yet. Would I be
better off using sip with them? Is there a server with better
response/bandwidth?
I admit that I am running a cvs head may 2004 prior to 1.x.x release.
Could that be the problem?
Regards,
John


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Re: [Asterisk-Users] Anyone noticed Voipjet voice quality problems?

[Scott Wolfe]

I had problems with their West coast server so I switched to their East
coast server and better success.
-Scott

- Original Message - 
From: "Roman Zhovtulya" <[EMAIL PROTECTED]>
To: 
Sent: Wednesday, June 08, 2005 2:08 PM
Subject: [Asterisk-Users] Anyone noticed Voipjet voice quality problems?


> Dear all,
> I've noticed some significant voice quality deterioration when calling US
> landline via VoIPjet.com in the last week or so.
>
> Before that the quality was pretty good.
>
> Has anyone else experienced any voice quality problems with voipjet
> recently?
>
> Thanks,
> Roman
>
>
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[Asterisk-Users] Mitel working together with Asterisk??

[Scott Wolfe]

Hello all,

I am looking at the possibility integrating Asterisk with our current Mitel
200sx. If this is possible what physical connection is made between the
Mitel box and * box? Then can a user choose if a call is go out VoIP or not?



 Has anyone had any luck in doing this?



Thanks,

Scott

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[Asterisk-Users] All Circuits are Busy Now

[Scott Wolfe]

I have downloaded and installed [EMAIL PROTECTED] and I have installed X-Lite 
on my Windows machine and I am able to connect it to the Asterisk server. I went 
ahead an created an account on Broadvoice today and followed the directions on 
http://voip-info.org/wiki-Asterisk+settings+Broadvoice and 
http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup but 
when ever I try and make a call from Xlite I get the all circuits are Busy now 
recording. 
 
Do I need to create a Trunk or get rid of the one 
that's there? Currently listed is  the  
ZAP/g0 wich I think is for a hard line. Here is my 
current sip.conf and extensions.conf
 
Thanks for any tips. 
  -Scott
 
 
 
== sip.conf  
==
 
; Note: If your SIP devices are behind a NAT and 
your Asterisk;  server isn't, try adding "nat=1" to each peer 
definition to;  solve translation problems.
 
[general]
 
port = 
5060   ; Port to bind 
to (SIP is 5060)bindaddr = 0.0.0.0    ; Address to bind to 
(all addresses on 
machine)disallow=allallow=ulawallow=alawcontext = 
from-sip-external ; Send unknown SIP callers to this contextcallerid = 
Unknown
 
#include sip_nat.conf#include 
sip_additional.conf
 
register => xx@sip.broadvoice.com:pp:[EMAIL PROTECTED]/2197
 
[sip.broadvoice.com]type=peeruser=phonehost=sip.broadvoice.comfromdomain=sip.broadvoice.comfromuser=xxsecret=ppusername=xxinsecure=verycontext=from-broadvoiceauthname=xxdtmfmode=inbanddtmf=inbandauthuser=xx;Disable 
canreinvite if you are behind a 
NATcanreinvite=noquality=yes
=== Extensions.conf ===
; I only addedd:
 
[VOIP-OUT]exten => _9NXXNXX, 1, dial(SIP/[EMAIL PROTECTED],30) 
exten => _9NXXNXX, 2, congestion() exten => _9NXXNXX, 102, 
busy()
 
 
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Re: [Asterisk-Users] Broadvoice latest changes and still not working-An

[Scott Wolfe]

Just wondering. How are you getting this debug. I 
am having problems to and I cant seem to track it down.

  - Original Message - 
  From: 
  Joe 
  To: asterisk-users@lists.digium.com 
  
  Sent: Wednesday, March 09, 2005 10:41 
  AM
  Subject: [Asterisk-Users] Broadvoice 
  latest changes and still not working-An
  
  
   
  I’ve tried everything with the * box after this 
  weekend.  I have read every 
  document on the problems people are having with them after this weekend as 
  well, but none of them address my problem.
   
  I checked my settings in my sips which I have below as 
  well,  
  
   
  I have changed the host file a few times,  but this was new to me and I never had 
  modified it before.  I have and 
  the same results happened.
   
  I have always used the CHI proxy until this past 
  weekend.
   
  I get a 404 not found when the invite goes out.   
   
  Below is my debug for broadvoice,  which I think tells the whole 
  story,  but for the life of me, I 
  can not figure out where the 404 is coming from.
   
