RE: [Asterisk-Users] Problem with Avaya 4602 / SIP response 481

[Trevor G. Hammonds]

Trevor G. Hammonds wrote on Monday, 21 February 2005 2:54 PM:

> I suspect this may be related to the MWI indicator and the "mailbox="
> statement in my sip.conf file, as the last time I was using this
> phone with *, it was not set up to use voicemail. 

I have confirmed that this issue is directly related to the voice mail MWI.
After commenting out the "mailbox=" statement in my sip.conf, the
problem has not presented itself in over 10 hours.  

If anyone has any ideas or suggestions, I would appreciate hearing them.  I
really would like to use the MWI light.  

Sincerely,
Trevor Hammonds

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[Asterisk-Users] Custom Menu Not Working

[Chris Blake]

Greetings *`s,

I am having what appears to be a small problem, but the frustration is
erally getting to me, what am I doing wrong here ?

I used AMP to set up a custom menu, so if caller presses 1 it goes to
ext200, if caller presses 2 it goes to ext201 etc etc...

Now I have created a third option that when the caller presses 3 it must
play a sound and hang up.
No rocket science yet.

When adding the details in AMP for when caller dials 3, I have
referenced it using 'custom-myapp,s,1', and if I go to
'extensions_additional.conf' I see the following line under the rest of
menu item info that was created :

"exten => 3,1,Goto(custom-myapp,s,1) ;"

and in the extensions_custom.conf file I have 

[custom-myapp]
exten => 3,1,SayDigits(1234)
exten => 3,2,Hangup()

But when you call and press option 3, it hangs up immediately.
I have followed examples from the documentation, and this should be
working.

Any other places I can check where something is perhaps missing ?

Regards

--
Chris Blake 
Cell: 082 775 1492
Work: +27 11 782 0840
Fax : +27 11 782 0841
Mail: [EMAIL PROTECTED]

When the ax entered the forest, the trees said, "The handle is one of
us!" -- Turkish proverb


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Re: [Asterisk-Users] Minimal hardware requirements

[Liaan vd Merwe]

Hi
If you going to run digium hardware, please make sure
that machine does PCI 
2.2.. otherwise you will NOT be able to get the card
to work.
the slowest machine we found was a pIII that supported
pci 2.2
chow
Liaan

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

Sent: Tuesday, February 22, 2005 5:03 AM
Subject: RE: [Asterisk-Users] Minimal hardware
requirements


> We did our proof on concept on a celeron 600 - which
was fine to run 2 
> software and 2 hardware phones off.
>
> Original testing was with sjphone and xlite.
(software phones)
>
> We didn't fit a TDM card until we set up our first
box, which was a dual 
> p3-1000.
>
> Later,
>
> PaulH
>
> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On
Behalf Of Rudolf 
> Ladyzhenskii
> Sent: Tuesday, 22 February 2005 11:35 AM
> To: Asterisk Users Mailing List - Non-Commercial
Discussion
> Subject: [Asterisk-Users] Minimal hardware
requirements
>
> Hi, all
>
> I am doing "prrof of concept" system. I will have
two IP phones connected 
> to Asterisk box. Box itself will have 1 PSTN
conenction and one analog 
> phone conenction. A basic minimal configuration.
>
> At the moment I am planning to use an old PII-350
with 128M of RAM I have 
> lying around. I can not test anything yet, as I am
waiting for phones to 
> arrive, so question is will that be enough to
demonstrate?
>
> Thanks,
> Rudolf
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[Asterisk-Users] route outgoing call

[Altus Snyman]

Good day all
I registered at a few sip server in different countries
Now I want to route outgoing calls for that country threw that sip
server and all the others there my own pstn,ZAP card.I already
registered asterisk with them.
How would my extensions.conf look.This is what I have but no matter what
it still goes there my server.We dial 9+countrycode to get to that
country.So on the pbx 0944... will go to the UK.
Here is what I have.Please help me correct this

ignorepat => 0

;UK 

exten => _0944.,1,Dial(SIP/0${EXTEN:[EMAIL PROTECTED],50)
exten => _0944.,2,Congestion
;USA
.
.
;--Germany
.
.
;--All other
exten => _0.,1,Dial(Zap/1/${EXTEN:1})
exten => _0.,2,Dial(Zap/2/${EXTEN:1})
exten => _0.,3,Congestion




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Re: [Asterisk-Users] Noise during calls

[Larry Hendrickson]

On Mon, 21 Feb 2005 23:26:24 -0600
"Anton Krall" <[EMAIL PROTECTED]> wrote:

> Guys..
>  
> Anybody ever had problems with noise on calls after certain amount of
> minutes? It happening to me since yesterday.. after you place or get a
> call using your SIP Phones and X100P cards, after around 3 or 4
> minutes of talk, I get noise like interference and either they ant
> hear me or I cant hear them it doesnt stop until I hangup and
> place the call again...
>  
> Anybody had these problems?
>  
> 

I had similar problems before. For me, at least, it always turned out to
be a firewall/connectivity issue. Is there any software firewall turned
on or anything that might be blocking packets?

--Larry
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[Asterisk-Users] Textron voip gateway

[kolo sos]

hi,

anybody have tried to connect from asterisk PBX to
textron voip gateway?...a 4FXO gateway..h323 capable...

=
Kolosos
Philippines



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RE: [Asterisk-Users] Outbound Caller ID on PRI

[Paul Hales]

What did the Telco have to do? I want to get our outbound callid working 
properly...

PaulH 

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon
Sent: Friday, 18 February 2005 4:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Outbound Caller ID on PRI

As it turns out, it was a telco configuration problem all along. I wasted a day 
for nothing...


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

Sent: Friday, February 18, 2005 2:38 PM
Subject: Re: [Asterisk-Users] Outbound Caller ID on PRI


> Rod Bacon wrote:
>> Would someone mind doing an intense debug on their ISDN PRI and see what 
>> LEN (length) the calling number field is being sent? Maybe everyone is 
>> sending 14 characters, and my Telco is just fussier than most.
>
> Not asterisk related but on the Nortel Opt 81C switches that I maintain, 
> the CLID is sent out on the PRI-ISDN span d-channels as (520) 873- 
> which I believe is 14 characters.  It shows up on my caller ID unit at 
> home as (520) 873- which is 14 characters.
>
> The telco I am connected to is Time Warner and their CO switch is a Nortel 
> DMS-500 running NI2 compatible software.  This is in Arizona USA.
>
> George Cohn
>
>
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Re: [Asterisk-Users] sip wifi phone?

[Paradise Dove]

what about senao SI-7800H?
this is the link:
http://www.senao.com.tw/english/product/product_wireless01_outdoor_1.asp?pgtl=Wireless&tp1id=02&tp2id=06&proid=000131


On Mon, 21 Feb 2005 23:42:30 -0600, Kristian Kielhofner <[EMAIL PROTECTED]> 
wrote:
> Kurt Fankhauser wrote:
> > Sounds like I'm going to have to wait and hope some new phones are
> > released.
> 
> Kurt,
> 
> Check out my message from October:
> 
> http://lists.digium.com/pipermail/asterisk-users/2004-October/067769.html
> 
> and here is a link to the Broadcom page:
> 
> http://www.broadcom.com/products/product.php?product_id=BCM1160&category_id=45
> 
> I really, really wish someone, anyone, would start cranking out some
> devices based off of these chips.  Linksys is really into Broadcom.  Why
> not them? (As long as they don't have a blue plastic case!)
> 
> --
> Kristian Kielhofner
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Re: [Asterisk-Users] sip wifi phone?

[Kristian Kielhofner]

Kurt Fankhauser wrote:
Sounds like I'm going to have to wait and hope some new phones are
released.
Kurt,
Check out my message from October:
http://lists.digium.com/pipermail/asterisk-users/2004-October/067769.html
and here is a link to the Broadcom page:
http://www.broadcom.com/products/product.php?product_id=BCM1160&category_id=45
	I really, really wish someone, anyone, would start cranking out some 
devices based off of these chips.  Linksys is really into Broadcom.  Why 
not them? (As long as they don't have a blue plastic case!)

--
Kristian Kielhofner
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[Asterisk-Users] Noise during calls

[Anton Krall]

Guys..
 
Anybody ever had problems with noise on calls after certain amount of
minutes? It happening to me since yesterday.. after you place or get a call
using your SIP Phones and X100P cards, after around 3 or 4 minutes of talk,
I get noise like interference and either they ant hear me or I cant hear
them it doesnt stop until I hangup and place the call again...
 
Anybody had these problems?
 
__
Anton Krall
 

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[Asterisk-Users] Automating calls

[PHP Mechanic]

Hi,
I wish to initate calls from a web interface, by clicking on a link and then 
connecting to the automatic outgoing call by picking up an analogue phone.

I've got one fxs and one fxo and I wish to automate the call using a call 
file (which I can do now). How can I pick up a handset and connect to this 
call I've made when it's ringing?

Can someone point me as to how I may be able to do this.
Thanks, Oliver 

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RE: [Asterisk-Users] sip wifi phone?

[Paul Hales]

That's my plan as well.

PaulH 

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kurt Fankhauser
Sent: Tuesday, 22 February 2005 6:05 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] sip wifi phone?

Sounds like I'm going to have to wait and hope some new phones are released.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm
Sent: Monday, February 21, 2005 7:55 PM
To: Asterisk Users
Subject: Re: [Asterisk-Users] sip wifi phone?


Its not flaky at all. We have 2. The only bad thing is its lack of power. I'm 
not that too familiar with WiFi devices but it only has about 2hrs worth of 
talk time and about 10hrs of standby time. I'm not really sure on the standby 
time, but it had a full battery when I left it on my desk at 5 on fri; came 
back on Monday and it was dead.

-Matthew


> From: Kurt Fankhauser <[EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion 
> 
> Date: Mon, 21 Feb 2005 20:34:18 -0800
> To: 
> Subject: [Asterisk-Users] sip wifi phone?
> 
> Does anyone know of any sip wifi phones? Only one i can find that is 
> redily availiable is the zyxel prestige 2000w and from what i hear it 
> is flaky.
>  
> Kurt Fankhauser
> WaveLinc
> HYPERLINK "http://www.wavelinc.com/"www.wavelinc.com
> 114 S. Walnut St.
> Bucyrus, OH 44820
> 419-562-6405
>  
> 
> --
> No virus found in this outgoing message.
> Checked by AVG Anti-Virus.
> Version: 7.0.300 / Virus Database: 266.3.0 - Release Date: 2/21/2005
>  
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Version: 7.0.300 / Virus Database: 266.3.0 - Release Date: 2/21/2005
 

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Checked by AVG Anti-Virus.
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RE: [Asterisk-Users] sip wifi phone?

[Race Vanderdecken]

Tell me about.

I was on a project once that tried to use DELL PDA's with a soft phone
in them to be wi-fi back to asterisk. Unless it sat in a cradle,
attached to a wall outlet, you couldn't make it through a 10 minute
call.

Lucky if we got the PDA thing to stay awake for a call to come in.
One guy suggested that we let it sleep and then wake it up when we here
another phone ringing.

2.5 GHz sucks a bunch of juice in wi-fi. Any luck with 100 metre
Blue-Tooth phones anyone?


Race "The Tyrant" Vanderdecken



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Boehm
Sent: Monday, February 21, 2005 10:55 PM
To: Asterisk Users
Subject: Re: [Asterisk-Users] sip wifi phone?

Its not flaky at all. We have 2. The only bad thing is its lack of
power.
I'm not that too familiar with WiFi devices but it only has about 2hrs
worth
of talk time and about 10hrs of standby time. I'm not really sure on the
standby time, but it had a full battery when I left it on my desk at 5
on
fri; came back on Monday and it was dead.

-Matthew


> From: Kurt Fankhauser <[EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Date: Mon, 21 Feb 2005 20:34:18 -0800
> To: 
> Subject: [Asterisk-Users] sip wifi phone?
> 
> Does anyone know of any sip wifi phones? Only one i can find that is
> redily availiable is the zyxel prestige 2000w and from what i hear it
is
> flaky.
>  
> Kurt Fankhauser
> WaveLinc
> HYPERLINK "http://www.wavelinc.com/"www.wavelinc.com
> 114 S. Walnut St.
> Bucyrus, OH 44820
> 419-562-6405 
>  
> 
> -- 
> No virus found in this outgoing message.
> Checked by AVG Anti-Virus.
> Version: 7.0.300 / Virus Database: 266.3.0 - Release Date: 2/21/2005
>  
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Re: [Asterisk-Users] ZAP FXS vs ethernet FXS

[Jerry]

On Feb 21, 2005, at 3:12 PM, Jon Gabrielson wrote:
What are the advantages/disadvantages of using
a ZAP FXS port versus using one of the many
small ethernet FXS devices on the market.  The
ZAP FXS talks directly to asterisk over PCI.  Is this
an advantage?  The ethernet devices I assume
speak either iax2 or sip, does this cripple the
functionality of the attached FXS device for things
like callwaiting,callerid,distinctive ring, etc...
Does anyone have experience with both types
of devices and would recommend one over the
other?
Using a Zap device you are TDM in and out of the card.
If you are attaching a FAX or an alarm circuit then this is definately 
the way to go. Most analog gateways other than the IAXy speak SIP or 
MGCP. They do still provide your standard features and then some. For 
best results make certain the ethernet links from the gateways to the 
server are full duplex and switched, no hubs.

Depending on number of ports required, my preference is to use a 
channel bank, not the cheapest solution always but a highly reliable 
one.

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RE: [Asterisk-Users] sip wifi phone?

[Mathew McKernan]

Title: Message








Hi Kurt,

 

I agree the Zyxel WiFi 2000w is a very
difficult phone to use. Extremerely laggy in the interface etc.

 

I just sent back 5 to my supplier as they
were not very good. The only option I can see is a Cisco based one, but you
need a mint to buy it.

 

Thanks

 

Mathew



 











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kurt Fankhauser
Sent: Tuesday, 22 February 2005
3:34 PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] sip wifi
phone?



 



Does anyone know of any sip wifi phones? Only one i can find
that is redily availiable is the zyxel prestige 2000w and from what i hear it
is flaky.





 



Kurt Fankhauser

WaveLinc
www.wavelinc.com
114 S. Walnut St.
Bucyrus, OH
 44820
419-562-6405 



 










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RE: [Asterisk-Users] sip wifi phone?

[Kurt Fankhauser]

Sounds like I'm going to have to wait and hope some new phones are
released.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Boehm
Sent: Monday, February 21, 2005 7:55 PM
To: Asterisk Users
Subject: Re: [Asterisk-Users] sip wifi phone?


Its not flaky at all. We have 2. The only bad thing is its lack of
power. I'm not that too familiar with WiFi devices but it only has about
2hrs worth of talk time and about 10hrs of standby time. I'm not really
sure on the standby time, but it had a full battery when I left it on my
desk at 5 on fri; came back on Monday and it was dead.

-Matthew


> From: Kurt Fankhauser <[EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion 
> 
> Date: Mon, 21 Feb 2005 20:34:18 -0800
> To: 
> Subject: [Asterisk-Users] sip wifi phone?
> 
> Does anyone know of any sip wifi phones? Only one i can find that is 
> redily availiable is the zyxel prestige 2000w and from what i hear it 
> is flaky.
>  
> Kurt Fankhauser
> WaveLinc
> HYPERLINK "http://www.wavelinc.com/"www.wavelinc.com
> 114 S. Walnut St.
> Bucyrus, OH 44820
> 419-562-6405
>  
> 
> --
> No virus found in this outgoing message.
> Checked by AVG Anti-Virus.
> Version: 7.0.300 / Virus Database: 266.3.0 - Release Date: 2/21/2005
>  
> ___
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-- 
No virus found in this incoming message.
Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 266.3.0 - Release Date: 2/21/2005
 

-- 
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 266.3.0 - Release Date: 2/21/2005
 

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[Asterisk-Users] Asterisk Video Phones <-> Cisco Call manager 4.0

[Dinesh]

Hello All,

I have integrated my asterisk server to cisco call manager, now in the
process of doing video for asterisk.  I understand from the wiki that
asterisk supports Wooksung WVP-2000 SIP (hardware phone).  Are there any
others in the market, I mean hardware phones.

I know sometime last year cisco relased video for call manager 4.0.  Wanting
to know if someone has ventured into that end? Basically I want to set up
video phones in asterisk, that can talk to the callmanager.

Regards,

Dinesh Birlasekaran
Network Engineer,
ComIT, Institute of Molecular and Cell Biology
61 Biopolis Drive, Singapore 138673
HP : 92962676 DID : 65869804 Fax : 67791117 Email :
[EMAIL PROTECTED]
WWW: www.imcb.a-star.edu.sg 



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Re: [Asterisk-Users] sip wifi phone?

