Re: [Asterisk-Users] TE410P and SPANDSP

[Ma Zhiyong]

It looks like spandsp doesn't fit a busy fax server solution, usually this kind 
of solution is depending on some onborad dsp  card like eicon diva server.
But it still a good tool for faxing, and we can expect its t.38 function. Maybe 
a big surprise.


I have some experience using the TE410p and spandsp.  I think my max 
concurrent rxfax has been about 16 or so on a single E1 on a single cpu 
3.0Ghz P4.  I expect that it could handle at least 30 concurrent faxes, 
thereafter I think that the disk controller may start to impact performance. 
My experience of txfax is that it is very sensitive to system load.  When I 
was testing things I was able to get no more than three concurrent instances 
of txfax going on a system that was otherwise idle.  If I was receiving a 
fax at all via rxfax then I could reliably have no more than one instance of 
txfax.

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

Sent: Thursday, November 24, 2005 9:21 AM
Subject: [Asterisk-Users] TE410P and SPANDSP


> Hi, All
>   Does any one has successful experience use te410p and spandsp together?
>   Could they work well with all 120 channels receive/send fax at the same 
> time?
>
>   My practice is that rxfax always get broken fax page.
>
>   Help!





> ___
> --Bandwidth and Colocation sponsored by Easynews.com --
>
> 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 

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

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

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

Re: [Asterisk-Users] Asterisk as Softswitch

[Olle E. Johansson]

Somesh S Shanbhag wrote:
> Dear All,
> 
> Can I use Asterisk IP-PBX as Softswitch? If not, what
> is lacking in asterisk
> from not *becoming* softswitch?
> 
What is your definition of a softswitch?

/O
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Asterisk SIP architecture question

[Olle E. Johansson]

Matt Riddell wrote:
> Kevin P. Fleming wrote:
> 
>>Matt Riddell wrote:
>>
>>
>>>So how does Asterisk know that the media stream has been disconnected
>>>between
>>>the two remote hosts?
>>
>>It doesn't... nor does any other SIP softswitch. See my other reply for
>>a possible solution.
> 
> 
> I agree that you could code a fix, but saying my advice is bogus because you
> could code a fix for Asterisk to avoid it is slightly wrong.
> 
> The fact remains, if you need *very* accurate cdr's then you either don't do
> canreinvite=yes for the peer or you code something so that Asterisk notices
> that the rtp has stopped.  The fact remains that without these, the most
> accurate CDR is going to come from the provider.
> 

If the audio goes through asterisk without re-invites, you could use the
rtptimeouts to detect a dead phone.

/O
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Looking for Windows based Asterisk

[Stefan-Michael. Guenther (in-put GbR)]

Hi,

>Does anyone know of a Asterisk Manager Interface client application that can 
>run from a Windows XP machine to manage Asterisk installed on a Linux 
>Machine.
>
if you consider the IE to be a client application, you could use the Asterisk 
PBX Manager from Thirdlane (www.thirdlane.com).

Bye,

Stefan

-- 


in-put GbR - Das Linux-Systemhaus
Stefan-Michael Guenther
Moltkestrasse 49 D-76133 Karlsruhe
Tel./Fax : +49 (0)721 / 83044 - 98/93
http://www.in-put.de

 Schulungen  Installationen  
 Beratung   Support
  Voice-over-IP-Lösungen


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

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

Re: [Asterisk-Users] Asterisk DNS SRV lookups

[Olle E. Johansson]

David Thomas wrote:
> Does asterisk fully support DNS SRV lookups yet, or does it still only
> read the first SRV entry?
> Info on the wiki looked quite old, so I thought I better ask.
It's still accurate. We do not support DNS SRV lookups fully yet.

/O
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Invite with Replaces

[Olle E. Johansson]

< Arnaud > wrote:
> Does Asterisk 1.2 support INVITE with Replaces header (rfc 3891) ?
No.
But 1.3 will.

/O
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

RE: [Asterisk-Users] zaptel 1.2.0 on (Tettnang)

[Kong]

old zaptel 1.0 install nicely, how come the new one will have problems?

At 03:09 PM 11/24/2005, you wrote:

Check the location specified in the kernel Makefile, and validate that is
installs the modules to the propler /usr/lib/modules/bla blabla directory.

Nir S

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kong
Sent: Thursday, November 24, 2005 9:01 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] zaptel 1.2.0 on (Tettnang)

hi, i was able to do a "make linux26" without problem on my FC2 machine.

but when i tried a "make install" nothing has been install.

i had another machine running FC4, doens't have this problem. any ideas?

thank you.

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

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

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

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


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

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

Re: [Asterisk-Users] Asterisk 1.2 + Debian Sarge

[Dulmandakh Sukhbaatar]

Tzafrir Cohen wrote:


On Tue, Nov 22, 2005 at 02:00:11PM -0700, Matt wrote:
 

Looks like you need to install the kernel headers  package. While you 
are at it be sure that you have the kernel source package installed also.
   



 apt-get install kernel-headers-`uname -r`

should suffice.

 


my $0.02

Juanjo Portela wrote:

   


Dear Collegues

I am trying to compile the new version (Asterisk.1.2) with my debian
box and i get the following error when i compile the zaptel package:

radio:/usr/src/asterisk-1.2/zaptel-1.2.0# make
make: Warning: File `Makefile' has modification time 3.1e+08 s in the 
future
 



Hmmm... better setup your clock properly

 


And you should create  symlink to your kernel header
ln -s kernel-headers-`uname -r` linux

I had no problem with compilation.
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Re: Asterisk SIP architecture question

[Olle E. Johansson]

Kevin P. Fleming wrote:
> David Thomas wrote:
> 
>> Is the CDR accounting done based on SIP signaling? If a UA is talking
>> (RTP) to a third party PSTN gateway, isn't it at risk if say the UA
>> loses power. How will asterisk know the call has ended if it is not
>> involved in the media path. The idea is this.. I want to use
>> canreinvite =yes to force users to talk end-to-end to preserve
>> bandwidth, but I can see the potential for hung calls if asterisk
>> never get the BYE from a UA in the event the ATA gets unplugged or
>> somehow loses power.
> 
> 
> That is the case in every SIP network, Asterisk is not unique in that
> regard.
> 
> I would suggest that you could make a modification to chan_sip so that
> if the peer goes 'unreachable' (as determined by using qualify=yes) than
> any existing calls involved with that peer are immediately hung up; that
> would take care of this problem.


That would be a good addition. Optional of course.

/O
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Asterisk SIP architecture question

[Olle E. Johansson]

Kevin P. Fleming wrote:
> Matt Riddell wrote:
> 
>> So how does Asterisk know that the media stream has been disconnected
>> between
>> the two remote hosts?
> 
> 
> It doesn't... nor does any other SIP softswitch. See my other reply for
> a possible solution.

...or implement the SIP timer extension.

/O
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

RE: [Asterisk-Users] zaptel 1.2.0 on (Tettnang)

[Nir Simionovich - CTO]

Check the location specified in the kernel Makefile, and validate that is
installs the modules to the propler /usr/lib/modules/bla blabla directory.

Nir S 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kong
Sent: Thursday, November 24, 2005 9:01 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] zaptel 1.2.0 on (Tettnang)

hi, i was able to do a "make linux26" without problem on my FC2 machine.

but when i tried a "make install" nothing has been install.

i had another machine running FC4, doens't have this problem. any ideas?

thank you. 

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

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

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

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

[Asterisk-Users] Voicemail email format, please help!

[Ryan Pagquil]

Hi,


I'm now using Asterisk for my voicemail together with SER. 
They just work fine. When the user in SER is not registered the 
call will be forwarded to Asterisk and the caller will record his 
message. Then I also made asterisk to send the wav as attachment to 
its email. I try using two ip phones one is Xlite and the other is 
a hardware ip phone to call the voicemail. When asterisk sent the 
mail to me I found that the voicemail from the hardware ip phone 
has the display username and the number of the caller, but the 
Xlite voicemail only has the display username... then I checked the 
voicemail box of my username and check the message text that 
corresponds to the voicemail and found these:



hardware ip phone:



[message] origmailbox=810020

context=ser
macrocontext=
exten=u810020
priority=1
callerchan=SIP/mydomain.com-0018b368
callerid="test3" <103>
origdate=Thu Nov 17 11:22:09 AM GMT 2005
origtime=1132226529
duration=31

xlite:
[message] origmailbox=810020
context=ser
macrocontext=
exten=u810020
priority=1
callerchan=SIP/810020-e30c
callerid="810020"
origdate=Thu Nov 17 11:21:38 AM GMT 2005
origtime=1132226498
duration=31

How come does the hardware phone has the 103 on the callerid and 
xlite don't have its number? Is this a misconfiguration?


Thanks,

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

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

[Asterisk-Users] zaptel 1.2.0 on (Tettnang)

[Kong]

hi, i was able to do a "make linux26" without problem on my FC2 machine.

but when i tried a "make install" nothing has been install.

i had another machine running FC4, doens't have this problem. any ideas?

thank you. 


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

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

[Asterisk-Users] MTP Requirements for getting * talking to CCM for Voicemail

[Nathan Reeves]

Okay, finally run a server up with asterisk 1.2 and started work on getting CCM 4.1 talking to it to try and investigate the use of * for voicemail and possibly meetme conferences.  Using the notes on the voip wiki, I've managed to get it to a point where I can call * successfully and get into and leave a message in voicemail.  I've hit one issue with CheckMailboxExists not working quite right, but I'll look at that later.

 
My biggest issue at the moment is that I can't get external calls coming into CCM to be able to successfully get through to * without hanging up.  If I do a debug sip I can see that CCM sends a BYE message to * after the initial invite.  After running a trace on CCM I can see I get an error about an MTP resource failure (resource unavailable).  Looking at the config on the SIP Trunk I'd setup, I realise I didn't have an MTP included in the MGRL I'd placed the trunk in.  I've since amended this config but need to restart my devices to complete this.

 
I know MTP's being required was mentioned a while back.  Just wondering if anyone found a workaround for this, or if they just ended up including an MTP in the MGRL for the SIP Trunk.
 
Any suggestions gratefully accepted at this point.
 
Thanks
 
Nathan
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] hello

[Justin Tunney]

How are you gentlemen.

All your VoIP are belong to us.

Signed,
China
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] TE410P and SPANDSP

[Craig Guy]

I have some experience using the TE410p and spandsp.  I think my max 
concurrent rxfax has been about 16 or so on a single E1 on a single cpu 
3.0Ghz P4.  I expect that it could handle at least 30 concurrent faxes, 
thereafter I think that the disk controller may start to impact performance. 
My experience of txfax is that it is very sensitive to system load.  When I 
was testing things I was able to get no more than three concurrent instances 
of txfax going on a system that was otherwise idle.  If I was receiving a 
fax at all via rxfax then I could reliably have no more than one instance of 
txfax.


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


Sent: Thursday, November 24, 2005 9:21 AM
Subject: [Asterisk-Users] TE410P and SPANDSP



Hi, All
  Does any one has successful experience use te410p and spandsp together?
  Could they work well with all 120 channels receive/send fax at the same 
time?


  My practice is that rxfax always get broken fax page.

  Help!







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

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 


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

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

Re: [Asterisk-Users] hello

[pdhales]

> harry gaillac wrote:
> > hello
> 
> Hi there! How are you today?

Very well, thank you.

PaulH
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Loss of Registration for SIP Trunks

[Jerry Jones]

On Nov 24, 2005, at 12:06 AM, Scott Clements wrote:


HI List,

You'll have to pardon the newbieness of this question, I was  
editing the sip.conf file on my asterisk server yesterday, and now  
none of my asterisk trunks will connect. From my knowledge sip.conf  
does not effect registration, but there have been no other changes  
at all. Below is my sip.conf, and some other CLI info. If anone has  
some thoughts please let me know.



[general]

port = 5060   ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
disallow=all
allow=g729
allow=ulaw
allow=alaw
context=from-pstn
;context = from-sip-external ; Send unknown SIP callers to this  
context

callerid = Unknown
;dtmfmode=rfc2833
;relaxdtmf=yes

#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf




cee*CLI> sip show registry
HostUsername   Refresh State
This shows other servers to which asterisk has registered. I see no  
register statements in your sip.conf above.



cee*CLI> sip show peers
Name/usernameHostDyn Nat ACL Mask  
Port Status
sip-out-test/02  202.177.222.24  255.255.255.255   
5060 Unmonitored
127/127  (Unspecified)D  255.255.255.255   
0Unmonitored
126/126  (Unspecified)D  255.255.255.255   
0Unmonitored
These are sip clients registered to your asterisk server. I see no  
users listed in your sip.conf above, though I guess they are in your  
include files. I also looks like user sip-out-test has a hardcoded IP  
and is not set to dynamic so cannot really tell if it is registered  
or not from this info. Users 127 and 126 are not registered. None  
have a qualify to verify connectivity.


perhaps restoring to your previous config and editing more slowly  
will show where things broke:)







I have tried removing the trunks, confirmed the username and  
passwords for the trunks are ok. I am totally stumped as to what  
would cause it.


If anyone can help it'd be great :)

SCott
___
--Bandwidth and Colocation sponsored by Easynews.com --

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


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

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

RE: [Asterisk-Users] Eicon Diva Server query

[gw]

I would go with chan-capi-cm, as well as loading up the eicon drivers
first for the base drivers and utility set.

I have a few installations as such that are working flawlessly, and
Armin has done great work on the driver.

Regards,
Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Avi Miller
Sent: Wednesday, November 23, 2005 3:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Eicon Diva Server query

David Waugh wrote:
> Yes, you can use the Eicon Diva Range with 2.6 Kernels

Another question, considering the card should arrive tomorrow and I'd
like to try my hand at setting it up this weekend: Do I need to BRIstuff
Asterisk to get the Eicon Diva V-4BRI to work, or should I just need
chan_capi-cm?

Thanks,
Avi

--
National Manager - Special Projects

< Sydney / Melbourne / Canberra / Hobart / London />
   Walter Turnbull Bldg   T: +61 (0) 2 6233 0607
   44 Sydney Ave, F: +61 (0) 2 6233 0696
   Forrest,   W: http://www.squiz.net/
   ACT 2603

.>> Open Source  - Own it  -  Squiz.net ./>
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

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

RE: [Asterisk-Users] Virtual Modems Revisited

[Don Fanning]

Ah... Well I was sort of thinking more along the lines of trying to get
this to work into IAX or SIP.  But if you know for sure that the
modulation is broken... 

Just imagine... You'd be able to have a modem bank and save thousands of
dollars in leasing/purchasing a modem bank.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Underwood
Sent: Wednesday, November 23, 2005 6:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Virtual Modems Revisited

Don Fanning wrote:

>I brought this up a while back and althought there are pieces that 
>interface * into Fax Telephony applications, there hasn't been 
>something that works with plain old analog modems.
>
>Then I found this piece of code.  From my initial tests it looks solid,

>but I have no clue in how to interface this into asterisk.  I thought I

>would put this link up for other people to comment and try.
>
>http://fabrice.bellard.free.fr/linmodem.html
>
>Out of the box it works with soundcards.  I've been battling jack and 
>alsa for a week trying to get them to play nice just to reroute the 
>audio but I'm out of time in this regard.  So I thought I would toss it

>up and see what other people can come up with.
>
>Happy Holidays!
>Don
>  
>
Linmodem doesn't work out the box with anything. linmodem was abandoned
by its author at a very early stage, before any of its component parts
really worked.

It has a number of useful bits, which might be used as the basis for a
modem. It does not have a properly working code for any of the modem
standards.

I think Fabrice got busy, and with patent issues preventing wide
deployment of a V.34 modem finished the software just seemed like a
waste of time to him. He is one of the good guys of free DSP, and has
since produced several valuable things which are complete.

Steve

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

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


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

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

[Asterisk-Users] Asterisk as Softswitch

[Somesh S Shanbhag]

Dear All,

Can I use Asterisk IP-PBX as Softswitch? If not, what
is lacking in asterisk
from not *becoming* softswitch?

Thanks

Regards,
Somesh S. Shanbhag




__ 
Yahoo! Music Unlimited 
Access over 1 million songs. Try it free. 
http://music.yahoo.com/unlimited/
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Asterisk SIP architecture question

[Matt Riddell]

Kevin P. Fleming wrote:

> However, in the general case of not being concerned so much about the
> peer going away and losing CDR information for _one_ call, using
> reinvites does _not_ impact the quality of the softswitch's (Asterisk)
> CDRs.

Agreed.

-- 
Cheers,

Matt Riddell
___

http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://freevoip.gedameurope.com (Free Asterisk Voip Community)
http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)

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

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

Re: [Asterisk-Users] hello

[Matt Riddell]

harry gaillac wrote:
> hello

Hi there! How are you today?