  I have listed my sip file below as 
  well.
   
  Inbound calls work and I am 
  registered.
   
  Before we go into the debug,  I get this message when I reload my 
  configs files.
  
  Mar  9 13:23:48 NOTICE[3161]: chan_sip.c:6779 handle_response: 
  Outbound Registration: Expiry for sip.broadvoice.com is 1948 sec (Scheduling 
  reregistration in 1933000 ms)
   
   
  Below is the debug:
   
      
  -- Executing Dial("OSS/dsp", "SIP/[EMAIL PROTECTED]|30") in 
  new stack
  We're at outsideIPaddress port 
  14842
  Answering with preferred capability 0x4 
  (ulaw)
  12 headers, 8 lines
  Reliably Transmitting:
  INVITE sip:[EMAIL PROTECTED] 
  SIP/2.0
  Via: SIP/2.0/UDP 
  outsideIPaddress:5060;branch=z9hG4bK1572d8bc
  From: "asterisk" 
  ;tag=as6ed673e9
  To: 
  
  Contact: 
  
  Call-ID: 
  [EMAIL PROTECTED]
  CSeq: 102 INVITE
  User-Agent: Asterisk PBX
  Date: Wed, 09 Mar 2005 18:15:18 
  GMT
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, 
  REFER
  Content-Type: 
  application/sdp
  Content-Length: 164
   
  v=0
  o=root 17647 17647 IN IP4 
  outsideIPaddress
  s=session
  c=IN IP4 outsideIPaddress
  t=0 0
  m=audio 14842 RTP/AVP 0
  a=rtpmap:0 PCMU/8000
  a=silenceSupp:off - - - -
   (no NAT) 
  to 147.135.8.128:5060
      
  -- Called [EMAIL PROTECTED]
  asterisk1*CLI>
   
  Sip read:
  SIP/2.0 100 Trying
  Via: SIP/2.0/UDP 
  outsideIPaddress:5060;branch=z9hG4bK1572d8bc
  From: "asterisk" 
  ;tag=as6ed673e9
  To: 
  
  Call-ID: 
  [EMAIL PROTECTED]
  CSeq: 102 INVITE
   
   
  6 headers, 0 lines
  asterisk1*CLI>
   
  Sip read:
  SIP/2.0 404 
  Not Found
  Via: SIP/2.0/UDP 
  outsideIPaddress:5060;branch=z9hG4bK1572d8bc
  From: "asterisk" 
  ;tag=as6ed673e9
  To: 
  ;tag=SD4ou5a99-
  Call-ID: 
  [EMAIL PROTECTED]
  CSeq: 102 
  INVITE
  Content-Length: 0
   
   
  7 headers, 0 lines
      
  -- Got SIP response 404 "Not Found" back from 
  147.135.8.128
  Transmitting:
  ACK sip:[EMAIL PROTECTED] 
  SIP/2.0
  Via: SIP/2.0/UDP 
  outsideIPaddress:5060;branch=z9hG4bK1572d8bc
  From: "asterisk" 
  ;tag=as6ed673e9
  To: 
  ;tag=SD4ou5a99-
  Contact: 
  
  Call-ID: 
  [EMAIL PROTECTED]
  CSeq: 102 ACK
  User-Agent: Asterisk PBX
  Content-Length: 0
   
   (no NAT) 
  to 147.135.8.128:5060
      
  -- SIP/sip.broadvoice.com-2a2c is 
  circuit-busy
    == 
  Everyone is busy/congested at this time
      
  -- Executing Busy("OSS/dsp", "") in new 
  stack
  Destroying call 
  '[EMAIL PROTECTED]'
  asterisk1*CLI> hangup
    == Spawn 
  extension (default, 509, 102) exited non-zero on 
  'OSS/dsp'
   << 
  Hangup on console >>
   
   
  [sip.broadvoice.com]
  type=peer
  host=proxy.lax.broadvoice.com
  fromdomain=sip.broadvoice.com
  fromuser= BB
  username= BB
  ;authuser= BB
  secret= secret
  context=sip
  nat=no
  insecure=very
  dtmfmode=inband
   
  
  

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[Asterisk-Users] Ports/Protocals to Open in Firewall

[Scott Wolfe]

I am just getting started using Asterisk and would like to know what ports I
need to open in my firewall for incoming and outgoing calls. I am running a
Cisco Pix 506 and I am having problems using Xlite to make calls through
Asterisk => Broadvoice and I think this maybe due to not having the proper
protocols passed since I can use X-lite on its own ok from home behind my
Linksys Router (no Xlite).