[Matthew Boehm]

Its not flaky at all. We have 2. The only bad thing is its lack of power.
I'm not that too familiar with WiFi devices but it only has about 2hrs worth
of talk time and about 10hrs of standby time. I'm not really sure on the
standby time, but it had a full battery when I left it on my desk at 5 on
fri; came back on Monday and it was dead.

-Matthew


> From: Kurt Fankhauser <[EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Date: Mon, 21 Feb 2005 20:34:18 -0800
> To: 
> Subject: [Asterisk-Users] sip wifi phone?
> 
> Does anyone know of any sip wifi phones? Only one i can find that is
> redily availiable is the zyxel prestige 2000w and from what i hear it is
> flaky.
>  
> Kurt Fankhauser
> WaveLinc
> HYPERLINK "http://www.wavelinc.com/"www.wavelinc.com
> 114 S. Walnut St.
> Bucyrus, OH 44820
> 419-562-6405 
>  
> 
> -- 
> No virus found in this outgoing message.
> Checked by AVG Anti-Virus.
> Version: 7.0.300 / Virus Database: 266.3.0 - Release Date: 2/21/2005
>  
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RE: [Asterisk-Users] Unable to call FWD user via IAX servers

[Anton Krall]

Seems I have the same problem, when I call a FWD number or they call me, my
phone rings once and then the console says everybody was busy and they get
my voicemail...

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of david kwok
Sent: Lunes, 21 de Febrero de 2005 07:50 p.m.
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Unable to call FWD user via IAX servers

I have set up FWD via IAX service. I have tested the IAX service with 613,
echo test, and 612, saytime. It all works well.

However when ringing a FWD user, I got this error all the time:

Connected to Asterisk CVS-v1-0-02/01/05-09:34:45 currently running on chat
(pid = 8282) chat*CLI> Verbosity is at least 3
-- Executing SetCallerID("SIP/1001-a1fb", ""David Kwok"") in new stack
-- Executing Dial("SIP/1001-a1fb",
"IAX2/:[EMAIL PROTECTED]/268757|60|r") in new stack
-- Called X:[EMAIL PROTECTED]/268757
-- Call accepted by 65.39.205.121 (format ulaw)
-- Format for call is ulaw
-- IAX2/FWD/4 is busy
-- Hungup 'IAX2/FWD/4'
  == Everyone is busy/congested at this time
-- Executing Congestion("SIP/1001-a1fb", "") in new stack
  == Spawn extension (local, 393268757, 3) exited non-zero on
'SIP/1001-a1fb'
-- Executing Hangup("SIP/1001-a1fb", "") in new stack
  == Spawn extension (local, h, 1) exited non-zero on 'SIP/1001-a1fb'
Feb 22 12:44:29 NOTICE[8282]: chan_iax2.c:6188 iax2_poke_noanswer: Peer
'iaxtel' is now UNREACHABLE!
Feb 22 12:44:54 NOTICE[8282]: chan_iax2.c:5668 socket_read: Peer 'iaxtel' is
now REACHABLE!

This is the console message shows the connection to their 613 service:

-- Executing SetCallerID("SIP/1001-11c3", ""David Kwok"") in new stack
-- Executing Dial("SIP/1001-11c3",
"IAX2/X:[EMAIL PROTECTED]/613|60|r") in new stack
-- Called X:[EMAIL PROTECTED]/613
-- Call accepted by 65.39.205.121 (format ulaw)
-- Format for call is ulaw
-- IAX2/FWD/2 is ringing
-- IAX2/FWD/2 answered SIP/1001-11c3
-- Hungup 'IAX2/FWD/2'
  == Spawn extension (local, 393613, 2) exited non-zero on 'SIP/1001-11c3'
-- Executing Hangup("SIP/1001-11c3", "") in new stack
  == Spawn extension (local, h, 1) exited non-zero on 'SIP/1001-11c3'

Has anyone have the same experience?

David Kwok
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Re: [Asterisk-Users] Minimal hardware requirements

[jurgen]

(Thanks Paul) And that same box now has a TDM-400 card in it, all 4
ports used. Two ATs are registered with the server as well. Most of
the time, it doesn't even break a sweat. I would not want to use it
for anything close to production though.


On Tue, 22 Feb 2005 14:25:00 +1100, Paul Hales <[EMAIL PROTECTED]> wrote:
> And I forgot - once we were finished with the 600 we gave it to Jurgen.
> 
> Caring and sharing,
> 
> PaulH
> 
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales
> Sent: Tuesday, 22 February 2005 2:03 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Minimal hardware requirements
> 
> We did our proof on concept on a celeron 600 - which was fine to run 2 
> software and 2 hardware phones off.
> 
> Original testing was with sjphone and xlite. (software phones)
> 
> We didn't fit a TDM card until we set up our first box, which was a dual 
> p3-1000.
> 
> Later,
> 
> PaulH
> 
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rudolf 
> Ladyzhenskii
> Sent: Tuesday, 22 February 2005 11:35 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Minimal hardware requirements
> 
> Hi, all
> 
> I am doing "prrof of concept" system. I will have two IP phones connected to 
> Asterisk box. Box itself will have 1 PSTN conenction and one analog phone 
> conenction. A basic minimal configuration.
> 
> At the moment I am planning to use an old PII-350 with 128M of RAM I have 
> lying around. I can not test anything yet, as I am waiting for phones to 
> arrive, so question is will that be enough to demonstrate?
> 
> Thanks,
> Rudolf
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-- 
[EMAIL PROTECTED] is jurgen's gmail address.
Visit http://jurgen.ca/ for more yummy goodness.
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[Asterisk-Users] app_groupcount

[Mark Halverson]

Can someone show me an example of extensions.conf to do the following?

I have three broadvoice accounts, and I have four SIP phones in the house.
With broadvoice you are charged for making more than one call at a time on
the same SIP account…so I want to check to see if the 1st one is in use and
then roll to the second and third bv account.

Here is what I was thinking with extensions.conf:

exten => _1NXXNXX,1,SetGroup(bv1)
exten => _1NXXNXX,2,CheckGroup(1) 
exten => _1NXXNXX,3,Dial(SIP/[EMAIL PROTECTED])
exten => _1NXXNXX,103,SetGroup(bv2)
exten => _1NXXNXX,104,CheckGroup(1)
exten => _1NXXNXX,105,Dial(SIP/[EMAIL PROTECTED])
exten => _1NXXNXX,205,SetGroup)bv3)
exten => _1NXXNXX,206,CheckGroup(1)
exten => _1NXXNXX,207,Dial(SIP/[EMAIL PROTECTED])
exten => _1NXXNXX,307,Dial(IAX2/[EMAIL PROTECTED])

Yet, all the calls go to bv1 and never go to bv2 – also when I enter group
show channels it says 2 active calls but nothing in the channel, group or
category columns.

-Mark

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RE: [Asterisk-Users] Minimal hardware requirements

[Paul Hales]

And I forgot - once we were finished with the 600 we gave it to Jurgen.

Caring and sharing,

PaulH 

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales
Sent: Tuesday, 22 February 2005 2:03 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Minimal hardware requirements

We did our proof on concept on a celeron 600 - which was fine to run 2 software 
and 2 hardware phones off.

Original testing was with sjphone and xlite. (software phones)

We didn't fit a TDM card until we set up our first box, which was a dual 
p3-1000.

Later,

PaulH

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rudolf 
Ladyzhenskii
Sent: Tuesday, 22 February 2005 11:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Minimal hardware requirements

Hi, all

I am doing "prrof of concept" system. I will have two IP phones connected to 
Asterisk box. Box itself will have 1 PSTN conenction and one analog phone 
conenction. A basic minimal configuration.

At the moment I am planning to use an old PII-350 with 128M of RAM I have lying 
around. I can not test anything yet, as I am waiting for phones to arrive, so 
question is will that be enough to demonstrate?

Thanks,
Rudolf
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Re: [Asterisk-Users] Sangoma A101

[Michael Bielicki]

no, but you have to install the wanpipe from sangoma as well
you can get it at: frp.sangoma.com/linux/current_wanpipe


On Mon, 21 Feb 2005 08:09:45 +0200, Altus Snyman <[EMAIL PROTECTED]> wrote:
> Good day all
> Is there any difference in the sangoma zaptel.conf and zapata.conf then
> other cards
> Thanks
> altus
> 
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-- 
Michal Bielicki
http://www.asterisk.com.pl/
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Re: [Asterisk-Users] Mandrake & CAPI

[Craig Guy]

I've been using FC2 with Kernel 2.6.9, the hardest thing for me was getting
my capi startup script right, you should not have any capi related stuff in
modprobe.conf.  I have included my startup script.  If you are using a DID
or Point to Point line for the Fritz! then change protocol=2 to protocol=34
for the avmfritz driver.  The mISDN fritz! driver will support up to four
cards and I am successfully using both a Eicon Diva Server 4-BRI card (With
melware drivers) and Fritz! card in the same system.  The Fritz! must be
loaded first with the capi script, followed by divas_cfg for the Diva card.
The divas_cfg script must have its own modprobe capi line removed or
commented out.

Craig

#!/bin/bash
#
# System startup script for the isdn-capi subsystem

case "$1" in
 start)
echo -n "Starting mISDN and CAPI"
 modprobe capi
 modprobe mISDN_core
 modprobe mISDN_l1
 modprobe mISDN_l2
 modprobe l3udss1
 modprobe mISDN_capi
 modprobe mISDN_isac
 modprobe avmfritz protocol=2
;;

 stop)
echo -n "Stopping mISDN and CAPI"
rmmod avmfritz
 rmmod mISDN_isac
 rmmod mISDN_capi
 rmmod l3udss1
 rmmod mISDN_l2
 rmmod mISDN_l1
 rmmod mISDN_dtmf
 rmmod mISDN_core
 rmmod capi
 rmmod kernelcapi
;;
 restart)
$0 stop
$0 start
;;
 *)
echo "Usage:$0{start|stop|restart}"
;;
esac
exit 0


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

Sent: Tuesday, February 22, 2005 3:02 AM
Subject: RE: [Asterisk-Users] Mandrake & CAPI


> I was looking at the exercise as a bit of Linux lerning for myself, so I
> guess Mandrake 10.1 and mISDN? Does anyone have working examples?
> Ray
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Craig Guy
> Sent: 20 February 2005 23:57
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Mandrake & CAPI
>
>
> Or you could go to a 2.6 kernel and use the mISDN drivers.
>
> Craig
>
> - Original Message - 
> From: "Razza" <[EMAIL PROTECTED]>
> To: 
> Sent: Sunday, February 20, 2005 8:00 PM
> Subject: [Asterisk-Users] Mandrake & CAPI
>
>
> > All,
> > I have been trying to get CAPI4Linux working on my machine and being
> > frank am failing miserably! I am looking for any help available, I am
> > real newbie (so please be gentle) and choose to run Mandrake 9.2 as it
>
> > appears quite friendly (or so I thought!).
> >
> > I have been following the guidance found at
> > http://www.voip-info.org/wiki-Asterisk+How+to+connect+with+CAPI for
> > the AVM card (actually I have a BT Speedway - apparently the same
> > thing).
> >
> > I guess the best approach is to detail what I have done in tandem with
>
> > the guidance? So here we go -
> >
> > Type -
> > # modprobe capi
> >
> > Great! I get no response (which is expected!), so move to step 2
> > (http://www.voip-info.org/wiki-Asterisk+AVM+Fritz+CAPI+Driver+Install)
> >
> > Guidance states 'Download and install your kernel sources' - I
> > installed these as part of the original installation, so I'll ignore.
> >
> > I download and install the CAPI driver -
> > # cd /usr/src
> > # wget
> > ftp://ftp.avm.de/cardware/fritzcrd.pci/linux/suse.82/fcpci-suse8.2-03.
> > 11
> > .02.tar.gz
> > # tar -xzvf fcpci-suse8.2-03.11.02.tar.gz
> > # cd fritz
> > Great! Looking good!
> >
> > Guidance states modify the makefile in /usr/src/src.drv as follows -
> > Replace -
> >  CARD_PATH   = /lib/modules/`uname -r`/misc
> > with  -
> >  CARD_PATH   = /lib/modules/$(uname -r)/kernel/drivers/isdn/avmb1
> >
> > I am aware this chap is running Debian and I am running Mandrake, so
> > after searching decided to modify the line as such -
> >  CARD_PATH   = /lib/modules/2.4.22-10mdk/kernel/drivers/isdn/avmb1
> >
> > Guidance states modify the KRNLINCL lines for the correct include path
>
> > -
> >
> > KRNLINCL= /usr/src/kernel-headers-`uname -r`/include
> > #KRNLINCL= /lib/modules/`uname -r`/build/include
> > #KRNLINCL= /usr/src/linux/include
> >
> > And modify the lines as thus -
> > DEFINES = -DMODULE -D__KERNEL__ -DNDEBUG \
> >  -D__$(CARD)__ -DTARGET=\"$(CARD)\"
> > CCFLAGS = -c $(DEFINES) -O2 -Wall -I $(KRNLINCL)
> > With -
> > DEFINES = -DMODULE -DMODVERSIONS -D__KERNEL__ -DNDEBUG \
> >  -D__$(CARD)__ -DTARGET=\"$(CARD)\"
> > CCFLAGS = -c $(DEFINES) -march=i686 -O2 -Wall -I $(KRNLINCL) \
> >-include $(KRNLINCL)/linux/modversions.h
> >
> > Again aware of the Debian V's Mandrake configuration, I searched the
> > web and found the following guidance for Mandrake (using the google
> > translation feature - http://translate.google.com/translate?hl=en
> >  > gu
> >
> in.de/&prev=/search%3Fq%3Dcapi%2Bmandrake%26hl%3Den%26lr%3D%26rls%3DRNWE
> > ,RNWE:2004-35,RNWE:en>
> >
> &sl=de&u=htt

[Asterisk-Users] SIP registration timeout

[Larry Hendrickson]

Hi all,

I am using * as a PBX for a Broadvoice VoIP account. It had been working
well since about last November, although not perfectly (similar
disconnection problems, although I am pretty sure it had to do with my
PPPoE setup, but I think these issues were resolved). As of a few weeks
ago, though, I started having serious problems.

Basically, I can start up * and connect to Broadvoice and everything
works well for a few hours. At some point, I always get disconnected
from Broadvoice, and the following errors appear on the * command line
every few seconds:

<- start errors

Feb 21 11:31:36 NOTICE[485]: chan_sip.c:4027 sip_reg_timeout:--
Registration for '[EMAIL PROTECTED]@sip.broadvoice.com'
timed out, trying again

 Feb 21 11:31:42 WARNING[485]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED] for
seqno 334 (Critical Request)

 Feb 21 11:31:56 NOTICE[485]: chan_sip.c:4027 sip_reg_timeout:--
Registration for '[EMAIL PROTECTED]@sip.broadvoice.com'
timed out, trying again

 Feb 21 11:32:02 WARNING[485]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED] for
seqno 335 (Critical Request)

<- end errors

NOTE: "sip.broadvoice.com" is specified as an IP address in /etc/hosts
as per one of the Broadvoice set-up instructions for * I saw somewhere.
For this example, it is currently set to "147.135.4.128".

If I then change sip.broadvoice.com to another proxy IP (say
147.135.8.128), and issue a "sip reload" command to *, then * reconnects
to Broadvoice and everything is fine for another few hours.

If I immediately change back to the original proxy, then it fails to
register with errors as above. If I wait a few hours for the new proxy
to have this same problem, then I can switch back to the old proxy
(which will work for another few hours). 

I sent an email to Broadvoice about this, but didn't get any response
from the support staff.

I held off reporting this problem to make sure that it wasn't caused by
anything I could think of on my end (NTP is keeping my machine at the
correct time, Dynamic DNS is correctly updating my IP, etc.). SIP
reloads seem to work before this problem starts, but do not help after.
Anecdotally, it seems that this problem coincides with the SIP
registration timeout, but since the first registration works, why would
it have problems on the second or third?

Has anyone else been having this problem?

I am running Asterisk (1.0.5-r1) on Gentoo from a DSL connection in
Thailand. My ping times to the proxy is between about 230ms and 350ms
(I know this is quite long for VoIP, but when it works, it works fine
for my purposes). To handle the changing of IP addresses with my DSL, I
set up a local DNS server which dynamically updates the IP of an
internal hostname to the IP of the external firewall, and added a script
that executes a 'sip reload' whenever my IP changes (sip.conf has the
this DNS host name in the 'externip' field and 'nat' set to no). Forcing
a DSL router reboot and thus an IP address change to test this
solution seems to work well, and does not cause this problem.
Firewall is forwarding port 5060 to this machine (as well as my locall
specified RTP ports which are 5000-5007), and firewall seems to be
working (I have done tests in the past to make sure that the port
forwarding is working, and I don't see how the firewall would allow it
to connect for 2-3 hours and then cause problems).