-- 
Cheers,

Matt Riddell
___

http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://freevoip.gedameurope.com (Free Asterisk Voip Community)
http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)

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

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

Re: [Asterisk-Users] Cisco Callmanager & Asterisk for Voicemail revisited

[Guillermo Salas M]

El lun, 19-09-2005 a las 14:28 +1000, Shaun Ewing escribió:
> Some of you may remember back in May the thread on using Asterisk as a
> voicemail server for a Cisco Callmanager system.
> 

I'm looking for the same feature, please let me know where can I find
more resources and info about it.

Thank you very much.

> My own Callmanager system is integrated into an Asterisk server for
> voicemail (and other things). Back in May I was using H323 for
> integration, but since I've upgraded to CCM 4.1 I have switched over
> to SIP.
> 
> The integration with H323 required using Call forwarding to send the
> call to an extension on Asterisk. For example, extension 7443 would
> forward to 27443 on Asterisk which looked something like:
> exten => _27XXX,1,Voicemail(u${EXTEN:1})
> 
> Obviously setting this for each and every phone on Callmanager was not
> an option for any wide deployments, and Paul Davidson investigated
> some of the other options. Paul discovered that it was possible to
> setup a voicemail pilot, tick the voicemail box, etc. but you would
> lose the ability to have the caller ID information added to the
> voicemail.
> 
> This wasn't an option for us, as caller ID is quite important.
> 
> Up until now, I have continued with the custom extension option,
> setting up the appropriate call forwarding when new phones were added
> to the system. The trunk between CCM and Asterisk changed to SIP after
> the CCM upgrade, but everything else stayed the same until I revisited
> this today.
> 
> To summarise what I have accomplished:
> 
> Full voicemail integration between CCM and Asterisk with the following 
> features:
> - MWI
> - Voicemail on the CCM side is enabled by selecting Forward to
> 'Voicemail' rather than yucky custom extensions. Allows for wide
> deployment.
> - Messages are accessed by pressing the 'Messages' button on the CCM
> phones, or dialing the VM pilot number.
> - If a CCM user doesn't want to take a call, they can press the
> iDivert softkey to send to voicemail immediately.
> - CCM users can forward all calls to voicemail in the ccmuser pages,
> or by pressing CFwdAll and entering the pilot number or messages.
> - All the standard Asterisk voicemail features work just fine, eg: vm to 
> email.
> - more
> 
> Bugs with the setup:
> - If there's a SIP device registered with the Asterisk machine
> handling the voicemail, and the call path is something like: Sip
> Device -> Asterisk -> CCM. If the call subsequently reaches voicemail,
> Asterisk prompts for proxy authentication and CCM drops the call. This
> problem can be avoided by using usernames that don't match the caller
> id, eg: [sip7345], or having a machine dedicated to Voicemail.
> - That's all I've found
> 
> These options have been tested with Asterisk 1.0.8 and CCM 4.1(2)sr1.
> If this is something that people would be interested (and you made it
> this far), I'd be quite happy to whip up some instructions and add it
> to the wiki.
> 
> -Shaun
> ___
> --Bandwidth and Colocation sponsored by Easynews.com --
> 
> 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
-- 
Guillermo Salas M.
Telconet S.A. Manta
Calle 15 y Av. 24 Esq.
Phone : 593 5 262 8071
Mobile: 593 9 985 5138
SIP   : [EMAIL PROTECTED]
e-mail: [EMAIL PROTECTED]
www   : http://www.telconet.net
http://www.telcocarrier.net

Linux User: 255902

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

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

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

RE : [Asterisk-Users] In France asterisk never detect hang up. Why ?

[f6hqz-m]

Hello everybody  :-)

This are my first line french zapata.conf settings.
I have 3 like this, with only rx/tx gain a little bit different levels.
Running well.
Best Regards,
Francois BERGERET,
France.

usecallerid=yes
hidecallerid=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
rxgain=2
txgain=6
group=1
callgroup=1
pickupgroup=1
immediate=no
busydetect=yes
busycount=3
busypattern=500,500
signalling = fxs_ks
channel => 1

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de asterisk user
dupont
Envoyé : vendredi 18 novembre 2005 13:33
À : asterisk-users@lists.digium.com
Objet : [Asterisk-Users] In France asterisk never detect hang up. Why ?


Hello.

I am sorry my english is not good at all.

When i have a call from a fxo port of a tdm400p, asterisk waits one minute
before detecting that the caller has hang up his phone.

I have in my extension conf :
answer
background  (the prompt is 40 second long)
dial (on fxs port)  confgured for 30 seconds ringing.

if the caller hang up at the begining of the background prompt, asterisk
waits until he make ring the phone on the dial command for the all 30
secondes before detecting the hang up.

Do you know if there is a way to repair that ?

here is what i see on asterisk when the caller hang up IMMEDITALY after the
test prompt begins :

*CLI> -- Starting simple switch on 'Zap/4-1'
-- Executing Answer("Zap/4-1", "") in new stack
-- Executing NoOp("Zap/4-1", "0675458745") in new stack
-- Executing Set("Zap/4-1", "TIMEOUT(response)=20") in new stack
-- Response timeout set to 20
-- Executing BackGround("Zap/4-1", "barge") in new stack
-- Playing 'test' (language 'fr')
-- Executing Dial("Zap/4-1", "Zap/2|0675458745|30") in new stack
-- Called 2
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 answered Zap/4-1
-- Attempting native bridge of Zap/4-1 and Zap/2-1
-- Hungup 'Zap/2-1'
  == Spawn extension (reseau, s, 5) exited non-zero on 'Zap/4-1'
-- Executing Hangup("Zap/4-1", "") in new stack
  == Spawn extension (reseau, h, 1) exited non-zero on 'Zap/4-1'
-- Hungup 'Zap/4-1'


In my zapata.conf i have :

language=fr
default=fr
relaxdtmf=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
cidsignalling=v23
usecallerid=yes
group => 1
context=reseau
signalling=fxs_ks
callprogress=yes
busydetect=yes
callerid=asreceived
busycount=5
pulse=yes

In my zaptel.conf i have :

loadzone=fr
defaultzone=fr
fxoks=1-3
fxsks=4


If anyone can see what is wrong he will really help me.

thank you.
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

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

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

Re: [Asterisk-Users] ver1.2 installation problem

[Bartosz Wegrzyn - asterisk]

Thanks it helped

> Bartosz Wegrzyn - asterisk wrote:
>> I have fedora core 2 on this box.
>> I updated to the latest kernel but same problem.
>> My kernel is 2.6.10-1.771_FC2
>
> I had this problem on Ubuntu 5.04 when I checked out the 1-2 and 1-2-0
> cvs tags, just downloading the tar.gz from the asterisk site worked
> fine, however.
>
> cvs head which I was previously using also worked fine.
>
> --
> iseek computing - The Linux experts
> Phone: 08 8211 7722, Fax: 08 8211 7388, Web: http://www.iseek.biz
> ___
> --Bandwidth and Colocation sponsored by Easynews.com --
>
> 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
>

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

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

[Asterisk-Users] Querry about the modem

[Kunhikrishnan, Salil Geethanjaly (STSD)]

Hello 

Sorry to tell you that I am resending this mail because didn't get a 
reply for this query.

Salil

Hello

I have seen the article in digium site about the answering machine made 
using a softmodem and the zap library. I am using Fedora Core 2/3 system for 
doing this project. I was trying to find a PCI modem card with Intel 537 
chipset. I couldn't find any model with intel 537 chipset. Can any one please 
get me some insight into which model I can go for. Available models here in 
India are, 

Krypton
Dlink
Intex
Aztec
...

Do any of the above modem have this chipset. Or can I use these models 
for this purpose. 


Salil G. K.
kpfleming at  
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] Codec negotiation (not the same old stuff)

[Dan Austin]

Title: Codec negotiation (not the same old stuff)






I have a H.323 device, let's call it stupid, that supports all variants of G.729.  That should be

good, but no.  When it negotiates a call between Asterisk and a phone that supports all

varients of G.729, it gets it wrong.  Asterisk sends G.729A and the phone sends G.729,

at which point the device transcodes the call, but changes the packetization.


I'm looking for a way to get Asterisk to offer plain old G.729, so the device (stupid) can

just bridge the RTP.  My research shows this should not be required, since G.729 and

G.729A are compatible and don't need transcoding.


I'm trying to get an answer from the device manufacturer, but I do not have high hopes.


If it matters, I am using chan_ooh323 with Asterisk 1.2.0.


Thanks,

Dan




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

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

Re: [Asterisk-Users] ver1.2 installation problem

[Simon Lindsay]

Bartosz Wegrzyn - asterisk wrote:

I have fedora core 2 on this box.
I updated to the latest kernel but same problem.
My kernel is 2.6.10-1.771_FC2


I had this problem on Ubuntu 5.04 when I checked out the 1-2 and 1-2-0 
cvs tags, just downloading the tar.gz from the asterisk site worked 
fine, however.


cvs head which I was previously using also worked fine.

--
iseek computing - The Linux experts
Phone: 08 8211 7722, Fax: 08 8211 7388, Web: http://www.iseek.biz
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] What's the best way to stream and/or convert MP3 and WAV files?

[Matt Riddell]

Leo Burd wrote:
> Hello everyone,
> 
> I'm implementing an audioblog application and have some questions about
> how to best stream and/or convert MP3 and WAV files to be played by
> Asterisk.  Currently, I first copy the files from the server to my
> machine, convert them to Wav and play.  Unfortunately, this process is
> not very efficient at all.  Here are my questions:

Hehe, I have an AGI at the moment that connects to an RSS feed, parses it for
various mp3 files and then streams them.

> a) Is there any way to play MP3 files directly by Asterisk?

MP3Player.

> b) What is the best command line to be used to convert MP3 files into
> any format that can be played by Asterisk?  Shall I use sox or anything
> like that?  If so, what would be the proper parameters to pass?

Don't know sorry.

> c) Is there any way to stream MP3 files directly into Asterisk and still
> allow users to forward/pause/rewind the file while playing?

Not really.  What you're wanting is controlplayback, but when I tried it with
mp3 files, it didn't seem to work.

> d) Does Asterisk play any kind of WAV file?  Do I need to convert Wav
> files before playing them?

Should be ok, have you tried it?  What format are they in?

-- 
Cheers,

Matt Riddell
___

http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://freevoip.gedameurope.com (Free Asterisk Voip Community)
http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)

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

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

Re: [Asterisk-Users] Looking for Windows based Asterisk Management Client

[Tom Vile]

check out http://ipswitchboard.thorben.dk/ there is an asterisk
manager and other nice GUI 's for Windows

On 11/23/05, kchase <[EMAIL PROTECTED]> wrote:
>
> Greetings,
>
> Does anyone know of a Asterisk Manager Interface client application that can
> run from a Windows XP machine to manage Asterisk installed on a Linux
> Machine.
>
> I just know something like this exists but can't seem to find it out there.
>
> Thanks,
>
> KC
> ___
> --Bandwidth and Colocation sponsored by Easynews.com --
>
> 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
>
>


--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Phone: 978-203-3848 x205
Fax: 518-631-2856
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

RE: [Asterisk-Users] ver1.2 installation problem

[Bartosz Wegrzyn - asterisk]

I have fedora core 2 on this box.
I updated to the latest kernel but same problem.
My kernel is 2.6.10-1.771_FC2

aNY IDEAS

bART


> I had this problem with Fedora.  I updated the kernel to the latest one
> available for core 3 and changes the links to point to the new source
> code.  It worked fine then.
>
> Regards
>
> Lee
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz
> Wegrzyn - asterisk
> Sent: 23 November 2005 03:42
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] ver1.2 installation problem
>
> Hi,
>
> After I compile asterisk v.1.2 is tells me that last thing to do is to
> make install. Unfortunately it goes it to loop after I type make install
>
> this is the loop:
>
>  else \
> mv include/asterisk/version.h.tmp include/asterisk/version.h ; \
> fi rm -f include/asterisk/version.h.tmp build_tools/mkdep  -pipe  -Wall
> -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3
> -Iinclude -I../include -D_REENTRANT
> -D_GNU_SOURCE  -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS
> -fomit-frame-pointer  acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c
> asterisk.c ast_expr2.c ast_expr2f.c astmm.c autoservice.c buildinfo.c
> callerid.c cdr.c channel.c chanvars.c cli.c config.c cryptostub.c db.c
> devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c enum.c file.c frame.c
> fskmodem.c image.c indications.c io.c jitterbuf.c loader.c logger.c
> manager.c md5.c muted.c netsock.c pbx.c plc.c poll.c privacy.c rtp.c
> say.c sched.c slinfactory.c srv.c strcompat.c tdd.c term.c translate.c
> ulaw.c utils.c build_tools/make_version_h >
> include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp
> include/asterisk/version.h ; then echo; else \
> mv include/asterisk/version.h.tmp include/asterisk/version.h ; \
> fi rm -f include/asterisk/version.h.tmp build_tools/mkdep  -pipe  -Wall
> -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3
> -Iinclude -I../include -D_REENTRANT
> -D_GNU_SOURCE  -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS
> -fomit-frame-pointer  acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c
> asterisk.c ast_expr2.c ast_expr2f.c astmm.c autoservice.c buildinfo.c
> callerid.c cdr.c channel.c chanvars.c cli.c config.c cryptostub.c db.c
> devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c enum.c file.c frame.c
> fskmodem.c image.c indications.c io.c jitterbuf.c loader.c logger.c
> manager.c md5.c muted.c netsock.c pbx.c plc.c poll.c privacy.c rtp.c
> say.c sched.c slinfactory.c srv.c strcompat.c tdd.c term.c translate.c
> ulaw.c utils.c build_tools/make_version_h >
> include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp
> include/asterisk/version.h ; then echo; else \
> mv include/asterisk/version.h.tmp include/asterisk/version.h ; \
> fi rm -f include/asterisk/version.h.tmp build_tools/mkdep  -pipe  -Wall
> -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3
> -Iinclude -I../include -D_REENTRANT
> -D_GNU_SOURCE  -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS
> -fomit-frame-pointer  acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c
> asterisk.c ast_expr2.c ast_expr2f.c astmm.c autoservice.c buildinfo.c
> callerid.c cdr.c channel.c chanvars.c cli.c config.c cryptostub.c db.c
> devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c enum.c file.c frame.c
> fskmodem.c image.c indications.c io.c jitterbuf.c loader.c logger.c
> manager.c md5.c muted.c netsock.c pbx.c plc.c poll.c privacy.c rtp.c
> say.c sched.c slinfactory.c srv.c strcompat.c tdd.c term.c translate.c
> ulaw.c utils.c build_tools/make_version_h >
> include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp
> include/asterisk/version.h ; then echo; else \
> mv include/asterisk/version.h.tmp include/asterisk/version.h ; \
> fi rm -f include/asterisk/version.h.tmp build_tools/mkdep  -pipe  -Wall
> -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3
> -Iinclude -I../include -D_REENTRANT
> -D_GNU_SOURCE  -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS
> -fomit-frame-pointer  acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c
> asterisk.c ast_expr2.c ast_expr2f.c astmm.c autoservice.c buildinfo.c
> callerid.c cdr.c channel.c chanvars.c cli.c config.c cryptostub.c db.c
> devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c enum.c file.c frame.c
> fskmodem.c image.c indications.c io.c jitterbuf.c loader.c logger.c
> manager.c md5.c muted.c netsock.c pbx.c plc.c poll.c privacy.c rtp.c
> say.c sched.c slinfactory.c srv.c strcompat.c tdd.c term.c translate.c
> ulaw.c utils.c build_tools/make_version_h >
> include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp
> include/asterisk/version.h ; then echo; else \
> mv include/asterisk/version.h.tmp include/asterisk/version.h ; \
> fi rm -f include/asterisk/version.h.tmp build_tools/mkdep  -pipe  -Wall
> -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3
> -Iinclude -I../include -D_REENTRANT
> -D_GNU_SOURCE  -O6 -march=i686 -DZAPTEL_OPTIMIZA

RE: [Asterisk-Users] Satellite, SIP Invite 488 Codec Rejection, SIP Timing Issue??

[Aaron Clauson]

 Hi,

Thanks for the tip I'll try it out. That would explain some situations where
one of the peeople concerned was mucking around with the codec settings on
the PAP2 and managed to get some calls out.

It's a bit baffling how the Linksys devices will get INVITES through without
G.729 being set across non-satellite links and yet can't get the very same
INVITE through across a satellite link. Fair enough if it was the Linksys
generating the 488 during the INVITE negotiation but how does Asterisk even
know the difference??

