[Asterisk-Users] RE: Asterisk crashed so often

2004-03-05 Thread Freddi Hansen
From: Unavailable ID [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Date: Thu, 4 Mar 2004 16:33:12 -0800
Subject: [Asterisk-Users] Asterisk crashed so often
Reply-To: [EMAIL PROTECTED]
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FYI that I have experience the same problem many times.  The system is =
running RH 9.0 with Asterisk CVS-02/21/04.  Here is the output from the =
console:
I am using RH 9.0 too.
If I forget to do the:
export LD_ASSUME_KERNEL=2.4.1
before asterisk is started then it will handle less than 10 calls before it crashes.
Otherwise it seems rockstable in our enviroment with SIP and zaptel TP410P's when it's started 
Freddi
 



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Re: [Asterisk-Users] 3com NBX phones

2004-03-05 Thread Nicolas Bougues

On Thu, Mar 04, 2004 at 04:32:52PM -0500, Clif Jones wrote:
 I know a little history on the 3com SIP phones... We have about a dozen 
 of them
 where I work.  I'm not familiar with the NBX100 model number but the ones we
 have are labeled: P/N: 655005001.  The first ones didn't support SIP out 
 of the
 box and had to be upgraded with a new flash image.  I can't recall if 
 they came from
 the factory with H323 or the 3com proprietary IP protocol but the phones 
 look just
 like the 3com PBX phones you see in small businesses.  3com abandoned 
 the phone
 after spinning off the SIP division (Commworks?) and determining that 
 the phone
 hardware just didn't have the resources to continue work on SIP.  It is 
 a shame because
 these phones boot faster than any other IP phone I have seen and have a 
 good speakerphone.
 The image that we use is pre-RFC3261 but would probably work with Asterisk.
 

I have the same phones here (two of them). Got them directly from 3com
in late 2000 I think.

They were never distributed in the channel I think (at least not here
in Europe). Mine arrived SIP ready, and I flashed them once with a
standard TFTP procedure. Unfortunatly, 3com discontinued them, no more
firmware are available, AFAIK. My firmware is 1.0.1.21.0 SIP.

Unfortunatly I'm not able to find this latest firmware on file.

Note that the hardware is probably not the same as the standard NBX
phones : my SIP phones did feature an IR sensor to be used by a Palm
for automated dialing.

-- 
Nicolas Bougues
Axialys Interactive
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Re: [Asterisk-Users] Asterisk fault tolerance and a embedded hardware solution.....??

2004-03-05 Thread Kiss Karoly
Hello,

IMHO you have a problem with the hardware that Asterisk runs on.
You should really look around because there are a number of companies
selling intel based systems with a cPCI bus fully hot swap capable.
I think the only problem would be getting network adapters compatible with
* but then this is only a problem of drivers easily solved by a good
programmer.

If you test out Asterisk on a fully redundant box and you find problems I
think you'd be welcome to send in a patch to fix them so that * could
be used in enterprise computing instead of sending in a two page e-mail
with the problems we all know about !

Regards

Kiss Karoly

On Thu, 4 Mar 2004, Randall Shimizu wrote:

 Asterisk fault tolerance and a embedded hardware solution.??

 Has anyoone tried implement Asterisk as a hardware based solution similar to Soekris 
 firewall?


 Asterisk  fault tolerance: I ran across this posting about Asterisk
 and here is some interesting thoughts to ponder


 http://groups.google.com/groups?hl=enlr=ie=UTF-8oe=UTF-8selm=aca5dd1d9141c07addd9d3414e934380%40free.teranews.comrnum=14


 Not blow anyone's ASTERISK bubble BUT,,,

 Show me an Asterisk system that can:

 1) Have a communication bus that can survive the removal of the CPU,
 and
 still have calls in progress that remain active until the calling
 parties
 hang up.

 Difficult problem to solve. One would have to have some sort of
 parallel network connection. Perhaps one could have a buffering or
 cache solution.

 The CPU problem could be solved by a blade server or failover.

 2) I have yet to hear of any Asterisk box running a fully redundant
 CPU
 configuration. I bet this is possible.Especially with the newer hot
 swap
 cPCI bus systems and slave CPU cards.  Even better if the chassis has
 and
 embedded H.110, or equivalent in LAN/memory, switching bus.

 Yes could be solved.

 3) A redundant configuration where either CPU can talk to the
 communications
 boards (T1/E1), and LAN interfaces.  And which can address all boards
 in the
 system redundantly.

 Sounds like a job for Infiniband or a platform that has a switched
 crossbar architecture like IBM P-Series or Sun.

 4) A redundant configuration that has either shared system memory
 between
 the CPU's, or at least table copies between memory that hold all
 static and
 dynamic call information.

 5) A redundant configuration that can swap between system CPU's in
 less than
 20 seconds.

 6) A redundant configuration that can synchronize on, and share one,
 two ,
 and more network clocking signals.  Plus synchronize on a independent
 stratum 3 or greater clock source.

 7) And can support 1,000 or more endpoints (TDM and/or IP) without
 choking
 on it's own guts.

 8) A redundant configuration that can synchronize on, and share one,
 two ,
 and more network clocking signals.

 Well it's a lot to ask, but enterprise computing demands a lot.


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Re: [Asterisk-Users] Supervised transfer (almost) with GS phone

2004-03-05 Thread Brian Capouch
Stephen R. Besch wrote:
I have now tested a (previously suggested) method for doing supervised 
transfers using the Grandstream SIP phone. It isn't perfect, but it 
works and is very functional. Here are the steps:

When I try this, all goes well until, after putting the original caller 
on hold and then getting a dialtone, I dial another extension, and then 
get these errors on the CLI:

find_user: Call from user 'btel' rejected due to usage limit of 1
find_user: Call from user 'btel' rejected due to usage limit of 1
find_user: Call from user 'btel' rejected due to usage limit of 1
find_user: Call from user 'btel' rejected due to usage limit of 1
Then the Grandstream gives me a busy, and my orignal caller is a zombie.

What am I doing wrong?

Thx.

B.
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Re: [Asterisk-Users] Supervised transfer (almost) with GS phone

2004-03-05 Thread Konrad Gorski
Brian Capouch wrote:

Stephen R. Besch wrote:

I have now tested a (previously suggested) method for doing 
supervised transfers using the Grandstream SIP phone. It isn't 
perfect, but it works and is very functional. Here are the steps:

When I try this, all goes well until, after putting the original 
caller on hold and then getting a dialtone, I dial another extension, 
and then get these errors on the CLI:

find_user: Call from user 'btel' rejected due to usage limit of 1
find_user: Call from user 'btel' rejected due to usage limit of 1
find_user: Call from user 'btel' rejected due to usage limit of 1
find_user: Call from user 'btel' rejected due to usage limit of 1
Then the Grandstream gives me a busy, and my orignal caller is a zombie.

What am I doing wrong?

Thx.

B.
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check sip.conf:

incominglimit=1
outgoinglimit=1


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Re: [Asterisk-Users] problem to place calls to NIKOTEL

2004-03-05 Thread Jakob Strebel
Phillipp,

thank you for your help.

At 00:57 04.03.2004 +0100, Philipp von Klitzing wrote:
Hi!

 exten = _8.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,r)

Use this instead:
exten = _8.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,r)
this works, in the meantime I do understand also why it did send the wrong 
call setup packet.
The not working Dial command did not take the variables form the context 
nikotel in the sip.conf, because this context was not called.

I was a bit confused by the examples.
jakob  

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[Asterisk-Users] how to disable zap debug!!!

2004-03-05 Thread atif
how to disable this DEBUG information...
I am getting this on Asterisk CLI

---
Mar  5 16:18:17 DEBUG[426001]: chan_zap.c:3397 __zt_exception: Exception on 22, 
channel 4
Mar  5 16:18:17 DEBUG[426001]: chan_zap.c:2765 zt_handle_event: Got event On hook(1) 
on channel 4 (index 0)
Mar  5 16:18:17 DEBUG[426001]: chan_zap.c:1133 zt_disable_ec: disabled echo 
cancellation on channel 4
Mar  5 16:18:17 DEBUG[426001]: channel.c:2275 ast_channel_bridge: Didn't get a frame 
from channel: Zap/4-1
Mar  5 16:18:17 DEBUG[426001]: channel.c:2343 ast_channel_bridge: Bridge stops 
bridging channels Zap/3-1 and Zap/4-1
Mar  5 16:18:17 DEBUG[426001]: chan_zap.c:1715 zt_hangup: Hangup: channel: 4 index = 
0, normal = 22, callwait = -1, thirdcall = -1
Mar  5 16:18:17 DEBUG[426001]: chan_zap.c:1133 zt_disable_ec: disabled echo 
cancellation on channel 4
Mar  5 16:18:17 DEBUG[426001]: chan_zap.c:2095 zt_setoption: Set option TDD MODE, 
value: OFF(0) on Zap/4-1
Mar  5 16:18:17 DEBUG[426001]: chan_zap.c:1076 update_conf: Updated conferencing on 4, 
with 0 conference users
-- Hungup 'Zap/4-1'
---

--
Atif Rasheed
Convergence (Business Systems)
http://www.convergence.com.pk
--
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[Asterisk-Users] Can't find capi.conf syntax to use 2 controllers

2004-03-05 Thread Prima Informatica
Hello,

I'm trying to configure asterisk with 2 controller (one is a AVM fxusb, 
the other is AVM fcpci). Each controller is bound to a separate BRI ISDN 
line.

The two modules are correctly loaded and configured by capinit:

# capiinit status
1 fxusb  running  fritz-usbA1 3.10-02 2
2 fcpci  running  fritz-pciA1 3.11-02 0xe800 10
I've configure capi.conf as follows:

[interfaces]
msn=041534,041534
incomingmsn=*
controller=1,2
softdtmf=0
devices=2,2
Asterisk reports:

*CLI capi info
Contr1: 2 B channels total, 2 B channels free.
Contr2: 2 B channels total, 2 B channels free.
So, apparently everthing is fine, but incoming calls form second msn are 
not detected by asterisk.

I not able to find where support for multiple controllers is documented.

May anyone help please?

Thanks in advance.

Luca Azzalini



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Re: [Asterisk-Users] how to disable zap debug!!!

2004-03-05 Thread Jean-Marc V. Liotier
On Fri, 2004-03-05 at 12:18, atif wrote:
 how to disable this DEBUG information...

I would have intuitively said 'zap no debug' but apparently the 'no
debug' is not implemented for zap although it exist for sip, iax, h323,
skinny and mgcp. Should we consider this absence as a bug worthy of a
wishlist item ?



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Re: [Asterisk-Users] how to disable zap debug!!!

2004-03-05 Thread Tilghman Lesher
On 2004 Mar 05, at 05:18, atif wrote:

how to disable this DEBUG information...
I am getting this on Asterisk CLI
---
Mar  5 16:18:17 DEBUG[426001]: chan_zap.c:3397 __zt_exception: 
Exception on 22, channel 4
Edit /etc/asterisk/logger.conf.  On the line beginning with console,
remove the debug item, then issue the command logger restart.
-Tilghman

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Re: [Asterisk-Users] segfault and backtrace info

2004-03-05 Thread Tilghman Lesher
On 2004 Mar 04, at 16:12, Philipp von Klitzing wrote:

#0  0x080570a6 in ast_queue_frame (chan=0x810a770, fin=0x41dfd0cc,
lock=1)
at channel.c:368
368 cur = chan-pvt-readq;
(gdb) bt
Post a 'bt full' to bugs.digium.com.

-Tilghman

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[Asterisk-Users] ?Application Hardware Recommendation

2004-03-05 Thread Hopper
Greetings All!