Is it possible that the SIP server is having problems with
re-registration when the IP address changes (even when the client
reports this change as per my solution above)? (I find that my DSL
connection changes IP address unusually frequently... possibly as much
as every few hours.) 

Any suggestions would be _greatly_ appreciated! 

--Larry

< begin sip.conf
[general]
externip=
bindaddr = 0.0.0.0
port=5060
localnet=192.168.0.0/255.255.255.0
disallow=all
allow=gsm
allow=slinear
allow=ulaw
allow=alaw
context=incomingvoip
dtmfmode=inband
register => \
[EMAIL PROTECTED]:mypassword:[EMAIL PROTECTED]/s

tos=lowdelay
srvlookup=yes
nat=no

[broadvoice]
type=friend
;type=peer
username=3235551212
fromuser=3235551212
secret=mypassword
host=sip.broadvoice.com
port=5060
context=broadvoice
fromdomain=sip.broadvoice.com
canreinvite=no
dtmfmode=inband
insecure=very
disallow=all
allow=ulaw
deny=0.0.0.0/0.0.0.0
permit=147.135.0.128/32
permit=147.135.4.128/32
permit=147.135.8.128/32
permit=147.135.12.128/32
nat=no
qualify=yes
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RE: [Asterisk-Users] Minimal hardware requirements

[Paul Hales]

We did our proof on concept on a celeron 600 - which was fine to run 2 software 
and 2 hardware phones off.

Original testing was with sjphone and xlite. (software phones)

We didn't fit a TDM card until we set up our first box, which was a dual 
p3-1000.

Later,

PaulH

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rudolf 
Ladyzhenskii
Sent: Tuesday, 22 February 2005 11:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Minimal hardware requirements

Hi, all

I am doing "prrof of concept" system. I will have two IP phones connected to 
Asterisk box. Box itself will have 1 PSTN conenction and one analog phone 
conenction. A basic minimal configuration.

At the moment I am planning to use an old PII-350 with 128M of RAM I have lying 
around. I can not test anything yet, as I am waiting for phones to arrive, so 
question is will that be enough to demonstrate?

Thanks,
Rudolf
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[Asterisk-Users] Asterisk-oh323

[Oswaldo Arratia]

Hi,
Thanks to Andrew Kochetkoff for sending Asterisk-oh323 files while
inaccessnetworks web page was down.

Now, I have a problem when compiling Asterisk-oh323 versions 0.7.0 or 0.7.1.
I get the following error:

/usr/include/_G_config.h:52: confused by earlier errors, bailing out
make[1]: *** [chan_oh323.o] Error 1
make[1]: Leaving directory `/usr/src/asterisk-oh323-0.7.0/asterisk-driver'
make: *** [subdirs_build] Error 1


I have followed these instructions I've found on prior e-mails:

*/
Get oh323 from
http://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries/openh323-J
anus_patch4-src-tar.gz
Get pwlib from
http://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries/pwlib-Janu
s_patch4-src-tar.gz
Get asterisk-oh323 from
http://www.inaccessnetworks.com/projects/asterisk-oh323/download/asterisk-oh
323-0.6.5.tar.gz


Untar the files
#tar zxvf openh323-Janus_patch4-src-tar.gz
#tar zxvf pwlib-Janus_patch4-src-tar.gz
#tar zxvf asterisk-oh323-0.6.5.tar.gz
#tar zxvf asterisk-1.0.3.tar.gz


Install Pwlib
#cd pwlib
#./configure && make clean && make opt && make install && ldconfig


Patch and Install OpenH323
#cd openh323
#patch -p1 < ../asterisk-oh323-0.6.5/openh323_1.13.5-make.patch
#./configure && make clean && make opt && make install && ldconfig


Asterisk
#cd asterisk-1.0.3
#make && make install && make samples


Asterisk-oh323
#cd asterisk-oh323-0.6.5
Edit the Makefile
#make && make install && ldconfig

/*


Instead of working with version 0.6.5 I am trying versions 0.7.0 and 0.7.1
and I get the very same error on both.

I am currently using Asterisk CVS-HEAD-09/16/04-13:09:53

Anybody has a clue of what's going on here??

Thanks for you help/tip.


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RE: [Asterisk-Users] Unable to call FWD user via IAX servers

[Race Vanderdecken]

Please let me know the answer to this one.

I set up FWD today and I am having the same problem.

Thanks for the iax debug tip.

Race Vanderdecken

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Webb
Sent: Monday, February 21, 2005 9:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Unable to call FWD user via IAX servers

Any chance that is a bad number??? I do not see anything that would
cause this unless there is a problem with the number you are trying to
dial.

Maybe do am iax debug to get more info?? 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of david kwok
Sent: Monday, February 21, 2005 8:50 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Unable to call FWD user via IAX servers

I have set up FWD via IAX service. I have tested the IAX service with
613, echo test, and 612, saytime. It all works well.

However when ringing a FWD user, I got this error all the time:

Connected to Asterisk CVS-v1-0-02/01/05-09:34:45 currently running on
chat (pid = 8282) chat*CLI> Verbosity is at least 3
-- Executing SetCallerID("SIP/1001-a1fb", ""David Kwok"") in new
stack
-- Executing Dial("SIP/1001-a1fb",
"IAX2/:[EMAIL PROTECTED]/268757|60|r") in new stack
-- Called X:[EMAIL PROTECTED]/268757
-- Call accepted by 65.39.205.121 (format ulaw)
-- Format for call is ulaw
-- IAX2/FWD/4 is busy
-- Hungup 'IAX2/FWD/4'
  == Everyone is busy/congested at this time
-- Executing Congestion("SIP/1001-a1fb", "") in new stack
  == Spawn extension (local, 393268757, 3) exited non-zero on
'SIP/1001-a1fb'
-- Executing Hangup("SIP/1001-a1fb", "") in new stack
  == Spawn extension (local, h, 1) exited non-zero on 'SIP/1001-a1fb'
Feb 22 12:44:29 NOTICE[8282]: chan_iax2.c:6188 iax2_poke_noanswer: Peer
'iaxtel' is now UNREACHABLE!
Feb 22 12:44:54 NOTICE[8282]: chan_iax2.c:5668 socket_read: Peer
'iaxtel' is now REACHABLE!

This is the console message shows the connection to their 613 service:

-- Executing SetCallerID("SIP/1001-11c3", ""David Kwok"") in new
stack
-- Executing Dial("SIP/1001-11c3",
"IAX2/X:[EMAIL PROTECTED]/613|60|r") in new stack
-- Called X:[EMAIL PROTECTED]/613
-- Call accepted by 65.39.205.121 (format ulaw)
-- Format for call is ulaw
-- IAX2/FWD/2 is ringing
-- IAX2/FWD/2 answered SIP/1001-11c3
-- Hungup 'IAX2/FWD/2'
  == Spawn extension (local, 393613, 2) exited non-zero on
'SIP/1001-11c3'
-- Executing Hangup("SIP/1001-11c3", "") in new stack
  == Spawn extension (local, h, 1) exited non-zero on 'SIP/1001-11c3'

Has anyone have the same experience?

David Kwok
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RE: [Asterisk-Users] Unable to call FWD user via IAX servers

[Robert Webb]

Any chance that is a bad number??? I do not see anything that would
cause this unless there is a problem with the number you are trying to
dial.

Maybe do am iax debug to get more info??

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of david kwok
Sent: Monday, February 21, 2005 8:50 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Unable to call FWD user via IAX servers

I have set up FWD via IAX service. I have tested the IAX service with
613, echo test, and 612, saytime. It all works well.

However when ringing a FWD user, I got this error all the time:

Connected to Asterisk CVS-v1-0-02/01/05-09:34:45 currently running on
chat (pid = 8282) chat*CLI> Verbosity is at least 3
-- Executing SetCallerID("SIP/1001-a1fb", ""David Kwok"") in new
stack
-- Executing Dial("SIP/1001-a1fb",
"IAX2/:[EMAIL PROTECTED]/268757|60|r") in new stack
-- Called X:[EMAIL PROTECTED]/268757
-- Call accepted by 65.39.205.121 (format ulaw)
-- Format for call is ulaw
-- IAX2/FWD/4 is busy
-- Hungup 'IAX2/FWD/4'
  == Everyone is busy/congested at this time
-- Executing Congestion("SIP/1001-a1fb", "") in new stack
  == Spawn extension (local, 393268757, 3) exited non-zero on
'SIP/1001-a1fb'
-- Executing Hangup("SIP/1001-a1fb", "") in new stack
  == Spawn extension (local, h, 1) exited non-zero on 'SIP/1001-a1fb'
Feb 22 12:44:29 NOTICE[8282]: chan_iax2.c:6188 iax2_poke_noanswer: Peer
'iaxtel' is now UNREACHABLE!
Feb 22 12:44:54 NOTICE[8282]: chan_iax2.c:5668 socket_read: Peer
'iaxtel' is now REACHABLE!

This is the console message shows the connection to their 613 service:

-- Executing SetCallerID("SIP/1001-11c3", ""David Kwok"") in new
stack
-- Executing Dial("SIP/1001-11c3",
"IAX2/X:[EMAIL PROTECTED]/613|60|r") in new stack
-- Called X:[EMAIL PROTECTED]/613
-- Call accepted by 65.39.205.121 (format ulaw)
-- Format for call is ulaw
-- IAX2/FWD/2 is ringing
-- IAX2/FWD/2 answered SIP/1001-11c3
-- Hungup 'IAX2/FWD/2'
  == Spawn extension (local, 393613, 2) exited non-zero on
'SIP/1001-11c3'
-- Executing Hangup("SIP/1001-11c3", "") in new stack
  == Spawn extension (local, h, 1) exited non-zero on 'SIP/1001-11c3'

Has anyone have the same experience?

David Kwok
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RE: [Asterisk-Users] Zap call bridge drops randomly

[mattf]

Would enabling Busydetect really help if Asterisk thinks it detects an
On-Hook?

MATT---

-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED]
Sent: Monday, February 21, 2005 7:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Zap call bridge drops randomly


> We have a call redirection system setup inhouse to send calls from an
> incoming line on a T1 to an external dialed out number:
>   Zap(call comes in) -> Asterisk -> Zap(call dials out)
> 
> The call comes in on a Robbed-bit T1 (d4/ami EM Wink) and goes out on a
PRI.
> We are using Asterisk release 1.0.5
> 
> Randomly the calls will drop less than a minute into the call. The Debug
> messages at the end of the call always say something like this: (incoming
> call on 58, outgoing on 73) 
> 
> Feb 21 11:37:28 DEBUG[16804]: Exception on 63, channel 58
> Feb 21 11:37:28 DEBUG[16804]: Got event On hook(1) on channel 58 (index 0)
> Feb 21 11:37:28 DEBUG[16804]: disabled echo cancellation on channel 58
> Feb 21 11:37:28 DEBUG[16804]: Unlinking slave 73 from 58 
> 
> The strange thing is that the person that called in did not hang up, in
fact
> they are usually talking when this call goes dead. Only about 10% of the
> calls that redirect have this happen to them, and it seems to be random as
> to which ones drop. Calls that dial out on either T1 and calls that come
in
> and are not redirected never seem to have these problems.
> 
> I have callprogress=no and busydetect=no but that doesn't seem to help.
> 
> Anyone have an idea on this?
> 
> Is there any way to make Asterisk less sensitive to hangups if that's even
> the cause?
> 
> Just looking for some feedback before I post on the bugtracker.

You might try:
busydetect=yes
busycount=6

in the top section of zapata.conf and see if that helps. I've not
tried this on a T1, but it certainly clears up the same issue on
the TDM pstn lnterfaces.


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[Asterisk-Users] Canadian DIDs...

[Mohit Muthanna]

Anybody know a good IAX provider for Canadian DIDs?

Mohit.


-- 
Mohit Muthanna [mohit (at) muthanna (uhuh) com]
"There are 10 types of people. Those who understand binary, and those
who don't."
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Re: [Asterisk-Users] sip wifi phone?

[cory]

These are the currently available wireless IP Phones that we are aware of.

Pulver
Zyxel 2000
Hitachi IP5000
Clipcomm CP-100E (Traditional desktop phone that is WIFI extensible with
an optional PCMCIA wireless card)

We are also currently in the midst of testing a new wireless phone from a
Canadian firm called Wiztel.

There are a few other vendors bringing WIFI VOIP phones to market in the
next few months.

Cory Andrews
VOIPSupply.com

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[Asterisk-Users] Multiple multiline sip phones ringing.

[C F]

how would one dial multiple multiline sip phones (cisco 7960) and
making sure that all the phones ring on the next available line
appearance?
I'm currently using the local channel to accomplish this but I'm
having some trouble. Here is the configs:
each cisco 7960 phone has six registrations in sip.conf, 1XX1 thru
1XX6, normaly when an extension is dialed the following happens:

exten => 101,1,Macro(stdext,${EXTEN})
[macro-stdext]
;arg1 = ${EXTEN}
exten => s,1,Diall(sip/${ARG1}1,45,r)
exten => s,2,Goto(s-u,1)
exten => s,102,Dial(sip/${ARG1}2,45,r)
exten => s,103,Goto(s-u,1)
exten => s,203,Dial(sip/${ARG1}3,45,r)
exten => s,204,Goto(s-u,1)
exten => s,304,Dial(sip/${ARG1}4,45,r)
exten => s,305,Goto(s-u,1)
exten => s,405,Dial(sip/${ARG1}5,45,r)
exten => s,406,Goto(s-u,1)
exten => s,506,Dial(sip/${ARG1}6,45,r)
exten => s,507,Goto(s-u,1)
exten => s-b,1,Voicemail(b${ARG1})
exten => s-u,1,Voicemail(u${ARG1})

when I want to ring mutiple phones this is what I do:

[rollover]
exten => _1XX,1,Dial(SIP/${EXTEN}1,45,r)
exten => _1XX,102,Dial(SIP/${EXTEN}2,45,r)
exten => _1XX,203,Dial(SIP/${EXTEN}3,45,r)
exten => _1XX,304,Dial(SIP/${EXTEN}4,45,r)
exten => _1XX,405,Dial(SIP/${EXTEN}5,45,r)
exten => _1XX,506,Dial(SIP/${EXTEN}6,45,r)

[default]
exten => 160,1,Dial(Local/[EMAIL PROTECTED]&Local/[EMAIL PROTECTED])

the problem is that if:
1. A phone is not registered the dial fails.
2. It doens't hand back nicely to exten 160,2 in default (it first
waits for the timeout)
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[Asterisk-Users] LiveVoip digit loss

[Ed Greenberg]

Receiving calls from LiveVoip DIDs results in dropped DTMF digits.
I'm using SIP, not IAX, and I've tried this without a dtmfmode and with 
dtmfmode in all the various permutations. Note that LiveVoip does not 
instruct us to put any dtmfmod statement in.

The server is set to do ulaw and I've verified that it is doing so.
LiveVoip originally suggested that I go from IAX to SIP to resolve this. I 
did. No resolution.

Questions:
1. What's the default for dtmfmode if we don't the statement in there?
2. Any suggestions on whether this is my problem of LiveVoip?
Thanks,
 
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[Asterisk-Users] Unable to call FWD user via IAX servers

[david kwok]

I have set up FWD via IAX service. I have tested the IAX service with 
613, echo test, and 612, saytime. It all works well.

However when ringing a FWD user, I got this error all the time:
Connected to Asterisk CVS-v1-0-02/01/05-09:34:45 currently running on 
chat (pid = 8282)
chat*CLI>
Verbosity is at least 3
   -- Executing SetCallerID("SIP/1001-a1fb", ""David Kwok"") in new stack
   -- Executing Dial("SIP/1001-a1fb", 
"IAX2/:[EMAIL PROTECTED]/268757|60|r") in new stack
   -- Called X:[EMAIL PROTECTED]/268757
   -- Call accepted by 65.39.205.121 (format ulaw)
   -- Format for call is ulaw
   -- IAX2/FWD/4 is busy
   -- Hungup 'IAX2/FWD/4'
 == Everyone is busy/congested at this time
   -- Executing Congestion("SIP/1001-a1fb", "") in new stack
 == Spawn extension (local, 393268757, 3) exited non-zero on 
'SIP/1001-a1fb'
   -- Executing Hangup("SIP/1001-a1fb", "") in new stack
 == Spawn extension (local, h, 1) exited non-zero on 'SIP/1001-a1fb'
Feb 22 12:44:29 NOTICE[8282]: chan_iax2.c:6188 iax2_poke_noanswer: Peer 
'iaxtel' is now UNREACHABLE!
Feb 22 12:44:54 NOTICE[8282]: chan_iax2.c:5668 socket_read: Peer 
'iaxtel' is now REACHABLE!