Aaron

> -Original Message-
> From: Jason p [mailto:[EMAIL PROTECTED] 
> Sent: 24 November 2005 02:25
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List - 
> Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Satellite, SIP Invite 488 Codec 
> Rejection, SIP Timing Issue??
> 
> I had the same problem when we were setting up these boxes 
> after katrina. What i found is that they will only do one 
> G729 session at a time. so that mesg that your showing is 
> that its trying to register  two chans as 729. what i did to 
> get around this was to turn off fource prefered codec on one 
> line. This threw me for a loop also but trust me this is the 
> fix, and yes you can only make one 729 call at a time.
> 
> 
> Jason Price
> 
> 
> On 11/23/05, Aaron Clauson <[EMAIL PROTECTED]> wrote: 
> 
>   Hi,
>   
>   I have a very strange Asterisk SIP call signalling 
> problem that is proving
>   extremely difficult to track down. The problem is that 
> any SIP INVITE
>   request that is coming into Asterisk over a satellite 
> connection from a 
>   Linksys Router or PAP2 is getting a "Not Acceptable 
> Here (codec error)" from
>   Asterisk. I've done all the normal checks on the 
> allowed codecs in sip.conf
>   but to no avail.
>   
>   I've even gone as far as writing a basic SIP stack to 
> authenticate and send 
>   the INVITE request to Asterisk with exactly the same 
> SDP payload to let me
>   brute force different options in the SDP request to try 
> an narrow it down
>   that way. The preplexing thing from that length 
> exercise is that if exactly 
>   the same INVITE request comes in from my app across the 
> same satellite
>   connection to Asterisk it gets 200 Ok'ed but coming 
> from the Linksys PAP2 or
>   WRT54GP2 it gets 488 Codec Not Acceptable Here'ed.
>   
>   The first time this happened we went through all the 
> usual checks and got 
>   nowhere and the person drifted off and it was put down 
> to something speicifc
>   to that set up/connection. But now it's cropped up 
> again with a different
>   person who also just happens to be on a satellite 
> connection but from a 
>   different provider, although it is possible both 
> providers use the same
>   infrastructure. In both cases incoming calls to the 
> Linksys devices worked
>   correctly it's just the outgoing calls from the devices 
> to Asterisk that are 
>   getting the rejection. In the second case we can't put 
> it down to something
>   to do with the connection because the person has a 
> Vonage service working no
>   problems across the same satellite link we are getting 
> the rejection on. 
>   
>   The SIP trace is below and I'm wondering if anybody has 
> ever seen something
>   similar. The only thing I can think of is that it's 
> somehow a timing issue I
>   can't see how it can be a codec issue since the exactly 
> the same SDP payload 
>   will get OK'ed if coming from my app. Is the Asterisk 
> SIP stack sensitive to
>   the any timings in the INVITE request? It seems highly 
> unlikely but I just
>   can't think of anything else.
>   
>   INVITE sip:[EMAIL PROTECTED] SIP/2.0

>   Via: SIP/2.0/UDP 192.168.1.248:5061;branch=z9hG4bK-3b91173f
>   From: XXX ;tag=831f2cca367c3ddfo1 
>   To: 
>   Call-ID: [EMAIL PROTECTED]
>   CSeq: 103 INVITE
>   Max-Forwards: 70
>   Proxy-Authorization: Digest 
>   
> username="XXX",realm="asterisk",nonce="489bfe04",uri="sip:018X
> [EMAIL PROTECTED]",al
>   gorithm=MD5,response="22f566e03a225047469d73bec5ab640c" 
>   Contact: XXX 
>   Expires: 240
>   User-Agent: Linksys/PAP2-3.1.3(LS)
>   Content-Length: 424
>   Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
>   Supported: x-sipura
>   Content-Type: application/sdp 
>   
>   v=0
>   o=- 418210 418210 IN IP4 192.168.1.248
>   s=-
>   c=IN IP4 192.168.1.248
>   t=0 0
>   m=audio 16450 RTP/AVP 0 2 4 8 18 96 97 98 100 101
>   a=rtpmap:0 PCMU/8000
>   a=rtpmap:2 G726-32/8000
>   a=rtpmap:4 G723/8000
>   a=rtpmap:8 PCMA/8000
>   a=rtpmap:18 G729a/8000
>   a=rtpmap:96 G726-40/8000
>   a=rtpmap:97 G726-24/8000
>   a=rtpmap:98 G726-16/8000
>   a=rtpmap:100 NSE/8000 
>   a=rtpmap:101 telephone-event/8000
>   a=fmtp:101 0-15
> 

[Asterisk-Users] Looking for Windows based Asterisk Management Client

[kchase]

Greetings,
 
Does anyone know of a Asterisk Manager 
Interface client application that can run from a Windows XP machine to manage 
Asterisk installed on a Linux Machine.
 
I just know something like this exists but can't 
seem to find it out there.
 
Thanks,
 
KC
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] QSig and MD110

[Tim Rayner]

Hi Roger,

We've solved this with the MD110 sending calls to cisco VoIP gateways.  
The method is to set the Minimum and Maximum call length for this number 
range on the MD110 - and to configure the destination route to only send 
the call when the minimum length is reached (sometimes called en-block 
sending).  If your asterisk number range is 1500 - 1599


define your minimum and maximum length.

NANLS:EXL=15,MIN=4,MAX=4;

Also - make sure that your Route definition for this destination does 
the enblock.


RODDI:DEST=15,ROU=45,ADC=1x;

The first parameter of ADC causes the call to wait for minimum length 
before sending it to the route - the other parameters should stay at 
their pervious values.


I hope this helps - its worked well for us.

Tim.



Hi,
I have one Asterisk linked to a MD110 (Ericsson PBX) using a TE100P. 
I'm using the QSIG  ( Asterisk 1.2).
From * I can make calls elsewhere. But when the calling is coming 
from MD, the Asterisk is answering the call at the first digit it 
receives. The dial plain is waiting for a four digits long string (my 
extension plan). So it send back a hangup as a invalid dial.
How can I do to let Asterisk wait for the next digits without answer 
the call?. The MD is programmed to not wait a chunk of digits from 
the user,  to get a channel, and start sending the numbers.
(I know I could do a IVR style configuration - answer and let the 
user choose the extension, but it is not my intention).

Sincerely,
Roger.



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

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

Re: [Asterisk-Users] Satellite, SIP Invite 488 Codec Rejection, SIP Timing Issue??

[Jason p]

Trust me this is on the ATA. set both lines to use 729 but dont fource
them to only use that codec (in the ata config) I spent days trying to
figure this out the first time i ran accross it, and after that config
change on the ata i haven't had problems. I have seen this on most of
the sipura's that are in the linksys routers.

JasonOn 11/23/05, Kevin P. Fleming <[EMAIL PROTECTED]> wrote:
Aaron Clauson wrote:> m=audio 16450 RTP/AVP 0 2 4 8 18 96 97 98 100 101> a=rtpmap:0 PCMU/8000> a=rtpmap:2 G726-32/8000> a=rtpmap:4 G723/8000> a=rtpmap:8 PCMA/8000> a=rtpmap:18 G729a/8000
> a=rtpmap:96 G726-40/8000> a=rtpmap:97 G726-24/8000> a=rtpmap:98 G726-16/8000> a=rtpmap:100 NSE/8000I don't know what this (NSE) is, but Asterisk certainly doesn't support it.The only way we can debug this is by getting a complete 'sip debug' and
'set verbose' console trace; read the bug posting guidelines atbugs.digium.com and open a bug there with the required information.___
--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users

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

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

Re: [Asterisk-Users] Asterisk SIP architecture question

[Kevin P. Fleming]

Matt Riddell wrote:


The fact remains, if you need *very* accurate cdr's then you either don't do
canreinvite=yes for the peer or you code something so that Asterisk notices
that the rtp has stopped.  The fact remains that without these, the most
accurate CDR is going to come from the provider.


OK, I'll agree with that, except for one thing: if the provider notices 
that the RTP has stopped and wants to kill the call, it will send BYE to 
Asterisk and Asterisk will close the channels and update the CDR. The 
only time this will be an issue is if _both_ ends disappear and never 
send any signaling to Asterisk.


However, in the general case of not being concerned so much about the 
peer going away and losing CDR information for _one_ call, using 
reinvites does _not_ impact the quality of the softswitch's (Asterisk) CDRs.

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

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

Re: [Asterisk-Users] Satellite, SIP Invite 488 Codec Rejection, SIP Timing Issue??

[Jason p]

I had the same problem when we were setting up these boxes after
katrina. What i found is that they will only do one G729 session at a
time. so that mesg that your showing is that its trying to
register  two chans as 729. what i did to get around this was to
turn off fource prefered codec on one line. This threw me for a loop
also but trust me this is the fix, and yes you can only make one 729
call at a time.


Jason PriceOn 11/23/05, Aaron Clauson <[EMAIL PROTECTED]> wrote:
Hi,I have a very strange Asterisk SIP call signalling problem that is provingextremely difficult to track down. The problem is that any SIP INVITErequest that is coming into Asterisk over a satellite connection from a
Linksys Router or PAP2 is getting a "Not Acceptable Here (codec error)" fromAsterisk. I've done all the normal checks on the allowed codecs in sip.confbut to no avail.I've even gone as far as writing a basic SIP stack to authenticate and send
the INVITE request to Asterisk with exactly the same SDP payload to let mebrute force different options in the SDP request to try an narrow it downthat way. The preplexing thing from that length exercise is that if exactly
the same INVITE request comes in from my app across the same satelliteconnection to Asterisk it gets 200 Ok'ed but coming from the Linksys PAP2 orWRT54GP2 it gets 488 Codec Not Acceptable Here'ed.The first time this happened we went through all the usual checks and got
nowhere and the person drifted off and it was put down to something speicifcto that set up/connection. But now it's cropped up again with a differentperson who also just happens to be on a satellite connection but from a
different provider, although it is possible both providers use the sameinfrastructure. In both cases incoming calls to the Linksys devices workedcorrectly it's just the outgoing calls from the devices to Asterisk that are
getting the rejection. In the second case we can't put it down to somethingto do with the connection because the person has a Vonage service working noproblems across the same satellite link we are getting the rejection on.
The SIP trace is below and I'm wondering if anybody has ever seen somethingsimilar. The only thing I can think of is that it's somehow a timing issue Ican't see how it can be a codec issue since the exactly the same SDP payload
will get OK'ed if coming from my app. Is the Asterisk SIP stack sensitive tothe any timings in the INVITE request? It seems highly unlikely but I justcan't think of anything else.INVITE 
sip:[EMAIL PROTECTED] SIP/2.0Via: SIP/2.0/UDP 192.168.1.248:5061;branch=z9hG4bK-3b91173fFrom: XXX ;tag=831f2cca367c3ddfo1
To: Call-ID: [EMAIL PROTECTED]CSeq: 103 INVITEMax-Forwards: 70Proxy-Authorization: Digest
username="XXX",realm="asterisk",nonce="489bfe04",uri="sip:[EMAIL PROTECTED]",algorithm=MD5,response="22f566e03a225047469d73bec5ab640c"
Contact: XXX Expires: 240User-Agent: Linksys/PAP2-3.1.3(LS)Content-Length: 424Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFERSupported: x-sipuraContent-Type: application/sdp
v=0o=- 418210 418210 IN IP4 192.168.1.248s=-c=IN IP4 192.168.1.248t=0 0m=audio 16450 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000a=rtpmap:2 G726-32/8000a=rtpmap:4 G723/8000a=rtpmap:8 PCMA/8000a=rtpmap:18 G729a/8000a=rtpmap:96 G726-40/8000a=rtpmap:97 G726-24/8000a=rtpmap:98 G726-16/8000a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15a=ptime:30a=sendrecvSIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP192.168.1.248:5061;branch=z9hG4bK-3b91173f;received=xxx;rport=5061From: xxx ;tag=831f2cca367c3ddfo1
To: ;tag=as17d663fbCall-ID: [EMAIL PROTECTED]CSeq: 103 INVITEUser-Agent: asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFYContact: Proxy-Authenticate: Digest realm="asterisk", nonce="48554be3"Content-Length: 0
ACK sip:[EMAIL PROTECTED] SIP/2.0Via: SIP/2.0/UDP 192.168.1.248:5061;branch=z9hG4bK-c341696bFrom: xxx <
sip:[EMAIL PROTECTED]>;tag=831f2cca367c3ddfo1To: ;tag=as50c8f92dCall-ID: [EMAIL PROTECTED]
CSeq: 102 ACKMax-Forwards: 70Proxy-Authorization: Digestusername="xxx",realm="asterisk",nonce="3cb4e5eb",uri="sip:[EMAIL PROTECTED]
",algorithm=MD5,response="d4438aec627cefa82b6388a3b0c2cb1f"Contact: xxx User-Agent: Linksys/PAP2-3.1.3(LS)Content-Length: 0
INVITE sip:[EMAIL PROTECTED] SIP/2.0Via: SIP/2.0/UDP 192.168.1.248:5061;branch=z9hG4bK-3b91173fFrom: xxx <
sip:[EMAIL PROTECTED]>;tag=831f2cca367c3ddfo1To: Call-ID: [EMAIL PROTECTED]
CSeq: 103 INVITEMax-Forwards: 70Proxy-Authorization: Digestusername="xxx",realm="asterisk",nonce="489bfe04",uri="sip:[EMAIL PROTECTED]",al
gorithm=MD5,response="22f5

Re: [Asterisk-Users] Satellite, SIP Invite 488 Codec Rejection, SIP Timing Issue??

[Kevin P. Fleming]

Aaron Clauson wrote:


m=audio 16450 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000


I don't know what this (NSE) is, but Asterisk certainly doesn't support it.

The only way we can debug this is by getting a complete 'sip debug' and 
'set verbose' console trace; read the bug posting guidelines at 
bugs.digium.com and open a bug there with the required information.

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

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

Re: [Asterisk-Users] Asterisk SIP architecture question

[Matt Riddell]

Kevin P. Fleming wrote:
> Matt Riddell wrote:
> 
>> So how does Asterisk know that the media stream has been disconnected
>> between
>> the two remote hosts?
> 
> It doesn't... nor does any other SIP softswitch. See my other reply for
> a possible solution.

I agree that you could code a fix, but saying my advice is bogus because you
could code a fix for Asterisk to avoid it is slightly wrong.

The fact remains, if you need *very* accurate cdr's then you either don't do
canreinvite=yes for the peer or you code something so that Asterisk notices
that the rtp has stopped.  The fact remains that without these, the most
accurate CDR is going to come from the provider.

-- 
Cheers,

Matt Riddell
___

http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://freevoip.gedameurope.com (Free Asterisk Voip Community)
http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)

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

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

Re: [Asterisk-Users] reverse lookup when dialing an extension?

[Kevin P. Fleming]

Jeremy Koski wrote:

I'm using Cisco 7960 phones with asterisk. When I dial extension 200 
from my phone, it displays on the screen that I'm dialing 200. Is there 
a way to have the phone look up the callerid value in sip.conf and use 
that information instead of the dialed extension number?


At this time, no. The target of the call is not a 'peer' (which has 
CallerID), it's a 'user' (which doesn't). We have some semi-firm ideas 
on how to move in this direction, but it will take some planning and 
coding for the next major release to be able to support 'connected 
party' identification.

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

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

Re: [Asterisk-Users] Virtual Modems Revisited

[Steve Underwood]

Don Fanning wrote:


I brought this up a while back and althought there are pieces that
interface * into Fax Telephony applications, there hasn't been something
that works with plain old analog modems.

Then I found this piece of code.  From my initial tests it looks solid,
but I have no clue in how to interface this into asterisk.  I thought I
would put this link up for other people to comment and try.

http://fabrice.bellard.free.fr/linmodem.html

Out of the box it works with soundcards.  I've been battling jack and
alsa for a week trying to get them to play nice just to reroute the
audio but I'm out of time in this regard.  So I thought I would toss it
up and see what other people can come up with.

Happy Holidays!
Don
 

Linmodem doesn't work out the box with anything. linmodem was abandoned 
by its author at a very early stage, before any of its component parts 
really worked.


It has a number of useful bits, which might be used as the basis for a 
modem. It does not have a properly working code for any of the modem 
standards.


I think Fabrice got busy, and with patent issues preventing wide 
deployment of a V.34 modem finished the software just seemed like a 
waste of time to him. He is one of the good guys of free DSP, and has 
since produced several valuable things which are complete.


Steve

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

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

RE: [Asterisk-Users] Re: manager interface behavior

[Mark Edwards]

Title: Message



Are 
you also implementing the "ping" keepalive as part of your 
app?
 