Thanks,
Scott

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[Asterisk-Users] Broadvoice Config proper??

[Scott Wolfe]

I am having a bugger of a time getting Broadvoice 
to work. While using the configuration below I get "the device you are using is 
not registered to place calls on the network.."  I am using xlite to 
dial through [EMAIL PROTECTED] to Broadvoice 
service. If I configure xlite with the settings listed in my account and dial 
stright out (without broadvoice) I can place calls just fine. 
 
I have configured my /etc/hosts file to point 
147.135.8.128 to sip.broadvoice.com
 
Any tips you think of would be great. Thanks for 
looking. 
 
 
sip.conf file
 
register => [EMAIL PROTECTED]:PP:[EMAIL PROTECTED]/200
 
[sip.broadvoice.com]type=peeruser=phonehost=sip.broadvoice.comfromdomain=sip.broadvoice.comfromuser=XXsecret=PPusername=XXauthname=XXinsecure=verycontext=from-pstndtmfmode=inbanddtmf=inband;Disable 
canreinvite if you are behind a NATcanreinvite=nonat=yes
 
extensions.conf fileexten => 
_NXX,1,dial(SIP/[EMAIL PROTECTED],30) 
; Dial Broadvoice for 30 secondsexten => _NXXNXX,1,dial(SIP/[EMAIL PROTECTED],30) 
; Dial Broadvoice for 30 seconds
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[Asterisk-Users] Broadvoice hangs-up / disconnects after about 30 deconds

[Scott Wolfe]

I have just installed * from the latest CVS and I 
can make calls via X-Lite to outside numbers but for only 30 to 35 Seconds at a 
time then Broadvoice will hang up on me but X-Lite will not know it. Once I hang 
up X-Lite then I will get the following error message:
 
Spawn extension (default, Number-I-Dialed, 1) 
exited non-zero on 'SIP/200-6a22'-- Got SIP response 404 "Not Found" back 
from 147.135.0.128
 
I have also just installed a TDM22B and I get the same thing when I use 
a regular analog phone with it. Call goes through then dropped after about 30 
seconds.
 
Extension to Extension calls work just fine. 
Can anyone see anything wrong with my config files? I have tried using 
a direct proxy name but I always get a "404 Not found" right away.
 
Sip.confregister => [EMAIL PROTECTED]:PP:[EMAIL PROTECTED]/200
 
[sip.broadvoice.com];type=friendtype=peerhost=sip.broadvoice.comusername=425XXXsecret=PPfromdomain=sip.broadvoice.comfromuser=425XXXinsecure=very;context=from-broadvoicecontext=from-pstndtmfmode=inbandcanreinvite=noqualify=yesuser=phone
 
[200]type=friendsecret=010101auth=md5nat=yeshost=dynamicreinvite=nocanreinvite=nodtmfmode=inbandcallerid="Fred 
F"<200>dissallow=all
 
Extensions.conf[default]
 
exten => 1000,1,Dial,Zap/1|20 exten => 
1000,2,Voicemail,u1000exten => 1000,3,Hangupexten => 
1000,102,Voicemail,b1000exten => 1000,103,Hangup
 
exten => 2000,1,Dial,Zap/2|20exten => 
2000,2,Voicemail,u2000exten => 2000,3,Hangupexten => 
2000,102,Voicemail,b2000exten => 2000,103,Hangup
 
exten => _NXXNXX, 1, dial(SIP/[EMAIL PROTECTED],30) 
; Dial Broadvoice for 30 secondsexten => _NXXNXX, 2, congestion() ; 
No answer, nothingexten => _NXXNXX, 102, busy() ; 
Busy
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[Asterisk-Users] Broadvoice hangs-up / disconnects after about 30 deconds

[Scott Wolfe]

I have just installed * from the latest CVS and I can make calls via X-Lite
to outside numbers but for only 30 to 35 Seconds at a time then Broadvoice
will hang up on me but X-Lite will not know it. Once I hang up X-Lite then I
will get the following error message:

Spawn extension (default, Number-I-Dialed, 1) exited non-zero on
'SIP/200-6a22'
-- Got SIP response 404 "Not Found" back from 147.135.0.128


I have also just installed a TDM22B and I get the same thing when I use a
regular analog phone with it. Call goes through then dropped after about 30
seconds.

Extension to Extension calls work just fine.