This is the console message shows the connection to their 613 service:
   -- Executing SetCallerID("SIP/1001-11c3", ""David Kwok"") in new stack
   -- Executing Dial("SIP/1001-11c3", 
"IAX2/X:[EMAIL PROTECTED]/613|60|r") in new stack
   -- Called X:[EMAIL PROTECTED]/613
   -- Call accepted by 65.39.205.121 (format ulaw)
   -- Format for call is ulaw
   -- IAX2/FWD/2 is ringing
   -- IAX2/FWD/2 answered SIP/1001-11c3
   -- Hungup 'IAX2/FWD/2'
 == Spawn extension (local, 393613, 2) exited non-zero on 'SIP/1001-11c3'
   -- Executing Hangup("SIP/1001-11c3", "") in new stack
 == Spawn extension (local, h, 1) exited non-zero on 'SIP/1001-11c3'

Has anyone have the same experience?
David Kwok
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Re: [Asterisk-Users] Minimal hardware requirements

[Roderick A. Anderson]

Rudolf Ladyzhenskii wrote:
Hi, all
I am doing "prrof of concept" system. I will have two IP phones connected to 
Asterisk box. Box itself will have 1 PSTN conenction and one analog phone conenction. A 
basic minimal configuration.
At the moment I am planning to use an old PII-350 with 128M of RAM I have lying around. I can not test anything yet, as I am waiting for phones to arrive, so question is will that be enough to demonstrate?
I am trying a similar setup and did get one error about needing at least 
256 MByte RAM.  I'm doing this very sporadically so I can't remember 
which application gave the error.  US$50.00 at Office Max and I have a , 
hopefully , < $500 home phone system.  Which is being used to proto type 
a system for the office.

Rod
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[Asterisk-Users] sip wifi phone?

[Kurt Fankhauser]

Title: Message



Does anyone know of 
any sip wifi phones? Only one i can find that is redily availiable is the zyxel 
prestige 2000w and from what i hear it is flaky.
 
Kurt Fankhauser
WaveLincwww.wavelinc.com114 S. Walnut 
St.Bucyrus, OH 44820419-562-6405 
 


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Re: [Asterisk-Users] voice recognition xml

[Steve Underwood]

Hi,
That much exists in a well standardised form (VoiceXML at 
http://www.w3c.org/Voice/) and several not so well standardised forms. 
It is what comes beyond the basic XML that needs to be implemented.

Regards,
Steve
beonice wrote:
Dean,
I'd be very interested in helping with this effort.
I've worked with both SGML and XML in the past (I used
to work at SoftQuad in Toronto, one of the original
providers of SGML and HTML tools), and have written
several DTDs, both for SGML and XML.
I think it would be fun to work on an XML interchange
design for voice recognition ... please let me know if
your contact would be interested.
Cheers,
Maya Kurup
 

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Re: [Asterisk-Users] IAX channel unable to create

[Pedro]

First off - 

change:
exten => _2000,1,Dial(IAX/master:[EMAIL PROTECTED]/[EMAIL PROTECTED])

to:
exten => _2000,1,Dial(IAX2/master:[EMAIL PROTECTED]/[EMAIL PROTECTED])

On Mon, 21 Feb 2005 13:00:39 -0500, kurt x <[EMAIL PROTECTED]> wrote:
> I have two * boxes running two differnet versions of *.
>  Box A is running:
> 
> Asterisk CVS-HEAD-07/14/04-16:28:29 built by
> [EMAIL PROTECTED] on a i686 running Linux
> 
> Box B is running:
> 
> Asterisk 1.0.3 built by [EMAIL PROTECTED] on a i386 running FreeBSD
> 
> I can make a IAX call from B to A but not from A to B.
> When I try to make a call from A to B I get these messages:
> 
> Feb 21 12:48:12 WARNING[-1233155152]: channel.c:1860 ast_request: No
> channel type registered for 'IAX'
> Feb 21 12:48:12 NOTICE[-1233155152]: app_dial.c:696 dial_exec: Unable
> to create channel of type 'IAX'
> Feb 21 12:48:14 WARNING[-1116300368]: chan_sip.c:673 retrans_pkt:
> Maximum retries exceeded on call
> [EMAIL PROTECTED]
> for seqno 1 (Non-critical Response)
> 
> My box A iax.conf:
> [general]
> port=5036
> bindport=5036
> bandwidth=low
> allow=ulaw
> disallow=lpc10
> jitterbuffer=no
> tos=lowdelay
> 
> [slave]
> type=friend
> secret=4435
> context=voice-mail
> defaultip=192.168.2.232
> qualify=yes
> 
> My Box A extension.conf
> [voice-mail]
> exten => _2000,1,Dial(IAX/master:[EMAIL PROTECTED]/[EMAIL PROTECTED])
> 
> My box B iax.conf
> [general]
> port=5036
> bindport=5036
> bandwidth=low
> allow=ulaw
> disallow=lpc10
> tos=lowdelay
> 
> [master]
> type=friend
> secret=4435
> context=home
> defaultip=192.168.1.2
> qualify=yes
> 
> My Box B extension.conf
> [home]
> exten => _24xx,1,Dial(IAX2/slave:[EMAIL PROTECTED]/[EMAIL PROTECTED])
> 
> Thanks in advance
> 
> Kurt
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RE: [Asterisk-Users] Why can't I make inter IAX calls between 2 Asterisk servers

[Joseph]

Try to analyze this link: Asterisk - Dual -Server:
http://www.voip-info.org/wiki-Asterisk+-+dual+servers

#Joseph

On Mon, 2005-02-21 at 15:41 -0700, [EMAIL PROTECTED] wrote:
> Hello,
>  Can anyone help with this please?
>  
> thx,
> chuks
> 
> 
> 
>  Original Message 
> Subject: [Asterisk-Users] Why can't I make inter IAX calls
> between
> 2 Asterisk servers
> From: [EMAIL PROTECTED]
> Date: Mon, February 21, 2005 11:04 am
> To: asterisk-users@lists.digium.com
> 
> Hello,
> two questions: 
>  
> 1: How can I open/enable network connection to B?
> scenerio:
> I have 2 Asterisk servers, A and B, running Fedora Core1 on my
> local network. B refuses any network connection attempts from
> A, i.e. I can't even telnet or FTP to B from A, but I can to A
> from B. This makes B refuse any IAX connection attempt from
> A. 
>  
>  
> 2: what's wrong with my configurations, why can't I dial A
> from B, and vice versa?
> scenerio:
> A and B each has an analog device connected to their
> Zap/1 channels, on extensions 2000 and 3000 respectively. I am
> trying to make IAX calls to each extension from the other, i.e
> call 3000 (on B) from A, and call 2000 (on A) from B. I get
> two different errors. While calling ext 2000 (on B) from A,
> connection was refused because of problem 1 above. While
> calling ext 3000 (on A) from B, it says context/extension does
> not exist on A. Here are my config files:
>  
> A's extension.config
>  
> [internal]
> exten => 3000,1,DIAL(IAX2/chuks:[EMAIL PROTECTED]/3000) 
> exten => 3000,2,congestion
> include -> from-iax
>  
> [from-iax]
> exten => s,1,Wait(2)
> exten => s,2,Answer
> exten => 2000,3,Dial(Zap/1,20)
>  
> NB: A's zapata.conf points to the internal context
>  
> A's iax.conf
>  
> [general]
> port=5036
> bandwidth=high
> disallow=lpc10  
> tos=lowdelay
>  
> [michael]
> type=friend
> secret=password
> auth=plaintext
> host=192.168.1.107
> context=from-iax
> allow=all
> trunk=yes
>  
>  
> B's extension.config
>  
> [internal]
> exten =>
> 2000,1,DIAL(IAX2/michael:[EMAIL PROTECTED]/2000)  ;A is
> on 192.168.1.103
> exten => 2000,2,congestion
> include -> from-iax
>  
> [from-iax]
> exten => s,1,Wait(2)
> exten => s,2,Answer
> exten => 3000,3,Dial(Zap/1,20)
>  
> NB: B's zapata.conf points to the internal context
>  
> B's iax.conf
>  
> [general]
> port=5036
> bandwidth=high
> disallow=lpc10  
> tos=lowdelay
>  
> [chuks]
> type=friend
> secret=password
> auth=plaintext
> host=192.168.1.103
> context=from-iax
> allow=all
> trunk=yes
>  
>  
> At least I thought I'd hear A ring when I dial 2000 from B,
> instead, I get the congestion (busy) tone. Can anyone tell me
> what I'm doing wrong? If I can open up B's network connectios,
> I know I'll get the same problem each way. 
>  
> thx,
> chuks
> [EMAIL PROTECTED]
>  
>  
> 
>  
> 
> __
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Re: [Asterisk-Users] voice recognition xml

[beonice]

Dean,

I'd be very interested in helping with this effort.
I've worked with both SGML and XML in the past (I used
to work at SoftQuad in Toronto, one of the original
providers of SGML and HTML tools), and have written
several DTDs, both for SGML and XML.

I think it would be fun to work on an XML interchange
design for voice recognition ... please let me know if
your contact would be interested.

Cheers,
Maya Kurup

--- dean collins <[EMAIL PROTECTED]> wrote:

> Anyone here technical enough to design a voice
> recognition voice xml
> interchange for asterisk please email me; I've been
> speaking with a
> contact of mine that is in the voice recognition
> space and he is
> interested in 'donating' some technical support to
> the Asterisk
> community to assist with this project.
> 
>  
> 
> This can only help benefit the Asterisk Community if
> this comes off.
> 
>  
> 
> If this got up and running it would mean that
> Asterisk users would be
> able to offer voice recognition capabilities to
> their clients (or on
> their own installations) in an on-net ASP
> capability.
> 
>  
> 
> Email me and I'll send you the details of the
> working group.
> 
>  
> 
>  
> 
> Cheers,
> 
> Dean
> 
>  
> 
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[Asterisk-Users] NAT-helping outbound proxy

[ds-lists]

Hi,

We're deploying a small VoIP solution for a group of teleworkers.
Naturally, this exposes us to all sorts of fun, most of which we seem to
have working properly. However, some NAT issues are still bugging us and
we have noticed that often these situations didn't exist when users were
connected directly to our VoIP provider, voiptalk.org.

They have something which they call a NAT-Traversal Gateway (see item 6 at
http://www.voiptalk.org/products/voiptalkfaq.html), which one configures
as the outgoing proxy, using port 5065.

Does anyone have any idea what this NAT-Traversal Gateway could be?
Naturally, I'm asking this in the hope that I can install something
similar on our server to solve our NAT issues.

Thanks in advance for your help.

Best regards,

David Shirley
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Re: [Asterisk-Users] Q.SIG support in CVS

[Matt Fredrickson]

On Sat, Feb 19, 2005 at 11:17:14AM +0100, Kurt Bauer wrote:
> So all I have to do is change the line 'switchtype=euroisdn' in zapata.conf 
> to 'switchtype=Q.SIG' ??? I still haven't got the point how to tell my * 
> Box that it is now Q.SIG aware :-o

Well, switchtype=qsig, not switchtype=Q.SIG.  You'll also probably want to
be using either libpri-matt or use the patch agains CVS head in bug 3554.


Matthew Fredrickson
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RE: [Asterisk-Users] Minimal hardware requirements

[Marco Castillo]

I have two comments:
a. It maybe doesn't work because of the PCI specifications the box support.
If was manufactured before Jan 2000, it is quite probably that it won't
recognize the Digium cards.
b. From the point of view of load, I see no problems, I think the specs of
the machine are enough for such a small system.
Hope it helps.

Marco


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rudolf
Ladyzhenskii
Sent: Monday, February 21, 2005 6:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Minimal hardware requirements


Hi, all

I am doing "prrof of concept" system. I will have two IP phones connected to
Asterisk box. Box itself will have 1 PSTN conenction and one analog phone
conenction. A basic minimal configuration.

At the moment I am planning to use an old PII-350 with 128M of RAM I have
lying around. I can not test anything yet, as I am waiting for phones to
arrive, so question is will that be enough to demonstrate?

Thanks,
Rudolf
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[Asterisk-Users] Re: list SNR

[David Josephson]

John Novack writes
Dare I suggest that a MUCH better job of documenting would go a long way 
towards eliminating the problems  you mention?

Now I realize that programmers are much more interested in writing code 
than documentation, as well as moving on to the next hot feature than 
making sure the current set work well, but . . .

I have found the Asterisk handbook Version 2, to be kind, poor, and the 
Mahler book sold for a small fortune obsolete as well as lacking. Most 
of what is published in that book is a rehash of what is available on 
line free for the taking.

Hi John - same experience here. Actually there were a couple of good 
efforts in this direction which have now merged. There are two volumes 
up now and more on the way, see www.asteriskdocs.org. Join and help make 
it happen. I have suggested that another volume ought to be about 
interfacing to carriers, legacy switches and subscriber equipment, etc.
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RE: [Asterisk-Users] Minimal hardware requirements

[Rudolf Ladyzhenskii]

Thanks.

I guess, I will have to try it and see. Mine is one of those small form factor 
COMPAQ boxes. I will try to get full specs from COMPAQ/HP. 

What about load Asterisk puts on processor if you do, for example, IP-IP call 
and IP-PSTN call? Since I will use Polycom phones, I will use SIP.

Rudolf

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John Novack
( Mozilla - portable )
Sent: Tuesday, February 22, 2005 11:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Minimal hardware requirements


Rudolf Ladyzhenskii wrote:

>Hi, all
>
>I am doing "prrof of concept" system. I will have two IP phones connected to 
>Asterisk box. Box itself will have 1 PSTN conenction and one analog phone 
>conenction. A basic minimal configuration.
>
>At the moment I am planning to use an old PII-350 with 128M of RAM I have 
>lying around. I can not test anything yet, as I am waiting for phones to 
>arrive, so question is will that be enough to demonstrate?
>
>Thanks,
>Rudolf
>
>  
>
Depends.

If you plan on using the TDM400 with one each FXS and FXO, the MB needs 
to have PCI Ver 2.2 slots, or the card won't be seen

Any MB made after 2000 probably is OK


John Novack



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[Asterisk-Users] How do I install Skinny support for non sip cisco phones

[Paul A Brown]

I have a server setup that runs sip no problem. I want to try a cisco phone.
how do I 

a) Tell if I have skinny support loaded
b) Load it onto a debian system
Many thanks
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Re: [Asterisk-Users] Minimal hardware requirements

[Howard Lowndes]

On Tue, 2005-02-22 at 11:41, John Novack ( Mozilla - portable ) wrote:
> Rudolf Ladyzhenskii wrote:
> 
> >Hi, all
> >
> >I am doing "prrof of concept" system. I will have two IP phones connected to 
> >Asterisk box. Box itself will have 1 PSTN conenction and one analog phone 
> >conenction. A basic minimal configuration.
> >
> >At the moment I am planning to use an old PII-350 with 128M of RAM I have 
> >lying around. I can not test anything yet, as I am waiting for phones to 
> >arrive, so question is will that be enough to demonstrate?
> >
> >Thanks,
> >Rudolf
> >
> >  
> >
> Depends.
> 
> If you plan on using the TDM400 with one each FXS and FXO, the MB needs 
> to have PCI Ver 2.2 slots, or the card won't be seen
> 
> Any MB made after 2000 probably is OK

Well, I have a PII 300 of about 2000 vintage and that didn't work with
the TDM400P card.  The PCI 2.2 spec was announced in Jan 2000 so it
would take a while for it to filter thru to actual mobos.

> 
> 
> John Novack
> 
> 
> 
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[Asterisk-Users] zaprtc on Debian Sarge 2.4.27

[Philipp von Klitzing]

Hi there,

since I found a couple of reports with complaints concerning zaprtc I 
thought that one or the other user might be glad to know that it works 
indeed. All that was necessary was to copy all *.h files from /zaptel 
into /zaptelrtc and then do "make" followed by "make load".

Of course you'll need to make sure to not have rtc compiled into your 
kernel, or loaded as a module.