Mark

  
  -Original Message-From: Bill Michaelson 
  [mailto:[EMAIL PROTECTED] Sent: Thursday, 24 November 2005 9:09 
  AMTo: asterisk-users@lists.digium.comSubject: 
  [Asterisk-Users] Re: manager interface behaviorsnacktime 
  wrote:On 11/23/05, Bill Michaelson <[EMAIL PROTECTED]> wrote:

  > I'm working on a manager client that I designed to hold open TCP
> connection to asterisk while it is running for varoius purposes.  After
> being puzzled by unexpected behavior, I realized that the server closes
> the connection after it completes an "originate" action - or at least it
> does in the case of my test transactions.
>
> I solicit opinions: is this a feature or a bug?
  
I've never seen that behavior and I've written several clients for the
manager api.  I guess it's possible that a particular combination of
variables in the request could trigger an error that makes asterisk do
that.   I would try issuing the same originate by telneting in
manually and see what happens.  That way you can positively rule out
your client being the one that's disconnecting.

to which I reply:That's the first thing I did, and it confirmed 
  the behavior (see below).  To be precise, the disconnect occurs after the 
  Newchannel report.  So I infer that you think it is inappropriate.  
  I've recoded the client so that it immediately reconnects.  Anybody 
  actually tried this?  I can imagine that the developer might have assumed 
  that such a request would likely come from a transient client, and that it 
  would be helpful to terminate the connection.  But if so, I don't think 
  it's the right decision.  Maybe it's just an oversight.  Any other 
  opinions?  I'm too lazy to read the server side 
  code.[EMAIL PROTECTED]:~> telnet hack.cosi.com 5038Trying 
  192.168.10.26...Connected to hack.cosi.com.Escape character is 
  '^]'.Asterisk Call Manager/1.0action: loginusername: 
  billsecret: dontellResponse: SuccessMessage: Authentication 
  acceptedaction: originatecallerid: 00context: 
  defaultpriority: 1exten: 212channel: Local/762Response: 
  SuccessMessage: Originate successfully queuedEvent: 
  NewchannelPrivilege: call,allChannel: 
  Local/[EMAIL PROTECTED],2State: RingCallerID: 
  CallerIDName: Uniqueid: 
  1132773921.72Connection closed by foreign host.[EMAIL PROTECTED]:~> 
  
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

RE: [Asterisk-Users] Virtual Modems Revisited

[Don Fanning]

Whoops... Sorry.. Mailer delay. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Don
Fanning
Sent: Wednesday, November 23, 2005 5:59 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Virtual Modems Revisited

I brought this up a while back and althought there are pieces that
interface * into Fax Telephony applications, there hasn't been something
that works with plain old analog modems.

Then I found this piece of code.  From my initial tests it looks solid,
but I have no clue in how to interface this into asterisk.  I thought I
would put this link up for other people to comment and try.

http://fabrice.bellard.free.fr/linmodem.html

Out of the box it works with soundcards.  I've been battling jack and
alsa for a week trying to get them to play nice just to reroute the
audio but I'm out of time in this regard.  So I thought I would toss it
up and see what other people can come up with.

Happy Holidays!
Don

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

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


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

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

RE: [Asterisk-Users] IAXmodem

[Miguel Soto]

Hi:
Well I tried to connect to instances of asterisk (hylafax & iaxmodem),
but I have a problem. I tried to send a fax from one of them, a message
appears:

-- Registered IAX2 '4000' (AUTHENTICATED) at 127.0.0.1:33270
-- Accepting AUTHENTICATED call from 127.0.0.1:
   > requested format = slin,
   > requested prefs = (),
   > actual format = slin,
   > host prefs = (slin),
   > priority = mine
-- Executing Dial("IAX2/4000-4", 
"IAX2/linux:[EMAIL PROTECTED]/[EMAIL PROTECTED]") in new stack
-- Called linux:[EMAIL PROTECTED]/[EMAIL PROTECTED]
Nov 23 18:34:43 NOTICE[9903]: chan_iax2.c:2818 auto_congest:
Auto-congesting 
call due to slow response
-- IAX2/192.169.0.99:4569-5 is circuit-busy
-- Hungup 'IAX2/192.169.0.99:4569-5'
  == Everyone is busy/congested at this time (1:0/1/0)
Nov 23 18:34:53 WARNING[9942]: pbx.c:2405 __ast_pbx_run: Timeout, but no

rule 't' in context 'contextFax'

Do you know why "Auto-congesting" appears?

Regards

Miguel


DETAILS:

My configuration files:

**First computer, from where I send the fax.

iax.conf
allow=slinear

Register=> linux:[EMAIL PROTECTED]
...
[4000]
username=4000
type=friend
secret=password
record_out=Never
record_in=Never
qualify=5000
port=4569
notransfer=yes
[EMAIL PROTECTED]
host=dynamic
context=contextFax
callerid=modem4000<4000>

extensions.conf
[contextFax]
extent=>3000,1,Dial(IAX2/linux:[EMAIL PROTECTED]/[EMAIL PROTECTED]
tFax)
extent=>4000,1,Dial,IAX2/4000


I send the fax with:

sendfax -f "[EMAIL PROTECTED]" -R -r "Subject" -c "coverpage
comments" -x "Your company" -d "[EMAIL PROTECTED]" README.txt

***Second computer

 iax.conf
[3000]
username=3000
type=friend
secret=password
record_out=Never
record_in=Never
qualify=5000
port=4569
notransfer=yes
[EMAIL PROTECTED]
host=dynamic
context=contextFax
callerid=modem3000<3000>

extensions.conf 
[contextFax]
extent=>3000,1,Dial,IAX2/3000





-Original Message-
From: Lee Howard [mailto:[EMAIL PROTECTED] 
Sent: Tuesday, November 22, 2005 23:21
To: Miguel Soto
Subject: Re: [Asterisk-Users] IAXmodem

You can fax to another instance of IAXmodem+HylaFAX.

Lee.


Miguel Soto wrote:

>Hi:
>Sorry if I write you directly and not to the email list. But I am 
>having some problems receiving mails from the list.
>
>Well, my question is how can I test my Asterisk, IAXModem, Hylafax
>configuration without being connected to the PSTN, is there any
>virtual fax?  
>
>
>Regards
>Miguel Soto
>
>-Original Message-
>From: Lee Howard [mailto:[EMAIL PROTECTED] 
>Sent: Monday, November 21, 2005 20:53
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [Asterisk-Users] IAXmodem
>
>Miguel Soto wrote:
>
>  
>
>>I am trying to configure the IAXmodem/ Hylafax server. I have a 
>>question related, where do you define the extension for the fax? In 
>>the extensions.conf or iax.conf file?
>>
>>
>>
>
>You define the extension in extensions.conf, but you also have to grant

>access permissions in iax.conf.
>
>Lee.
>___
>--Bandwidth and Colocation sponsored by Easynews.com --
>
>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
>
>
>
>  
>




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

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

[Asterisk-Users] Virtual Modems Revisited

[Don Fanning]

I brought this up a while back and althought there are pieces that
interface * into Fax Telephony applications, there hasn't been something
that works with plain old analog modems.

Then I found this piece of code.  From my initial tests it looks solid,
but I have no clue in how to interface this into asterisk.  I thought I
would put this link up for other people to comment and try.

http://fabrice.bellard.free.fr/linmodem.html

Out of the box it works with soundcards.  I've been battling jack and
alsa for a week trying to get them to play nice just to reroute the
audio but I'm out of time in this regard.  So I thought I would toss it
up and see what other people can come up with.

Happy Holidays!
Don

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

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

[Asterisk-Users] Re: sip URL peering

[Wolfgang S. Rupprecht]

Patrick <[EMAIL PROTECTED]> writes:
> Shouldn't the last line in exten-peers.conf be:
> exten => _**XXX.,1,Macro(dial-outgoing,SIP/${EXTEN:[EMAIL PROTECTED])
>   ^^^
> Similar to the previous line sipbroker line:
> exten => _**999.,1,Macro(dial-outgoing,SIP/${EXTEN:[EMAIL PROTECTED])

Thanks for looking this over.  Extra eyes always help.

The last line is a bit different from the ones above it.  In the
normal case the ${EXTEN:5} was meant to strip the 5 chars in the
routing prefix "**999" and only pass on the base number to the 
remote sip server.

The catch-all sipbroker line is meant to have the 4 of those 5 chars
passed off to sipbroker so that they can examine the routing prefix
and route the call.  This should only happen for the prefixes added
between the time I last updated the file and whenever a new prefix was
added to sipbroker.  The reason the first "*" needs to be stripped is
that sipbroker wants to see the prefix codes as "*999", with only one
"*".  Asterisk along with my Sipura phone seem to use *XX codes for
their own purposes and I didn't feel comfortable enough putting the
dial prefix codes in potentially clashing real-estate.  (Comments
suggestions are very welcome.  I've got very little telco/telecom
experience and am just "winging it".)

The one thing I think I do have a minor error on is in the dial-out
macro.  I copied it from somewhere, but the last "s-." line looks more
wrong the longer I look at it.  I think it should really be "_s-." and
not "s-.".

-wolfgang
-- 
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Re: Asterisk SIP architecture question

[David Liu]

If you use qualify=yes to determine whether that device is alive or not, 
then it won't be very accurate as every now and then, the device may 
fail to reply to the SIP OPTIONS packet due to reasons other than it is 
really offline. 

If you are linked to a PSTN GW, I would believe that GW will monitor the 
RTP stream and then initiate a BYE if it sees no RTP packets coming in.  
That way Asterisk will receive the proper disconnect signal in a 
canreinvite=yes scenario.


David

Kevin P. Fleming wrote:


David Thomas wrote:


Is the CDR accounting done based on SIP signaling? If a UA is talking
(RTP) to a third party PSTN gateway, isn't it at risk if say the UA
loses power. How will asterisk know the call has ended if it is not
involved in the media path. The idea is this.. I want to use
canreinvite =yes to force users to talk end-to-end to preserve
bandwidth, but I can see the potential for hung calls if asterisk
never get the BYE from a UA in the event the ATA gets unplugged or
somehow loses power.



That is the case in every SIP network, Asterisk is not unique in that 
regard.


I would suggest that you could make a modification to chan_sip so that 
if the peer goes 'unreachable' (as determined by using qualify=yes) 
than any existing calls involved with that peer are immediately hung 
up; that would take care of this problem.

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

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




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

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

[Asterisk-Users] Satellite, SIP Invite 488 Codec Rejection, SIP Timing Issue??

[Aaron Clauson]

Hi,

I have a very strange Asterisk SIP call signalling problem that is proving
extremely difficult to track down. The problem is that any SIP INVITE
request that is coming into Asterisk over a satellite connection from a
Linksys Router or PAP2 is getting a "Not Acceptable Here (codec error)" from
Asterisk. I've done all the normal checks on the allowed codecs in sip.conf
but to no avail. 

I've even gone as far as writing a basic SIP stack to authenticate and send
the INVITE request to Asterisk with exactly the same SDP payload to let me
brute force different options in the SDP request to try an narrow it down
that way. The preplexing thing from that length exercise is that if exactly
the same INVITE request comes in from my app across the same satellite
connection to Asterisk it gets 200 Ok'ed but coming from the Linksys PAP2 or
WRT54GP2 it gets 488 Codec Not Acceptable Here'ed. 

The first time this happened we went through all the usual checks and got
nowhere and the person drifted off and it was put down to something speicifc
to that set up/connection. But now it's cropped up again with a different
person who also just happens to be on a satellite connection but from a
different provider, although it is possible both providers use the same
infrastructure. In both cases incoming calls to the Linksys devices worked
correctly it's just the outgoing calls from the devices to Asterisk that are
getting the rejection. In the second case we can't put it down to something
to do with the connection because the person has a Vonage service working no
problems across the same satellite link we are getting the rejection on.  

The SIP trace is below and I'm wondering if anybody has ever seen something
similar. The only thing I can think of is that it's somehow a timing issue I
can't see how it can be a codec issue since the exactly the same SDP payload
will get OK'ed if coming from my app. Is the Asterisk SIP stack sensitive to
the any timings in the INVITE request? It seems highly unlikely but I just
can't think of anything else.

INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.248:5061;branch=z9hG4bK-3b91173f
From: XXX ;tag=831f2cca367c3ddfo1
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 103 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest
username="XXX",realm="asterisk",nonce="489bfe04",uri="sip:[EMAIL PROTECTED]",al
gorithm=MD5,response="22f566e03a225047469d73bec5ab640c"
Contact: XXX 
Expires: 240
User-Agent: Linksys/PAP2-3.1.3(LS)
Content-Length: 424
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 418210 418210 IN IP4 192.168.1.248
s=-
c=IN IP4 192.168.1.248
t=0 0
m=audio 16450 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv




SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.1.248:5061;branch=z9hG4bK-3b91173f;received=xxx;rport=5061
From: xxx ;tag=831f2cca367c3ddfo1
To: ;tag=as17d663fb
Call-ID: [EMAIL PROTECTED]
CSeq: 103 INVITE
User-Agent: asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: 
Proxy-Authenticate: Digest realm="asterisk", nonce="48554be3" 
Content-Length: 0





ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.248:5061;branch=z9hG4bK-c341696b
From: xxx ;tag=831f2cca367c3ddfo1
To: ;tag=as50c8f92d
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest
username="xxx",realm="asterisk",nonce="3cb4e5eb",uri="sip:[EMAIL PROTECTED]",al
gorithm=MD5,response="d4438aec627cefa82b6388a3b0c2cb1f"
Contact: xxx 
User-Agent: Linksys/PAP2-3.1.3(LS)
Content-Length: 0





INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.248:5061;branch=z9hG4bK-3b91173f
From: xxx ;tag=831f2cca367c3ddfo1
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 103 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest
username="xxx",realm="asterisk",nonce="489bfe04",uri="sip:[EMAIL PROTECTED]",al
gorithm=MD5,response="22f566e03a225047469d73bec5ab640c"
Contact: xxx 
Expires: 240
User-Agent: Linksys/PAP2-3.1.3(LS)
Content-Length: 424
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 418210 418210 IN IP4 192.168.1.248
s=-
c=IN IP4 192.168.1.248
t=0 0
m=audio 16450 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=rtpma

Re: [Asterisk-Users] ASTCC - card in use

[Darren Wiebe]

Ronald Wiplinger wrote:

Is there a solution for the problem that the card in use flag is set, 
after the user hang up?


Yes, there is a patch.  This was fixed in cvs quite a while ago.

Put this:
$SIG{HUP}  = 'ignore_hup';

sub ignore_hup {
   print STDERR "\nHUP received!\n\n";
}


just after the "use POSIX qw(ceil floor);" line



The flag remains set, if the user hang up, after the price for the 
call will be announced.
It is bad (for the business), because this happens most of the time 
only for NEW users!


Solutions?
1. Do we need the flag at all, if we use the phone number as card 
number anyway? If we can use the phone number as card number, we could 
omit it. However, I plan that I will have other premium features, 
where I need to punch in a card number!!!


The flag keeps one card from having more than 1 simultaneous call.  If 
you don't want to use it, a bit of work in astcc.agi would disable it.



2. Could we use a extension number to reset the flag?


You could this this using an agi script.

3. Could we use a cron job to reset the flag, if the extension number 
is not in a call? This one sounds for me most reliable, but at the 
moment the most complicated one to figure it out by the cron job and 
reset it from there.


This would be fairly easy.  You would need a script that ran the 
appropriate sql command when called.




Is a solution available?


Yes, it is.

Good Luck Ronald, I haven't talked to you for a long time. :-)

Darren Wiebe
[EMAIL PROTECTED]



bye

Ronald
___
--Bandwidth and Colocation sponsored by Easynews.com --

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



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

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

[Asterisk-Users] ASTCC - card in use

[Ronald Wiplinger]

Is there a solution for the problem that the card in use flag is set, 
after the user hang up?


The flag remains set, if the user hang up, after the price for the call 
will be announced.
It is bad (for the business), because this happens most of the time only 
for NEW users!


Solutions?
1. Do we need the flag at all, if we use the phone number as card number 
anyway? If we can use the phone number as card number, we could omit it. 
However, I plan that I will have other premium features, where I need to 
punch in a card number!!!

2. Could we use a extension number to reset the flag?
3. Could we use a cron job to reset the flag, if the extension number is 
not in a call? This one sounds for me most reliable, but at the moment 
the most complicated one to figure it out by the cron job and reset it 
from there.