I just found Asterisk and am new this list.

I have a home office that I have an old DOS based IVR Telephone System
(Telepro) running on an old 386 pc and Dialogic D41D board.  I would like 
to upgrade
to something new and came across Asterisk.  I have read the documentation 
and it does
appear very flexible, but I thought I would run my questions by the list, 
before I jump in.

My current setup looks something like this.

Extensions
When someone enters an extension it will either be routed to an outside
number, for example
Joe is at ext 400.  Caller enters 400 and is placed on hold while phone
system dials Joe's Cell phone if Joe does not answer phone system tries his
home number.  If Joe answers, phone system announce, I have a call for you,
Press 1 to Accept Press 2 To Decline.  Joe presses 1, call is transferred
to him.  Joe presses 2, call is sent to next number, if no further numbers,
call is sent vm.
Or virtually, for example

Caller presses 1 for general sales.  Screen Prompt appears stating Sales
Call Accept/Busy.  Pickup phone click accept and call is released.  Click
busy or prompt times out (no response) call is routed to next number or to vm
Does Asterisk make sense as a replacement for this system?

What hardware would you recommend?

I have two incoming centrix lines.

Thanks for any info.

~Hopper

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Re: [Asterisk-Users] how to disable zap debug!!!

2004-03-05 Thread John Fraizer
Tilghman Lesher wrote:

 On 2004 Mar 05, at 05:18, atif wrote:

 how to disable this DEBUG information...
 I am getting this on Asterisk CLI

 ---
 Mar  5 16:18:17 DEBUG[426001]: chan_zap.c:3397 __zt_exception:
 Exception on 22, channel 4


 Edit /etc/asterisk/logger.conf.  On the line beginning with console,
 remove the debug item, then issue the command logger restart.

 -Tilghman
That should be logger reload and not restart.

John

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[Asterisk-Users] H323 termination to Cisco 5300

2004-03-05 Thread Senad Jordanovic
Hi,

While terminating calls to Cisco 5300 the called party hears converstion
all OK.
However, calling party hears periodic short bursts of interferance
and/or lost packets noise.

I can see on CLI this:

Mar  5 14:35:53 DEBUG[458773]: rtp.c:943 ast_rtp_raw_write: Difference
is 3760, ms is 490
Mar  5 14:35:53 DEBUG[458773]: rtp.c:943 ast_rtp_raw_write: Difference
is 3200, ms is 420
Mar  5 14:35:55 DEBUG[458773]: rtp.c:943 ast_rtp_raw_write: Difference
is 704, ms is 108
Mar  5 14:35:55 DEBUG[458773]: rtp.c:943 ast_rtp_raw_write: Difference
is 952, ms is 139

And so on...

I have come across above message before but I just can not remember what
it is for...

Any pointers would be greatly appreciated.

Ta
SJ  


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Re: [Asterisk-Users] how to disable zap debug!!!

2004-03-05 Thread Tilghman Lesher
On 2004 Mar 05, at 08:28, John Fraizer wrote:

Tilghman Lesher wrote:

 On 2004 Mar 05, at 05:18, atif wrote:

 how to disable this DEBUG information...
 I am getting this on Asterisk CLI

  
---
 Mar  5 16:18:17 DEBUG[426001]: chan_zap.c:3397 __zt_exception:
 Exception on 22, channel 4


 Edit /etc/asterisk/logger.conf.  On the line beginning with  
console,
 remove the debug item, then issue the command logger restart.

That should be logger reload and not restart.
Ah, yes.  Blame me for always typing logtabretabenter

-Tilghman

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RE: [Asterisk-Users] 3com NBX phones

2004-03-05 Thread daryl
 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Nicolas Bougues
 Sent: Friday, March 05, 2004 3:17 AM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] 3com NBX phones
 
 
[...]
 Note that the hardware is probably not the same as the 
 standard NBX phones : my SIP phones did feature an IR sensor 
 to be used by a Palm for automated dialing.

That's actually an option on the better NBX phones...your is probably a
2102-IR or similar, and has been since at least when I did my last NBX
rollout about a year and a half ago.  What seems different is that you
could flash it at all.  When connecting to an NBX, these phones grab
their firmware from the NBX they pin up to.  I suppose there is a
flashable area on the phone that is used as a boot loader in NBX mode,
and probably to store the whole image when flashed with SIP.

Can anyone confirm these are the same phones?  Because I still have
boxes of them somewhere too (that seems to be a common thread here).
Daryl
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[Asterisk-Users] Ahead SIPPS and Asterisk

2004-03-05 Thread Carlton J. O'Riley
Has anyone gotten Ahead's SIPPS softphone to work with Asterisk?  I get it
to register, but when I dial into a number as soon as voice is to be
connected I get a Warning 399 SDP body missing message and then a BYE
disconnecting the call.  The setup I have works great with Xten's x-pro, but
can't get it to work with SIPPS.  Any hints?
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[Asterisk-Users] OT: Snom 105

2004-03-05 Thread Eric Hendrickson
Greets,

Apologies for the OT post. I'm working with a Snom 105 and can't seem to 
find the Administrator's Manual for this phone on Snom's website. Does 
anyone know where to find this document? Anyone know how to perform a 
factory reset on this device? After upgrading the firmware to 2.03o, 
it appears that I'm locked out of the administrative menu.

--

Eric Hendrickson
Sr. Solutions Analyst
Iophase Inc.
19200 Von Karman, Ste. 400
Irvine, California 92612
949-608-1770 x202
888-627-6273


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Re: [Asterisk-Users] 3com NBX phones

2004-03-05 Thread Clif Jones
The IR device is a 3rd-party piece of hardware from Extended System (now 
owned by
iFoundry).  The SIP phone looks like all of the other 3com IP phones 
that I have seen
and turning it over with the front of the phone facing up the connectors 
go from left to
right as follows:
1. Handset connector
2. IRDA (serial) RJ-45 connector
3. PC Ethernet RJ-45 connector
4. Wall Ethernet RJ-45 connector
5. Power adapter

Maybe this will help in comparing the units.  I have posted my last SIP 
firmware
(with appropriate disclaimers) to the list but it is held up in 
moderator no man's land.

[EMAIL PROTECTED] wrote:

-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
Nicolas Bougues
Sent: Friday, March 05, 2004 3:17 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] 3com NBX phones

   

[...]
 

Note that the hardware is probably not the same as the 
standard NBX phones : my SIP phones did feature an IR sensor 
to be used by a Palm for automated dialing.
   

That's actually an option on the better NBX phones...your is probably a
2102-IR or similar, and has been since at least when I did my last NBX
rollout about a year and a half ago.  What seems different is that you
could flash it at all.  When connecting to an NBX, these phones grab
their firmware from the NBX they pin up to.  I suppose there is a
flashable area on the phone that is used as a boot loader in NBX mode,
and probably to store the whole image when flashed with SIP.
Can anyone confirm these are the same phones?  Because I still have
boxes of them somewhere too (that seems to be a common thread here).
Daryl
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Re: [Asterisk-Users] 3com NBX phones

2004-03-05 Thread Rob Fugina
On Fri, Mar 05, 2004 at 10:25:49AM -0500, Clif Jones wrote:
 The IR device is a 3rd-party piece of hardware from Extended System (now 
 owned by
 iFoundry).  The SIP phone looks like all of the other 3com IP phones 
 that I have seen
 and turning it over with the front of the phone facing up the connectors 
 go from left to
 right as follows:
 1. Handset connector
 2. IRDA (serial) RJ-45 connector
 3. PC Ethernet RJ-45 connector
 4. Wall Ethernet RJ-45 connector
 5. Power adapter
 
 Maybe this will help in comparing the units.  I have posted my last SIP 
 firmware
 (with appropriate disclaimers) to the list but it is held up in 
 moderator no man's land.

How about part numbers, model numbers, revisions, etc, from the labelling
on the phone?

Rob

-- 
Rob Fugina, Systems Guy
[EMAIL PROTECTED] -- http://www.geekthing.com
My firewall filters MS Office attachments.

We aren't surrounded. We're in a target-rich environment.
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[Asterisk-Users] Not able to dial 9 to get out with SIP Grandstream BudgeTone-100 or SIP softphone

2004-03-05 Thread Stephen Foster








Hi everyone,

 I
am having problems dialing 9 to get an external line with my SIP
phones or SIP clients. I have been looking for months on websites, sitting in
MIRC rooms, and reading * documentation but I cannot seem to find a solution.



My asterisk box is sitting directly on the internet ( NO NAT ) with a firewall. I have also tested this box on
my LAN and I have the same issue ( this is not a
firewall issue ). I am using a T-100P card and an Adtran
Total Access unit for all my analog phones which for now is all I use.



My Grand stream SIP phone works fine for calling internal
extensions with no problems at all. When I try and dial 9 and a
number, after a wait of a few seconds I get  404
 displayed on the screen and a busy signal. I have tried to tweak
everything I know within the dial plan, but I always seem to have the same
issue. 



I previously tried to attach my sip and extensions.conf
but the email is too big for the mailing list. I have pasted small sections of
them below.



Id very much appreciate any help anyone can provide.



SIP Conf



[gs01]

type=friend

username=gs01

secret=pass

nat=1

host=dynamic

qualify=yes

dtmfmode=info

canreinvite=no



EXTENSIONS.CONF



[general]

static=yes

writeprotect=no



[globals]

CONSOLE=Console/dsp

TRUNK=Zap/g2
RINGOUT=Zap/14Zap/7Zap/8Zap/9Zap/10Zap/11Zap/12



[trunkint]

exten
= _9011.,1,Dial(${TRUNK}/www${EXTEN:1})

exten
= _9011.,2,Congestion



[trunkld]

exten
= _91NXXNXX,1,Dial(${TRUNK}/www${EXTEN:1})

exten
= _91NXXNXX,2,Congestion



[trunklocal]

exten
= _9NXXNXX,1,Dial(${TRUNK}/www${EXTEN:1})

exten
= _9NXXNXX,2,Congestion

exten
= 9411,1,Dial(${TRUNK}/www${EXTEN:1})

exten
= 9411,2,Congestion

exten
= 9911,1,Dial(${TRUNK}/www${EXTEN:1})

exten
= 9911,2,Congestion



[local]

;trusted
users only!

ignorepat
= 9

include
= default

include
= parkedcalls

include
= trunklocal

include
= trunktollfree

include
= trunkint

include
= trunkld

include
= phones

include
= voicemail

include
= recording



[macro-stdexten]

exten
= s,1,Dial(${ARG2},20)

exten
= s,2,Voicemail2(u${ARG1})

exten
= s,3,Goto(default,s,1)

exten
= s,102,Voicemail2(b${ARG1})

exten
= s,103,Goto(default,s,1)



[phones]

exten
= 200,1,Macro(stdexten,200,Zap/10)



;SIP
phones

;Grandstream
Phones

exten
= 210,1,Dial(SIP/gs01)

exten
= 222,1,Dial(SIP/bradwell)

exten
= _64xx,1,Dial(SIP/gs${EXTEN:2}|20)

exten
= _64xx,2,Voicemail2(u${ARG1})

exten
= _64xx,3,Congestion

exten
= _64xx,102,Voicemail2(b${ARG1})

exten
= _64xx,103,Congestion



[sipstart]

include
= phones

include
= voicemail

include
= default

include
= trunklocal

include
= trunktollfree



Thanks,

 Steve [EMAIL PROTECTED]










Re: [Asterisk-Users] OT: Snom 105

2004-03-05 Thread Eric Hendrickson
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Thanks Sven. I think I've made things worse now :-0  Playing around with
some key combinations while booting the phone, I think I've disabled the
display. I see the Snom logo briefly at the bootloader but after that
nothing is displayed on the panel. The backlight comes on when I pick up
the receiver and the phone does work--just no display.
Any suggestions? ... Other than to quit mucking around in places unknown.