Cheers, Philipp


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Re: [Asterisk-Users] Minimal hardware requirements

[John Novack ( Mozilla - portable )]

Rudolf Ladyzhenskii wrote:
Hi, all
I am doing "prrof of concept" system. I will have two IP phones connected to 
Asterisk box. Box itself will have 1 PSTN conenction and one analog phone conenction. A 
basic minimal configuration.
At the moment I am planning to use an old PII-350 with 128M of RAM I have lying 
around. I can not test anything yet, as I am waiting for phones to arrive, so 
question is will that be enough to demonstrate?
Thanks,
Rudolf
 

Depends.
If you plan on using the TDM400 with one each FXS and FXO, the MB needs 
to have PCI Ver 2.2 slots, or the card won't be seen

Any MB made after 2000 probably is OK
John Novack

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[Asterisk-Users] Minimal hardware requirements

[Rudolf Ladyzhenskii]

Hi, all

I am doing "prrof of concept" system. I will have two IP phones connected to 
Asterisk box. Box itself will have 1 PSTN conenction and one analog phone 
conenction. A basic minimal configuration.

At the moment I am planning to use an old PII-350 with 128M of RAM I have lying 
around. I can not test anything yet, as I am waiting for phones to arrive, so 
question is will that be enough to demonstrate?

Thanks,
Rudolf
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RE: [Asterisk-Users] asterisk - oh323 driver

[Oswaldo Arratia]

Hi,
Is there anybody out there that can e-mail me the following 
files?
 
Get oh323 fromhttp://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries/openh323-Janus_patch4-src-tar.gzGet 
pwlib fromhttp://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries/pwlib-Janus_patch4-src-tar.gzGet 
asterisk-oh323 fromhttp://www.inaccessnetworks.com/projects/asterisk-oh323/download/asterisk-oh323-0.6.5.tar.gz
 
 
 
inaccessnetworks web page is going down every 2 
minutes  and I am trying to make h323 work.
 
I'd really appreciate if anyone can send them to me 
off-list.  
 Thanks!!
 


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of João 
AmaroSent: Thursday, January 06, 2005 11:40 AMTo: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: Re: 
[Asterisk-Users] asterisk - oh323 driver
-BEGIN PGP SIGNED MESSAGE-Hash: 
SHA1Kanuri,If you want to use the last stable release of 
asterisk (1.0.3), youshould do it:(don't forget to read the README 
)Get oh323 fromhttp://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries/openh323-Janus_patch4-src-tar.gzGet 
pwlib fromhttp://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries/pwlib-Janus_patch4-src-tar.gzGet 
asterisk-oh323 fromhttp://www.inaccessnetworks.com/projects/asterisk-oh323/download/asterisk-oh323-0.6.5.tar.gzUntar 
the files#tar zxvf openh323-Janus_patch4-src-tar.gz#tar zxvf 
pwlib-Janus_patch4-src-tar.gz#tar zxvf asterisk-oh323-0.6.5.tar.gz#tar 
zxvf asterisk-1.0.3.tar.gzInstall Pwlib#cd pwlib#./configure 
&& make clean && make opt && make install && 
ldconfigPatch and Install OpenH323#cd openh323#patch -p1 
< ../asterisk-oh323-0.6.5/openh323_1.13.5-make.patch#./configure 
&& make clean && make opt && make install && 
ldconfigAsterisk#cd asterisk-1.0.3#make && make 
install && make samplesAsterisk-oh323#cd 
asterisk-oh323-0.6.5Edit the Makefile#make && make install 
&& ldconfigHope it helps,Contact-me of line if it don't 
workJoão AmaroKanuri, Seshu (Company 
IT) wrote:|| Rafael,|| Thanks for the detailed instructions. 
This really helps everyone| looking fix this nagging issue.|| Seshu 
Kanuri|| -Original Message- From:| [EMAIL PROTECTED]| 
[mailto:[EMAIL PROTECTED]] 
On Behalf Of| Rafael J. Risco G.V. Sent: Thursday, January 06, 2005 9:45 AM 
To:| Asterisk Users Mailing List - Non-Commercial Discussion Subject:| 
Re: [Asterisk-Users] asterisk - oh323 driver|| Hi I am using oh323-0.7.1 
with asterisk cvs head version and works| great for me (Linux Fedora1), see 
details below:|| Requirements: 
PWLIB    
:    pwlib-v1_6_6-src.tar.gz  (orJanus_Patch)| 
OpenH323    
:    openh323-v1_13_5-src.tar.gz (or Janus_Patch)| 
Inaccessnetworks-asterisk-oh323    :    
asterisk-oh323-0.7.1.tar.gz|| Sources:| http://www.inaccessnetworks.com/projects/asterisk-oh323/download/asteris|  
k-oh323-0.7.1.tar.gz| http://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries/openh3|  
23-Janus_patch4-src-tar.gz| http://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries/pwlib-|  
Janus_patch4-src-tar.gz|| Note: asterisk-oh323-0.7.1 must be used with 
Asterisk CVS Head...|| Installation: tar -zxvf 
asterisk-oh323-0.7.1.tar.gz tar -zxvf| pwlib-Janus_patch4-src-tar.gz tar 
-zxvf| openh323-Janus_patch4-src-tar.gz|| cd pwlib ./configure 
make|| cd openh323 patch -p1 <| 
/root/asterisk-oh323-0.7.0/openh323_1.13.5-make.patch (pach to| 
openh323)|| cd openh323 ./configure make opt|| ASTERISK CVS 
Head: ---|| export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot 
cvs| login    - the password is 
anoncvs. cvs checkout zaptel libpri| asterisk|| cd zaptel/ make 
clean; make install|| cd libpri/ make clean make install|| cd 
asterisk/ make clean make install make samples make progdocs|| Finally 
install ASTERISK OH323 channel driver:| 
-- cd| asterisk-oh323-0.7.1 vi 
Makefile  ( check paths according with your| system )|| 
PWLIBDIR=/root/pwlib OPENH323DIR=/root/openh323| 
ASTERISKINCDIR=/root/asterisk/include| 
ASTERISKMODDIR=/usr/lib/asterisk/modules| ASTERISKETCDIR=/etc/asterisk 
OH323WRAPLIBDIR=/usr/local/lib| SSLINCDIR=/usr/include/openssl 
SSLLIBDIR=/usr/lib|| Compiling --- Type "make" to build the 
oh323wrap library| and the ASTERISK OH323 channel driver.|| Type 
"make install" to install the binaries.|| Add to your LD_LIBRARY_PATH 
the path where the oh323wrap library| was installed (or edit your 
/etc/ld.so.conf file, add  the library| path, and run 
"ldconfig").|| Hope it helps|| Rafael Risco| 
|| NOTICE: If 
received in error, please destroy and notify sender.| Sender does not waive 
confidentiality or privilege, and use is| 
prohibited.||-BEGIN PGP SIGNATURE-Version: GnuPG v1.2.4 
(GNU/Linux)iD8DBQFB3WnZJUm/Bor63CERAoLjAKCYOZsUNE3uVxd0COgOkHi2nDVE2wCfX+fpt0iiPQYJesHaZ2upDytUzvg==Jb1m-END 
PGP SIGNATURE-

Re: [Asterisk-Users] Snom phone hint exten question

[Paul Fielding]

- Original Message - 
From: "James Bean" <[EMAIL PROTECTED]>

I am going to now sit in a corner and go quietly insane while playing
the banyo with no strings.
I'd probably go insane, too, if I was trying to figure out how the heck to 
play a banyo

;)
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Re: [Asterisk-Users] Suggestion for noise reduction on Asterisk-Users

[Kristian Kielhofner]

Rich Adamson wrote:
This same topic comes up about every month or two, and the exact same
words are used over and over again. The last run at this was on the -dev
list about one/two months ago and shouldn't be hard to find.
If memory serves anywhere near correct (which is a stretch), lots of
folks agreed basically on:
 - all new signups to the list be sent a 'how-to' list of where to
   find answers, not to use html, etc, etc.
 - monthly reminders sent,
 - auto-response to certain posts,
Let's not repeat the same thing yet again.
Rich,
	I know.  I read the entire thread.  Where is all/any of that?  I would 
like to add my original suggestion of moving the list subscribe 
information down a couple of levels AFTER some basic documentation. 
That is what my original thread said.  I feel that would help the most.

--
Kristian Kielhofner
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Re: [Asterisk-Users] Suggestion for noise reduction on Asterisk-Users

[Rich Adamson]

> > This might be one for Digium, but I would like to see some type of 
> > Wiki that people would have to wade through before they would get the 
> > information on how to subscribe to the list.
> >
> >
> > How many more times do we have to read posts of "I just downloaded 
> > the Asterisk and now how do I talk on it?"
> >
> > Let me know what you think...
> >
> > -- 
> > Kristian Kielhofner
> 
> Dare I suggest that a MUCH better job of documenting would go a long way 
> towards eliminating the problems  you mention?
> 
> Now I realize that programmers are much more interested in writing code 
> than documentation, as well as moving on to the next hot feature than 
> making sure the current set work well, but . . .
> 
> I have found the Asterisk handbook Version 2, to be kind, poor, and the 
> Mahler book sold for a small fortune obsolete as well as lacking. Most 
> of what is published in that book is a rehash of what is available on 
> line free for the taking.

This same topic comes up about every month or two, and the exact same
words are used over and over again. The last run at this was on the -dev
list about one/two months ago and shouldn't be hard to find.

If memory serves anywhere near correct (which is a stretch), lots of
folks agreed basically on:
 - all new signups to the list be sent a 'how-to' list of where to
   find answers, not to use html, etc, etc.
 - monthly reminders sent,
 - auto-response to certain posts,

Let's not repeat the same thing yet again.


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Re: [Asterisk-Users] Zap call bridge drops randomly

[Rich Adamson]

> We have a call redirection system setup inhouse to send calls from an
> incoming line on a T1 to an external dialed out number:
>   Zap(call comes in) -> Asterisk -> Zap(call dials out)
> 
> The call comes in on a Robbed-bit T1 (d4/ami EM Wink) and goes out on a PRI.
> We are using Asterisk release 1.0.5
> 
> Randomly the calls will drop less than a minute into the call. The Debug
> messages at the end of the call always say something like this: (incoming
> call on 58, outgoing on 73) 
> 
> Feb 21 11:37:28 DEBUG[16804]: Exception on 63, channel 58
> Feb 21 11:37:28 DEBUG[16804]: Got event On hook(1) on channel 58 (index 0)
> Feb 21 11:37:28 DEBUG[16804]: disabled echo cancellation on channel 58
> Feb 21 11:37:28 DEBUG[16804]: Unlinking slave 73 from 58 
> 
> The strange thing is that the person that called in did not hang up, in fact
> they are usually talking when this call goes dead. Only about 10% of the
> calls that redirect have this happen to them, and it seems to be random as
> to which ones drop. Calls that dial out on either T1 and calls that come in
> and are not redirected never seem to have these problems.
> 
> I have callprogress=no and busydetect=no but that doesn't seem to help.
> 
> Anyone have an idea on this?
> 
> Is there any way to make Asterisk less sensitive to hangups if that's even
> the cause?
> 
> Just looking for some feedback before I post on the bugtracker.

You might try:
busydetect=yes
busycount=6

in the top section of zapata.conf and see if that helps. I've not
tried this on a T1, but it certainly clears up the same issue on
the TDM pstn lnterfaces.


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RE: [Asterisk-Users] IAX ATA's

[Michael Giagnocavo]

ATAs with the PA168 - a very popular chip with quite a few of Chinese
manufacturers.

-Michael

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Lewis
Sent: Monday, February 21, 2005 4:36 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] IAX ATA's

Are their any good chooses for IAX Adapters?


-Thanks

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Re: [Asterisk-Users] Asterisk@Home 0.6 Released [Follow Me]

[Joel Vandal]

Hi,
Does it's possible to get more information about your design ?
Thanks,
--
Joel Vandal
- Original Message - 
From: "Race Vanderdecken" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>; "'Asterisk Users Mailing List - 
Non-Commercial Discussion'" 
Sent: Thursday, February 17, 2005 5:10 PM
Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] 0.6 Released [Follow Me]


I have a design that works for "Follow-Me" and "Find-Me" is anyone is
interested.
I can help you with the code, but don't ask me to check-it in to the
CVS.
Race "The Tyrant" Vanderdecken
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nitesh
Divecha
Sent: Thursday, February 17, 2005 2:51 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] 0.6 Released [Follow Me]
Hello All,
And thanks for the [EMAIL PROTECTED] 0.6! It works awesome.
Any plans to implement "Follow Me" feature?
Nitesh
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Re: [Asterisk-Users] Suggestion for noise reduction on Asterisk-U sers

[Kristian Kielhofner]

Colin Anderson wrote:
Lets point them to google site:voip-info.org 
or site:lists.digium.com then.  We do a lot of that once they get on the 
list.  Why not before?

OK, Ollie J if you are listening maybe you might consider appending those
links to your monthly or weekly list etiquitte reminders. Post them daily,
even. Seems to me that posting 1 daily reminder with links on the top 20
problems is better than 20 posts with those top 20 questions by newbies. At
worst, it's 1 extra email and at best you may save those 20 posts. Something
like:
MOST COMMONLY ASKED QUESTIONS:
---
1.How to light up the line indicator on your Snom:
http://www.google.ca/search?sourceid=navclient&ie=UTF-8&rls=GGLD,GGLD:2004-2
3,GGLD:en&q=snom+190+hint+site%3Alists%2Edigium%2Ecom
2. Fax modems and Asterisk:
http://www.google.ca/search?hl=en&rls=GGLD%2CGGLD%3A2004-23%2CGGLD%3Aen&q=fa
x+modems+site%3Alists.digium.com&meta=
3. How come Asterisk doesn't behave like my Meridian with "inbound lines?"
http://www.google.ca/search?hl=en&rls=GGLD%2CGGLD%3A2004-23%2CGGLD%3Aen&q=ke
y+system+site%3Alists.digium.com&meta=
4. How do I record calls?
http://www.google.ca/search?hl=en&rls=GGLD%2CGGLD%3A2004-23%2CGGLD%3Aen&q=za
pbarge+monitor+calls+site%3Alists.digium.com&meta=
FWIW, I find sometimes that meta-discussions like this and things like top
posting, distro advocacy etc take as much bandwidth as noob questions. 
Colin,
Thanks.  That is what I was looking for.
	And yes, these types of posts take up bandwidth just like newbie 
questions.  The difference is that newbies will keep coming, and coming, 
and coming until we just can't stand it anymore and go elsewhere.  You 
will find that with Linux and OSS you can't go anywhere without running 
into a distro war, BSD's vs. BSD's, BSD's vs. Linux, etc.  I won't even 
touch on those issues...

--
Kristian Kielhofner
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RE: [Asterisk-Users] Suggestion for noise reduction on Asterisk-U sers

[Colin Anderson]

>Lets point them to google site:voip-info.org 
>or site:lists.digium.com then.  We do a lot of that once they get on the 
>list.  Why not before?

OK, Ollie J if you are listening maybe you might consider appending those
links to your monthly or weekly list etiquitte reminders. Post them daily,
even. Seems to me that posting 1 daily reminder with links on the top 20
problems is better than 20 posts with those top 20 questions by newbies. At
worst, it's 1 extra email and at best you may save those 20 posts. Something
like:

MOST COMMONLY ASKED QUESTIONS:
---

1.How to light up the line indicator on your Snom:

http://www.google.ca/search?sourceid=navclient&ie=UTF-8&rls=GGLD,GGLD:2004-2
3,GGLD:en&q=snom+190+hint+site%3Alists%2Edigium%2Ecom

2. Fax modems and Asterisk:

http://www.google.ca/search?hl=en&rls=GGLD%2CGGLD%3A2004-23%2CGGLD%3Aen&q=fa
x+modems+site%3Alists.digium.com&meta=

3. How come Asterisk doesn't behave like my Meridian with "inbound lines?"

http://www.google.ca/search?hl=en&rls=GGLD%2CGGLD%3A2004-23%2CGGLD%3Aen&q=ke
y+system+site%3Alists.digium.com&meta=

4. How do I record calls?

http://www.google.ca/search?hl=en&rls=GGLD%2CGGLD%3A2004-23%2CGGLD%3Aen&q=za
pbarge+monitor+calls+site%3Alists.digium.com&meta=

FWIW, I find sometimes that meta-discussions like this and things like top
posting, distro advocacy etc take as much bandwidth as noob questions. 

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Re: [Asterisk-Users] bridging iaxtel calls to PSTN

[John Novack]

And yet another "helpful"
comment to clog up the list.

Some people use HTML
Some people top post
Some people don't read too well
Some people aren't as skilled as others in searching.
GET OVER IT


JMO

John Novack


Jens Vagelpohl wrote:

On Feb 21, 2005, at 23:36, [EMAIL PROTECTED] wrote:
  
  
  Hello,

 actually I did, but nobody responded to that.

  
  
Maybe people would look at it if you stopped sending HTML mail.
  
  
jens
  
  
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Re: [Asterisk-Users] Asterisk@home Linux has no KDE

[Dalon Westergreen]

use another pc.