Is a solution available?

bye

Ronald
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Re: sip URL peering

[Patrick]

On Wed, 2005-11-23 at 15:18 -0800, Wolfgang S. Rupprecht wrote:
> "Wolfgang S. Rupprecht" writes:
> > If there is enough interest, maybe the greater asterisk community
> > could adopt some semi-official mapping tables.  I'd be willing to
> > periodically generate a flat mapping file and an extension.conf
> > dialplan snippet from sipbroker's list or whatever else is deemed more
> > neutral or useful if there was any interest in such.
> 
> Just to try to get the ball rolling, I put together an asterisk config
> file that allows folks to "direct dial" other open sip servers using
> the same prefix codes as sipbroker.  Sipbroker also encourages folks
> to add listings for their sip servers, so in theory everyone here
> could join in the fun.  I'll update these files periodically, so they
> should track sipbroker's web page as folks add themselves.
> 
>   http://www.wsrcc.com/wolfgang/ftp/exten-peers.conf   (asterisk conf file)
>   http://www.wsrcc.com/wolfgang/ftp/sip-peers.txt  (raw mapping file)
>   http://www.wsrcc.com/wolfgang/ftp/dial-out.conf  (dial-out macro)

Shouldn't the last line in exten-peers.conf be:
exten => _**XXX.,1,Macro(dial-outgoing,SIP/${EXTEN:[EMAIL PROTECTED])
  ^^^
Similar to the previous line sipbroker line:
exten => _**999.,1,Macro(dial-outgoing,SIP/${EXTEN:[EMAIL PROTECTED])

Regards,
Patrick
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] TE410P and SPANDSP

[Ma Zhiyong]

Hi, All
   Does any one has successful experience use te410p and spandsp together?
   Could they work well with all 120 channels receive/send fax at the same time?

   My practice is that rxfax always get broken fax page.

   Help!___
--Bandwidth and Colocation sponsored by Easynews.com --

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

RE: [Asterisk-Users] Agent Logoff

[Marcus Deluigi \(intern\)]

 
> > That helped a little.
> > Thanks a lot!
> >
> > Is there any chance to determine the agent id (defined in 
> agents.conf) 
> > of a caller?
> 
> If I'm understanding you correctly, you seem to be under the 
> impression that you can only use  
> RemoveQueueMember/AddQueueMember on agents that are defined 
> in agents.conf?  If so, there is no such relationship.  The 
> agent id you pass to these commands doesn't  have to be 
> defined anywhere..

Thanks, but that's not the case. 
I just want to automatically register the agents to the same extension
as their Agent ID.
I wrote a small Perl AGI script, but it would be more easy and
convenient to have this directly in extension.conf

So, since the agents don't enter an extension on their login, it would
make sense if they would not need to enter one on their logoff. But it
seems that that's not possible, since I have no way to automatically
send 0-length extension to Asterisk.
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] 1.2.0 voicemail: unable to create lock file?

[Patrick]

Hi all,

Anyone know if the following message is something that should not
happen? I'm running asterisk as user/group asterisk/asterisk.

-- Executing VoiceMailMain("SIP/1003-6e02", "s1003") in new stack
Unable to create lock file
'/var/spool/asterisk/voicemail/default/1003/Old': No such file or
directory
Unable to create lock file
'/var/spool/asterisk/voicemail/default/1003/Old': No such file or
directory

The perms on /var/spool/asterisk/voicemail/default/1003 are:

drwx--  3 asterisk asterisk 4096 Nov 22 18:19 1003

-rwx--  1 asterisk asterisk 52844 Nov 22 18:19 busy.wav
-rwx--  1 asterisk asterisk 19884 Nov 22 18:19 greet.wav
drwx--  2 asterisk asterisk  4096 Nov 24 01:57 INBOX
-rwx--  1 asterisk asterisk 56684 Nov 22 18:19 unavail.wav

If I create the Old directory by hand it no longer complains. Shouldn't
it create the directory by itself?

Regards,
Patrick

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

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

Re: [Asterisk-Users] reverse lookup when dialing an extension?

[C F]

I don't know of any way of doing it thru asterisk, but you can test to
see if 200 exists in the 7960 directory xml file, maybe it will take
the value from there, but I might be wrong. I don't have one in front
of me to test it.

On 11/23/05, Jeremy Koski <[EMAIL PROTECTED]> wrote:
>
>
> I'm using Cisco 7960 phones with asterisk. When I dial extension 200 from
> my phone, it displays on the screen that I'm dialing 200. Is there a way
> to have the phone look up the callerid value in sip.conf and use that
> information instead of the dialed extension number?
>
>
> ___
> --Bandwidth and Colocation sponsored by Easynews.com --
>
> 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
>
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] QSig and MD110

[Rogerio Ferreira da Cunha]

Hi,
I have one Asterisk linked to a MD110 (Ericsson PBX) using a TE100P. I'm 
using the QSIG  ( Asterisk 1.2).
From * I can make calls elsewhere. But when the calling is coming from 
MD, the Asterisk is answering the call at the first digit it receives. 
The dial plain is waiting for a four digits long string (my extension 
plan). So it send back a hangup as a invalid dial.
How can I do to let Asterisk wait for the next digits without answer the 
call?. 
The MD is programmed to not wait a chunk of digits from the user,  to 
get a channel, and start sending the numbers.
(I know I could do a IVR style configuration - answer and let the user 
choose the extension, but it is not my intention).

Sincerely,
Roger.



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

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

[Asterisk-Users] reverse lookup when dialing an extension?

[Jeremy Koski]

I'm using Cisco 7960 phones with asterisk. When I dial extension 200 from 
my phone, it displays on the screen that I'm dialing 200. Is there a way 
to have the phone look up the callerid value in sip.conf and use that 
information instead of the dialed extension number?



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

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

Re: [Asterisk-Users] IAX clients

[Stefan Reuter]

On Wed, 2005-11-23 at 16:29 -0700, Jason Becker wrote:
> http://www.hem.za.org/jiaxclient/

Thanks for the pointer.
I should have been more clear with my request: What I am looking for is
a pure Java implementation. JIAXClient is a solution that is ok for many
use cases but is unacceptable in others as it merely wraps the native
library via JNI.

=Stefan


signature.asc
Description: This is a digitally signed message part
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] IAX clients

[Jason Becker]

Stefan Reuter wrote:

Yes it would be really interesting if there are any IAX libraries for
Java that are available under an open source license and that we might
improve further.
There is a growing demand for such a thing (for example see
http://forums.digium.com/viewtopic.php?t=2431)
Would be cool if we can create kind of a defacto standard, i.e.
something that everybody uses.


http://www.hem.za.org/jiaxclient/

(No affiliation.)

Regards,

--
Jason Becker
Director & CEO
Coalescent Systems Inc.
Enabling Open Source Telephony
403.244.8089
www.coalescentsystems.ca
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] Re: sip URL peering

[Wolfgang S. Rupprecht]

Klaus Darilion <[EMAIL PROTECTED]> writes:
> There is a new ietf WG to come which deals with peering issues. It's
> called SPEER (formerly VOIPEER)
>
> The list archive is at
> http://darkwing.uoregon.edu/~llynch/voipeer/
>
> minutes from last ietf meeting:
> http://www3.ietf.org/proceedings/05nov/minutes/voipeer.html

It looks interesting, but these things always seem to be scuttled or
reduced to glacial progress by the telecom interests.

VOIP peering isn't something that should require years of meeting to
make happen.

-wolfgang
-- 
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Clearwire and Asterisk

[Sean Kennedy]

Justin,

I can tell you that I haven't been able to get a clearwire rep out to my 
location to demonstrate their lines to us.  I keep calling, telling them 
what i want to do.  And they tell me that clearwire is a great service, 
and that one of they will relay my questions to a sales rep who will get 
back in touch with me soon.  And I never hear from them again.


In my mind, that doesn't seem like a 'good thing'.  YMMV

Sean

Justin Newman wrote:


Has anyone had problems using Clearwire, VOIP, and/or Asterisk?
Just curious...
 



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

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

Re: [Asterisk-Users] Re: manager interface behavior

[snacktime]

On 11/23/05, Bill Michaelson <[EMAIL PROTECTED]> wrote:
>  snacktime wrote:
>
> On 11/23/05, Bill Michaelson <[EMAIL PROTECTED]> wrote:
>
>
>  > I'm working on a manager client that I designed to hold open TCP
> > connection to asterisk while it is running for varoius purposes. After
> > being puzzled by unexpected behavior, I realized that the server closes
> > the connection after it completes an "originate" action - or at least it
> > does in the case of my test transactions.
> >
> > I solicit opinions: is this a feature or a bug?
>
>  I've never seen that behavior and I've written several clients for the
> manager api. I guess it's possible that a particular combination of
> variables in the request could trigger an error that makes asterisk do
> that. I would try issuing the same originate by telneting in
> manually and see what happens. That way you can positively rule out
> your client being the one that's disconnecting.
>
>
>  to which I reply:
>
>  That's the first thing I did, and it confirmed the behavior (see below).
> To be precise, the disconnect occurs after the Newchannel report.  So I
> infer that you think it is inappropriate.  I've recoded the client so that
> it immediately reconnects.  Anybody actually tried this?  I can imagine that
> the developer might have assumed that such a request would likely come from
> a transient client, and that it would be helpful to terminate the
> connection.  But if so, I don't think it's the right decision.  Maybe it's
> just an oversight.  Any other opinions?  I'm too lazy to read the server
> side code.
>

Anything to see in the debug logs when you did the originate?  I'd
probably file this as a bug.  I've never had the originate command
make the server drop the connection like that, and I've never heard
anywhere that it would be normal behavior.

Chris
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Asterisk + WiFi Phones

[Cory Andrews]

David - the "2nd generation" crop of WLAN handsets will start coming to 
market shortly, and vendors are promising improvements.   UTStarCom has 
the F3000 coming in December, which will have according to their spec


   * WEP (64 and 128 bit )/WPA/MD5 Auth
   * Handover/Roaming between different AP and SSID

There is also 802.11n on the horizon, which promises a four-fold 
improvement in performance over current 802.11g. 


Cory J Andrews
Partner / Purchasing
+++
VOIPSupply.com - Everything you need for VOIP
454 Sonwil Drive
Buffalo, NY 14225
+++
tf voice - 800-398-VOIP X22
l voice - 716.630.1555 X22
f - 716.630.1548
e - [EMAIL PROTECTED]
AIM - b2Cory



David Tillman wrote:

Thank you to everyone for the input. It may be to our advantage to
install a Wi-Fi mesh
with handoff as we will eventually put data-terminals on our
fork-trucks. In the meantime,
we have warehouse managers carrying cell phones for comms to the office.

I'm going to drop five Snom or Grandstream phones around the warehoue which will
help some.

I think trying to use the WiFi VOIP without robust handover will be a
mistake as the
users will always be walking around the warehouse or riding a forktruck.

Thanks,
-dave
___
--Bandwidth and Colocation sponsored by Easynews.com --

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


  

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

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

Re: [Asterisk-Users] IAX clients

[Stefan Reuter]

Yes it would be really interesting if there are any IAX libraries for
Java that are available under an open source license and that we might
improve further.
There is a growing demand for such a thing (for example see
http://forums.digium.com/viewtopic.php?t=2431)
Would be cool if we can create kind of a defacto standard, i.e.
something that everybody uses.

=Stefan



signature.asc
Description: This is a digitally signed message part
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] Re: sip URL peering

[Wolfgang S. Rupprecht]

"Wolfgang S. Rupprecht" writes:
> If there is enough interest, maybe the greater asterisk community
> could adopt some semi-official mapping tables.  I'd be willing to
> periodically generate a flat mapping file and an extension.conf
> dialplan snippet from sipbroker's list or whatever else is deemed more
> neutral or useful if there was any interest in such.

Just to try to get the ball rolling, I put together an asterisk config
file that allows folks to "direct dial" other open sip servers using
the same prefix codes as sipbroker.  Sipbroker also encourages folks
to add listings for their sip servers, so in theory everyone here
could join in the fun.  I'll update these files periodically, so they
should track sipbroker's web page as folks add themselves.

  http://www.wsrcc.com/wolfgang/ftp/exten-peers.conf   (asterisk conf file)
  http://www.wsrcc.com/wolfgang/ftp/sip-peers.txt  (raw mapping file)
  http://www.wsrcc.com/wolfgang/ftp/dial-out.conf  (dial-out macro)

-wolfgang
-- 
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] astman make error

[Fred Blaise]

On Wed, 2005-11-23 at 19:58 +0100, Fred Blaise wrote:
> Hi all
> 
> I am having an issue when trying to 'make astman'. Asterisk 1.2.0 here,
> from source, on debian sarge. Everything else working fine (only SIP
> setup anyway)
> 
> deafneuron:/opt/asterisk-1.2.0/utils# make astman
> cc -DNO_AST_MM   -c -o astman.o astman.c
> In file included from /usr/include/asterisk/manager.h:28,
>  from astman.c:41:
> /usr/include/asterisk/lock.h: In function `ast_mutex_init':
> /usr/include/asterisk/lock.h:517: error: `PTHREAD_MUTEX_RECURSIVE'
> undeclared (first use in this function)
> /usr/include/asterisk/lock.h:517: error: (Each undeclared identifier is
> reported only once
> /usr/include/asterisk/lock.h:517: error: for each function it appears
> in.)
> make: *** [astman.o] Error 1
I added -D_GNU_SOURCE to the CFLAGS.. did the trick.

> 
> thanks for any input
> 
> ___
> --Bandwidth and Colocation sponsored by Easynews.com --
> 
> 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


signature.asc
Description: This is a digitally signed message part
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Asterisk + WiFi Phones

[David Tillman]

Thank you to everyone for the input. It may be to our advantage to
install a Wi-Fi mesh
with handoff as we will eventually put data-terminals on our
fork-trucks. In the meantime,
we have warehouse managers carrying cell phones for comms to the office.

I'm going to drop five Snom or Grandstream phones around the warehoue which will
help some.

I think trying to use the WiFi VOIP without robust handover will be a
mistake as the
users will always be walking around the warehouse or riding a forktruck.

Thanks,
-dave
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Asterisk + WiFi Phones

[Cory Andrews]

Yeah the 480i-CT is a nice product, and not subject to the 
inconsistencies of WIFI.  What impressed me about the Engenius handsets 
was the range.  250,000 square feet coverage, or 3,000 acre coverage in 
an outdoor application, off a single radio AP, that's a large footprint.


Cory J Andrews
Partner / Purchasing
+++
VOIPSupply.com - Everything you need for VOIP
454 Sonwil Drive
Buffalo, NY 14225
+++
tf voice - 800-398-VOIP X22
l voice - 716.630.1555 X22
f - 716.630.1548
e - [EMAIL PROTECTED]
AIM - b2Cory



Michael Graves wrote:

FWIW, the cordless handsets on the Aastra 480i CT work well. I've waled
about while on a call, to a distahce of approx 300 feet before the call
quality started to suffer. The phone supports up to 8 (I think)
handsets, treating each as a separate extension. It can page between
handsets without involving Asterisk. It supports up to 9 line
appearances, which means that all the cordless and the base units can
be on a call at one time.

The only thing that I hated about the Engenius phones was the fact that
they didn't support a plain 2.5mm connector for a headset.

The 480i CT is pretty awesome.

Michael

On Wed, 23 Nov 2005 17:05:48 -0500, Cory Andrews wrote:

  
I concur with Michael, the current crop of WIFI phones on the market do 
have their individual quirks, and you will likely encounter issues using 
consumer grade access points.  If you have some money to throw at this, 
and want a real slick, industrial grade solution that will integrate 
with SIP, I would recommend you check out the Durafon 1X or Durafon 4X 

>from Engenius.  We have seen these implemented and SIP enabled using 
  
your garden variety ATA or FXS gateway.  These are not WIFI phones, they 
use conventional Digital spread spectrum with frequency hopping, and 
have a very cool, full duplex 2 way radio function as well.  The 
handsets are bulletproof as well, very rugged and well designed.


These were designed for relatively "extreme" environments, and they work 
very well, but are not inexpensive.


Cory J Andrews
Partner / Purchasing
+++
VOIPSupply.com - Everything you need for VOIP
454 Sonwil Drive
Buffalo, NY 14225
+++
tf voice - 800-398-VOIP X22
l voice - 716.630.1555 X22
f - 716.630.1548
e - [EMAIL PROTECTED]
AIM - b2Cory



Michael Graves wrote:


On Wed, 23 Nov 2005 15:33:49 -0600, David Tillman wrote:

  
  

This is only slightly related to Asterisk (in that we are using
Asterisk as our PBX),
so feel free to contact me off-list.

I need to hear from someone who has practical experience using the combo of
Asterisk, WiFi, and WiFi VOIP phones like the Hitachi-Cable IPC-5000.

In particular, I need to know if scattering consumer grade access
points (Linksys,
D-Link) throughout a warehouse is going to be satisfactory, or if we
should use some
kind of mesh access point from Orinoco or Strix. (In hopes of a better handoff.)