Sven Fischer wrote:

| Hi,
|
| the default value to come back to admin mode for all snom phones is
 (for
| times zero). You are right, there is an admin manual for snom200
available
| only at the moment. Please see that for questions.
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.2 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
iD8DBQFASKlDbziSUSbgS2kRAllhAJ4+yp0V4EdALVHn3FN1VwIPa7HFswCeOPyz
1yF8nwRBGm+XVUvWAxS6R+E=
=+yu1
-END PGP SIGNATURE-
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Re: [Asterisk-Users] OT: Snom 105

2004-03-05 Thread Mike Machado
I could not find one either, but I did find one for the 200 series
phones, which seemed similar. That is where I found the default admin
password once I locked myself out. Check the 200 series product page.

On Fri, 2004-03-05 at 07:18, Eric Hendrickson wrote:
 Greets,
 
 Apologies for the OT post. I'm working with a Snom 105 and can't seem to 
 find the Administrator's Manual for this phone on Snom's website. Does 
 anyone know where to find this document? Anyone know how to perform a 
 factory reset on this device? After upgrading the firmware to 2.03o, 
 it appears that I'm locked out of the administrative menu.

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Re: [Asterisk-Users] flash button on GS101

2004-03-05 Thread Dave Cotton
On Thu, 2004-03-04 at 23:21, Steven Critchfield wrote:
 On Fri, 2004-03-05 at 02:04, dkwok wrote:
  Has anyone using the flash button on GS101 to access call waiting?
  
  My experience is that it does not work. I read in the list that it may 
  need to tweak the flash duration to under 100msec. Has anyone have any 
  solution?
 
 GS, as in Grandstream Budgetone? If so, then the flash is sent out of
 band. It has been mentioned that the quality can be questionable and the
 flash button may not be functional due to quality control issues. Please
 verify with a sniff of the wire using something like ethereal to verify
 whether or not the button press actually hits the wire in a meaningful
 manner.

This is the end of what my GS is sending when I press flash so at least
the button is working :-

 INFO..Use
  r-Agent: Grandstream BT100 1.0.4.46..Max-Forwards: 70..Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIB
  E..Content-Type: application/dtmf-relay..Content-Length:
25Signal=16..Duration=40875

but is it meaningful?
-- 
Dave Cotton [EMAIL PROTECTED]

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Re: [Asterisk-Users] OT: Snom 105

2004-03-05 Thread Eric Hendrickson
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Yes, I can get to the webpage. And the phone plays its happy tune after
it boots up. Just no display on the LCD panel.
Sven Fischer wrote:

| Eric,
|
| the phone is working ? Can you access the webpage ?
|
| regards,
|
| Sven
|
| On Friday 05 March 2004 17:22, Eric Hendrickson wrote:
|
|Thanks Sven. I think I've made things worse now :-0  Playing around with
|some key combinations while booting the phone, I think I've disabled the
|display. I see the Snom logo briefly at the bootloader but after that
|nothing is displayed on the panel. The backlight comes on when I pick up
|the receiver and the phone does work--just no display.
|
|Any suggestions? ... Other than to quit mucking around in places unknown.
|
|Sven Fischer wrote:
|| Hi,
||
|| the default value to come back to admin mode for all snom phones is
|
| (for
|
|| times zero). You are right, there is an admin manual for snom200
|
|available
|
|| only at the moment. Please see that for questions.
|
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|
- --
Eric Hendrickson
Sr. Solutions Analyst
Iophase Inc.
19200 Von Karman, Ste. 400
Irvine, California 92612
949-608-1770 x202
888-627-6273
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.2 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
iD8DBQFASLQbbziSUSbgS2kRAt+WAJ4yrw6libJTqJD/czxo8w7HAe+FmwCfcO0y
p7MDqoiM0rfSX0pvJAFy97k=
=G0u6
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[Asterisk-Users] Kernel - TE410P

2004-03-05 Thread Tomica Crnek



Hi 
everyone,

Is there something 
that needs to be done with RedHat 9 kernel prior to installing TE410P and 
loading wct4xxp zap module? I mean, is there a kernel patch or something else 
that must be installed, or is it enough just to compile zaptel 
driver?

thanks
Tomica



[Asterisk-Users] dropped calls

2004-03-05 Thread Paulo Loureiro
Hello list,

I'm getting droped calls on an asterisk installation. When on GS phone
dials another one, the call is dropped after some (usually random) time
but most of the tome within 3 to 20 seconds.
I think the cause is stated on the logs, see bellow, and is related with
the message  Didn't get a frame from channel: SIP/3805-df43, but I
can't figure why.


asterisk logs:
-
Mar  5 15:57:26 DEBUG[1116957488]: build_route: Contact hop:
sip:192.168.60.106
Mar  5 15:57:26 VERBOSE[1217669936]: -- SIP/3805-df43 answered
SIP/-08122450
Mar  5 15:57:26 VERBOSE[1217669936]: -- Attempting native bridge of
SIP/-08122450 and SIP/3805-df43
Mar  5 15:57:26 DEBUG[1116957488]: Stopping retransmission on
'[EMAIL PROTECTED]' of Response\ 25663: Found
Mar  5 15:57:26 DEBUG[1217669936]: Difference is 4032, ms is 524
Mar  5 15:57:26 DEBUG[1217669936]: Ooh, format changed from UNKN to ULAW
Mar  5 15:57:38 DEBUG[1217669936]: Didn't get a frame from channel:
SIP/3805-df43
Mar  5 15:57:38 DEBUG[1217669936]: Bridge stops bridging channels
SIP/-08122450 and SIP/3805-df43
Mar  5 15:57:38 DEBUG[1217669936]: find_user() - decrement outUse
counter
Mar  5 15:57:38 DEBUG[1217669936]:  is not a local user
Mar  5 15:57:38 VERBOSE[1217669936]:   == Spawn extension (local, 3805,
1) exited non-zero on 'SIP/-0812245\0'
-

The scenario:
1 server (redhat 9),  asterisk (stable) and a 2 x hisax ISDN BRI.
One of the BRI boards is used to dial out (ppp) on one channel and a
mgetty on the other channel. The other board is in ptp and used by *.
The phones are Grandstream BT101 and Handytone and are all on a switched
network (3 procurve switches, stacked).

The configs are ok, since the same files on another server work ok (no
dropped calls), but I can post them if needed.


Any help will be greatly appreciated.

Thanks in advance,



--- Paulo Loureiro


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RE: [Asterisk-Users] Kernel - TE410P

2004-03-05 Thread mattf
compiling zaptel is enough, 

I have one and it loaded up just fine on RH9, make sure you modprobe zaptel,
modprobe wct4xxp and then ztcfg -vvv before you try starting the first time.

MATT---

-Original Message-
From: Tomica Crnek [mailto:[EMAIL PROTECTED]
Sent: Friday, March 05, 2004 12:20 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Kernel - TE410P


Hi everyone,
 
Is there something that needs to be done with RedHat 9 kernel prior to
installing TE410P and loading wct4xxp zap module? I mean, is there a kernel
patch or something else that must be installed, or is it enough just to
compile zaptel driver?
 
thanks
Tomica
 
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[Asterisk-Users] SIP = Zaptel TDM400P issue

2004-03-05 Thread Dustin Mulcahey
Situation:
SIP phone A calls Asterisk.
Asterisk forwards to another SIP agent B.
The SIP agent B forwards A back to an Asterisk extension that is mapped
to a TDM400P channel

Once all of this has transpired, there is no audio channel between the
SIP phone A and the TDM400P.

If A is not registered with Asterisk performs the same sequence, a half-
duplex connection results in which A can hear the TDM400P, but not the
other way around.

If the above procedure is repeated in reverse, that is:
TDM400P dials the extension of SIP agent B.
SIP agent B forwards TDM400P to the asterisk extension of SIP A.

A full duplex connection results, and everything is normal.

Any ideas?  Elaborations needed?

===sip.conf===
[general]
port = 5060 ; Port to bind to
bindaddr = 192.168.0.104; Address to bind to
context = from-sip-internal
disallow=all
allow=ulaw

[201]
username=201
type=friend
secret=password
host=dynamic
dtmfmode=inband
===

201 is SIP phone A.

-Dustin Mulcahey

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Re: [Asterisk-Users] Not able to dial 9 to get out with SIP Grandstream BudgeTone-100 or SIP softphone

2004-03-05 Thread Tim Robinson
From a brief look, it seems you do not have a context= in your sip.conf 
file for the extension.  If you don't put a contxt in, I don't know what 
it assumes, and it will not include the contexts you have set up to 
define external access.

From looking at your dialplan, if you put context=local into the [gs01] 
entry in sip.conf, you should be able to make outbound calls from this 
extension, as you will be forced into 'local' context and will be able 
se see all the external access contexts you have defined.

Let us know how you get on...

Rgds
Tim Robinson
Basingstoke, UK
Stephen Foster wrote:
Hi everyone,

I am having problems dialing 9 to get an 
external line with my SIP phones or SIP clients. I have been looking for 
months on websites, sitting in MIRC rooms, and reading * documentation 
but I cannot seem to find a solution.

 

My asterisk box is sitting directly on the internet ( NO NAT ) with a 
firewall. I have also tested this box on my LAN and I have the same 
issue ( this is not a firewall issue ). I am using a T-100P card and an 
Adtran Total Access unit for all my analog phones which for now is all I 
use.

 

My Grand stream SIP phone works fine for calling internal extensions 
with no problems at all. When I try and dial 9 and a number, after a 
wait of a few seconds I get  404  displayed on the screen and a busy 
signal. I have tried to tweak everything I know within the dial plan, 
but I always seem to have the same issue.

 

I previously tried to attach my sip and extensions.conf but the email is 
too big for the mailing list. I have pasted small sections of them below.

 

Id very much appreciate any help anyone can provide.

 

SIP Conf

 

[gs01]

type=friend

username=gs01

secret=pass

nat=1

host=dynamic

qualify=yes

dtmfmode=info

canreinvite=no

 

EXTENSIONS.CONF

 

[general]

static=yes

writeprotect=no

 

[globals]

CONSOLE=Console/dsp

TRUNK=Zap/g2 RINGOUT=Zap/14Zap/7Zap/8Zap/9Zap/10Zap/11Zap/12

 

[trunkint]

exten = _9011.,1,Dial(${TRUNK}/www${EXTEN:1})

exten = _9011.,2,Congestion

 

[trunkld]

exten = _91NXXNXX,1,Dial(${TRUNK}/www${EXTEN:1})

exten = _91NXXNXX,2,Congestion

 

[trunklocal]

exten = _9NXXNXX,1,Dial(${TRUNK}/www${EXTEN:1})

exten = _9NXXNXX,2,Congestion

exten = 9411,1,Dial(${TRUNK}/www${EXTEN:1})

exten = 9411,2,Congestion

exten = 9911,1,Dial(${TRUNK}/www${EXTEN:1})

exten = 9911,2,Congestion

 

[local]

;trusted users only!

ignorepat = 9

include = default

include = parkedcalls

include = trunklocal

include = trunktollfree

include = trunkint

include = trunkld

include = phones

include = voicemail

include = recording

 

[macro-stdexten]

exten = s,1,Dial(${ARG2},20)

exten = s,2,Voicemail2(u${ARG1})

exten = s,3,Goto(default,s,1)

exten = s,102,Voicemail2(b${ARG1})

exten = s,103,Goto(default,s,1)

 

[phones]

exten = 200,1,Macro(stdexten,200,Zap/10)

 

;SIP phones

;Grandstream Phones

exten = 210,1,Dial(SIP/gs01)

exten = 222,1,Dial(SIP/bradwell)

exten = _64xx,1,Dial(SIP/gs${EXTEN:2}|20)

exten = _64xx,2,Voicemail2(u${ARG1})

exten = _64xx,3,Congestion

exten = _64xx,102,Voicemail2(b${ARG1})

exten = _64xx,103,Congestion

 

[sipstart]

include = phones

include = voicemail

include = default

include = trunklocal

include = trunktollfree

 

Thanks,

Steve[EMAIL PROTECTED]

 



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RE: [Asterisk-Users] Ahead SIPPS and Asterisk

2004-03-05 Thread Craig Waddington
What sort of phone are you trying to call?