Can anyone see anything wrong with my config files? I have tried using a
direct proxy name but I always get a "404 Not found" right away.

Sip.conf
register =>
[EMAIL PROTECTED]:PP:[EMAIL PROTECTED]/200

[sip.broadvoice.com]
;type=friend
type=peer
host=sip.broadvoice.com
username=425XXX
secret=PP
fromdomain=sip.broadvoice.com
fromuser=425XXX
insecure=very
;context=from-broadvoice
context=from-pstn
dtmfmode=inband
canreinvite=no
qualify=yes
user=phone

[200]
type=friend
secret=010101
auth=md5
nat=yes
host=dynamic
reinvite=no
canreinvite=no
dtmfmode=inband
callerid="Fred F"<200>
dissallow=all


Extensions.conf
[default]

exten => 1000,1,Dial,Zap/1|20
exten => 1000,2,Voicemail,u1000
exten => 1000,3,Hangup
exten => 1000,102,Voicemail,b1000
exten => 1000,103,Hangup

exten => 2000,1,Dial,Zap/2|20
exten => 2000,2,Voicemail,u2000
exten => 2000,3,Hangup
exten => 2000,102,Voicemail,b2000
exten => 2000,103,Hangup

exten => _NXXNXX, 1, dial(SIP/[EMAIL PROTECTED],30) ; Dial
Broadvoice for 30 seconds
exten => _NXXNXX, 2, congestion() ; No answer, nothing
exten => _NXXNXX, 102, busy() ; Busy

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[Asterisk-Users] No Echo But Broken Audio

[Scott Wolfe]

Hi there,
  I have suceseffuly installed Asterisk with a TDM22B and two sip phones
(Polycom300 and Xlite). My provider is Broadvoice and I am having a heck of
a time with the audio. All of my calls are broken up and sometimes really
tough to hear.

My CLI> 'sip show peers'  has me usually at OK(128ms) connection with
Broadvoice.

I do have the same issue with my Analog phone plugged into my TDM22B.

Any suggestions on where to look to debug this would be great.

This is an install from CVS-HEAD on 3-21-2005. I'll post my sip.conf just in
cast anyone sees anything odd in it.

Thanks,
  -Scott

[general]
localnet = 10.0.1.0/255.255.255.0
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes


register =>
[EMAIL PROTECTED]::[EMAIL PROTECTED]

[sip.broadvoice.com]
type=friend
host=sip.broadvoice.com
username=4252XX
secret=
fromdomain=sip.broadvoice.com
fromuser=4252XX
insecure=very
context=from-broadvoice
dtmfmode=inband
canreinvite=no
qualify=yes

[200] ;Xlite Phone
type=friend
secret=
auth=md5
nat=yes
host=dynamic
reinvite=no
canreinvite=no
dtmfmode=inband
callerid="Fred Flinstone"<200>
dissallow=all

[201] ;Polycom 300 Phone
type=peer
username=201polycom
password=
host=dynamic
dtmfmode=rfc2833
defaultip=10.0.1.253
context=default
callerid="Bob" <201>
mailbox=201
progressinband=no

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Re: [Asterisk-Users] No Echo But Broken Audio

[Scott Wolfe]

Thanks.
In fact SIP <->SIP is fine as well as SIP<->FXS. Although I am not sure
what Codec Polycom is using I will try and force the Polycom to use AWAL.
Thanks for the advice. My machine is a Dual P4 as well with 1gig of memory.
-Scott

- Original Message - 
From: "Adam Goryachev" <[EMAIL PROTECTED]>
To: "Scott Wolfe" <[EMAIL PROTECTED]>; "Asterisk Users Mailing List -
Non-Commercial Discussion" 
Sent: Tuesday, March 22, 2005 7:10 PM
Subject: Re: [Asterisk-Users] No Echo But Broken Audio