I tried using the consumer access points (Linksys WAP-54G) and had not
end of troubles. Firstly, they all have to have the same SSID or not
hand-off at all. Secondly, I found that the WLAN bandwidth would
occasionally just go away for a few seconds, which caused calls to
simply go silent for 3-4 seconds unexpectedly. Never long enough to
drop the call...but enough to force me to ask the other person to
repeat themselves. Lastly, the volume level presented by the WIP-5000
was never enough for my tastes.

I found that the range around each access point was very limited. Bear
in mind that consumer access points often don't put out the full
allowable RF power. If you use third party firmware hacks to up the RF
output you'll find that they start to through a lot of spurious
interference as well. Not the best quality radios I guess.

I eventually switched to using a Astra 480i CT desk phone with a couple
of corless handsets. It's been great.

Michael Graves

--
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]

o713-861-4005
o800-905-6412
c713-201-1262
fwd 54245



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

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


  
  

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

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





--
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc. 

Re: [Asterisk-Users] Asterisk + WiFi Phones

[Michael Graves]

FWIW, the cordless handsets on the Aastra 480i CT work well. I've waled
about while on a call, to a distahce of approx 300 feet before the call
quality started to suffer. The phone supports up to 8 (I think)
handsets, treating each as a separate extension. It can page between
handsets without involving Asterisk. It supports up to 9 line
appearances, which means that all the cordless and the base units can
be on a call at one time.

The only thing that I hated about the Engenius phones was the fact that
they didn't support a plain 2.5mm connector for a headset.

The 480i CT is pretty awesome.

Michael

On Wed, 23 Nov 2005 17:05:48 -0500, Cory Andrews wrote:

>I concur with Michael, the current crop of WIFI phones on the market do 
>have their individual quirks, and you will likely encounter issues using 
>consumer grade access points.  If you have some money to throw at this, 
>and want a real slick, industrial grade solution that will integrate 
>with SIP, I would recommend you check out the Durafon 1X or Durafon 4X 
>from Engenius.  We have seen these implemented and SIP enabled using 
>your garden variety ATA or FXS gateway.  These are not WIFI phones, they 
>use conventional Digital spread spectrum with frequency hopping, and 
>have a very cool, full duplex 2 way radio function as well.  The 
>handsets are bulletproof as well, very rugged and well designed.
>
>These were designed for relatively "extreme" environments, and they work 
>very well, but are not inexpensive.
>
>Cory J Andrews
>Partner / Purchasing
>+++
>VOIPSupply.com - Everything you need for VOIP
>454 Sonwil Drive
>Buffalo, NY 14225
>+++
>tf voice - 800-398-VOIP X22
>l voice - 716.630.1555 X22
>f - 716.630.1548
>e - [EMAIL PROTECTED]
>AIM - b2Cory
>
>
>
>Michael Graves wrote:
>> On Wed, 23 Nov 2005 15:33:49 -0600, David Tillman wrote:
>>
>>   
>>> This is only slightly related to Asterisk (in that we are using
>>> Asterisk as our PBX),
>>> so feel free to contact me off-list.
>>>
>>> I need to hear from someone who has practical experience using the combo of
>>> Asterisk, WiFi, and WiFi VOIP phones like the Hitachi-Cable IPC-5000.
>>>
>>> In particular, I need to know if scattering consumer grade access
>>> points (Linksys,
>>> D-Link) throughout a warehouse is going to be satisfactory, or if we
>>> should use some
>>> kind of mesh access point from Orinoco or Strix. (In hopes of a better 
>>> handoff.)
>>> 
>>
>> I tried using the consumer access points (Linksys WAP-54G) and had not
>> end of troubles. Firstly, they all have to have the same SSID or not
>> hand-off at all. Secondly, I found that the WLAN bandwidth would
>> occasionally just go away for a few seconds, which caused calls to
>> simply go silent for 3-4 seconds unexpectedly. Never long enough to
>> drop the call...but enough to force me to ask the other person to
>> repeat themselves. Lastly, the volume level presented by the WIP-5000
>> was never enough for my tastes.
>>
>> I found that the range around each access point was very limited. Bear
>> in mind that consumer access points often don't put out the full
>> allowable RF power. If you use third party firmware hacks to up the RF
>> output you'll find that they start to through a lot of spurious
>> interference as well. Not the best quality radios I guess.
>>
>> I eventually switched to using a Astra 480i CT desk phone with a couple
>> of corless handsets. It's been great.
>>
>> Michael Graves
>>
>> --
>> Michael Graves   [EMAIL PROTECTED]
>> Sr. Product Specialist  www.pixelpower.com
>> Pixel Power Inc. [EMAIL PROTECTED]
>>
>> o713-861-4005
>> o800-905-6412
>> c713-201-1262
>> fwd 54245
>>
>>
>>
>> ___
>> --Bandwidth and Colocation sponsored by Easynews.com --
>>
>> 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
>>
>>
>>   
>___
>--Bandwidth and Colocation sponsored by Easynews.com --
>
>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
>
>

--
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]

o713-861-4005
o800-905-6412
c713-201-1262
fwd 54245



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

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] Asterisk DNS SRV lookups

[David Thomas]

Does asterisk fully support DNS SRV lookups yet, or does it still only
read the first SRV entry?
Info on the wiki looked quite old, so I thought I better ask.

regards
David
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Modem Connections to PPP Server

[Casey Boone]

I think that timing issues will kill you, but if it were going to work 
you would want to use ulaw all the way around as your codec.


a better option would be to use more traditional terminal server/remote 
access server type hardware off of an actual copper pstn line.


you can pick up terminal servers off of ebay if cost is an issue, such 
as lucent max 4048s and cisco 5400s.  voip just is not a good way to 
carry a modem call


Casey Boone

Denis Vella wrote:

Hi,
 
I'm trying to use modems with Asterisk+VoIP Gateways in an attempt 
at providing an Internet service. 
 
Home_PC-->Modem-->PSTN-->VoIP_Gateway_FXO-->Ethernet-->Asterisk-->Ethernet-->VoIP_Gateway_FXS-->Modem-->PPP_Server-->Internet
 
I've been trying to use G711u and G711a codecs on the VoIP Gateways but, 
so far, no joy.  Has anyone got this to work?

Any pointers to setting this up?
 
Thanks,
Denis 



The information contained in this email is confidential and may be 
privileged. It is intended for the addressee only, if you are not the 
intended recipient please notify the sender and delete the email 
immediately. The contents of this email must not be disclosed or copied 
without the senders consent.

We cannot accept any responsibility for viruses.
Any views expressed in this message are those of the individual sender, 
except where the sender specifically states them to be the view of 
Philip Toledo Limited





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

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

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

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

RE: [Asterisk-Users] Asterisk + WiFi Phones

[Diseyi Diffa]

You can try Zyxel Wifi phones
They do the JOB cost bout 200 bucks

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Cory Andrews
Sent: Wednesday, November 23, 2005 2:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk + WiFi Phones

I concur with Michael, the current crop of WIFI phones on the market do 
have their individual quirks, and you will likely encounter issues using 
consumer grade access points.  If you have some money to throw at this, 
and want a real slick, industrial grade solution that will integrate 
with SIP, I would recommend you check out the Durafon 1X or Durafon 4X 
from Engenius.  We have seen these implemented and SIP enabled using 
your garden variety ATA or FXS gateway.  These are not WIFI phones, they 
use conventional Digital spread spectrum with frequency hopping, and 
have a very cool, full duplex 2 way radio function as well.  The 
handsets are bulletproof as well, very rugged and well designed.

These were designed for relatively "extreme" environments, and they work 
very well, but are not inexpensive.

Cory J Andrews
Partner / Purchasing
+++
VOIPSupply.com - Everything you need for VOIP
454 Sonwil Drive
Buffalo, NY 14225
+++
tf voice - 800-398-VOIP X22
l voice - 716.630.1555 X22
f - 716.630.1548
e - [EMAIL PROTECTED]
AIM - b2Cory



Michael Graves wrote:
> On Wed, 23 Nov 2005 15:33:49 -0600, David Tillman wrote:
>
>   
>> This is only slightly related to Asterisk (in that we are using
>> Asterisk as our PBX),
>> so feel free to contact me off-list.
>>
>> I need to hear from someone who has practical experience using the combo
of
>> Asterisk, WiFi, and WiFi VOIP phones like the Hitachi-Cable IPC-5000.
>>
>> In particular, I need to know if scattering consumer grade access
>> points (Linksys,
>> D-Link) throughout a warehouse is going to be satisfactory, or if we
>> should use some
>> kind of mesh access point from Orinoco or Strix. (In hopes of a better
handoff.)
>> 
>
> I tried using the consumer access points (Linksys WAP-54G) and had not
> end of troubles. Firstly, they all have to have the same SSID or not
> hand-off at all. Secondly, I found that the WLAN bandwidth would
> occasionally just go away for a few seconds, which caused calls to
> simply go silent for 3-4 seconds unexpectedly. Never long enough to
> drop the call...but enough to force me to ask the other person to
> repeat themselves. Lastly, the volume level presented by the WIP-5000
> was never enough for my tastes.
>
> I found that the range around each access point was very limited. Bear
> in mind that consumer access points often don't put out the full
> allowable RF power. If you use third party firmware hacks to up the RF
> output you'll find that they start to through a lot of spurious
> interference as well. Not the best quality radios I guess.
>
> I eventually switched to using a Astra 480i CT desk phone with a couple
> of corless handsets. It's been great.
>
> Michael Graves
>
> --
> Michael Graves   [EMAIL PROTECTED]
> Sr. Product Specialist  www.pixelpower.com
> Pixel Power Inc. [EMAIL PROTECTED]
>
> o713-861-4005
> o800-905-6412
> c713-201-1262
> fwd 54245
>
>
>
> ___
> --Bandwidth and Colocation sponsored by Easynews.com --
>
> 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
>
>
>   
___
--Bandwidth and Colocation sponsored by Easynews.com --

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


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

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

[Asterisk-Users] Invite with Replaces

[< Arnaud >]

Does Asterisk 1.2 support INVITE with Replaces header (rfc 3891) ?
thanks
- Arnaud
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] Re: manager interface behavior

[Bill Michaelson]

snacktime wrote:
On 11/23/05, Bill Michaelson <[EMAIL PROTECTED]> wrote:


  > I'm working on a manager client that I designed to hold open TCP
> connection to asterisk while it is running for varoius purposes.  After
> being puzzled by unexpected behavior, I realized that the server closes
> the connection after it completes an "originate" action - or at least it
> does in the case of my test transactions.
>
> I solicit opinions: is this a feature or a bug?
  


I've never seen that behavior and I've written several clients for the
manager api.  I guess it's possible that a particular combination of
variables in the request could trigger an error that makes asterisk do
that.   I would try issuing the same originate by telneting in
manually and see what happens.  That way you can positively rule out
your client being the one that's disconnecting.


to which I reply:

That's the first thing I did, and it confirmed the behavior (see
below).  To be precise, the disconnect occurs after the Newchannel
report.  So I infer that you think it is inappropriate.  I've recoded
the client so that it immediately reconnects.  Anybody actually tried
this?  I can imagine that the developer might have assumed that such a
request would likely come from a transient client, and that it would be
helpful to terminate the connection.  But if so, I don't think it's the
right decision.  Maybe it's just an oversight.  Any other opinions? 
I'm too lazy to read the server side code.

[EMAIL PROTECTED]:~> telnet hack.cosi.com 5038
Trying 192.168.10.26...
Connected to hack.cosi.com.
Escape character is '^]'.
Asterisk Call Manager/1.0
action: login
username: bill
secret: dontell

Response: Success
Message: Authentication accepted

action: originate
callerid: 00
context: default
priority: 1
exten: 212
channel: Local/762

Response: Success
Message: Originate successfully queued

Event: Newchannel
Privilege: call,all
Channel: Local/[EMAIL PROTECTED],2
State: Ring
CallerID: 
CallerIDName: 
Uniqueid: 1132773921.72

Connection closed by foreign host.
[EMAIL PROTECTED]:~> 


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

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

Re: [Asterisk-Users] Asterisk + WiFi Phones

[Cory Andrews]

I concur with Michael, the current crop of WIFI phones on the market do 
have their individual quirks, and you will likely encounter issues using 
consumer grade access points.  If you have some money to throw at this, 
and want a real slick, industrial grade solution that will integrate 
with SIP, I would recommend you check out the Durafon 1X or Durafon 4X 
from Engenius.  We have seen these implemented and SIP enabled using 
your garden variety ATA or FXS gateway.  These are not WIFI phones, they 
use conventional Digital spread spectrum with frequency hopping, and 
have a very cool, full duplex 2 way radio function as well.  The 
handsets are bulletproof as well, very rugged and well designed.


These were designed for relatively "extreme" environments, and they work 
very well, but are not inexpensive.


Cory J Andrews
Partner / Purchasing
+++
VOIPSupply.com - Everything you need for VOIP
454 Sonwil Drive
Buffalo, NY 14225
+++
tf voice - 800-398-VOIP X22
l voice - 716.630.1555 X22
f - 716.630.1548
e - [EMAIL PROTECTED]
AIM - b2Cory



Michael Graves wrote:

On Wed, 23 Nov 2005 15:33:49 -0600, David Tillman wrote:

  

This is only slightly related to Asterisk (in that we are using
Asterisk as our PBX),
so feel free to contact me off-list.

I need to hear from someone who has practical experience using the combo of
Asterisk, WiFi, and WiFi VOIP phones like the Hitachi-Cable IPC-5000.

In particular, I need to know if scattering consumer grade access
points (Linksys,
D-Link) throughout a warehouse is going to be satisfactory, or if we
should use some
kind of mesh access point from Orinoco or Strix. (In hopes of a better handoff.)



I tried using the consumer access points (Linksys WAP-54G) and had not
end of troubles. Firstly, they all have to have the same SSID or not
hand-off at all. Secondly, I found that the WLAN bandwidth would
occasionally just go away for a few seconds, which caused calls to
simply go silent for 3-4 seconds unexpectedly. Never long enough to
drop the call...but enough to force me to ask the other person to
repeat themselves. Lastly, the volume level presented by the WIP-5000
was never enough for my tastes.

I found that the range around each access point was very limited. Bear
in mind that consumer access points often don't put out the full
allowable RF power. If you use third party firmware hacks to up the RF
output you'll find that they start to through a lot of spurious
interference as well. Not the best quality radios I guess.

I eventually switched to using a Astra 480i CT desk phone with a couple
of corless handsets. It's been great.

Michael Graves

--
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]

o713-861-4005
o800-905-6412
c713-201-1262
fwd 54245



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

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


  

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

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

AW: [Asterisk-Users] Asterisk 1.2.0 AddOn's compile error with MySQL5.0.15

[Rainer Maier]

Hi Matt,

I did not move the whole asterisk directory I just put a link to it. (ln -s
/usr/src/asterisk-1.2.0 /usr/src/asterisk)
Then I tried to compile but the error stayed.
I also tried with MySQL 4.1.15 and had the same error.

I am getting to the point where I think I might have not all nessecary
packets installed.
I got only two. (apt-get install libssl-dev zlib1g-dev)
Would I need more ?