I use SIPPS and * and it works fine, it just wont work when you call
Windows Messenger for some reason. I can call X-lite, POTS, GS phones no
problems.

I use the same config as X-lite, in SIPPS if you click on the spanner or
press F9 to go into configuration and click on network.

You should see aquired? 

Click modify and ensure Gateway is ticked and you have the IP of your
server in there, for some reason it defaults as redirect. On mine I
leave Dial prefix blank.

Click modify again then Ok, which will take you to the config section
for the user, choose use authentication, fill in the blanks, I found
also putting the realm in made it work correctly.

Hope that helps in some way.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Carlton J.
O'Riley
Sent: 05 March 2004 15:17
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Ahead SIPPS and Asterisk

Has anyone gotten Ahead's SIPPS softphone to work with Asterisk?  I get
it
to register, but when I dial into a number as soon as voice is to be
connected I get a Warning 399 SDP body missing message and then a BYE
disconnecting the call.  The setup I have works great with Xten's x-pro,
but
can't get it to work with SIPPS.  Any hints?
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Re: [Asterisk-Users] G.729 vs. G.729 pass thru

2004-03-05 Thread Unavailable ID
Derek,

Can you use fax with G.729?  I know that only ULAW codec can use for fax but
I don't know that if you can use fax with G.729 or not.

BTW, what service provider that you are using?  Quality can sometime depend
on provider too.

Thanks.

- Original Message - 
From: Derek Samford [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; [EMAIL PROTECTED]
Sent: Thursday, March 04, 2004 9:12 PM
Subject: RE: [Asterisk-Users] G.729 vs. G.729 pass thru


On all tests, I've run, fantastic. I haven't had any issues with voice
quality at all, even on analog lines.

Derek
-Original Message-
From: Unavailable ID [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 04, 2004 9:03 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] G.729 vs. G.729 pass thru

Darek,

Thank you for the info.

How is the sound quality when you are using with G.729 codec?  What's
your
thought?

Thanks.

- Original Message - 
From: Derek Samford [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; [EMAIL PROTECTED]
Sent: Thursday, March 04, 2004 2:58 PM
Subject: RE: [Asterisk-Users] G.729 vs. G.729 pass thru


As long as you have a IDE drive available, and mounted when you install
it,
it will work. This includes CD ROM's...It's what I did.
Funkiness with the registration process.
As far as pass through goes, from what I understand, it *should*, but
when
you have licensed binaries, from what I've seen, it doesn't. It's
actually
used 2 licenses. I plan on figuring that out next.

Derek


From: Unavailable ID [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 04, 2004 5:33 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] G.729 vs. G.729 pass thru

Hello everyone,

If you don't have Digium card but you want to use G.729 codec, do you
need a
license for it?

If the VoIP termination point supports G.729 and you are using sip phone
(soft/hard phone), can you use the G.729 pass thru or you have to buy
the
license?

Have anyone test it with SCSI system? Seems like it only work on machine
with IDE disk.

Thanks.
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RE: [Asterisk-Users] dropped calls

2004-03-05 Thread Ross Donaldson
There is new firmware that may help http://www.grandstream.com/BETATEST/.
Grandstream acknowledges this problem. They say it is a codec issue with
asterisk. I don't know if this update addresses this problem but it may be
worth a try.

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Paulo Loureiro
 Sent: Friday, March 05, 2004 10:26 AM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] dropped calls
 
 Hello list,
 
 I'm getting droped calls on an asterisk installation. When on GS phone
 dials another one, the call is dropped after some (usually 
 random) time
 but most of the tome within 3 to 20 seconds.
 I think the cause is stated on the logs, see bellow, and is 
 related with
 the message  Didn't get a frame from channel: SIP/3805-df43, but I
 can't figure why.
 
 
 asterisk logs:
 -
 Mar  5 15:57:26 DEBUG[1116957488]: build_route: Contact hop:
 sip:192.168.60.106
 Mar  5 15:57:26 VERBOSE[1217669936]: -- SIP/3805-df43 answered
 SIP/-08122450
 Mar  5 15:57:26 VERBOSE[1217669936]: -- Attempting native 
 bridge of
 SIP/-08122450 and SIP/3805-df43
 Mar  5 15:57:26 DEBUG[1116957488]: Stopping retransmission on
 '[EMAIL PROTECTED]' of Response\ 25663: Found
 Mar  5 15:57:26 DEBUG[1217669936]: Difference is 4032, ms is 524
 Mar  5 15:57:26 DEBUG[1217669936]: Ooh, format changed from 
 UNKN to ULAW
 Mar  5 15:57:38 DEBUG[1217669936]: Didn't get a frame from channel:
 SIP/3805-df43
 Mar  5 15:57:38 DEBUG[1217669936]: Bridge stops bridging channels
 SIP/-08122450 and SIP/3805-df43
 Mar  5 15:57:38 DEBUG[1217669936]: find_user() - decrement outUse
 counter
 Mar  5 15:57:38 DEBUG[1217669936]:  is not a local user
 Mar  5 15:57:38 VERBOSE[1217669936]:   == Spawn extension 
 (local, 3805,
 1) exited non-zero on 'SIP/-0812245\0'
 -
 
 The scenario:
 1 server (redhat 9),  asterisk (stable) and a 2 x hisax ISDN BRI.
 One of the BRI boards is used to dial out (ppp) on one channel and a
 mgetty on the other channel. The other board is in ptp and used by *.
 The phones are Grandstream BT101 and Handytone and are all on 
 a switched
 network (3 procurve switches, stacked).
 
 The configs are ok, since the same files on another server work ok (no
 dropped calls), but I can post them if needed.
 
 
 Any help will be greatly appreciated.
 
 Thanks in advance,
 
 
 
 --- Paulo Loureiro
 
 
 

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RE: [Asterisk-Users] Ahead SIPPS and Asterisk

2004-03-05 Thread Carlton J. O'Riley
I'll try to look over my config again.  Not sure I put a realm in, but
everything else seemed fine.  I get the acquired message and I see the SIP
messages flowing on the Asterisk server, but once any sound needs to be
sent, it dies.  I'm using g711ulaw and I was calling into an announcement
menu to test that I have setup on the server.  The entry in my sip.conf is
exactly what I use for X-Pro, so that is why I am confused by the missing
SDP message.

-Original Message-
Message: 8
Subject: RE: [Asterisk-Users] Ahead SIPPS and Asterisk
Date: Fri, 5 Mar 2004 18:25:05 -
From: Craig Waddington [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]

What sort of phone are you trying to call?

I use SIPPS and * and it works fine, it just wont work when you call
Windows Messenger for some reason. I can call X-lite, POTS, GS phones no
problems.

I use the same config as X-lite, in SIPPS if you click on the spanner or
press F9 to go into configuration and click on network.

You should see aquired?=20

Click modify and ensure Gateway is ticked and you have the IP of your
server in there, for some reason it defaults as redirect. On mine I
leave Dial prefix blank.

Click modify again then Ok, which will take you to the config section
for the user, choose use authentication, fill in the blanks, I found
also putting the realm in made it work correctly.

Hope that helps in some way.
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RE: [Asterisk-Users] Ahead SIPPS and Asterisk

2004-03-05 Thread Carlton J. O'Riley
I'll try to look over my config again.  Not sure I put a realm in, but
everything else seemed fine.  I get the acquired message and I see the SIP
messages flowing on the Asterisk server, but once any sound needs to be
sent, it dies.  I'm using g711ulaw and I was calling into an announcement
menu to test that I have setup on the server.  The entry in my sip.conf is
exactly what I use for X-Pro, so that is why I am confused by the missing
SDP message.

-Original Message-
Message: 8
Subject: RE: [Asterisk-Users] Ahead SIPPS and Asterisk
Date: Fri, 5 Mar 2004 18:25:05 -
From: Craig Waddington [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]

What sort of phone are you trying to call?

I use SIPPS and * and it works fine, it just wont work when you call
Windows Messenger for some reason. I can call X-lite, POTS, GS phones no
problems.

I use the same config as X-lite, in SIPPS if you click on the spanner or
press F9 to go into configuration and click on network.

You should see aquired?=20

Click modify and ensure Gateway is ticked and you have the IP of your
server in there, for some reason it defaults as redirect. On mine I
leave Dial prefix blank.

Click modify again then Ok, which will take you to the config section
for the user, choose use authentication, fill in the blanks, I found
also putting the realm in made it work correctly.

Hope that helps in some way.
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[Asterisk-Users] Ahead SIPPS and Asterisk

2004-03-05 Thread Carlton J. O'Riley
Has anyone gotten Ahead's SIPPS softphone to work with Asterisk?  I get it
to register, but when I dial into a number as soon as voice is to be
connected I get a Warning 399 SDP body missing message and then a BYE
disconnecting the call.  The setup I have works great with Xten's x-pro, but
can't get it to work with SIPPS.  Any hints?
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RE: [Asterisk-Users] dropped calls

2004-03-05 Thread Paulo Loureiro
Hello,

I'll try that, but why on earth gs phones with the same firmware on
another * server, work with no problem?

I've failed to state I'm using zaprtc, since there is no digium hardware
on the server. Does it matter?

Thanks,

--- Paulo.


On Fri, 2004-03-05 at 18:49, Ross Donaldson wrote:
 There is new firmware that may help http://www.grandstream.com/BETATEST/.
 Grandstream acknowledges this problem. They say it is a codec issue with
 asterisk. I don't know if this update addresses this problem but it may be
 worth a try.
 
  -Original Message-
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of 
  Paulo Loureiro
  Sent: Friday, March 05, 2004 10:26 AM
  To: [EMAIL PROTECTED]
  Subject: [Asterisk-Users] dropped calls
  
  Hello list,
  
  I'm getting droped calls on an asterisk installation. When on GS phone
  dials another one, the call is dropped after some (usually 
  random) time
  but most of the tome within 3 to 20 seconds.
  I think the cause is stated on the logs, see bellow, and is 
  related with
  the message  Didn't get a frame from channel: SIP/3805-df43, but I
  can't figure why.
  
  
  asterisk logs:
  -
  Mar  5 15:57:26 DEBUG[1116957488]: build_route: Contact hop:
  sip:192.168.60.106
  Mar  5 15:57:26 VERBOSE[1217669936]: -- SIP/3805-df43 answered
  SIP/-08122450
  Mar  5 15:57:26 VERBOSE[1217669936]: -- Attempting native 
  bridge of
  SIP/-08122450 and SIP/3805-df43
  Mar  5 15:57:26 DEBUG[1116957488]: Stopping retransmission on
  '[EMAIL PROTECTED]' of Response\ 25663: Found
  Mar  5 15:57:26 DEBUG[1217669936]: Difference is 4032, ms is 524
  Mar  5 15:57:26 DEBUG[1217669936]: Ooh, format changed from 
  UNKN to ULAW
  Mar  5 15:57:38 DEBUG[1217669936]: Didn't get a frame from channel:
  SIP/3805-df43
  Mar  5 15:57:38 DEBUG[1217669936]: Bridge stops bridging channels
  SIP/-08122450 and SIP/3805-df43
  Mar  5 15:57:38 DEBUG[1217669936]: find_user() - decrement outUse
  counter
  Mar  5 15:57:38 DEBUG[1217669936]:  is not a local user
  Mar  5 15:57:38 VERBOSE[1217669936]:   == Spawn extension 
  (local, 3805,
  1) exited non-zero on 'SIP/-0812245\0'
  -
  
  The scenario:
  1 server (redhat 9),  asterisk (stable) and a 2 x hisax ISDN BRI.
  One of the BRI boards is used to dial out (ppp) on one channel and a
  mgetty on the other channel. The other board is in ptp and used by *.
  The phones are Grandstream BT101 and Handytone and are all on 
  a switched
  network (3 procurve switches, stacked).
  