> On Tue, 2005-03-22 at 17:58 -0800, Scott Wolfe wrote:
> > Hi there,
> >   I have suceseffuly installed Asterisk with a TDM22B and two sip phones
> > (Polycom300 and Xlite). My provider is Broadvoice and I am having a heck
of
> > a time with the audio. All of my calls are broken up and sometimes
really
> > tough to hear.
> >
> > My CLI> 'sip show peers'  has me usually at OK(128ms) connection with
> > Broadvoice.
> >
> > I do have the same issue with my Analog phone plugged into my TDM22B.
> >
> > Any suggestions on where to look to debug this would be great.
> >
> > This is an install from CVS-HEAD on 3-21-2005. I'll post my sip.conf
just in
> > cast anyone sees anything odd in it.
>
> 1) Try with CVS-STABLE
> 2) Try calling from the sip phone to asterisk itself (ie, no internet
> path or PSTN)
> 3) Try calling from an FXS port, and to an FXO port, and between FXS/FXO
> and FXO/SIP and FXS/SIP and even SIP/SIP
> 4) What codecs are you using, I found that forcing the polycoms to use
> ALAW, and my PRI was ALAW also, then it improved the audio quality.
> Though this was with a very small number of calls, on a pretty beefy
> Intel PIV based server I never realised ALAW<->ULAW would make so
> much of a difference.
>
> If none of that helps, then let us all know the outcome of each test,
> and someone else will likely offer some additional advice...
>
> Regards,
> Adam
>
> -- 
>  -- 
> Adam Goryachev
> Website Managers
> Ph:  +61 2 8304 [EMAIL PROTECTED]
> Fax: +61 2 9345 4396www.websitemanagers.com.au
>

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Re: [Asterisk-Users] Newbie pointers

[Scott Wolfe]

Kerry,
  I know you have been having issues with Broadvoice registration. Have you
been able to work them all out? I was having the same issues to so I had to
go the CVS-HEAD version but would like to use @home because if its console
and cdr functionality.

Thanks,
  -Scott


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

Sent: Thursday, March 24, 2005 8:33 AM
Subject: RE: [Asterisk-Users] Newbie pointers


> <>
>
> We have been writing some How-To guides and will be doing different
product
> reviews as well. So far, we have had a very good response. Check out our
> site you will find some things to help get your started.
>
> http://www.geekgazette.com
>
> -Kerry
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Fred Blaise
> Sent: Thursday, March 24, 2005 3:46 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Newbie pointers
>
> Hello all
>
> I have come to Asterisk with no previous telco experience.
> As I will be playing with Asterisk really soon, I would like to have some
> pointers as to some tutorials in telco that could help me get into all
this.
> I am quite a beginner, don't forget :)
>
> Thanks a lot!
>
> Best,
> fred
>
>
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Re: [asterisk-users] incoming call popup

[Scott Wolfe]

ASTassistant can do this as well. www.astassistant.com

-Scott

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

Sent: Wednesday, March 05, 2008 5:48 AM
Subject: Re: [asterisk-users] incoming call popup


> On Tue, Mar 4, 2008 at 7:18 PM, marek cervenka <[EMAIL PROTECTED]> wrote:
>>  can you recommend "clean&simple&stable" solution for incoming call popup
>>  (in browser)?
>
> ADM http://adm.hamnett.org/ can invoke browsers when a call arrives.
>
> raj
>
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Re: [asterisk-users] queue logging

[Scott Wolfe]

You could ASTassistant to see this. Its Freeware.
www.astassistant.com

  - Original Message - 
  From: Arjan Kroon | Mobillion 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Sent: Wednesday, April 09, 2008 1:01 AM
  Subject: [asterisk-users] queue logging


  Hi,

   

  I' using with asterisk a queue with tree members and round robin.

  When a caller enters this queue and it is connecting to one of the members, 
is there a possibility to see which member the caller is connected to?

   

  And is there a way to see in de application to see if the connection from the 
caller to one of the members was successful of not successful?

   

  I know you can see it in de queue. log.

  But I want to know if I can see it also in the hangup (h) in de application?

   

  Kind Regards



   



--


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Re: [asterisk-users] Queue stats

[Scott Wolfe]

Jay,
  You could try ASTassistant. It has Queue information at a glance.
http://www.astassistant.com


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

Sent: Thursday, July 26, 2007 7:37 AM
Subject: [asterisk-users] Queue stats


> Greetings, list!
>
> My boss would like some statistics on how many calls are answered out of
> specific queues during a given time period, and I'm not sure how exactly
> to obtain those stats.  Here's how our queue system works.
>
> 1) Call comes in and enters our 'ring' queue where the phones ring for
> 15 seconds (caller hears the standard ring tone).
>
> 2) After 15 seconds, the caller falls into our 'music on hold' queue, a
> message is played and the caller hears our music on hold while the
> phones are rung again.
>
> 3) After 30 seconds, if the caller is still in our 'moh' queue, they
> drop out of queue and immediately re-enter the 'moh' queue again until
> the call is answered or the caller hangs up.
>
> How can I find out how many calls are answered out of each queue during
> certain times (1st shift, 2nd shift, etc...)?  Also, I'm curious how I
> can track how many times a call repeats the 'moh' queue.
>
> Thanks in advance,
> Jay
>
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Re: [asterisk-users] Queue stats

[Scott Wolfe]

What do you want? Maybe I can write it into ASTassistant.