--dalon

On Mon, 21 Feb 2005 15:56:26 -0600, Kiran Vahaja <[EMAIL PROTECTED]> wrote:
> Then how can i use web interface to configure?
> 
> On Mon, 21 Feb 2005 15:30:55 -0600, Kristian Kielhofner <[EMAIL PROTECTED]> 
> wrote:
> > Kiran Vahaja wrote:
> > > Hi Folks,
> > >
> > > I installed [EMAIL PROTECTED] on my PC. It went through the installation
> > > and all. But now i get a command line login window. Doesn't it has a
> > > KDE or some other type of OS GUI (i am not talking about [EMAIL PROTECTED]
> > > web GUI)? After i login, just the command line interface comes out.
> > > Any command to type here to get Linux OS GUI?
> > >
> > > Thanks,
> > > Kiran
> >
> > It probably doesn't have a GUI because you shouldn't use X on an
> > Asterisk server.  Asterisk does not play nice with X:
> >
> > http://voip-info.org/wiki-Asterisk+X11
> >
> > --
> > Kristian Kielhofner
> >
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Re: [Asterisk-Users] Suggestion for noise reduction on Asterisk-U sers

[Kristian Kielhofner]

Colin Anderson wrote:
Problem is, Wikis in general suck and voip-info.org in particular is quite
useless except as a random clicky-clicky exercise. You ever use the search
on voip-info.org? It's almost like someone goofed setting it up and the
search results are ordered least relevant first. 

News flash: The reason the newbie questions appear on the list in the first
place is because it's too hard to find
it on the web. Of course, Google is far better, and Mr. Critchfield takes
great pains to tell us so (and he's right) so there is the argument that if
you can't use Google properly you are wholly unqualified to run Asterisk.
Unfortunately, that type of elitism is exactly what marginalizes open source
/ Linux / RMS type of endeavors. 

Noob questions may seem like the death by a thousand cuts, but the
alternative is that there *are* no noobs and that means eventually that
"Netcraft Confirms Asterisk is Dying*" jokes will start flying.
*yeah, yeah, I know. I was trying to make a point. 
Colin,
	I know the search sucks.  Lets point them to google site:voip-info.org 
or site:lists.digium.com then.  We do a lot of that once they get on the 
list.  Why not before?

--
Kristian Kielhofner
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Re: [Asterisk-Users] bridging iaxtel calls to PSTN

[Jens Vagelpohl]

On Feb 21, 2005, at 23:36, [EMAIL PROTECTED] wrote:
Hello,
 actually I did, but nobody responded to that.
Maybe people would look at it if you stopped sending HTML mail.
jens
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Re: [Asterisk-Users] Suggestion for noise reduction on Asterisk-Users

[Kristian Kielhofner]

John Novack wrote:
Dare I suggest that a MUCH better job of documenting would go a long way 
towards eliminating the problems  you mention?

Now I realize that programmers are much more interested in writing code 
than documentation, as well as moving on to the next hot feature than 
making sure the current set work well, but . . .

I have found the Asterisk handbook Version 2, to be kind, poor, and the 
Mahler book sold for a small fortune obsolete as well as lacking. Most 
of what is published in that book is a rehash of what is available on 
line free for the taking.

Just my opinion, of course.
John Novack
That's exactly it.  The Wiki has been the only place that seems to be 
somewhat consistenly updated with changes in HEAD and STABLE.  That's 
because it can be updated by members of the community.  Point people 
there before you can even get the source or join the list and I think 
things will be better off for all (hence my post).

--
Kristian Kielhofner
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Re: [Asterisk-Users] * Call Monitoring

[Jens Vagelpohl]

BTW, I did need to suid the zttool-cli command to root, as the normal 
BB
user doesn't have the needed permissions. I haven't looked into this,
but if anyone has a suggestion on a better way to do this, feel free 
to
let me know.
It's called "sudo"
jens
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RE: [Asterisk-Users] bridging iaxtel calls to PSTN

[Rich Adamson]

iaxtel is not working and hasn't been for some time. All of us that
have 700 numbers get the same response that you are. 


> Hello,
>  actually I did, but nobody responded to that. So, here it is one more time:
> ___
> Hello,
>  can someone tell me what's wrong with this? I can't make toll free calls via 
> iaxtel. Here's 
the definition in my extensions.conf
>  
> [iaxtel-trunks]
> ;
> ;outbound 1-700 and toll free calls go via iaxtel
> ;be sure to include the iaxtel-trunks context in dialing context
> ;add function here to continue ring tone when 9 is dialed
> ;
> ignorepat=>9
> exten => _91700XXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]:1)
> exten => _91888NXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]:1)
> exten => _91877NXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]:1)
> exten => _91866NXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]:1)
> exten => _91800NXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]:1)
>  
> Note: IAXTEL_INFO is already defined as username:password
>  
> and here's my iax.conf
>  
> [general]
> port=5036
> bandwidth=high
> disallow=lpc10
> tos=lowdelay
>  
> ;to register with iaxtel
> register => username:[EMAIL PROTECTED]
>  
>  
> ;
> ; Trust Caller*ID Coming from iaxtel.com
> ;
> [iaxtel]
> type=friend
> context=from-iaxtel
> auth=cleartext
> ;inkeys=iaxtel
>  
>  
> when i make an 800 number call for instance, registration goes through and 
> iaxtel can find me. 
But there is an endless silence, sort of like an
> endles loop, and the only output I see is a "timeout on Zap/1-1" and it tries 
> the whole thing 
again...and goes on forever, and the call never goes
> through. Is there anything wrong with my configuration above?
>  
> thx,
> Chuks.
> 
>  Original Message 
> Subject: RE: [Asterisk-Users] bridging iaxtel calls to PSTN
> From: "Rich Adamson" <[EMAIL PROTECTED]>
> Date: Mon, February 21, 2005 3:16 pm
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> 
> I think someone did reply to this. Don't bother trying to use iaxtel for
> your connections. Its down far more then its up.
> 
> If you want help setting up an iax connection directly between your two
> systems, then post what you've got that pertains to this from iax.conf
> and extensions.conf. No one is going to be able to help you based on A
> doesn't talk to B words.
> 
> 
> > could you help me out with this? I have a posting on this list, bu nobody 
> > has replied yet.
> Titled "why can't I make IAX calls
> > between 2 asrterisk servers"? I'd appreciate.
> >  
> > -chuks.
> >
> >  Original Message 
> > Subject: Re: [Asterisk-Users] bridging iaxtel calls to PSTN
> > From: "Michael Graves" <[EMAIL PROTECTED]>
> > Date: Mon, February 21, 2005 7:49 am
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> > 
> >
> > On Sun, 20 Feb 2005 02:43:46 -0700, [EMAIL PROTECTED] wrote:
> >
> > >Hello,
> > > I just started using asterisk, and have a question. I have  setup two
> > >asterisk servers, A and B. A has a Digium TDM400 11B card (1 FXO and 1
> > >FSX modules) and is connected to the PSTN. B has same, but is NOT
> > >connected to PSTN. I want to configure B to call A via iaxtel, and
> > >connect to the PSTN using A's line. How can I configure iaxtel dial
> > >plan for B in extensions.conf? I want to be able to make a call to
> > >local US number (where A is located) from B, using iaxtel. Can anyone
> > >please help me? All I have seen so far is just making calls from A to B
> > >and vice versa using the iaxtel 1700 number, but I haven't seen any
> > >examples of how to bridge the iaxtel calls to PSTN. Help please.
> > >
> > >
> > >chuks.
> > >
> > >NB: I don't mean toll free number, I mean just local dialing.
> > >
> >
> > Don't bother with IAXTel. It's very frequently down. Just have each
> > server register with the other and trunk between them. That way you
> > just use dialplan logic to make the A place a call on B's resource. The
> > wiki has a good section on trunking between servers ova IAX.
> >
> > Michael
> >
> > --
> > Michael Graves   [EMAIL PROTECTED]
> > Sr. Product Specialist  www.pixelpower.com
> > Pixel Power Inc. [EMAIL PROTECTED]
> >
> > o713-861-4005
> > o800-905-6412
> > c713-201-1262
> >
> > ___
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> ---End of Original Message-
> 
> ___
> Asterisk-Us

RE: [Asterisk-Users] Suggestion for noise reduction on Asterisk-U sers

[Colin Anderson]

>   This wiki should cover most of the basic stuff that gets asked over
and 
>over again just to help reduce the amount of repetition that most of you 
>have probably noticed takes place here.

Problem is, Wikis in general suck and voip-info.org in particular is quite
useless except as a random clicky-clicky exercise. You ever use the search
on voip-info.org? It's almost like someone goofed setting it up and the
search results are ordered least relevant first. 

News flash: The reason the newbie questions appear on the list in the first
place is because it's too hard to find
it on the web. Of course, Google is far better, and Mr. Critchfield takes
great pains to tell us so (and he's right) so there is the argument that if
you can't use Google properly you are wholly unqualified to run Asterisk.
Unfortunately, that type of elitism is exactly what marginalizes open source
/ Linux / RMS type of endeavors. 

Noob questions may seem like the death by a thousand cuts, but the
alternative is that there *are* no noobs and that means eventually that
"Netcraft Confirms Asterisk is Dying*" jokes will start flying.

*yeah, yeah, I know. I was trying to make a point. 
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Re: [Asterisk-Users] Suggestion for noise reduction on Asterisk-Users

[John Novack]

Kristian Kielhofner wrote:
Hello all,
This might be one for Digium, but I would like to see some type of 
Wiki that people would have to wade through before they would get the 
information on how to subscribe to the list.

How many more times do we have to read posts of "I just downloaded 
the Asterisk and now how do I talk on it?"

Let me know what you think...
--
Kristian Kielhofner
Dare I suggest that a MUCH better job of documenting would go a long way 
towards eliminating the problems  you mention?

Now I realize that programmers are much more interested in writing code 
than documentation, as well as moving on to the next hot feature than 
making sure the current set work well, but . . .

I have found the Asterisk handbook Version 2, to be kind, poor, and the 
Mahler book sold for a small fortune obsolete as well as lacking. Most 
of what is published in that book is a rehash of what is available on 
line free for the taking.

Just my opinion, of course.
John Novack
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[Asterisk-Users] Problem with Avaya 4602 / SIP response 481

[Trevor G. Hammonds]

I have an Avaya 4602 IP phone that was previously working with Asterisk.  It
was being used elsewhere for several months, and I recently set it up again
to work with Asterisk.  Everything works fine for several minutes -- I am
able to receive and make calls as expected.  However, after a few minutes,
and every few minutes thereafter, I get the following message on the
console:

-- Got SIP response 481 "Call Does Not Exist" back from 192.168.1.98

After this message, the phone no longer works properly -- if at all.  When
attempting to dial, the phone seems to ignore my dial plan, accepting no
more than five digits (I get the tone burst indicating that it is done
accepting digits), and it does not transmit anything to Asterisk.
Attempting to call the phone's extension results in a busy condition (SIP
response 486).  Soft resetting the phone restores it to working order for
several minutes.  

I have searched Google and the Wiki for answers to this problem, and see
that Brian Elton has experienced this exact situation.  However, I saw no
resolution posted to the list, and do not have Brian's contact information.


I suspect this may be related to the MWI indicator and the "mailbox="
statement in my sip.conf file, as the last time I was using this phone with
*, it was not set up to use voicemail. 

Any ideas or assistance will be greatly appreciated. 

Sincerely,
Trevor Hammonds


SIP Debug output:

Sip read:
SIP/2.0 481 Call Does Not Exist
Call-ID: [EMAIL PROTECTED]
CSeq: 102 NOTIFY
From: "asterisk" ;tag=as5860bf17
To: ;tag=cad443b1cd74b1e
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK69c2e97b
Content-Length: 0
Contact: 
User-Agent: Elite 1.0 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26


9 headers, 0 lines
-- Got SIP response 481 "Call Does Not Exist" back from 192.168.1.98
Destroying call '[EMAIL PROTECTED]'
Destroying call '[EMAIL PROTECTED]'

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[Asterisk-Users] Brian Elton / Avaya 4602

[Trevor G. Hammonds]

I would like to get in contact with Brian Elton.  He posted information to
this list regarding problems with an Avaya 4602, late last year.  I am now
experiencing a similar issue, and would like to know if/how it was resolved.


Thank you.  

Sincerely,
Trevor Hammonds

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[Asterisk-Users] Zap call bridge drops randomly

[mattf]

Hello,

We have a call redirection system setup inhouse to send calls from an
incoming line on a T1 to an external dialed out number:
  Zap(call comes in) -> Asterisk -> Zap(call dials out)

The call comes in on a Robbed-bit T1 (d4/ami EM Wink) and goes out on a PRI.
We are using Asterisk release 1.0.5

Randomly the calls will drop less than a minute into the call. The Debug
messages at the end of the call always say something like this: (incoming
call on 58, outgoing on 73) 

Feb 21 11:37:28 DEBUG[16804]: Exception on 63, channel 58
Feb 21 11:37:28 DEBUG[16804]: Got event On hook(1) on channel 58 (index 0)
Feb 21 11:37:28 DEBUG[16804]: disabled echo cancellation on channel 58
Feb 21 11:37:28 DEBUG[16804]: Unlinking slave 73 from 58 

The strange thing is that the person that called in did not hang up, in fact
they are usually talking when this call goes dead. Only about 10% of the
calls that redirect have this happen to them, and it seems to be random as
to which ones drop. Calls that dial out on either T1 and calls that come in
and are not redirected never seem to have these problems.

I have callprogress=no and busydetect=no but that doesn't seem to help.

Anyone have an idea on this?

Is there any way to make Asterisk less sensitive to hangups if that's even
the cause?

Just looking for some feedback before I post on the bugtracker.

Thanks,

MATT---
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[Asterisk-Users] some questions about busy detection

[Warren Burstein]

I'm going to be hooking FXS lines on a TDM400 to a PBX which doesn't 
drop line voltage at the end of a call, so I'm going to have to use busy 
detection.  A few questions -

The tones are taken from the tones specified by the zone in zaptel.conf, 
right? Which tones cause hangup?

The PBX may not use the national standard tones.  Does anyone have any 
suggestion for how I can determine what tones it uses if the information 
is not in the PBX manual?  Back when I used to use Dialogic cards, there 
was a program called PBXpert (or something like that) which would 
measure all the PBX's tones.  But I think all I need is to record a tone 
and get a program that would tell me what tones/cadence are in the 
sample.  Cadence I could probably figure out from any graphical sound 
editor, tones too if it's single-frequency, but what if it's 
dual-frequency?  Is there a sound editor (free, or at least with an 
evaluation version that has this feature) that would do this for me?

thanks
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[Asterisk-Users] FWD problem

[Anton Krall]

Guys.
 
Im using IAX and FWD and I think everything is setup fine.. someobdy just
tried calling me but my phone jus ran once and sent them straight to the
voicemail.. the logs show this:
 
-- Accepting AUTHENTICATED call from 65.39.205.121:
   > requested format = ulaw,
   > requested prefs = (),
   > actual format = ulaw,
   > host prefs = (alaw|ulaw|ilbc|gsm),
   > priority = mine
-- Executing Dial("IAX2/[EMAIL PROTECTED]:4569-2", "sip/casa1|20|r")
in new stack
-- Called casa1
-- SIP/casa1-cfa7 is ringing
-- Accepting AUTHENTICATED call from 65.39.205.121:
   > requested format = ulaw,
   > requested prefs = (),
   > actual format = ulaw,
   > host prefs = (alaw|ulaw|ilbc|gsm),
   > priority = mine
-- Executing Dial("IAX2/[EMAIL PROTECTED]:4569-3", "sip/casa1|20|r")
in new stack
-- Called casa1
-- Got SIP response 486 "Busy" back from 10.0.0.12
-- SIP/casa1-8c26 is busy
  == Everyone is busy/congested at this time (1:1/0/0)
-- Executing VoiceMail("IAX2/[EMAIL PROTECTED]:4569-3", "b301") in
new stack
-- Playing '/var/spool/asterisk/voicemail/sip/301/busy' (language 'en')
  == Spawn extension (fromiaxfwd, 613602, 1) exited non-zero on
'IAX2/[EMAIL PROTECTED]:4569-2'
-- Hungup 'IAX2/[EMAIL PROTECTED]:4569-2'
-- Playing 'vm-intro' (language 'en')
-- Playing 'beep' (language 'en')
-- Recording the message
-- x=0, open writing:
/var/spool/asterisk/voicemail/sip/301/INBOX/msg0001 format: wav, 0x81424d0
-- User hung up
F
 
Is there some context problem here?
 
__
Anton Krall
 

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Re: [Asterisk-Users] Suggestion for noise reduction on Asterisk-Users

[Kristian Kielhofner]

Rich Adamson wrote:
The same has been proposed several times over the last nine months,
both on -users and -dev, and its simply been ignored.
Rich,
	I think that I said something like that somewhere in the message.  But 
it turned into much more than I thought it would, so I am not sure...