  The configs are ok, since the same files on another server work ok (no
  dropped calls), but I can post them if needed.
  
  
  Any help will be greatly appreciated.
  
  Thanks in advance,
  
  
  
  --- Paulo Loureiro
  
  
  
 
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Netmania

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[Asterisk-Users] Asterisk fault tolerance and a embedded hardware solution.....??

2004-03-05 Thread Randall Shimizu
Hello,

IMHO you have a problem with the hardware that Asterisk runs on.
You should really look around because there are a number of companies
selling intel based systems with a cPCI bus fully hot swap capable.
I think the only problem would be getting network adapters compatible with
* but then this is only a problem of drivers easily solved by a good
programmer.

If you test out Asterisk on a fully redundant box and you find problems I
think you'd be welcome to send in a patch to fix them so that * could
be used in enterprise computing instead of sending in a two page e-mail
with the problems we all know about !

Ok Sorry, I if the email was a bit provocative. I was just trying to get some 
suggestions  thoughts about hardware and fault tolerance with Asterisk. 

Regards

Kiss Karoly

On Thu, 4 Mar 2004, Randall Shimizu wrote:

 Asterisk fault tolerance and a embedded hardware solution.??

 Has anyoone tried implement Asterisk as a hardware based solution similar to Soekris 
 firewall?


 Asterisk  fault tolerance: I ran across this posting about Asterisk
 and here is some interesting thoughts to ponder



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[Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #3012 - 11 msgs

2004-03-05 Thread Carlton O'Riley
I'm at a little bit of a loss here.  I'm going to enclose my SIP output for
this session that hopefully someone knows why I get the SDP not available
message when using SIPPS to Asterisk.  It registers great and when I call
SIPPS it rings, but when it answers I get the same problem with the SDP not
available message.  Any help would be greatly appreciated.

Thanks...

-  SIP Session messages ---
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via:   SIP/ 2.0/ UDP  192.168.3.69
;branch=z9hG4bKnp492417237-4578560e192.168.3.69
From:   sip:[EMAIL PROTECTED] ;tag=1d59b0b2
To:   sip:[EMAIL PROTECTED]
Call-ID:   [EMAIL PROTECTED]
CSeq:   1  INVITE
User-Agent:   Ahead SIPPS IP Phone Version 2.0.45.18
Expires:   180
Accept:   application/ sdp
Content-Type:   application/ sdp
Content-Length:   243
Contact:   sip:[EMAIL PROTECTED]
v=0
o=SIPPS 492417159 492417162 IN IP4 192.168.3.69
s=SIP call
c=IN IP4 192.168.3.69
t=0 0
m=audio 1 RTP/AVP 0 8 97 2 3
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000



SIP/2.0 407 Proxy Authentication Required
Via:   SIP/ 2.0/ UDP  192.168.3.69
;branch=z9hG4bKnp492417237-4578560e192.168.3.69
From:   sip:[EMAIL PROTECTED] ;tag=1d59b0b2
To:   sip:[EMAIL PROTECTED] ;tag=as6b4bb6b5
Call-ID:   [EMAIL PROTECTED]
CSeq:   1  INVITE
User-Agent:   Asterisk PBX
Allow:   INVITE
Allow:   ACK
Allow:   CANCEL
Allow:   OPTIONS
Allow:   BYE
Allow:   REFER
Contact:   sip:[EMAIL PROTECTED]
Proxy-Authenticate:   Digest  realm=asterisk,nonce=6c08deed
Content-Length:   0

INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via:   SIP/ 2.0/ UDP  192.168.3.69
;branch=z9hG4bKnp492418173-4587980e192.168.3.69
From:   sip:[EMAIL PROTECTED] ;tag=1d59b0b2
To:   sip:[EMAIL PROTECTED]
Call-ID:   [EMAIL PROTECTED]
CSeq:   2  INVITE
Proxy-Authorization:   Digest
username=101,realm=asterisk,uri=sip:192.168.3.69,nonce=6c08deed,nc=
0001,response=f058d443e9cebe0b23ca1adf0db21afb
Content-Type:   application/ sdp
Content-Length:   243
Date:   Fri, 05 Mar 2004 19:16:01 GMT
Contact:   sip:[EMAIL PROTECTED]
Expires:   180
Accept:   application/ sdp
User-Agent:   Ahead SIPPS IP Phone Version 2.0.45.18
v=0
o=SIPPS 492417159 492417162 IN IP4 192.168.3.69
s=SIP call
c=IN IP4 192.168.3.69
t=0 0
m=audio 1 RTP/AVP 0 8 97 2 3
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000



SIP/2.0 100 Trying
Via:   SIP/ 2.0/ UDP  192.168.3.69
;branch=z9hG4bKnp492418173-4587980e192.168.3.69
From:   sip:[EMAIL PROTECTED] ;tag=1d59b0b2
To:   sip:[EMAIL PROTECTED] ;tag=as176f35b3
Call-ID:   [EMAIL PROTECTED]
CSeq:   2  INVITE
User-Agent:   Asterisk PBX
Allow:   INVITE
Allow:   ACK
Allow:   CANCEL
Allow:   OPTIONS
Allow:   BYE
Allow:   REFER
Contact:   sip:[EMAIL PROTECTED]
Content-Length:   0

SIP/2.0 200 OK
Via:   SIP/ 2.0/ UDP  192.168.3.69
;branch=z9hG4bKnp492418173-4587980e192.168.3.69
From:   sip:[EMAIL PROTECTED] ;tag=1d59b0b2
To:   sip:[EMAIL PROTECTED] ;tag=as176f35b3
Call-ID:   [EMAIL PROTECTED]
CSeq:   2  INVITE
User-Agent:   Asterisk PBX
Allow:   INVITE
Allow:   ACK
Allow:   CANCEL
Allow:   OPTIONS
Allow:   BYE
Allow:   REFER
Contact:   sip:[EMAIL PROTECTED]
Content-Type:   application/ sdp
Content-Length:   235
v=0
o=root 32181 32181 IN IP4 192.168.3.53
s=session
c=IN IP4 192.168.3.53
t=0 0
m=audio 16246 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
m=video 11206 RTP/AVP



CANCEL sip:[EMAIL PROTECTED] SIP/2.0
Via:   SIP/ 2.0/ UDP  192.168.3.69
;branch=z9hG4bKnp492418173-4587980e192.168.3.69
From:   sip:[EMAIL PROTECTED] ;tag=1d59b0b2
To:   sip:[EMAIL PROTECTED] ;tag=as176f35b3
Call-ID:   [EMAIL PROTECTED]
CSeq:   2  CANCEL
Warning:   399   SDP not available
Date:   Fri, 05 Mar 2004 19:16:01 GMT
Content-Length:   0
User-Agent:   Ahead SIPPS IP Phone Version 2.0.45.18

SIP/2.0 200 OK
Via:   SIP/ 2.0/ UDP  192.168.3.69
;branch=z9hG4bKnp492418173-4587980e192.168.3.69
From:   sip:[EMAIL PROTECTED] ;tag=1d59b0b2
To:   sip:[EMAIL PROTECTED] ;tag=as176f35b3
Call-ID:   [EMAIL PROTECTED]
CSeq:   2  CANCEL
User-Agent:   Asterisk PBX
Allow:   INVITE
Allow:   ACK
Allow:   CANCEL
Allow:   OPTIONS
Allow:   BYE
Allow:   REFER
Contact:   sip:[EMAIL PROTECTED]
Content-Length:   0

SIP/2.0 200 OK
Via:   SIP/ 2.0/ UDP  192.168.3.69
;branch=z9hG4bKnp492418173-4587980e192.168.3.69
From:   sip:[EMAIL PROTECTED] ;tag=1d59b0b2
To:   sip:[EMAIL PROTECTED] ;tag=as176f35b3
Call-ID:   [EMAIL PROTECTED]
CSeq:   2  INVITE
User-Agent:   Asterisk PBX
Allow:   INVITE
Allow:   ACK
Allow:   CANCEL
Allow:   OPTIONS
Allow:   BYE
Allow:   REFER
Contact:   sip:[EMAIL PROTECTED]
Content-Type:   application/ sdp
Content-Length:   235
v=0
o=root 32181 32181 IN IP4 192.168.3.53
s=session
c=IN IP4 192.168.3.53
t=0 0
m=audio 16246 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
m=video 11206 RTP/AVP



SIP/2.0 200 OK
Via:   SIP/ 2.0/ UDP 

Re: [Asterisk-Users] dropped calls

2004-03-05 Thread Bartosz Jozwiak
I have couple of GS phone and CISCO 7960.
The funny thing is that two of that GS phone keep disconnecting and also
CISCO 7960 phone keeps disconnecting.
But the problem appear month ago! This is really strange!

Bart



 Hello,

 I'll try that, but why on earth gs phones with the same firmware on
 another * server, work with no problem?

 I've failed to state I'm using zaprtc, since there is no digium hardware
 on the server. Does it matter?

 Thanks,

 --- Paulo.


 On Fri, 2004-03-05 at 18:49, Ross Donaldson wrote:
  There is new firmware that may help
http://www.grandstream.com/BETATEST/.
  Grandstream acknowledges this problem. They say it is a codec issue with
  asterisk. I don't know if this update addresses this problem but it may
be
  worth a try.
 
   -Original Message-
   From: [EMAIL PROTECTED]
   [mailto:[EMAIL PROTECTED] On Behalf Of
   Paulo Loureiro
   Sent: Friday, March 05, 2004 10:26 AM
   To: [EMAIL PROTECTED]
   Subject: [Asterisk-Users] dropped calls
  
   Hello list,
  
   I'm getting droped calls on an asterisk installation. When on GS phone
   dials another one, the call is dropped after some (usually
   random) time
   but most of the tome within 3 to 20 seconds.
   I think the cause is stated on the logs, see bellow, and is
   related with
   the message  Didn't get a frame from channel: SIP/3805-df43, but I
   can't figure why.
  
  
   asterisk logs:
   -
   Mar  5 15:57:26 DEBUG[1116957488]: build_route: Contact hop:
   sip:192.168.60.106
   Mar  5 15:57:26 VERBOSE[1217669936]: -- SIP/3805-df43 answered
   SIP/-08122450
   Mar  5 15:57:26 VERBOSE[1217669936]: -- Attempting native
   bridge of
   SIP/-08122450 and SIP/3805-df43
   Mar  5 15:57:26 DEBUG[1116957488]: Stopping retransmission on
   '[EMAIL PROTECTED]' of Response\ 25663: Found
   Mar  5 15:57:26 DEBUG[1217669936]: Difference is 4032, ms is 524
   Mar  5 15:57:26 DEBUG[1217669936]: Ooh, format changed from
   UNKN to ULAW
   Mar  5 15:57:38 DEBUG[1217669936]: Didn't get a frame from channel:
   SIP/3805-df43
   Mar  5 15:57:38 DEBUG[1217669936]: Bridge stops bridging channels
   SIP/-08122450 and SIP/3805-df43
   Mar  5 15:57:38 DEBUG[1217669936]: find_user() - decrement outUse
   counter
   Mar  5 15:57:38 DEBUG[1217669936]:  is not a local user
   Mar  5 15:57:38 VERBOSE[1217669936]:   == Spawn extension
   (local, 3805,
   1) exited non-zero on 'SIP/-0812245\0'
   -
  
   The scenario:
   1 server (redhat 9),  asterisk (stable) and a 2 x hisax ISDN BRI.
   One of the BRI boards is used to dial out (ppp) on one channel and a
   mgetty on the other channel. The other board is in ptp and used by *.
   The phones are Grandstream BT101 and Handytone and are all on
   a switched
   network (3 procurve switches, stacked).
  