Scott
http://www.astassistant.com


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

Sent: Wednesday, August 29, 2007 2:35 PM
Subject: Re: [asterisk-users] Queue stats


-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Ed Nuñez wrote:
> Can anyone recommend a good commercial solution for queue statistics?

http://queuemetrics.loway.it/

- --
Kind Regards,

Matt Riddell
Director
___

http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.7 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFG1ea0DQNt8rg0Kp4RAqlwAJ94aTwBccuYEfO8rRUWQd9bJVakMgCgt4cA
q6DH4NZFflCgxgqe1KX5iko=
=sCz/
-END PGP SIGNATURE-

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Re: [asterisk-users] Asterisk 1.2.13 and presence

[Scott Wolfe]

ASTassistant will let you see which if your IAX/SIP clients are registered / 
busy / DND. Its Free.

http://www.astassistant.com/


-Scott

- Original Message - 
From: "Alejandro Cabrera Obed" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List" 
Sent: Friday, September 21, 2007 7:18 AM
Subject: [asterisk-users] Asterisk 1.2.13 and presence


> Dear people, is it possible to have presence using Asterisk 1.2.13 / SIP
> with Linux/Debian Etch???
>
> I'd like to see if my intranet contacts are available, busy,
> disconnected
>
> Thanks a lot
>
> Alejandro
>
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Re: [asterisk-users] Asterisk 1.2.13 and presence

[Scott Wolfe]

I'm pretty sure that I wrote that its FREE.
-Scott



- Original Message - 
From: "Tzafrir Cohen" <[EMAIL PROTECTED]>
To: 
Sent: Friday, September 21, 2007 11:46 AM
Subject: Re: [asterisk-users] Asterisk 1.2.13 and presence


> On Fri, Sep 21, 2007 at 11:25:59AM -0700, Scott Wolfe wrote:
>> will let you see which if your IAX/SIP clients are registered /
>> busy / DND.
>>
>
> And I assume that you have tested it with Asterisk 1.2.13 on Debian Etch
> and can actually answer the OP, rather than spamming your non-free
> product here.
>
> Among the soft phones included with Etch, linphone should have support
> for public/subscript. Not usre about others. Twinkle only has this
> uspported in later versions (e.g: the one currently in Sid).
>
> But then again, there are simpler ways of getting this information.
> Through asterisk -rx or the manager interface.
>
> -- 
>   Tzafrir Cohen
> icq#16849755jabber:[EMAIL PROTECTED]
> +972-50-7952406   mailto:[EMAIL PROTECTED]
> http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
>
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Re: [asterisk-users] Asterisk API Manager

[Scott Wolfe]

Write a application  to log the information to a DB, then have all other 
clients connect to the database for the status. Unless I am missing 
something.

-Scott

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

Sent: Wednesday, November 28, 2007 11:29 AM
Subject: Re: [asterisk-users] Asterisk API Manager


If each client connected only once, subsequently made a request every
minute, and then disconnected only when finished, the load might be more
reasonable.  It can be a little harder to write that kind of client
though :)

Mojo

Mojo with Horan & Company, LLC wrote:
> So you'd be making 100 connections/minute, which is pretty relentless.
> That's like five connections, five requests sent, five responses
> received, and five disconnects, /every/ three seconds.
>
> And the likelihood of all 100 users to be spread out evenly over a
> minute doesn't seem very high.  I think your box would be pretty busy
> with that.
>
> Astmanproxy would be indicated.
>
> Moj
>
>
> Anthony Chapellier wrote:
>
>> However I wanted to get periodic infos about queued users (position in
>> queue) only. So I thought I could make a program sending periodic
>> requests to asterisk manager. Is it really bad to bother asterisk
>> manager with frequently and periodic requests sent by mutiple users (we
>> could say maybe 100 users with a request every minute) ?
>>
>> Moises Silva a écrit :
>>
>>
>>> Yes, but you should use astmanproxy instead and don't bother Asterisk
>>> with multiple manager connections.
>>>
>>> On Nov 27, 2007 8:24 AM, Anthony Chapellier <[EMAIL PROTECTED]> wrote:
>>>
>>>
>>>
 Hi,

 Does Asterisk manager allow multiple clients to connect to an Asterisk
 instance using the same user account ?