Also here is the the complete compiler run 

---
sv5000:/usr/src/asterisk-addons-1.2.0# make clean
rm -f *.so *.o .depend
make -C format_mp3 clean
make[1]: Gehe in Verzeichnis »/usr/src/asterisk-addons-1.2.0/format_mp3«
rm -f *.o *.so *~
make[1]: Verlasse Verzeichnis »/usr/src/asterisk-addons-1.2.0/format_mp3«
sv5000:/usr/src/asterisk-addons-1.2.0# make
./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/local/mysql/include
-I/usr/local/mysql/include/mysql`ls *.c`
make -C format_mp3 all
make[1]: Gehe in Verzeichnis »/usr/src/asterisk-addons-1.2.0/format_mp3«
gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations   -D_REENTRANT -D_GNU_SOURCE  -O6-c -o common.o
common.c
gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations   -D_REENTRANT -D_GNU_SOURCE  -O6-c -o
dct64_i386.o dct64_i386.c
gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations   -D_REENTRANT -D_GNU_SOURCE  -O6-c -o
decode_ntom.o decode_ntom.c
gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations   -D_REENTRANT -D_GNU_SOURCE  -O6-c -o layer3.o
layer3.c
gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations   -D_REENTRANT -D_GNU_SOURCE  -O6-c -o tabinit.o
tabinit.c
gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations   -D_REENTRANT -D_GNU_SOURCE  -O6-c -o
interface.o interface.c
gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations   -D_REENTRANT -D_GNU_SOURCE  -O6-c -o
format_mp3.o format_mp3.c
gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations   -D_REENTRANT -D_GNU_SOURCE  -O6  -shared -Xlinker
-x -o format_mp3.so common.o dct64_i386.o decode_ntom.o layer3.o tabinit.o
interface.o format_mp3.o
make[1]: Verlasse Verzeichnis »/usr/src/asterisk-addons-1.2.0/format_mp3«
cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/local/mysql/include
-I/usr/local/mysql/include/mysql  -c -o app_saycountpl.o
app_saycountpl.c
cc -shared -Xlinker -x -o app_saycountpl.so app_saycountpl.o
cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/local/mysql/include
-I/usr/local/mysql/include/mysql  -c -o cdr_addon_mysql.o
cdr_addon_mysql.c
cc -shared -Xlinker -x -o cdr_addon_mysql.so cdr_addon_mysql.o -lmysqlclient
-lz-L/usr/local/mysql/lib -L/usr/local/mysql/lib/mysql  
cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/local/mysql/include
-I/usr/local/mysql/include/mysql  -c -o app_addon_sql_mysql.o
app_addon_sql_mysql.c
cc -shared -Xlinker -x -o app_addon_sql_mysql.so app_addon_sql_mysql.o
-lmysqlclient -lz-L/usr/local/mysql/lib -L/usr/local/mysql/lib/mysql  
cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/local/mysql/include
-I/usr/local/mysql/include/mysql  -c -o res_config_mysql.o
res_config_mysql.c
res_config_mysql.c: In function 'realtime_mysql':
res_config_mysql.c:117: warning: incompatible implicit declaration of
built-in function 'snprintf'
res_config_mysql.c: In function 'realtime_multi_mysql':
res_config_mysql.c:224: warning: incompatible implicit declaration of
built-in function 'snprintf'
res_config_mysql.c: In function 'update_mysql':
res_config_mysql.c:313: warning: incompatible implicit declaration of
built-in function 'snprintf'
res_config_mysql.c: In function 'config_mysql':
res_config_mysql.c:376: warning: incompatible implicit declaration of
built-in function 'snprintf'
res_config_mysql.c: In function 'realtime_mysql_status':
res_config_mysql.c:648: warning: incompatible implicit declaration of
built-in function 'snprintf'
res_config_mysql.c:650: warning: incompatible implicit declaration of
built-in function 'snprintf'
res_config_mysql.c:652: warning: incompatible implicit declaration of
built-in function 'snprintf'
res_config_mysql.c:656: warning: incompatible implicit declaration of
built-in function 'snprintf'
cc -shared -Xlinker -x -o res_config_mysql.so res_config_mysql.o
-lmysqlclient -lz-L/usr/local/mysql/lib -L/usr/local/mysql/lib/mysql  
rm app_saycountpl.o
sv5000:/usr/src/asterisk-addons-1.2.0# 

---

How could I load the addons anyhow ?
Or does asterisk it by himself ?
How do I see which add ons are loaded ? 
By "show modules" ?

Best regards
Rainer

-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von
[EMAIL PROTECTED]
Gesendet: Die

Re: [Asterisk-Users] Asterisk + WiFi Phones

[Michael Graves]

On Wed, 23 Nov 2005 15:57:25 -0600, David Tillman wrote:

>On 11/23/05, Michael Graves <[EMAIL PROTECTED]> wrote:
>
>> I eventually switched to using a Astra 480i CT desk phone with a couple
>> of corless handsets. It's been great.
>
>My first thought was to use a cordless phone and a Sipura ATA. But this
>is a 100,000 sqft warehouse with a freezer section in the middle so the range
>isn't quite there. Next choice is WiFi phones.
>
>This should be interesting.

If a normal cordless phone isn't cutting it then you'll need a handfull
of access points to make wifi functional. Perhaps you should try the
Engenious high power corldess phones. I used to use those and they
worked well.

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
fwd 54245



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

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

Re: [Asterisk-Users] Asterisk + WiFi Phones

[David Tillman]

On 11/23/05, Michael Graves <[EMAIL PROTECTED]> wrote:

> I eventually switched to using a Astra 480i CT desk phone with a couple
> of corless handsets. It's been great.

My first thought was to use a cordless phone and a Sipura ATA. But this
is a 100,000 sqft warehouse with a freezer section in the middle so the range
isn't quite there. Next choice is WiFi phones.

This should be interesting.

-dave
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

RE: [Asterisk-Users] Asterisk + WiFi Phones

[trixter aka Bret McDanel]

Depending on load, and this shouldnt be a problem with low call volume,
wifi is half duplex, so you have to take that into account.  

I however use a ipaq h5500 with a softphone running and it works fine
over wifi, while that isnt quite the equipment you requested info on, I
do not have any problems at least with asterisk 1.0.7 and above


On Wed, 2005-11-23 at 13:42 -0800, Diseyi Diffa wrote:
> Well I am using a wifi phone running sip and works fine.   Using commercial
> grade access point. With Prizim card.
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of David Tillman
> Sent: Wednesday, November 23, 2005 1:34 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Asterisk + WiFi Phones
> 
> This is only slightly related to Asterisk (in that we are using
> Asterisk as our PBX),
> so feel free to contact me off-list.
> 
> I need to hear from someone who has practical experience using the combo of
> Asterisk, WiFi, and WiFi VOIP phones like the Hitachi-Cable IPC-5000.
> 
> In particular, I need to know if scattering consumer grade access
> points (Linksys,
> D-Link) throughout a warehouse is going to be satisfactory, or if we
> should use some
> kind of mesh access point from Orinoco or Strix. (In hopes of a better
> handoff.)
> 
> Thanks,
> -dave
> ___
> --Bandwidth and Colocation sponsored by Easynews.com --
> 
> 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
> 
> 
> ___
> --Bandwidth and Colocation sponsored by Easynews.com --
> 
> 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
-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group


signature.asc
Description: This is a digitally signed message part
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] Asterisk cisco FXO

[Diseyi Diffa]

Has anyone has any luck setting up cisco MC3800 with asterisk.  I looked at
the example on the voip-info.org site, but know luck.  Someone pls help



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

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

Re: [Asterisk-Users] Asterisk + WiFi Phones

[Michael Graves]

On Wed, 23 Nov 2005 15:33:49 -0600, David Tillman wrote:

>This is only slightly related to Asterisk (in that we are using
>Asterisk as our PBX),
>so feel free to contact me off-list.
>
>I need to hear from someone who has practical experience using the combo of
>Asterisk, WiFi, and WiFi VOIP phones like the Hitachi-Cable IPC-5000.
>
>In particular, I need to know if scattering consumer grade access
>points (Linksys,
>D-Link) throughout a warehouse is going to be satisfactory, or if we
>should use some
>kind of mesh access point from Orinoco or Strix. (In hopes of a better 
>handoff.)

I tried using the consumer access points (Linksys WAP-54G) and had not
end of troubles. Firstly, they all have to have the same SSID or not
hand-off at all. Secondly, I found that the WLAN bandwidth would
occasionally just go away for a few seconds, which caused calls to
simply go silent for 3-4 seconds unexpectedly. Never long enough to
drop the call...but enough to force me to ask the other person to
repeat themselves. Lastly, the volume level presented by the WIP-5000
was never enough for my tastes.

I found that the range around each access point was very limited. Bear
in mind that consumer access points often don't put out the full
allowable RF power. If you use third party firmware hacks to up the RF
output you'll find that they start to through a lot of spurious
interference as well. Not the best quality radios I guess.

I eventually switched to using a Astra 480i CT desk phone with a couple
of corless handsets. It's been great.

Michael Graves

--
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]

o713-861-4005
o800-905-6412
c713-201-1262
fwd 54245



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

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

RE: [Asterisk-Users] Asterisk + WiFi Phones

[Diseyi Diffa]

Well I am using a wifi phone running sip and works fine.   Using commercial
grade access point. With Prizim card.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Tillman
Sent: Wednesday, November 23, 2005 1:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk + WiFi Phones

This is only slightly related to Asterisk (in that we are using
Asterisk as our PBX),
so feel free to contact me off-list.

I need to hear from someone who has practical experience using the combo of
Asterisk, WiFi, and WiFi VOIP phones like the Hitachi-Cable IPC-5000.

In particular, I need to know if scattering consumer grade access
points (Linksys,
D-Link) throughout a warehouse is going to be satisfactory, or if we
should use some
kind of mesh access point from Orinoco or Strix. (In hopes of a better
handoff.)

Thanks,
-dave
___
--Bandwidth and Colocation sponsored by Easynews.com --

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


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

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

[Asterisk-Users] Asterisk + WiFi Phones

[David Tillman]

This is only slightly related to Asterisk (in that we are using
Asterisk as our PBX),
so feel free to contact me off-list.

I need to hear from someone who has practical experience using the combo of
Asterisk, WiFi, and WiFi VOIP phones like the Hitachi-Cable IPC-5000.

In particular, I need to know if scattering consumer grade access
points (Linksys,
D-Link) throughout a warehouse is going to be satisfactory, or if we
should use some
kind of mesh access point from Orinoco or Strix. (In hopes of a better handoff.)

Thanks,
-dave
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] Modem Connections to PPP Server

[Denis Vella]

Title: Message



Hi,
 
I'm trying to use 
modems with Asterisk+VoIP Gateways in an attempt at providing 
an Internet service.  
 
Home_PC-->Modem-->PSTN-->VoIP_Gateway_FXO-->Ethernet-->Asterisk-->Ethernet-->VoIP_Gateway_FXS-->Modem-->PPP_Server-->Internet
 
I've been trying to 
use G711u and G711a codecs on the VoIP Gateways but, so far, no joy.  
Has anyone 
got this to work?
Any pointers to 
setting this up?

 
Thanks,
Denis The information contained in this email is confidential and may be privileged. It is intended for the addressee only, if you are not the intended recipient please notify the sender and delete the email immediately. The contents of this email must not be disclosed or copied without the senders consent. We cannot accept any responsibility for viruses. Any views expressed in this message are those of the individual sender, except where the sender specifically states them to be the view of Philip Toledo Limited
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Agent Logoff

[snacktime]

On 11/22/05, Marcus Deluigi (intern) <[EMAIL PROTECTED]> wrote:
> That helped a little.
> Thanks a lot!
>
> Is there any chance to determine the agent id (defined in agents.conf)
> of a caller?

If I'm understanding you correctly, you seem to be under the
impression that you can only use  RemoveQueueMember/AddQueueMember on
agents that are defined in agents.conf?  If so, there is no such
relationship.  The agent id you pass to these commands doesn't  have
to be defined anywhere..

  Chris
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Eicon Diva Server query

[Armin Schindler]

On Thu, 24 Nov 2005, Avi Miller wrote:
> David Waugh wrote:
> > Yes, you can use the Eicon Diva Range with 2.6 Kernels
> 
> Another question, considering the card should arrive tomorrow and I'd like to
> try my hand at setting it up this weekend: Do I need to BRIstuff Asterisk to
> get the Eicon Diva V-4BRI to work, or should I just need chan_capi-cm?

Just chan_capi-cm with plain Asterisk.

Armin
 
> Thanks,
> Avi
> 
> -- 
> National Manager - Special Projects
> 
> < Sydney / Melbourne / Canberra / Hobart / London />
> Walter Turnbull Bldg   T: +61 (0) 2 6233 0607
> 44 Sydney Ave, F: +61 (0) 2 6233 0696
> Forrest,   W: http://www.squiz.net/
> ACT 2603
> 
> .>> Open Source  - Own it  -  Squiz.net ./>
> ___
> --Bandwidth and Colocation sponsored by Easynews.com --
> 
> 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
> 
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] Not receiving fax

[Wayne Gemmell]

Hi all

I'm having trouble receiving faxes using rxfax. Could somebody please browse 
my log file and give me a swift kick in the right direction? I've also added 
my zapata.conf file at the end. 

I tried adjusting the rxgain to 15 (as mentioned in the archives) but that 
didn't seem to make a difference. I've turned of my busydetect in case it is 
hanging up too soon.

I'm running asterisk 1.2 and spandsp 0.0.2pre21


Nov 23 22:34:51 DEBUG[3374] chan_sip.c: Stopping retransmission on 
'[EMAIL PROTECTED]' of Request 102: Match Found
Nov 23 22:34:59 DEBUG[3374] chan_sip.c: Auto destroying call 
'[EMAIL PROTECTED]'
Nov 23 22:35:43 DEBUG[3373] dsp.c: dsp busy pattern set to 0,0
Nov 23 22:35:43 VERBOSE[6125] logger.c: -- Starting simple switch on 
'Zap/1-1'
Nov 23 22:35:43 NOTICE[6125] chan_zap.c: Got event 18 (Ring Begin)...
Nov 23 22:35:44 NOTICE[6125] chan_zap.c: Got event 2 (Ring/Answered)...
Nov 23 22:35:46 NOTICE[6125] chan_zap.c: Got event 18 (Ring Begin)...
Nov 23 22:35:46 NOTICE[6125] chan_zap.c: Got event 2 (Ring/Answered)...
Nov 23 22:35:47 NOTICE[6125] chan_zap.c: Got event 18 (Ring Begin)...
Nov 23 22:35:47 NOTICE[6125] chan_zap.c: Got event 2 (Ring/Answered)...
Nov 23 22:35:49 NOTICE[6125] chan_zap.c: Got event 18 (Ring Begin)...
Nov 23 22:35:49 DEBUG[6125] pbx.c: Expression result is '0'
Nov 23 22:35:49 VERBOSE[6125] logger.c: -- Executing GotoIf("Zap/1-1", 
"0?from-pstn-reghours|s|1:") in new stack
Nov 23 22:35:49 DEBUG[6125] pbx.c: Not taking any branch
Nov 23 22:35:49 DEBUG[6125] pbx.c: Expression result is '0'
Nov 23 22:35:49 VERBOSE[6125] logger.c: -- Executing GotoIf("Zap/1-1", 
"0?from-pstn-afthours|s|1:") in new stack
Nov 23 22:35:49 DEBUG[6125] pbx.c: Not taking any branch
Nov 23 22:35:49 VERBOSE[6125] logger.c: -- Executing GotoIfTime("Zap/1-1", 
"7:55-17:05|mon-fri|*|*?from-pstn-reghours|s|1:") in new stack
Nov 23 22:35:49 VERBOSE[6125] logger.c: -- Executing Goto("Zap/1-1", 
"from-pstn-afthours|s|1") in new stack
Nov 23 22:35:49 VERBOSE[6125] logger.c: -- Goto (from-pstn-afthours,s,1)
Nov 23 22:35:49 DEBUG[6125] pbx.c: Expression result is '0'
Nov 23 22:35:49 VERBOSE[6125] logger.c: -- Executing GotoIf("Zap/1-1", 
"0?from-pstn-afthours-nofax|s|1:2") in new stack
Nov 23 22:35:49 VERBOSE[6125] logger.c: -- Goto (from-pstn-afthours,s,2)
Nov 23 22:35:49 VERBOSE[6125] logger.c: -- Executing Answer("Zap/1-1", "") 
in new stack
Nov 23 22:35:49 DEBUG[6125] chan_zap.c: Took Zap/1-1 off hook
Nov 23 22:35:49 DEBUG[6125] chan_zap.c: Enabled echo cancellation on channel 1
Nov 23 22:35:49 DEBUG[6125] chan_zap.c: Engaged echo training on channel 1
Nov 23 22:35:49 VERBOSE[6125] logger.c: -- Executing Wait("Zap/1-1", "1") 
in new stack
Nov 23 22:35:50 VERBOSE[6125] logger.c: -- Executing SetVar("Zap/1-1", 
"intype=aa_1") in new stack
Nov 23 22:35:50 VERBOSE[6125] logger.c: -- Executing Cut("Zap/1-1", 
"intype=intype|-|1") in new stack
Nov 23 22:35:50 DEBUG[6125] pbx.c: Expression result is '0'
Nov 23 22:35:50 VERBOSE[6125] logger.c: -- Executing GotoIf("Zap/1-1", 
"0?7:9") in new stack
Nov 23 22:35:50 VERBOSE[6125] logger.c: -- Goto (from-pstn-afthours,s,9)
Nov 23 22:35:50 DEBUG[6125] pbx.c: Expression result is '0'
Nov 23 22:35:50 VERBOSE[6125] logger.c: -- Executing GotoIf("Zap/1-1", 
"0?10:12") in new stack
Nov 23 22:35:50 VERBOSE[6125] logger.c: -- Goto (from-pstn-afthours,s,12)
Nov 23 22:35:50 DEBUG[6125] pbx.c: Expression result is '0'
Nov 23 22:35:50 VERBOSE[6125] logger.c: -- Executing GotoIf("Zap/1-1", 
"0?13:15") in new stack
Nov 23 22:35:50 VERBOSE[6125] logger.c: -- Goto (from-pstn-afthours,s,15)
Nov 23 22:35:50 VERBOSE[6125] logger.c: -- Executing Goto("Zap/1-1", 
"aa_1|s|1") in new stack
Nov 23 22:35:50 VERBOSE[6125] logger.c: -- Goto (aa_1,s,1)
Nov 23 22:35:50 WARNING[6125] ast_expr2.fl: ast_yyerror(): syntax error: 
syntax error, unexpected TOK_EQ, expecting TOK_MINUS or TOK_COMPL or TOK_LP 
or TOKEN; Input:
 = ANSWER
 ^
Nov 23 22:35:50 WARNING[6125] ast_expr2.fl: If you have questions, please 
refer to doc/README.variables in the asterisk source.
Nov 23 22:35:50 DEBUG[6125] pbx.c: Expression result is '0'
Nov 23 22:35:50 VERBOSE[6125] logger.c: -- Executing GotoIf("Zap/1-1", 
"0?4") in new stack
Nov 23 22:35:50 DEBUG[6125] pbx.c: Not taking any branch
Nov 23 22:35:50 VERBOSE[6125] logger.c: -- Executing Answer("Zap/1-1", "") 
in new stack
Nov 23 22:35:50 VERBOSE[6125] logger.c: -- Executing Wait("Zap/1-1", "1") 
in new stack
Nov 23 22:35:51 VERBOSE[6125] logger.c: -- Executing SetVar("Zap/1-1", 
"LOOPED=1") in new stack
Nov 23 22:35:51 DEBUG[6125] pbx.c: Expression result is '0'
Nov 23 22:35:51 VERBOSE[6125] logger.c: -- Executing GotoIf("Zap/1-1", 
"0?hang|1") in new stack
Nov 23 22:35:51 DEBUG[6125] pbx.c: Not taking any branch
Nov 23 22:35:51 VERBOSE[6125] logger.c: -- Executing SetVar("Zap/1-1", 
"DIR-CONTEXT=general") in new stack
Nov 23 22:35:51 VERBOSE[6125] logger.c: -

Re: [Asterisk-Users] [Asterisk-Dev] hello

[Steve Blair]

Yes?

harry gaillac wrote:


hello






___ 
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger 
Téléchargez cette version sur http://fr.messenger.yahoo.com

___
Asterisk-Dev mailing list
Asterisk-Dev@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-dev
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-dev

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

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
 


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

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

Re: [Asterisk-Users] Agent Logoff

[Anthony Rodgers]

Others may know better than me, but I don't think so...

Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp


On Nov 22, 2005, at 11:02 PM, Marcus Deluigi ((intern)) wrote:


That helped a little.
Thanks a lot!

Is there any chance to determine the agent id (defined in agents.conf)
of a caller?

> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Anthony Rodgers
> Sent: Wednesday, November 23, 2005 2:11 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Agent Logoff
>
> Hi Marcus,
>
> Here's what we do:
>
> [agent-login]
> exten => s,1,NoOp(${AgentUser})
> exten => s,2,AddQueueMember(${AgentContext}|${AgentChannel})
> exten => s,3,Wait(1)
> exten => s,4,Playback(agent-loginok)
> exten => s,5,Hangup
> exten => s,103,RemoveQueueMember(${AgentContext}|${AgentChannel})
> exten => s,104,Wait(1)
> exten => s,105,Playback(agent-loggedoff) exten => s,106,Hangup
>
> [help-desk]
> exten => s,1,Answer
> exten => s,2,SetMusicOnHold(default)
> exten => s,3,DigitTimeout,5
> exten => s,4,ResponseTimeout,10
> exten => s,5,SetVar(AgentContext=${CONTEXT})
> exten => s,6,SetVar(AgentChannel=${CHANNEL})
> exten => s,7,Cut(AgentChannel=AgentChannel,-,1)
> exten => s,8,Cut(AgentUser=AgentChannel,/,2)
> exten => s,9,NoOp(${AgentUser})
> ; help-desk agents
> exten => s,10,GotoIf($[$["${AgentUser}" = "davidb"] | $["$
> {AgentUser}" = "karenj"] | $["${AgentUser}" = 
"laurp"]]?agent-login,s,

> 1:)
> exten => s,11. (rest of queue)
>
> exten => 2313,1,Goto,help-desk|s|
> 1
>
> What this basically means is that if an agent calls the queue
> from their own phone, it logs them in or out as appropriate
> without the need for passwords or anything.
>
> Does this look like it might help?
>
> Regards,
> --
> Anthony Rodgers
> Business Systems Analyst
> District of North Vancouver
> Web: http://www.dnv.org
> RSS Feed: http://www.dnv.org/rss.asp
>
>
> On 22-Nov-05, at 10:01 PM, Marcus Deluigi ((intern)) wrote:
>
> >
> > Hi.
> >
> > Another agent question: is it possible to put an agentlogoff on an
> > extension, without dialing a '#' for a password?
> > Something like:
> > exten => 702,1,AgentCallbackLogin(,,'#'@interncall)
> > But it does not work, because asterisk does not except '#'
> as a valid
> > extension ...
> >
> > A perl skript would also help me ..
> >
> > Greetings,
> > Marcus
> >
> > ___
> > --Bandwidth and Colocation sponsored by Easynews.com --
> >
> > 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
>
> ___
> --Bandwidth and Colocation sponsored by Easynews.com --
>
> 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
>
___
--Bandwidth and Colocation sponsored by Easynews.com --

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


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

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

Re: [Asterisk-Users] Call parking on Polycom IP501

[Anthony Rodgers]

Hi Dave,

exten => callpark,1,Dial(SIP/1000) didn't work - invalid extension

exten => callpark,1,Transfer(1000) didn't work - the parker hung up, 
and the stall number announcement was made to the parked caller.


On Nov 22, 2005, at 10:34 PM, David Hindmarsh wrote:


Hi Guys,

What happened if you just sent the callpark extension to 1000

Regards,

David

 

> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Anthony Rodgers
> Sent: Wednesday, 23 November 2005 08:57
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Call parking on Polycom IP501
>
> Hi there,
>
> Instead of asking a question, I thought I'd post an answer. I
> got the Polycom IP501 'Park' softkey working with * by doing
> the following:
>
> features.conf:
>
> [general]
> parkext => 1000
> parkpos => 1001-1009
> context => parkedcalls
> parkingtime => 120
> transferdigittimeout => 3
> courtesytone = beep
>
> Nothing unusual there. Here's the neat bit:
>
> extensions.conf:
>
> [internal] ; or whatever the relevant context is for you -
> it's usually wherever your Polycom lives include =>
> parkedcalls exten =>
> callpark,1,ParkAndAnnounce(pbx-transfer:PARKED|120|SIP/
> ${DIALEDPEERNUMBER}|internal,${DIALEDPEERNUMBER},1)
>
> By using SIP DEBUG, I discovered that the Polycom attempts to
> re-invite the call to an extension called callpark. I
> couldn't get Park() to work (it announces the stall number to
> the parked caller, instead of the parker, for some reason),
> but using ParkAndAnnouce puts the parked call on hold, hangs
> up the parker and then immediately calls them back with an
> announcement of the stall number.
>
> Hope this helps someone out..
>
> Regards,
> --
> Anthony Rodgers
> Business Systems Analyst
> District of North Vancouver
> Web: http://www.dnv.org
> RSS Feed: http://www.dnv.org/rss.asp
>
> ___
> --Bandwidth and Colocation sponsored by Easynews.com --
>
> 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
>
>
> --
> No virus found in this incoming message.
> Checked by AVG Free Edition.
> Version: 7.1.362 / Virus Database: 267.13.5/178 - Release
> Date: 22/11/2005
> 
>

--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.362 / Virus Database: 267.13.5/178 - Release Date: 
22/11/2005

 

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

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


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

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

Re: [Asterisk-Users] Eicon Diva Server query

[Avi Miller]

David Waugh wrote:

Yes, you can use the Eicon Diva Range with 2.6 Kernels


Another question, considering the card should arrive tomorrow and I'd 
like to try my hand at setting it up this weekend: Do I need to BRIstuff 
Asterisk to get the Eicon Diva V-4BRI to work, or should I just need 
chan_capi-cm?


Thanks,
Avi

--
National Manager - Special Projects

< Sydney / Melbourne / Canberra / Hobart / London />
  Walter Turnbull Bldg   T: +61 (0) 2 6233 0607
  44 Sydney Ave, F: +61 (0) 2 6233 0696
  Forrest,   W: http://www.squiz.net/
  ACT 2603

.>> Open Source  - Own it  -  Squiz.net ./>
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Outgoing Calls

[Martin Joseph]

On Nov 23, 2005, at 11:14 AM, Michael wrote:

I am trying to route my calls through an outside IAX provider.  I am 
having a problem with which codec to use.  The only way I have 
successfully been able to make an outgoing call is if i do:


  disallow=all
   allow=g729

 in the sip.conf file (for my phones) and the iax.conf file.  The 
second I add one more codec to that list, for instance:


  disallow=all
   allow=g729
   allow=ulaw

 I get the following error in the CLI:

  Nov 23 10:56:35 NOTICE[3799]: channel.c:1703 ast_set_write_format: 
Unable to find a path from ulaw to g729
   Nov 23 10:56:35 NOTICE[3799]: channel.c:1736 ast_set_read_format: 
Unable to find a path from g729 to ulaw





Asterisk cannot translate from other codecs to g729 UNLESS you buy a 
license for that technology.  It can pass g729 along though.  it looks 
like your service provider "plainvoice" only supports G729?


I am a newb though so take this with a BIG grain of salt.

Marty

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

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

[Asterisk-Users] [Asterisk-Dev] hello

[harry gaillac]

hello






___ 
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger 
Téléchargez cette version sur http://fr.messenger.yahoo.com
___
Asterisk-Dev mailing list
Asterisk-Dev@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-dev
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev

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

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

[Asterisk-Users] PhoneCALL version 2.7-RC1 Released!

[Dustin Wildes]

Hello Everyone!
For all of you PhoneCALL users, we have a treat for you today as 
PhoneCALL 2.7-RC1 has been released!


We've worked hard to make this release as close to as bug-free as 
possible, but in the event you find a bug - PLEASE report it to the 
bugtracker.  It doesn't matter how small of a 'bug' or problem you think 
it is - all input helps and makes the program better for everyone.


The bug tracker is at:
http://bugs.vecsector.com

Get your copy of PhoneCALL in the Downloads section at:
http://www.vecsector.com/phonecall
http://www.vecsector.com/phonecall/modules.php?name=Downloads


Thanks!!
Dustin Wildes



INFO on 2.7-RC1

New System Features include:
-
 -- Better script handling of Arguments
 -- New Queue Configuration
 -- New Conference(MeetMe) configuration
 -- Defaults configuration for SIP/IAX/Voicemail
 -- Easy to use Installation Wizard
 -- Better Multi-Tenant Support
 --  More Security Enhances for user groups
 --  New user-login methods from accounts
 --  DID Manager implemented
 --  New Provider/Trunk manager
 --  More Advanced configuration options for accounts
 --  Beginning of Wizard API
 --  New Context Manager (for creating custom contexts)



BUG FIXES
-
---0001---
Warning: closedir(): supplied argument is not a valid Directory
---0002---
some settings in configs/generalsettings.php appear to have no affect / 
redundant

---0003---
debug (echo) statements in systemPrefs.php
---0004---
Site Name in system prefs doesn't appear to be used anywhere
---0005---
Make the $path option a configurable option
---0007---
Phonecall reports asterisk as not running while in fact the service is 
running

---0008---
saveconfig does not write correctly the arguments ?
---0009---
slashes,subject and body for voicemail - general settings
---00011---
Text Message field in voicemail config screen too short
---00013---
phonecall.sql file not compatible with mysql 4
---00014---
2 variables in generalsettings.php that look the same
---00015---
AEL
---00020---
Default account preferences for NEW accounts
---00022---
Update script causing top bar not to display for slow WAN
---00029---
Script with Multiple arguments posts to DB with ARG# 1 off
---00030---
Macro Copy
---00031---
Adding a new Extension does not bring up a screen to fill in arguments
---00032---
Arguments are not being processed properly with dropdown accounts


UPCOMING FEATURES
-
 --  Realtime Asterisk Support
 --  Statistics Support
 --  Realtime Monitoring & Status viewers
 --  More templates(template engine)
 --  More Wizards and Macro defaults
 --  Whatever else maybe entered into the bug tracker by the 
community(this means you! :)  )



Thanks to everyone for their feedback, contributions & support for 
getting us to 2.7-RC1!

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

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

Re: [Asterisk-Users] VoIPJet Support Contact

[Francesco Peeters]

On Wed, November 23, 2005 20:47, Chris Mason (Lists) said:
>
>>NON-DISCLOSURE: ALL CUSTOMERS USING VOIPJET'S SERVICE ARE SPECIFICALLY
>>PROHIBITED FROM DISCLOSING TO OTHERS THAT THEY USE VOIPJET'S SERVICE,
>>
>>
> I use Voipjet,
> I have used Voipjet...
>
> Did I mention I use Voipjet?
>
> I'd like to teach the world to sing (about using Voipjet)...
>
> So sue me Voipjet, or better still, refund the outstanding balance so I
> can use it with a service that doesn't make people agree to stupid
> unenforcable rules. Another LiveVoip in the making.
>
> --
> Chris Mason
>
>

/me tries to suppress a silly giggle...

/me fails miserably!

-- 
Francesco Peeters

GPG Key = AA69 E7C6 1D8A F148 160C  D5C4 9943 6E38 D5E3 7704
If your program doesn't recognize my signature, please visit
http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate.
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] hello

[harry gaillac]

hello






___ 
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger 
Téléchargez cette version sur http://fr.messenger.yahoo.com
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] LinksysOne.com (New SIP phone, and more)

[Cory Andrews]

Far as I know these products are going to be tied to the LinksysOne 
hosted services program, which you can find more information on at 
www.LinksysOne.com


Cory J Andrews
Partner / Purchasing
+++
VOIPSupply.com - Everything you need for VOIP
454 Sonwil Drive
Buffalo, NY 14225
+++
tf voice - 800-398-VOIP X22
l voice - 716.630.1555 X22
f - 716.630.1548
e - [EMAIL PROTECTED]
AIM - b2Cory



tracinet wrote:
Sounds like they are providing a Vonage-style service that is tied 
into the phones.  Not sure they will sell them "unlocked".  Looks cool 
though.


On 11/22/05, *Lenny Tropiano / asterisk.org  
Mailing list* <[EMAIL PROTECTED] > wrote:


Another IP phone possibility for Asterisk.

No, not the SPA941 (from the Linksys/Cisco/Sipura world)...

Don't know much about it... but found this.  Nothing on the datasheet
says what it'll support really.

http://newsroom.cisco.com/dlls/2005/eKits/Data_Sheet_IP_Manager_Phone.pdf


But I found this that also talked about it being "SIP based"

http://www.linksysinfo.org/modules.php?name=AvantGo&file=print&sid=438

http://www.linksysone.com

Everything they want that isn't in the SPA941 ...

PoE and integrated switch.  Color screen.  Price point $299 (estimated
list price).

Looks interesting.

--
Lenny Tropiano  E-mail: [EMAIL PROTECTED]

Partner, Networking
Specialist  Pager:  [EMAIL PROTECTED]

VoIPing,
LLCURL:http://www.voiping.com/
PO Box 867, Cedar Park, TX 78630-0867   Mobile: 512-698-VOIP [8647]
___
--Bandwidth and Colocation sponsored by Easynews.com
 --

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





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

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

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

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

Re: [Asterisk-Users] manager interface behavior

[snacktime]

On 11/23/05, Bill Michaelson <[EMAIL PROTECTED]> wrote:
> I'm working on a manager client that I designed to hold open TCP
> connection to asterisk while it is running for varoius purposes.  After
> being puzzled by unexpected behavior, I realized that the server closes
> the connection after it completes an "originate" action - or at least it
> does in the case of my test transactions.
>
> I solicit opinions: is this a feature or a bug?

I've never seen that behavior and I've written several clients for the
manager api.  I guess it's possible that a particular combination of
variables in the request could trigger an error that makes asterisk do
that.   I would try issuing the same originate by telneting in
manually and see what happens.  That way you can positively rule out
your client being the one that's disconnecting.

Chris
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] Cisco FXO hangup detection

[Eric Bishop]

I am using a Cisco 1760V with FXO card in Australia to provide ports into

Asterisk.



I was wondering if anyone out there has a config for the cisco to detect

the disconnect or hangup signal for Australian tones.



If the calling party hangs up while leaving a voice mail for example, it

takes around 15 seconds for the call to time out.  I believe the Cisco

can be configured to detect the hangup or disconnect tone, but l can't

find any details in my searching.
___
--Bandwidth and Colocation sponsored by Easynews.com --

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