   The configs are ok, since the same files on another server work ok (no
   dropped calls), but I can post them if needed.
  
  
   Any help will be greatly appreciated.
  
   Thanks in advance,
  
  
  
   --- Paulo Loureiro
  
  
  
 
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 -- 
 Cumprimentos,

 --- Paulo Loureiro
 Netmania

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Re: [Asterisk-Users] Simple * status

2004-03-05 Thread info-lists
Tim,
It looks interesting.. Are you willing to release the  source code?

Robert

Tim Sailer said:
 On Fri, Mar 05, 2004 at 01:29:38PM -0500, Tim Sailer wrote:
 Since there's not too much out there, I decided to take about 2 hrs and
 pound something into shape for a simple status for my * server.
 I wrote a perl script that parsed the output of 'sip show peers',
 'iax2 show peers', and 'show voicemail users' through the manager
 interface. It dumps the output to a few simple mysql tables, and
 the results are displayed on a web page. Now I can see some of the
 basic things.

 http://pbx.unslept.com/status.php

 Before anyone comments, I know it's rough and ugly looking, but this was
 just proof of concept for me, done over about 2 hours while trying to do
 my normal job, too. I'll keep poking at the CLI to see what other cool
 stuff I can pull out.

 Tim

 --

 Tim Sailer Coastal Internet, Inc.  
 Network and Systems Operations PO Box 726  
 http://www.buoy.comMoriches, NY 11955  
 [EMAIL PROTECTED]   (631) 399-2910  (888) 924-3728  

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[Asterisk-Users] ADSI and a SIP ATA

2004-03-05 Thread Doug Meredith
We are interested in deploying some ADSI phones, but we are currently
using Sipura SPA-2000s.  Everything I read on ADSI, says that it is
just audio, and you don't need a special channel bank or anything like
that.

But what about through the SPA?  I notice that zapata.conf appears to
have an option to turn on ADSI support.  This makes me think that it
won't work with any other channel.

I searched the mailing list, but the only thing I could find was from
Oct. 2002, at there was no definitive answer.

Anybody know if it will work?

Doug
-- 
Doug Meredith ([EMAIL PROTECTED])
SystemGuard - Oracle remote support
877-974-8273 (87-SYSGUARD)
506-854-7997
www.systemguard.com

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RE: [Asterisk-Users] G.729 vs. G.729 pass thru

2004-03-05 Thread Wes Marderness
I've had some small problems when trying to users features like
AbsoluteTimeout with pass thru. Other than that sound quality has been good.

Wes

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Unavailable
ID
Sent: Thursday, March 04, 2004 9:03 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] G.729 vs. G.729 pass thru


Darek,

Thank you for the info.

How is the sound quality when you are using with G.729 codec?  What's your
thought?

Thanks.

- Original Message -
From: Derek Samford [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; [EMAIL PROTECTED]
Sent: Thursday, March 04, 2004 2:58 PM
Subject: RE: [Asterisk-Users] G.729 vs. G.729 pass thru


As long as you have a IDE drive available, and mounted when you install it,
it will work. This includes CD ROM's...It's what I did.
Funkiness with the registration process.
As far as pass through goes, from what I understand, it *should*, but when
you have licensed binaries, from what I've seen, it doesn't. It's actually
used 2 licenses. I plan on figuring that out next.

Derek


From: Unavailable ID [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 04, 2004 5:33 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] G.729 vs. G.729 pass thru

Hello everyone,

If you don't have Digium card but you want to use G.729 codec, do you need a
license for it?

If the VoIP termination point supports G.729 and you are using sip phone
(soft/hard phone), can you use the G.729 pass thru or you have to buy the
license?

Have anyone test it with SCSI system? Seems like it only work on machine
with IDE disk.

Thanks.
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Re: [Asterisk-Users] Simple * status

2004-03-05 Thread Tim Sailer
On Fri, Mar 05, 2004 at 08:27:59PM +0100, [EMAIL PROTECTED] wrote:
 Tim,
 It looks interesting.. Are you willing to release the  source code?

Sure. let me clean it up a bit... OK, a LOT... and finish the comments,
and I'll have a download link for it sometime this weekend. I'll keep
the downloadable stuff up-to-date with the running version.

Tim

-- 

 Tim Sailer Coastal Internet, Inc.  
 Network and Systems Operations PO Box 726  
 http://www.buoy.comMoriches, NY 11955  
 [EMAIL PROTECTED]   (631) 399-2910  (888) 924-3728  

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RE: [Asterisk-Users] Ahead SIPPS and Asterisk

2004-03-05 Thread Craig Waddington
Your problem is what I experience with Messenger, when I call it.
Unfortunately I never bothered trying to work out the problem.

I like the SIPPS phone features, but it is ugly.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Carlton J.
O'Riley
Sent: 05 March 2004 18:48
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Ahead SIPPS and Asterisk

I'll try to look over my config again.  Not sure I put a realm in, but
everything else seemed fine.  I get the acquired message and I see the
SIP
messages flowing on the Asterisk server, but once any sound needs to be
sent, it dies.  I'm using g711ulaw and I was calling into an
announcement
menu to test that I have setup on the server.  The entry in my sip.conf
is
exactly what I use for X-Pro, so that is why I am confused by the
missing
SDP message.

-Original Message-
Message: 8
Subject: RE: [Asterisk-Users] Ahead SIPPS and Asterisk
Date: Fri, 5 Mar 2004 18:25:05 -
From: Craig Waddington [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]

What sort of phone are you trying to call?

I use SIPPS and * and it works fine, it just wont work when you call
Windows Messenger for some reason. I can call X-lite, POTS, GS phones no
problems.

I use the same config as X-lite, in SIPPS if you click on the spanner or
press F9 to go into configuration and click on network.

You should see aquired?=20

Click modify and ensure Gateway is ticked and you have the IP of your
server in there, for some reason it defaults as redirect. On mine I
leave Dial prefix blank.

Click modify again then Ok, which will take you to the config section
for the user, choose use authentication, fill in the blanks, I found
also putting the realm in made it work correctly.

Hope that helps in some way.
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RE: [Asterisk-Users] Simple * status

2004-03-05 Thread calvis
That is pretty cool.

I watched the light bulbs for a while now.  This is a useful tool that has
many possibilities.   That Tim guy is on the phone more than my teenage
daughter :)

calvis

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, March 05, 2004 11:28 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Simple * status

Tim,
It looks interesting.. Are you willing to release the  source code?

Robert

Tim Sailer said:
 On Fri, Mar 05, 2004 at 01:29:38PM -0500, Tim Sailer wrote:
 Since there's not too much out there, I decided to take about 2 hrs and
 pound something into shape for a simple status for my * server.
 I wrote a perl script that parsed the output of 'sip show peers',
 'iax2 show peers', and 'show voicemail users' through the manager
 interface. It dumps the output to a few simple mysql tables, and
 the results are displayed on a web page. Now I can see some of the
 basic things.

 http://pbx.unslept.com/status.php

 Before anyone comments, I know it's rough and ugly looking, but this was
 just proof of concept for me, done over about 2 hours while trying to do
 my normal job, too. I'll keep poking at the CLI to see what other cool
 stuff I can pull out.

 Tim

 --

 Tim Sailer Coastal Internet, Inc.  
 Network and Systems Operations PO Box 726  
 http://www.buoy.comMoriches, NY 11955  
 [EMAIL PROTECTED]   (631) 399-2910  (888) 924-3728  

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RE: [Asterisk-Users] Simple * status

2004-03-05 Thread Craig Waddington
Nice one thanks for sharing, I look forward to it.

This will be very handy for SIP call transfers. At the moment I blindly
transfer on sip.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Sailer
Sent: 05 March 2004 19:49
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Simple * status

On Fri, Mar 05, 2004 at 08:27:59PM +0100, [EMAIL PROTECTED] wrote:
 Tim,
 It looks interesting.. Are you willing to release the  source code?

Sure. let me clean it up a bit... OK, a LOT... and finish the comments,
and I'll have a download link for it sometime this weekend. I'll keep
the downloadable stuff up-to-date with the running version.

Tim

-- 


 Tim Sailer Coastal Internet, Inc.

 Network and Systems Operations PO Box 726

 http://www.buoy.comMoriches, NY 11955

 [EMAIL PROTECTED]   (631) 399-2910  (888) 924-3728



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[Asterisk-Users] Sipura SPA 200 Fax

2004-03-05 Thread Mark Messmore, Technical Support, University Telcom Inc.


Is anyone presently using the Sipura SPA 2000 for faxing?  I was about
to look into it and just figured that I would ask to see if anyone ran
into any snags, problems, etc.  Thanks.

Mark

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Re: [Asterisk-Users] Sipura SPA 200 Fax

2004-03-05 Thread Nate Carlson
On Fri, 5 Mar 2004, Mark Messmore, Technical Support, University Telcom Inc. wrote:
 Is anyone presently using the Sipura SPA 2000 for faxing?  I was about
 to look into it and just figured that I would ask to see if anyone ran
 into any snags, problems, etc.  Thanks.

I'm really trying.  :)

It works, sort of. Basically, about 1 in 4 faxes are going out without
errors. Of course, that's to an IAX peer, so I'm not sure if it's a 
problem with the IAX peer or with the Siupra.

I'm planning on testing with my Vonage line using a X100P, and see if I
get better results that way. (I can hook up to the Cisco ATA186 Vonage
gave me and fax without a problem.)


| nate carlson | [EMAIL PROTECTED] | http://www.natecarlson.com |
|   depriving some poor village of its idiot since 1981|

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Re: [Asterisk-Users] Sipura SPA 200 Fax

2004-03-05 Thread Brancaleoni Matteo
hi
 
 It works, sort of. Basically, about 1 in 4 faxes are going out without
 errors. Of course, that's to an IAX peer, so I'm not sure if it's a 
 problem with the IAX peer or with the Siupra.
check you IAX connection.
perhaps is using gsm and that could explain the failure
Faxes must be sent uncompressed, ie with [u-a]law as codecs.

I have 2 fax machines over SIP here (ulaw) and never missed an hit :)

-- 
Brancaleoni Matteo [EMAIL PROTECTED]
Espia - Emmegi Srl

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Re: [Asterisk-Users] G.729 vs. G.729 pass thru

2004-03-05 Thread Unavailable ID
Hi Wes,

Do you need to buy license when you are using pass thru.  How does it work?

I'm thinking about using pass thru for voip since the service provider has
g.279 codec.  Can you setup your * box connects to telco termination with
pass thru?

PBX =[T100P]= ASTERISK (*) =[G.729]= VOIP TERMINATION

Thanks.

- Original Message - 
From: Wes Marderness [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, March 05, 2004 11:42 AM
Subject: RE: [Asterisk-Users] G.729 vs. G.729 pass thru


 I've had some small problems when trying to users features like
 AbsoluteTimeout with pass thru. Other than that sound quality has been
good.