	Do you (or anyone) have proof that it has been offered up to someone at 
Digium - and their response, if any?

	Asterisk is open source, and communities are what drive open source.  I 
have seen many, many great technical minds drop off of the list because 
they (presumably) didn't want to have to wade through the repeats and 
previously asked/answered questions.

	I always put list traffic in seperate IMAP folders with procmail so I 
don't really have to read it if I don't want to.  I have gone from 
checking my Asterisk-Users folder many times an hour to maybe once every 
couple of days because hardly anything interesting shows up anymore. 
Someday I just might not check it all and give up on it.  I wonder how 
many more people have done just that - and we are all worse of because 
of it.  There are now many many more people asking questions that there 
are answering, and that's bad.

	P.S. - That is not to be taken as a threat in any way, I will probably 
never stop checking asterisk-users because I enjoy working with Asterisk 
way too much, but hopefully you can see my point...

--
Kristian Kielhofner
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RE: [Asterisk-Users] Why can't I make inter IAX calls between 2 Asterisk servers

[info]

Hello,
 Can anyone help with this please?
 
thx,
chuks
 Original Message Subject:
[Asterisk-Users] Why can't I make inter IAX calls between2 Asterisk
serversFrom: [EMAIL PROTECTED]Date: Mon, February 21, 2005
11:04 amTo: asterisk-users@lists.digium.com
Hello,
two questions: 
 
1: How can I open/enable network connection to
B?
scenerio:
I have 2 Asterisk servers, A and B, running Fedora
Core1 on my local network. B refuses any network connection
attempts from A, i.e. I can't even telnet or FTP to B from A, but
I can to A from B. This makes B refuse any IAX connection attempt
from A. 
 
 
2: what's wrong with my configurations, why
can't I dial A from B, and vice versa?
scenerio:
A and B each has an analog device connected to their
Zap/1 channels, on extensions 2000 and 3000 respectively. I am
trying to make IAX calls to each extension from the other, i.e call
3000 (on B) from A, and call 2000 (on A) from B. I get two
different errors. While calling ext 2000 (on B) from A, 
connection was refused because of problem 1 above. While calling ext
3000 (on A) from B, it says context/extension does not exist on A. Here
are my config files:
 
A's extension.config
 
[internal]
exten =>
3000,1,DIAL(IAX2/chuks:[EMAIL PROTECTED]/3000) 
exten => 3000,2,congestion
include -> from-iax
 
[from-iax]
exten => s,1,Wait(2)
exten => s,2,Answer
exten => 2000,3,Dial(Zap/1,20)
 
NB: A's zapata.conf points to the internal
context
 
A's iax.conf
 
[general]
port=5036
bandwidth=high
disallow=lpc10  
tos=lowdelay
 
[michael]
type=friend
secret=password
auth=plaintext
host=192.168.1.107
context=from-iax
allow=all
trunk=yes
 
 
B's extension.config
 
[internal]
exten =>
2000,1,DIAL(IAX2/michael:[EMAIL PROTECTED]/2000)  ;A is on
192.168.1.103
exten => 2000,2,congestion
include -> from-iax
 
[from-iax]
exten => s,1,Wait(2)
exten => s,2,Answer
exten => 3000,3,Dial(Zap/1,20)
 
NB: B's zapata.conf points to the internal
context
 
B's iax.conf
 
[general]
port=5036
bandwidth=high
disallow=lpc10  
tos=lowdelay
 
[chuks]
type=friend
secret=password
auth=plaintext
host=192.168.1.103
context=from-iax
allow=all
trunk=yes
 
 
At least I thought I'd hear A ring when I dial 2000 from B,
instead, I get the congestion (busy) tone. Can anyone tell me what I'm
doing wrong? If I can open up B's network connectios, I know I'll get
the same problem each way. 
 
thx,
chuks
[EMAIL PROTECTED]
 
 
 

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Re: [Asterisk-Users] Re: * > Mobile Phone > Mobile Network

[jurgen]

I'll second PaulH's recommendation of the Telular SX-5e units. They
plug into an FXO on the Asterisk machine (or wherever). Put anyone's
capped SIM into the thing, and you're communicating with the GSM
network. As a little added bonus, there's a serial port for sending
and receiving text messages. You can either send AT-commands to the
thing (it works like a modem), or get some software like SMS Server
Tools to do it for you.

There are also PRI-interface based units, but they're expensive and
really designed for high-end uses. Also expensive (but said to be
coming down in price soon) is a SIP-based box called the VoiceBlue.

jurgen


On Mon, 21 Feb 2005 15:43:43 +0300, AR Tarzi <[EMAIL PROTECTED]> wrote:
> I've used a Nokia 32 unattended (remote) for the past year or so.
> 
> "David Uzzell" <[EMAIL PROTECTED]> wrote in message
> news:[EMAIL PROTECTED]
> | Ok I have a question. Seen it come and go around the mailling list for a
> | while but never really seen an answer that seems to sort it out.
> |
> | What is needed is some interface from * > Mobile Phone > Mobile Network
> | Service.
> |
> | At this point all the providers in AUS that I have found are charging a
> | Premium Rate for Land Line > Mobile Network services.
> |
> | What I would like to do is be able to purchase a low rate Mobile SIM
> | that I can chuck into a Mobile Phone and have it setup so that I route
> | the Mobile calls through it.
> |
> | Rembering that most if not all mobile phones can be accessed via RS232
> | interface.
> |
> | Anyone done this or seen it done or know how to do it using * and whatever?
> |
> | Cheers
> | David
> | ___
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> 
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-- 
[EMAIL PROTECTED] is jurgen's gmail address.
Visit http://jurgen.ca/ for more yummy goodness.
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RE: [Asterisk-Users] bridging iaxtel calls to PSTN

[info]

Hello,
 actually I did, but nobody responded to that. So, here it is
one more time:
___
Hello,
 can someone tell me what's wrong with this? I can't make toll
free calls via iaxtel. Here's the definition in my extensions.conf
 
[iaxtel-trunks]
;
;outbound 1-700 and toll free calls go via iaxtel
;be sure to include the iaxtel-trunks context in dialing
context
;add function here to continue ring tone when 9 is dialed
;
ignorepat=>9
exten =>
_91700XXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]:1)
exten =>
_91888NXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]:1)
exten =>
_91877NXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]:1)
exten =>
_91866NXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]:1)
exten =>
_91800NXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]:1)

 
Note: IAXTEL_INFO is already defined as
username:password
 
and here's my iax.conf
 
[general]
port=5036
bandwidth=high
disallow=lpc10 
tos=lowdelay
 
;to register with iaxtel
register => username:[EMAIL PROTECTED]
 
 
;
; Trust Caller*ID Coming from iaxtel.com
;
[iaxtel]
type=friend
context=from-iaxtel
auth=cleartext
;inkeys=iaxtel
 
 
when i make an 800 number call for instance, registration goes
through and iaxtel can find me. But there is an endless silence, sort
of like an endles loop, and the only output I see is a "timeout on
Zap/1-1" and it tries the whole thing again...and goes on forever, and
the call never goes through. Is there anything wrong with my
configuration above?
 
thx,
Chuks.
 Original Message Subject: RE:
[Asterisk-Users] bridging iaxtel calls to PSTNFrom: "Rich Adamson"
<[EMAIL PROTECTED]>Date: Mon, February 21, 2005 3:16
pmTo: "Asterisk Users Mailing List - Non-Commercial
Discussion"I think
someone did reply to this. Don't bother trying to use iaxtel
foryour connections. Its down far more then its up.If you
want help setting up an iax connection directly between your
twosystems, then post what you've got that pertains to this from
iax.confand extensions.conf. No one is going to be able to help you
based on Adoesn't talk to B
words.> could you help me out
with this? I have a posting on this list, bu nobody has replied yet.
Titled "why can't I make IAX calls> between 2 asrterisk
servers"? I'd appreciate.>  > -chuks.>
>      Original Message >
    Subject: Re: [Asterisk-Users] bridging iaxtel calls to
PSTN>     From: "Michael Graves"
<[EMAIL PROTECTED]>>     Date: Mon, February 21,
2005 7:49 am>     To: "Asterisk Users Mailing List -
Non-Commercial Discussion">    
> >    
On Sun, 20 Feb 2005 02:43:46 -0700, [EMAIL PROTECTED]
wrote:> >     >Hello,>    
> I just started using asterisk, and have a question. I have
 setup two>     >asterisk servers, A and B. A
has a Digium TDM400 11B card (1 FXO and 1>     >FSX
modules) and is connected to the PSTN. B has same, but is NOT>
    >connected to PSTN. I want to configure B to call A
via iaxtel, and>     >connect to the PSTN using A's
line. How can I configure iaxtel dial>     >plan for
B in extensions.conf? I want to be able to make a call to>  
  >local US number (where A is located) from B, using iaxtel.
Can anyone>     >please help me? All I have seen so
far is just making calls from A to B>     >and vice
versa using the iaxtel 1700 number, but I haven't seen any>
    >examples of how to bridge the iaxtel calls to PSTN.
Help please.>     >>    
>>     >chuks.>    
>>     >NB: I don't mean toll free number, I
mean just local dialing.>     >> >
    Don't bother with IAXTel. It's very frequently down. Just
have each>     server register with the other and trunk
between them. That way you>     just use dialplan logic
to make the A place a call on B's resource. The>    
wiki has a good section on trunking between servers ova IAX.>
>     Michael> >    
-->     Michael Graves        
                 
[EMAIL PROTECTED]>     Sr. Product Specialist
                   
     www.pixelpower.com>     Pixel
Power Inc.                
               
[EMAIL PROTECTED]> >     o713-861-4005>
    o800-905-6412>     c713-201-1262>
>    
___>    
Asterisk-Users mailing list>    
Asterisk-Users@lists.digium.com>    
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[Asterisk-Users] IAX ATA's

[Tim Lewis]

Are their any good chooses for IAX Adapters?


-Thanks

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[Asterisk-Users] Call terminaison Tools

[Salomon Brevet]

What it is necessary to create a termination:
Asterisk server ?
card ?
type of network  ?
another software ?


Thank you.






Découvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails ! 
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Re: [Asterisk-Users] Suggestion for noise reduction on Asterisk-Users

[Rich Adamson]

The same has been proposed several times over the last nine months,
both on -users and -dev, and its simply been ignored.


  From: Kristian Kielhofner <[EMAIL PROTECTED]>
  Subject: [Asterisk-Users] Suggestion for noise reduction on Asterisk-Users
  Date: Mon, 21 Feb 2005 16:13:38 -0600 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 



> Hello all,
> 
>   This might be one for Digium, but I would like to see some type of Wiki 
> that people would have to wade through before they would get the 
> information on how to subscribe to the list.
> 
>   This wiki should cover most of the basic stuff that gets asked over and 
> over again just to help reduce the amount of repetition that most of you 
> have probably noticed takes place here.
> 
>   I understand that there is no good way to solve this problem, and 
> speaking of repetition, it has been covered before.  However, it seems 
> like list traffic has surged in the last several months and the HUGE 
> majority of it are questions that usually are answered with a link to 
> the voip-info.org wiki, because it has been asked before and someone 
> thought to add it.
> 
>   I think that this URL:
> 
> http://www.asterisk.org/index.php?menu=support
> 
>   Instead of saying "Blah, blah, blah and here are the lists", it should 
> say something like:
> 
> "...Digium is the corporate sponsor of Asterisk.  Support may be 
> purchased from us, and more information is available here...
> 
> Alternatively, free support may be obtained from the wonderful Asterisk 
> community.  The best starting point is the wiki at voip-info.org [link]. 
>   If your question cannot be answered there, Digium does provide mailing 
> lists where questions may be answered by members of the community on a 
> volunteer basis.  Please see [link] for more information."
> 
>   Where the second [link] would go to the wiki with clearly-stated posted 
> guidelines and some kind of whizzbang links that take care of a large 
> percentage of these typical newbie questions.
> 
>   I have nothing against newbies, but I know that I get tired of people 
> asking the same questions over and over, and sometimes I just don't feel 
> like saying the same thing that has been said over and over - "go to the 
> wiki first".  The problem with this is if we don't answer the question, 
> the person asking the question probably never solves his problem and 
> possibly ditches Asterisk altogether, or they just bang at it until they 
> either ditch Asterisk or waste a lot of time in the process.
> 
>   How many more times do we have to read posts of "I just downloaded the 
> Asterisk and now how do I talk on it?"
> 
>   Let me know what you think...
> 
> --
> Kristian Kielhofner
> ___
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RE: [Asterisk-Users] bridging iaxtel calls to PSTN

[Rich Adamson]

I think someone did reply to this. Don't bother trying to use iaxtel for
your connections. Its down far more then its up.

If you want help setting up an iax connection directly between your two
systems, then post what you've got that pertains to this from iax.conf
and extensions.conf. No one is going to be able to help you based on A
doesn't talk to B words.


> could you help me out with this? I have a posting on this list, bu nobody has 
> replied yet. 
Titled "why can't I make IAX calls
> between 2 asrterisk servers"? I'd appreciate.
>  
> -chuks.
> 
>  Original Message 
> Subject: Re: [Asterisk-Users] bridging iaxtel calls to PSTN
> From: "Michael Graves" <[EMAIL PROTECTED]>
> Date: Mon, February 21, 2005 7:49 am
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> 
> On Sun, 20 Feb 2005 02:43:46 -0700, [EMAIL PROTECTED] wrote:
> 
> >Hello,
> > I just started using asterisk, and have a question. I have  setup two
> >asterisk servers, A and B. A has a Digium TDM400 11B card (1 FXO and 1
> >FSX modules) and is connected to the PSTN. B has same, but is NOT
> >connected to PSTN. I want to configure B to call A via iaxtel, and
> >connect to the PSTN using A's line. How can I configure iaxtel dial
> >plan for B in extensions.conf? I want to be able to make a call to
> >local US number (where A is located) from B, using iaxtel. Can anyone
> >please help me? All I have seen so far is just making calls from A to B
> >and vice versa using the iaxtel 1700 number, but I haven't seen any
> >examples of how to bridge the iaxtel calls to PSTN. Help please.
> >
> >
> >chuks.
> >
> >NB: I don't mean toll free number, I mean just local dialing.
> >
> 
> Don't bother with IAXTel. It's very frequently down. Just have each
> server register with the other and trunk between them. That way you
> just use dialplan logic to make the A place a call on B's resource. The
> wiki has a good section on trunking between servers ova IAX.
> 
> Michael
> 
> --
> Michael Graves   [EMAIL PROTECTED]
> Sr. Product Specialist  www.pixelpower.com
> Pixel Power Inc. [EMAIL PROTECTED]
> 
> o713-861-4005
> o800-905-6412
> c713-201-1262
> 
> ___
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[Asterisk-Users] Suggestion for noise reduction on Asterisk-Users

[Kristian Kielhofner]

Hello all,
	This might be one for Digium, but I would like to see some type of Wiki 
that people would have to wade through before they would get the 
information on how to subscribe to the list.

	This wiki should cover most of the basic stuff that gets asked over and 
over again just to help reduce the amount of repetition that most of you 
have probably noticed takes place here.

	I understand that there is no good way to solve this problem, and 
speaking of repetition, it has been covered before.  However, it seems 
like list traffic has surged in the last several months and the HUGE 
majority of it are questions that usually are answered with a link to 
the voip-info.org wiki, because it has been asked before and someone 
thought to add it.

I think that this URL:
http://www.asterisk.org/index.php?menu=support
	Instead of saying "Blah, blah, blah and here are the lists", it should 
say something like:

"...Digium is the corporate sponsor of Asterisk.  Support may be 
purchased from us, and more information is available here...

Alternatively, free support may be obtained from the wonderful Asterisk 
community.  The best starting point is the wiki at voip-info.org [link]. 
 If your question cannot be answered there, Digium does provide mailing 
lists where questions may be answered by members of the community on a 
volunteer basis.  Please see [link] for more information."

	Where the second [link] would go to the wiki with clearly-stated posted 
guidelines and some kind of whizzbang links that take care of a large 
percentage of these typical newbie questions.

	I have nothing against newbies, but I know that I get tired of people 
asking the same questions over and over, and sometimes I just don't feel 
like saying the same thing that has been said over and over - "go to the 
wiki first".  The problem with this is if we don't answer the question, 
the person asking the question probably never solves his problem and 
possibly ditches Asterisk altogether, or they just bang at it until they 
either ditch Asterisk or waste a lot of time in the process.

	How many more times do we have to read posts of "I just downloaded the 
Asterisk and now how do I talk on it?"