 Thanks,

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>>>
>>>
>>>
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>>
>
>
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Re: [asterisk-users] Asterisk -> Streaming Audio Bridge

[Scott Wolfe]

Have you looked at ICEcast? http://www.icecast.org/index.php


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


Sent: Monday, February 26, 2007 6:17 PM
Subject: [asterisk-users] Asterisk -> Streaming Audio Bridge



Greetings,

Does anyone know of a tool that can act as a VoIP client and stream to a
streaming server such as shoutcast/icecast, etc.

I've got a client interested in doing basketball play by plays during
tourney season.  They have * in place now and the bandwidth to burn for
streaming out.  In the old world, I did an analog phone patch -> mixer ->
encoder -> streaming server.  What I'm thinking of is more along the lines
of a client that registers as a SIP/IAX client, answers the phone and
patches it to a streaming server.

Thoughts/suggestions?

Thanks

Eric

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[Asterisk-Users] ASTCC on Fedora 4 and MySQL 4.1.12

[Scott Wolfe]

I just installed the CVS 9-22 and am trying to get 
ASTCC up and running. I was able to get the web interface config running and it 
made the database but when I go to the brands page it says there is a problem 
with the table. Also when I save the config file through the intraface it wont 
save it to any location.
 
I want to set up a small CC application so if there 
is a better product to use please let me know.
 
Thanks,
Scott
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Re: [Asterisk-Users] ASTCC on Fedora 4 and MySQL 4.1.12

[Scott Wolfe]

Thanks for this. Interface works as it should now.
-Scott

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


Sent: Saturday, September 24, 2005 5:07 PM
Subject: Re: [Asterisk-Users] ASTCC on Fedora 4 and MySQL 4.1.12


Okay, after spending 12 hours on it I checked the thing that has bit me 
before.  Turn SElinux off.

OUCH!!  :-)

Darren Wiebe
[EMAIL PROTECTED]

Darren Wiebe wrote:

I fought with this one for hours last night.  I have to get it yet but 
I'm not sure what the problem is.  The permissions are all fine.


Any comments anyone?

Darren Wiebe
[EMAIL PROTECTED]

Scott Wolfe wrote:

I just installed the CVS 9-22 and am trying to get ASTCC up and running. 
I was able to get the web interface config running and it made the 
database but when I go to the brands page it says there is a problem 
with the table. Also when I save the config file through the intraface 
it wont save it to any location.
 I want to set up a small CC application so if there is a better product 
to use please let me know.

 Thanks,
Scott



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Re: [Asterisk-Users] Best Voip provider

[Scott Wolfe]

All of the providers given so far seem to have a 
limited simultaneous connections. As a business solution (multiple outgoing 
calls at one time) what are you guys using?
 
-Scott
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[Asterisk-Users] ASTCC - INUSE Flag

[Scott Wolfe]

I download and installed ASTCC over the weekend and 
I am having an issue where the INUSE flag will not get set back to 0 if the user 
drops a call while the balance is being played. All other times it seems to 
reset the flag correctly. 
 
I have tried both AGI and DeadAGI with the same 
results. 
 
Those of you using it for a while, how did you get 
around this?
 
Just for fun this is all I am doing in my 
astcc-exten.conf 
[incoming]exten => s,1,Answer;exten 
=> s,2,DeadAGI(astcc.agi)exten => s,2,AGI(astcc.agi)exten => 
s,3,Hangup
I did some Google search on this issue and saw 
someone else had a problem but no response. 
 
-Scott

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Re: [Asterisk-Users] ASTCC - INUSE Flag

[Scott Wolfe]

How do you you apply the patch?
 -Scott

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


Sent: Wednesday, October 05, 2005 9:31 PM
Subject: Re: [Asterisk-Users] ASTCC - INUSE Flag


On 10/5/05, Darren Wiebe <[EMAIL PROTECTED]> wrote:

Any developers out there that would like to look at this one?  It works
fine on Asterisk CVS-v1-0-09/22/05-22:23:34 on a i686 running Linux but
it does not work on the 1.2 betas.  I agree that the number should be
set aside then.  I wonder what the problem is.



http://bugs.digium.com/view.php?id=5400

Seems to fix the problem... please test and give feedback.

--
Nicolás Gudiño
Buenos Aires - Argentina
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Re: [Asterisk-Users] 2.6.13 zaptel incompability?