 Wes

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Unavailable
 ID
 Sent: Thursday, March 04, 2004 9:03 PM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] G.729 vs. G.729 pass thru


 Darek,

 Thank you for the info.

 How is the sound quality when you are using with G.729 codec?  What's your
 thought?

 Thanks.

 - Original Message -
 From: Derek Samford [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]; [EMAIL PROTECTED]
 Sent: Thursday, March 04, 2004 2:58 PM
 Subject: RE: [Asterisk-Users] G.729 vs. G.729 pass thru


 As long as you have a IDE drive available, and mounted when you install
it,
 it will work. This includes CD ROM's...It's what I did.
 Funkiness with the registration process.
 As far as pass through goes, from what I understand, it *should*, but when
 you have licensed binaries, from what I've seen, it doesn't. It's actually
 used 2 licenses. I plan on figuring that out next.

 Derek

 
 From: Unavailable ID [mailto:[EMAIL PROTECTED]
 Sent: Thursday, March 04, 2004 5:33 PM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] G.729 vs. G.729 pass thru

 Hello everyone,

 If you don't have Digium card but you want to use G.729 codec, do you need
a
 license for it?

 If the VoIP termination point supports G.729 and you are using sip phone
 (soft/hard phone), can you use the G.729 pass thru or you have to buy the
 license?

 Have anyone test it with SCSI system? Seems like it only work on machine
 with IDE disk.

 Thanks.
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Re: [Asterisk-Users] Simple * status

2004-03-05 Thread Bartosz Jozwiak
I am also looking forward to it.
It looks really nice!

bart
- Original Message - 
From: Craig Waddington [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, March 05, 2004 4:57 PM
Subject: RE: [Asterisk-Users] Simple * status


Nice one thanks for sharing, I look forward to it.

This will be very handy for SIP call transfers. At the moment I blindly
transfer on sip.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Sailer
Sent: 05 March 2004 19:49
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Simple * status

On Fri, Mar 05, 2004 at 08:27:59PM +0100, [EMAIL PROTECTED] wrote:
 Tim,
 It looks interesting.. Are you willing to release the  source code?

Sure. let me clean it up a bit... OK, a LOT... and finish the comments,
and I'll have a download link for it sometime this weekend. I'll keep
the downloadable stuff up-to-date with the running version.

Tim

-- 


 Tim Sailer Coastal Internet, Inc.

 Network and Systems Operations PO Box 726

 http://www.buoy.comMoriches, NY 11955

 [EMAIL PROTECTED]   (631) 399-2910  (888) 924-3728



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Re: [Asterisk-Users] Sipura SPA 200 Fax

2004-03-05 Thread Nate Carlson
On Fri, 5 Mar 2004, Brancaleoni Matteo wrote:
 check you IAX connection. perhaps is using gsm and that could explain
 the failure Faxes must be sent uncompressed, ie with [u-a]law as codecs.

Forgot to mention that - I've tried both ulaw and alaw all the way through
(sipura - asterisk, and asterisk - IAX peer).

 I have 2 fax machines over SIP here (ulaw) and never missed an hit :)

Just curious, which SIP peers (if they are public peers)? Is this over the
'net?


| nate carlson | [EMAIL PROTECTED] | http://www.natecarlson.com |
|   depriving some poor village of its idiot since 1981|

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RE: [Asterisk-Users] Sipura SPA 200 Fax

2004-03-05 Thread Justin Carlson
yes be sure you are using ULAW and I found that 9600 was the baud rate to go
with.  14400 seemed to be unreliable.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mark
Messmore, Technical Support, University Telcom Inc.
Sent: Friday, March 05, 2004 2:15 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Sipura SPA 200 Fax




Is anyone presently using the Sipura SPA 2000 for faxing?  I was about
to look into it and just figured that I would ask to see if anyone ran
into any snags, problems, etc.  Thanks.

Mark

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[Asterisk-Users] Dialogic supported well?

2004-03-05 Thread Alfred Werner

I'm new to asterisk and quite impressed by the feature list. I have a
D/4PCI already in hand. Is there any reason NOT to use this and buy a
digium card instead?

I basically want to set up a couple line analog system to check it out and
probably use as a a Soho setup for VM, access to a postgres database, and
to play with the VOIP stuff.

TIA,

Alfred Werner
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RE: [Asterisk-Users] G.729 vs. G.729 pass thru

2004-03-05 Thread Wes Marderness
Yes, I do something like that.
MediatrixFXO(1204)-Asterisk-MediatrixFXO(1204), I have bought license from
diguim for G.729. I do not really have a telco provider just an ISP. I use
if for a private network.

Wes

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Unavailable
ID
Sent: Friday, March 05, 2004 3:41 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] G.729 vs. G.729 pass thru


Hi Wes,

Do you need to buy license when you are using pass thru.  How does it work?

I'm thinking about using pass thru for voip since the service provider has
g.279 codec.  Can you setup your * box connects to telco termination with
pass thru?

PBX =[T100P]= ASTERISK (*) =[G.729]= VOIP TERMINATION

Thanks.

- Original Message -
From: Wes Marderness [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, March 05, 2004 11:42 AM
Subject: RE: [Asterisk-Users] G.729 vs. G.729 pass thru


 I've had some small problems when trying to users features like
 AbsoluteTimeout with pass thru. Other than that sound quality has been
good.

 Wes

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Unavailable
 ID
 Sent: Thursday, March 04, 2004 9:03 PM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] G.729 vs. G.729 pass thru


 Darek,

 Thank you for the info.

 How is the sound quality when you are using with G.729 codec?  What's your
 thought?

 Thanks.

 - Original Message -
 From: Derek Samford [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]; [EMAIL PROTECTED]
 Sent: Thursday, March 04, 2004 2:58 PM
 Subject: RE: [Asterisk-Users] G.729 vs. G.729 pass thru


 As long as you have a IDE drive available, and mounted when you install
it,
 it will work. This includes CD ROM's...It's what I did.
 Funkiness with the registration process.
 As far as pass through goes, from what I understand, it *should*, but when
 you have licensed binaries, from what I've seen, it doesn't. It's actually
 used 2 licenses. I plan on figuring that out next.

 Derek

 
 From: Unavailable ID [mailto:[EMAIL PROTECTED]
 Sent: Thursday, March 04, 2004 5:33 PM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] G.729 vs. G.729 pass thru

 Hello everyone,

 If you don't have Digium card but you want to use G.729 codec, do you need
a
 license for it?

 If the VoIP termination point supports G.729 and you are using sip phone
 (soft/hard phone), can you use the G.729 pass thru or you have to buy the
 license?

 Have anyone test it with SCSI system? Seems like it only work on machine
 with IDE disk.

 Thanks.
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RE: [Asterisk-Users] Sipura SPA 200 Fax

2004-03-05 Thread Nate Carlson
On Fri, 5 Mar 2004, Justin Carlson wrote:
 yes be sure you are using ULAW and I found that 9600 was the baud rate
 to go with.  14400 seemed to be unreliable.

Are you actually faxing over the 'net?

I'm using ULAW both from the Sipura - Asterisk and from Asterisk -
NuFone (IAX), and have the modem locked down to 9600, but most faxes still
fail.  :(


| nate carlson | [EMAIL PROTECTED] | http://www.natecarlson.com |
|   depriving some poor village of its idiot since 1981|

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RE: [Asterisk-Users] Dialogic supported well?

2004-03-05 Thread Kaydon Stanzione
 
Alfred,

I am in similar position. I took the route of going with Digium boards for
two primary reasons - they have excellent quality and offer the best in
customer and technical support. I am starting with an X100P just for testing
everything out (using IAXClient and Asterisk).

We have two Network Operation Centers - one running a T-3 and the other an
OC-3. We are looking to add VoIP and PABX solutions to our product line.
Once we can gain a better understanding with Asterisk and the Digium card,
we are upgrading to a TE405 with PRI (4 lines).

Digium is extremely knowledgeable and continues to provide us with valuable
information.

Good luck - call the guys at Digium (Malcolm or Greg) - they are very
helpful

kaydon





-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alfred Werner
Sent: Friday, March 05, 2004 5:05 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Dialogic supported well?


I'm new to asterisk and quite impressed by the feature list. I have a D/4PCI
already in hand. Is there any reason NOT to use this and buy a digium card
instead?

I basically want to set up a couple line analog system to check it out and
probably use as a a Soho setup for VM, access to a postgres database, and to
play with the VOIP stuff.

TIA,

Alfred Werner
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[Asterisk-Users] E100P / E1 dial out

2004-03-05 Thread Tobias F. Leucht
Hello *,

my setup of an Asterisk box with a TDM400P and an E100P went just
fine except that I cannot manage to place outgoing calls via the E1
interface (while incoming calls _do_ work). The CLI says:

-8-
Mar  6 00:42:25 DEBUG[-1147995216]: chan_sip.c:3593 build_route:
build_route: Contact hop: sip:[EMAIL PROTECTED]:5060;line=jet7pbic
 -- Executing Dial(SIP/265-7cf2, Zap/1/08948999807) in new stack
 -- Called 1/08948999807
Mar  6 00:42:25 DEBUG[-1263383632]: rtp.c:1008 ast_rtp_write:
Ooh, format changed from UNKN to ALAW
 -- Channel 1, span 1 got hangup
Mar  6 00:42:26 WARNING[-1263383632]: app_dial.c:337
wait_for_answer: Unable to forward voice
Mar  6 00:42:26 DEBUG[-1263383632]: chan_zap.c:2185
zt_setoption: Set option AUDIO MODE, value: ON(1) on Zap/1-1
Mar  6 00:42:26 DEBUG[-1263383632]: chan_zap.c:1715
zt_hangup: Hangup: channel: 1 index = 0, normal = 19,
callwait = -1, thirdcall = -1
Mar  6 00:42:26 DEBUG[-1263383632]: chan_zap.c:1133
zt_disable_ec: disabled echo cancellation on channel 1
Mar  6 00:42:26 DEBUG[-1263383632]: chan_zap.c:2095
zt_setoption: Set option TDD MODE, value: OFF(0) on Zap/1-1
Mar  6 00:42:26 DEBUG[-1263383632]: chan_zap.c:1076
update_conf: Updated conferencing on 1, with 0 conference
users
Mar  6 00:42:26 DEBUG[-1263383632]: chan_zap.c:2179
zt_setoption: Set option AUDIO MODE, value: OFF(0) on
Zap/1-1
Mar  6 00:42:26 DEBUG[-1263383632]: chan_zap.c:1133
zt_disable_ec: disabled echo cancellation on channel 1
  -- Hungup 'Zap/1-1'
-8-

corresponding configs run like:

---zaptel.conf---
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
fxoks=32-35
loadzone=us
defaultzone=us
---/zaptel.conf---

---zapata.conf---
[channels]

context=default
language=de
switchtype=euroisdn
signalling=pri_cpe
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
musiconhold=default

group=1
callgroup=1
pickupgroup=1
immediate=no

callerid=Zentrale (256) 36094-0
channel = 1
callerid=Joerg Czaika (256) 36094-245
channel = 2
callerid=Gerda Graeber-Otte (256) 36094-265
channel = 3
callerid=Petra Weigelt (256) 36094-217
channel = 4
callerid=Elfriede Gunzelmann (256) 36094-239
channel = 5

channel = 1-15,17-31

; TDM400B section
;
context=analog
signalling=fxo_ks
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=1.5
txgain=1.5
immediate=no
usecallerid=yes
callerid=analog Fax 200
channel = 32
---/zapata.conf---

and the corresponding extension in extensions.con:

-8-
exten = _0., 1,Dial(Zap/1/${EXTEN})
-8-

Hm, guess I just don't see the forest for the trees, do I? Any ideas?