Let me know what you think...
--
Kristian Kielhofner
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Re: [Asterisk-Users] SIP echo on LAN

[Michael George]

On Mon, Feb 21, 2005 at 08:42:33AM +, Julian J. M. wrote:
> Check your soundcard controls... maybe it's recording "what you hear"
> or PCM, thus sending it again to the other party.

Are you saying that when using a sound card with your softphone the PCM should
be set to 0?

I never knew that...

-- 
-M

There are 10 kinds of people in this world:
Those who can count in binary and those who cannot.
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RE: [Asterisk-Users] bridging iaxtel calls to PSTN

[info]

could you help me out with this? I have a posting on this list, bu
nobody has replied yet. Titled "why can't I make IAX calls between 2
asrterisk servers"? I'd appreciate.
 
-chuks.
 Original Message Subject: Re:
[Asterisk-Users] bridging iaxtel calls to PSTNFrom: "Michael
Graves" <[EMAIL PROTECTED]>Date: Mon, February 21, 2005 7:49
amTo: "Asterisk Users Mailing List - Non-Commercial
Discussion"On Sun,
20 Feb 2005 02:43:46 -0700, [EMAIL PROTECTED]
wrote:>Hello,> I just started using asterisk, and
have a question. I have  setup two>asterisk servers, A and
B. A has a Digium TDM400 11B card (1 FXO and 1>FSX modules) and
is connected to the PSTN. B has same, but is NOT>connected to
PSTN. I want to configure B to call A via iaxtel, and>connect to
the PSTN using A's line. How can I configure iaxtel dial>plan for
B in extensions.conf? I want to be able to make a call to>local
US number (where A is located) from B, using iaxtel. Can
anyone>please help me? All I have seen so far is just making
calls from A to B>and vice versa using the iaxtel 1700 number,
but I haven't seen any>examples of how to bridge the iaxtel
calls to PSTN. Help
please.>>>chuks.>>NB: I don't mean
toll free number, I mean just local dialing.>Don't
bother with IAXTel. It's very frequently down. Just have eachserver
register with the other and trunk between them. That way youjust use
dialplan logic to make the A place a call on B's resource. Thewiki
has a good section on trunking between servers ova
IAX.Michael--Michael Graves      
                   
[EMAIL PROTECTED]Sr. Product Specialist      
                 
 www.pixelpower.comPixel Power Inc.      
                   
     
[EMAIL PROTECTED]o713-861-4005o800-905-6412c713-201-1262___Asterisk-Users
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RE: [Asterisk-Users] Asterisk@home Linux has no KDE

[dean collins]

Kiran,
>From another pc on your network log into the web page by
http://ipaddress-of-the-asterisk-server/maint



Cheers,
Dean
p.s. RTFM


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kiran
Vahaja
Sent: Monday, February 21, 2005 4:05 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] [EMAIL PROTECTED] Linux has no KDE

Hi Folks,

I installed [EMAIL PROTECTED] on my PC. It went through the installation
and all. But now i get a command line login window. Doesn't it has a
KDE or some other type of OS GUI (i am not talking about [EMAIL PROTECTED]
web GUI)? After i login, just the command line interface comes out.
Any command to type here to get Linux OS GUI?

Thanks,
Kiran
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Re: [Asterisk-Users] Asterisk@home Linux has no KDE

[Kiran Vahaja]

Then how can i use web interface to configure?


On Mon, 21 Feb 2005 15:30:55 -0600, Kristian Kielhofner <[EMAIL PROTECTED]> 
wrote:
> Kiran Vahaja wrote:
> > Hi Folks,
> >
> > I installed [EMAIL PROTECTED] on my PC. It went through the installation
> > and all. But now i get a command line login window. Doesn't it has a
> > KDE or some other type of OS GUI (i am not talking about [EMAIL PROTECTED]
> > web GUI)? After i login, just the command line interface comes out.
> > Any command to type here to get Linux OS GUI?
> >
> > Thanks,
> > Kiran
> 
> It probably doesn't have a GUI because you shouldn't use X on an
> Asterisk server.  Asterisk does not play nice with X:
> 
> http://voip-info.org/wiki-Asterisk+X11
> 
> --
> Kristian Kielhofner
>
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Re: [Asterisk-Users] Asterisk H323 support

[Kuniyoshi Murata]

Hi,
(B
(BI'm using Asterisk-1.0.0 on Fedora Core 1
(B
(BDate: Mon, 21 Feb 2005 14:22:13 +0900 [zone:Tokyo],
(B[EMAIL PROTECTED] mentioned in msg:  that ...
(B
(B>   For channel asterisk-oh323-v0.6.5
(B>   need
(B>   openh323-Janus_patch4-src-tar.gz
(B>   pwlib-Janus_patch4-src-tar.gz
(B> 
(B
(BI tried this combination but openh323 fails in compiling (make clean, make
(Bopt).
(B
(BDate: Mon, 21 Feb 2005 00:20:39 -0800 (PST) [zone:-], [EMAIL PROTECTED]
(Bmentioned in msg:  that ...
(B
(B> with Openh323 - v1.12.2 and pwlib - v1.5.2 I use
(B> asterisk-oh323 v.0.6.3b and it works fine
(B
(BWhat version of Asterisk are you running? And on what os and distribution?
(B
(B--
(BKuniyoshi Murata.iChat/AIM:macwebcaster
(BEnglish-Japanese Interpreter mailto:[EMAIL PROTECTED]
(BMacintosh Webcast Specialisthttp://www.macwebcaster.com
(B
(B
(B
(B___
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Re: [Asterisk-Users] * Call Monitoring

[Daniel Corbe]

Yeah,  I'd be interested in porting your work so it runs under nagios.

Please post your results when you're finished.

-Daniel


On Tue, 22 Feb 2005 02:54:22 +1100, Adam Goryachev
<[EMAIL PROTECTED]> wrote:
> On Mon, 2005-02-21 at 08:00 -0500, Daniel Corbe wrote:
> > I've got a nagios plugin making sure the * box is up, but I would like
> > to do more than that.
> >
> > I need to make sure the PRIs connected to my box stay up and I need to
> > make sure calls are not failing for any reason.  Are there any *
> > monitoring packages like this?
> 
> Interesting you should ask this today...
> 
> I got to work this morning and was wondering why some of my calls were
> still diverting to my mobile.
> 
> Eventually I realised that they were diverting on no answer. A restart
> of asterisk, reload of modules etc made no differences, I couldn't do
> anything with the line. Eventually I worked out it was a telco problem
> (no dialtone/etc) so I logged the fault. I looked at zttool and it
> showed a red alarm... In around 10-20 minutes I hacked zttool.c and
> converted it into a very basic cli version (which doesn't need newt) and
> would just dump the current status of all the spans. Similar to what you
> see on screen when you first start zttool.
> 
> Then, I threw together some simple shell scripting to analyse/send the
> report to BigBrother (www.bb4.org). So far it is working nicely, by
> tomorrow night (yes, 27 hours after reporting it) hopefully my line
> should come back, and the alarm should change to OK...
> 
> I'll put the package etc onto www.deadcat.net (BB addons website) and
> drop a post here when it is done. Will also put it onto
> www.websitemanagers.com.au/asterisk/
> 
> BTW, I did need to suid the zttool-cli command to root, as the normal BB
> user doesn't have the needed permissions. I haven't looked into this,
> but if anyone has a suggestion on a better way to do this, feel free to
> let me know.
> 
> Regards,
> Adam
> 
> --
>  --
> Adam Goryachev
> Website Managers
> Ph:  +61 2 9345 4395[EMAIL PROTECTED]
> Fax: +61 2 9345 4396www.websitemanagers.com.au
> 
>
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[Asterisk-Users] FWD using IAX2

[Anton Krall]

Guys
 
Ive setup FWD using IAX according to all the docs and I tried the "give me a
call" url on FWD webpage and I do get the call but when asked to say my
name, I hear a voice saying it didnt get it.. seems my voice is not getting
thru to FWD... anybody had this problem while setting up FWD with IAX2?
 
 
 
__
Anton Krall
 

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Re: [Asterisk-Users] compiling cvs-head today?

[Rich Adamson]

> On Mon, 2005-02-21 at 12:11 -0600, Rich Adamson wrote:
> > > On Mon, 2005-02-21 at 06:36 -0600, Rich Adamson wrote:
> > > > Anyone having problems compiling the current cvs head this morning?
> > > > 
> > > > New cvs checkout on RH9, followed by appropriate make clean and make
> > > > install. System was running cvs head from Nov 23 with TDM card, PRI,
> > > > SIP phones on local wire, and IAX.
> > > 
> > > 
> > > > See http://bugzilla.redhat.com/bugzilla/> for instructions.
> > > > The bug is not reproduceable, so it is likely a hardware or OS problem
> > > > make: *** [say.o] Error 1
> > > 
> > > 
> > > > hash/hash.c:243: internal error: Segmentation fault
> > > > Please submit a full bug report,
> > > > with preprocessed source if appropriate.
> > > > See http://bugzilla.redhat.com/bugzilla/> for instructions.
> > > 
> > > Looks like a hardware problem as you had failures in different locations
> > > but both where a gcc seg fault. This means either your CPU is hot and
> > > starting to spit out randomness or your memory is failing and producing
> > > randomness. Could be something else like low power supply and therefor
> > > faulty writing/reading of data to/from memory. 
> > > 
> > > Any way around it looks like you are in for either a while of debugging
> > > hardware or a hardware replacement regiment.
> > 
> > Okay... this one is at a site 50 miles away where they are off on
> > holiday today. Guess I'll wait for someone to show up. ;)
> 
> If they are gone for holiday, it very well could be heat related. Try
> your compiles a few hours after they get into the office and see if the
> heat levels have changed.

This one is located in a data center with a fair air handler in place,
so more likely its a mem or power supply issue.

Rich


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Re: [ nocadm ] Re: [Asterisk-Users] HDLC Bad FCS / HDLC Abort

[Eric Wieling]

Alex G Robertson wrote:
Eric Wieling wrote:
 > Yes.  There are lots of messages in the mailing list archives regarding
 > this problem, some of them even include things to try.  You didn't see
 > these messages when you searched the mailing list archives?
 >
Yes, I've read then.
They say it can be caused by interruptions.
Did you confirm you are not running graphics?  (X, frame buffer, etc). 
 Did you confirm you have unmasked IDE interrupts (-u1 to haparm)?

--Eric
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[Asterisk-Users] why can't I make toll free calls via IAXTEL

[info]

Hello,
 can someone tell me what's wrong with this? I can't make toll
free calls via iaxtel. Here's the definition in my extensions.conf
 
[iaxtel-trunks]
;
;outbound 1-700 and toll free calls go via iaxtel
;be sure to include the iaxtel-trunks context in dialing context
;add function here to continue ring tone when 9 is dialed
;
ignorepat=>9
exten =>
_91700XXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]:1)
exten =>
_91888NXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]:1)
exten =>
_91877NXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]:1)
exten =>
_91866NXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]:1)
exten =>
_91800NXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]:1)

 
Note: IAXTEL_INFO is already defined as username:password
 
and here's my iax.conf
 
[general]
port=5036
bandwidth=high
disallow=lpc10 
tos=lowdelay
 
;to register with iaxtel
register => username:[EMAIL PROTECTED]
 
 
;
; Trust Caller*ID Coming from iaxtel.com
;
[iaxtel]
type=friend
context=from-iaxtel
auth=cleartext
;inkeys=iaxtel
 
 
when i make an 800 number call for instance, registration goes
through and iaxtel can find me. But there is an endless silence, sort
of like an endles loop, and the only output I see is a "timeout on
Zap/1-1" and it tries the whole thing again...and goes on forever, and
the call never goes through. Is there anything wrong with my
configuration above?
 
thx,
Chuks.

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Re: [Asterisk-Users] Asterisk@home Linux has no KDE

[Kristian Kielhofner]

Kiran Vahaja wrote:
Hi Folks,
I installed [EMAIL PROTECTED] on my PC. It went through the installation
and all. But now i get a command line login window. Doesn't it has a
KDE or some other type of OS GUI (i am not talking about [EMAIL PROTECTED]
web GUI)? After i login, just the command line interface comes out.
Any command to type here to get Linux OS GUI?
Thanks,
Kiran
It probably doesn't have a GUI because you shouldn't use X on an 
Asterisk server.  Asterisk does not play nice with X:

http://voip-info.org/wiki-Asterisk+X11
--
Kristian Kielhofner
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Re: [ nocadm ] Re: [Asterisk-Users] HDLC Bad FCS / HDLC Abort

[Alex G Robertson]

Sergey Kuznetsov wrote:
This is happens because of imperfect HDLC code. 
Do you mean the software? The source code?
[]s
--
Alex G Robertson
NOC - Microlink
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RE: [Asterisk-Users] compiling cvs-head today?

[David Brodbeck]

> -Original Message-
> From: Steven Critchfield [mailto:[EMAIL PROTECTED]

> Looks like a hardware problem as you had failures in 
> different locations
> but both where a gcc seg fault. This means either your CPU is hot and
> starting to spit out randomness or your memory is failing and 
> producing
> randomness. Could be something else like low power supply and therefor
> faulty writing/reading of data to/from memory. 
> 
> Any way around it looks like you are in for either a while of 
> debugging
> hardware or a hardware replacement regiment.

The first thing I usually do in these situations (after making sure the
machine's fans are all running and dust-free) is run MEMTEST-86.
http://www.memtest86.com/  It's not foolproof, but in my experience it
catches more memory problems than any other utility.
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Re: [Asterisk-Users] Queue Question

[Peter Svensson]

On Mon, 21 Feb 2005, Shaun Tierney wrote:

> Is there a way to prioritize calls in multiple queues based on hold time?  I
> have three queues set up on my Asterisk PBX with agents logged into all
> three queues.  I've noticed that sometimes calls in one queue will make it
> through in a couple minutes while another queue will be backed up with
> people having been on hold for 30+ minutes.  Is it possibly the fact that I
> am set for the rrmemory ring strategy?

We have solved the same problem by creating a single queue that all the 
calls are dumped in. Callers and agents are matched through various 
criteras. This treats all callers alike - the longers caller in the queue 
gets first dibs on any agents that become available if there is a match. 

Our queues are based on the "icd" queue system for Asterisk. I expect
we will submit the changes to the icd system to the icd team once we have 
cleaned up the implementation a bit. 

Peter


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[Asterisk-Users] South Korea DID wanted

[Justin Richards]

Sorry for the cross post, but I'm still trying to find a Seoul DID.  I
received an email from LiveVoip.com that said they have service in
South Korea, but when I called them they said they didn't offer such
service.

If you have the capability to offer a DID please let me know what your
pricing structure is.  This would be fairly low personal usage,
probably around 400 minutes a month.

Thank you!

Justin
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[Asterisk-Users] ZAP FXS vs ethernet FXS

[Jon Gabrielson]

What are the advantages/disadvantages of using
a ZAP FXS port versus using one of the many 
small ethernet FXS devices on the market.  The
ZAP FXS talks directly to asterisk over PCI.  Is this 
an advantage?  The ethernet devices I assume
speak either iax2 or sip, does this cripple the 
functionality of the attached FXS device for things
like callwaiting,callerid,distinctive ring, etc...

Does anyone have experience with both types
of devices and would recommend one over the 
other?


Thanks,


Jon.
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[Asterisk-Users] why can't I make toll free calls via IAXTEL

[info]

Hello,
 can someone tell me what's wrong with this? I can't make toll
free calls via iaxtel. Here's the definition in my extensions.conf
 
[iaxtel-trunks]
;
;outbound 1-700 and toll free calls go via iaxtel
;be sure to include the iaxtel-trunks context in dialing
context
;add function here to continue ring tone when 9 is dialed
;
ignorepat=>9
exten =>
_91700XXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]:1)
exten =>
_91888NXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]:1)
exten =>
_91877NXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]:1)
exten =>
_91866NXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]:1)
exten => _91800NXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]:1)

 
Note: IAXTEL_INFO is already defined as
username:password
 
and here's my iax.conf
 
[general]
port=5036
bandwidth=high
disallow=lpc10 
tos=lowdelay
 
;to register with iaxtel
register => username:[EMAIL PROTECTED]
 
 
;
; Trust Caller*ID Coming from iaxtel.com
;
[iaxtel]
type=friend
context=from-iaxtel
auth=cleartext
;inkeys=iaxtel
 
 
when i make an 800 number call for instance, registration goes
through and iaxtel can find me. But there is an endless silence, sort
of like an endles loop, and the only output I see is a "timeout on
Zap/1-1" and it tries the whole thing again...and goes on forever, and
the call never goes through. Is there anything wrong with my
configuration above?
 
thx,
Chuks.

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