[Scott Wolfe]

I am noticing that Zaptel will sometimes not have enough time to load all 
the way at bootup. If load by hand 'service zaptel start' then it loads just 
fine. I am running zaptel 1.2.0Beta1 on FC4 2.6.13-1. Since this is 
inconsistent I have not found a way around this yet.


-However -

On this same machine I am running swappable drives with CVS HEAD on one with 
FC4 2.6.13 (so the hardware is the same) and am not having any Zaptel issues 
at all.


-Scott





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


Sent: Friday, October 21, 2005 4:28 AM
Subject: Re: [Asterisk-Users] 2.6.13 zaptel incompability?



On Fri, 2005-10-21 at 12:11 +0200, Roy Sigurd Karlsbakk wrote:

>> i heard some talk about something in zaptel is currently incompatible
>> with 2.6.13.
>> is this so?
>> if so, will this be fixed soon?
>>
>
> zaptel 1.0.9.2's release notes has something about 2.6.13 kernels.

I was thinking more about the CVS HEAD version


My machines are totally CVS HEAD and have been running 2.6.13.x without
problems. At the moment they are running 2.6.14-rc4 also without any
problem. I _never_ use the distribution's kernel they are all from
kernel.org.


--
Dave Cotton <[EMAIL PROTECTED]>

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Re: [asterisk-users] iptables example

[Scott Wolfe]

I use BFD  on several of my servers. Works great. 
http://www.rfxnetworks.com/bfd.php 
  - Original Message - 
  From: Jeronimo Romero 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Sent: Tuesday, November 28, 2006 11:54 PM
  Subject: [asterisk-users] iptables example


  Hey everyone.  I recenty installed a server at a datacenter offsite and the 
thing is getting hammered with invalid ssh logins so I decided to use some 
iptables. 

  I included my ruleset here. I was wondering if I could get some feedback 
based on my ruleset from those of you using iptables in production systems.  It 
seems to be working but some critique would be appreciated.  Thanks

   

   

  #!/bin/sh

  # My system IP/set ip address of server

  SERVER_IP="x.x.x.x"

  # Flushing all rules

  iptables -F

  iptables -X

  # Setting default filter policy

  iptables -P INPUT DROP

  iptables -P OUTPUT DROP

  iptables -P FORWARD DROP

   

   

  # Allow unlimited traffic on loopback

  iptables -A INPUT -i lo -j ACCEPT

  iptables -A OUTPUT -o lo -j ACCEPT

   

  # Allow incoming ssh only from secure hosts

  iptables -A INPUT -p tcp -s x.x.x.x -d $SERVER_IP --sport 513:65535 --dport 
22 -m state --state NEW,ESTABLISHED -j ACCEPT

  iptables -A INPUT -p tcp -s x.x.x.x  -d $SERVER_IP --sport 513:65535 --dport 
22 -m state --state NEW,ESTABLISHED -j ACCEPT

   

  #Allow http & Asterisk Related Traffic

  iptables -A INPUT -p tcp -i eth0 --dport 80 -m state --state NEW -j ACCEPT

  # SIP on UDP 

  iptables -A INPUT -p udp -m udp --dport 5004:5082 -j ACCEPT

  # IAX2- 

  iptables -A INPUT -p udp -m udp --dport 4569 -j ACCEPT

  # IAX - 

  iptables -A INPUT -p udp -m udp --dport 5036 -j ACCEPT

  # RTP - the media stream

  iptables -A INPUT -p udp -m udp --dport 1:2 -j ACCEPT

   

  iptables -A INPUT -j DROP

  iptables -A OUTPUT -j ACCEPT



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Re: [Asterisk-Users] Authorization by ip

[Scott Wolfe]

Does this work the same with IAX?


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


Sent: Monday, March 27, 2006 12:44 PM
Subject: Re: [Asterisk-Users] Authorization by ip



meaning, when you put
host=dynamic
in sip.conf, it doesn't matter what ip the client comes from.
if you put instead:
host=www.xxx.yyy.zzz
then it _does_ matter where the client comes from, it must be that IP.

Giovanni Miano wrote:

You can use in sip.conf tag "host"

host=192.168.1.1 

2006/3/27, Sam Tam <[EMAIL PROTECTED]  >:

Can somebody send me a config of how to authorize SIP client by IP?

Sam



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--
Giovanni Miano




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--
Mojo <[EMAIL PROTECTED]>
Office Manger, Horan & Company, LLC
(907) 747- x112
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