TIA,

Tobias F. Leucht

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Re: [Asterisk-Users] 3com NBX phones

2004-03-05 Thread admin
When an NBX100 is upgraded a .tar file is uploaded and installed on the box.
Inside that tar file is the firmware for the phones which is downloaded when
the phone boots.  If someone can provide the last SIP firmware I will
replace the phone firmware in the tar file with the SIP code and see if the
phone can take it.  I see alot of RMA's go through our office so no loss if
it kills the phone and the NBX also retains its previous versions to boot
to.


- Original Message - 
From: Derek Bruce [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, March 04, 2004 4:22 PM
Subject: Re: [Asterisk-Users] 3com NBX phones


 The 3com phones can't be flashed... they download their firmware image
from
 the NBX call processor when they power on...
 However, if a SIP image you can have the phone download the image from a
 linux box using the bootloader provided by Tim Hogard at
 http://web.abnormal.com/~thogard/nbx100.shtml


 - Original Message -
 From: Steve Totaro [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Sent: Thursday, March 04, 2004 1:54 PM
 Subject: Re: [Asterisk-Users] 3com NBX phones


  They did make a SIP phone and are about to release new SIP phones and a
 new
  product line.  The old SIP phones look identical to the NBX phones but I
 am
  not sure about the guts.  Possibly the 2102 could be flashed into a
1002.
  Here is an ebay auction but these phones are really hard to come by.  I
  would love to hear if a 2102 can be flashed.
 
 

http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=3082407185category=11909
 
  - Original Message -
  From: Ejay Hire [EMAIL PROTECTED]
  To: [EMAIL PROTECTED]
  Sent: Thursday, March 04, 2004 12:34 PM
  Subject: RE: [Asterisk-Users] 3com NBX phones
 
 
   The original NBX100 phones spoke a proprietary voice-over-l2
   ethernet protocol, but would upgrade to ip connectivity with
   a liscense key on the NBX PBX box.  There was an optional
   software package that would let the NBX talk to an h323
   gateway but it ran on nt and was rather klunky.
  
   These originally came out in 97 or 98, so sip functionality
   in the originals is rather unlikely.
  
   -e
  
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
   Of Rob Fugina
Sent: Thursday, March 04, 2004 1:58 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] 3com NBX phones
   
On Thu, Mar 04, 2004 at 02:38:13PM -0500, Tim Sailer
   wrote:
 Does anyone know if the 3Com NBX 2102 series phones with
  
with * ? There
 are a crapload (a very precise measurement) on eBay, but
   I
can't figure
 out what protocols they talk.
   
I believe they did make a version that spoke SIP, but
   they're rare.
The ones on eBay are most likely 100% proprietary.
   
Rob
   
--
Rob Fugina, Systems Guy
[EMAIL PROTECTED] -- http://www.geekthing.com
My firewall filters MS Office attachments.
   
 666,000,000 -- The number of the megabeast.
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[Asterisk-Users] SIP and distinctive ring

2004-03-05 Thread Matt McIntyre
Has anyone implemented distinctive ring for SIP devices in Asterisk? My
searches revealed that there was a patch created at one time but I can't
tell if it was accepted or not.

Basically I have a Sipura analog adapter that I would like to have ring
differently for internal calls vs external calls.

Thanks guys,

Matt

^
!   Matt McIntyre (KF4FGZ) 
! Certified Novell Administrator
! (336) 272-9139 (Campus telephone)
! (336) 215-7199 (Mobile telephone) - Please note the
change
! (336) 272-9139 (Facsimile)
! E-MAIL: [EMAIL PROTECTED]
! AIM: MixMANJaVa
! ICQ: 11956085
^



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Re: [Asterisk-Users] SIP and distinctive ring

2004-03-05 Thread Nicolas Gudino
On Fri, 2004-03-05 at 21:04, Matt McIntyre wrote:
 Has anyone implemented distinctive ring for SIP devices in Asterisk? My
 searches revealed that there was a patch created at one time but I can't
 tell if it was accepted or not.
 
 Basically I have a Sipura analog adapter that I would like to have ring
 differently for internal calls vs external calls.
 
 Thanks guys,
 
 Matt  

Hi Matt,

Try with:

exten = 1000,1,SetVar(ALERT_INFO=Bellcore-r3)


-- 
Nicolas Gudino [EMAIL PROTECTED]
House Internet S.R.L.

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RE: [Asterisk-Users] Dialogic supported well?

2004-03-05 Thread Scott Stingel
Hi Alfred-

I'd like to echo Kaydon's very positive comments regarding asterisk and
Digium with one or two caveats.  I've had lots of experience building
systems with Dialogic boards (analog and E1), and more recently a few
systems built with Digium's quad E1 boards (the E400P and now the TE410P),
so I've had a chance to compare them under similar application load,
especially high-volume IVR and also calling card apps.

If you'll be running commercial apps, I would recommend that you do a lot of
testing, especially load testing, with the types of applications you'll be
running.  Dialogic boards, although incredibly expensive, do have lots of
horsepower built in for the purposes of encoding and decoding the voice
streams, decoding single and DTMF codes, voice energy and cadence detection,
etc.  Digium boards rely 100% on the processor that they run in to perform
these functions.  I was a little disappointed to find that I can (so far)
reliably only handle 4 E1's in a (very) high-volume IVR app.  In the past
I've run Dialogic-based systems which handled much more load (but also which
cost 4 times as much in hardware!)  Although Digium's newer TE410P board is
capable of bus mastering, I found that it made little difference in the
number of channels I could run.  

So, I guess my point is not to be over-optimistic in deciding the number of
channels that you can run.  Processors are relatively cheap these days, so
when in doubt, opt for more processors and spread the load!

I also agree that Digium's support has been great.  I do wish they would
spend more time in two areas:  (a) doing some documentation for the boards
they sell - even a one-page setup sheet would be nice (the TE410 boards
arrive with nothing, not even an explanation of what the jumpers/switches on
the board do)!  (b) improving the an apparent bug or shortcoming in the PRI
driver code, which results in framing errors not being dealt with properly.
This really only effects very high volume systems, but it needs attention.
(I've discussed these with Mark)

Anyway, I don't want to dissuade you from choosing Digium and asterisk, they
are super accomplishments, just want you to manage your expectations a bit.
Test, test, test!

Good luck in your projects
Scott Stingel
President 

Emerging Voice Technology Inc.

Email:  scott at evtmedia.com 
URL:www.evtmedia.com  

 

-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
Kaydon Stanzione
Sent: Friday, March 05, 2004 10:31 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Dialogic supported well?

 
Alfred,

I am in similar position. I took the route of going with 
Digium boards for
two primary reasons - they have excellent quality and offer the best in
customer and technical support. I am starting with an X100P 
just for testing
everything out (using IAXClient and Asterisk).

We have two Network Operation Centers - one running a T-3 and 
the other an
OC-3. We are looking to add VoIP and PABX solutions to our 
product line.
Once we can gain a better understanding with Asterisk and the 
Digium card,
we are upgrading to a TE405 with PRI (4 lines).

Digium is extremely knowledgeable and continues to provide us 
with valuable
information.

Good luck - call the guys at Digium (Malcolm or Greg) - they are very
helpful

kaydon





-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of 
Alfred Werner
Sent: Friday, March 05, 2004 5:05 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Dialogic supported well?


I'm new to asterisk and quite impressed by the feature list. I 
have a D/4PCI
already in hand. Is there any reason NOT to use this and buy a 
digium card
instead?

I basically want to set up a couple line analog system to 
check it out and
probably use as a a Soho setup for VM, access to a postgres 
database, and to
play with the VOIP stuff.

TIA,

Alfred Werner
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[Asterisk-Users] Internet Phone Concept Question

2004-03-05 Thread Greg Kedrovsky
Hey, all. I'm new to this Asterisk stuff and have a general concept
question about making calls and whatnot over the net. 

I have a 4-port TDM card and a 1-port x100p card for incoming. All is
configured and working fine. I have a _very_ simple configuration (start
simple, add bells and whistles later). 

I have a cable modem hook-up and access the internet with a download
speed of 128k and upload speed of 32k. My Asterisk server sits on a LAN,
under a Freesco router running on an Pentium I machine (10BaseT cards
because). 

I live in Costa Rica and would like to utilize the internet, if
possible, to call family and friends in the U.S.A. Can I do that with
Asterisk? Can I do that with standard analog phones through Asterisk?
Can I do that without having another Asterisk machine State-side? 

If you have a link that would explain the concepts to me, that would be
fine. Or if you could kinda prime the pump for me so I can get the
ball rolling on my end - that'd be very much appreciated, too.

Thanks ahead of time. 

-Greg

-- 
Mutt 1.4.1i on Slackware 9.1 Linux
Curridabat, San Jose, Costa Rica
http://www.greg-and-sue.com/screenshot.jpg
Yahoo Instant Messenger ID: gregkedro
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Re: [Asterisk-Users] newbie

2004-03-05 Thread Nicholas Bachmann
Andrew McRory wrote:

I can offer some links that helped me...
[...]

If anyone has other links I'd appreciate them!
 

Don't forget http://www.asteriskdocs.org/ !

Nick

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[Asterisk-Users] Call roll-over question...

2004-03-05 Thread Brian R. Swan
I have another question for the group.  I'm trying to make the following 
happen on my Cisco phone:

I have two lines configured, 2001 and 3001.  If I'm talking on 2001 and 
someone tries to call me on 2001 I'd like the call to roll over to 3001 and 
then if I don't answer, it goes to Voice mail.  I was able to accomplish this 
using the following sequence in extensions.conf (I'm doing this from memory, 
so I hope I got it right).

exten = 2001,1,Dial(SIP/2001,20)
exten = 2001,2,Dial(SIP/3001,20)
exten = 2001,3,Voicemail(u2001)

Now, while this works if I'm talking on 2001, the obvious problem is that if 
none of the extensions are busy it will ring 2001 for 4 ring, then head over 
to 3001 for 4 rings until going to voice mail.  So, I then tried the 
following:

exten = 2001,1,ChanIsAvail(SIP/2001)
exten = 2001,2,Dial(SIP/2001,20)
exten = 2001,3,Voicemail(u2001)
exten = 2001,102,Transfer(3001)
exten = 2001,203,Voicemail(2001)
exten = 3001,1,Dial(SIP/3001,20)
exten = 3001,2,Voicemail(u2001)

Which seems (to my newbie eyes) that it should work, but... it doesn't.  If I 
pickup 2001 and call from another extension it goes straight to voice mail.

Both extensions (2001,3001) work on their own, so I'm certain that they are 
configured correctly.  Also, I have call waiting shut off on the cisco 
phone (so it should reject the SIP call to 2001 as busy). 

...any ideas?

Thanks!
Swannie 

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Re: [Asterisk-Users] zaptel on Debian

2004-03-05 Thread Duane
Hermann Wecke wrote:
After trying and trying to compile and make Asterisk run on a Debian
box, I gave up and picked another HD with RH 9 on it. No headaches. Only 1
build was necessary to build and run *.
The problem I found with debian is how they decided how to do the linux 
header files for everyone in /usr/include/linux either libc6-dev or 
linux-kernel-headers packages...

The fix for me since I roll my own kernels, after a lot of buggering 
about and head banging on the desk, the solution was rather simple...

cd /usr/src
ln -s linux-2.4.25 linux
then just build the cvs zaptel modules as per documents...

of course a couple of `uname -a` in the Makefile would have saved me all 
the headaches but anyways, 6 hours later and all the wiser...

--
Best regards,
 Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net!
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[Asterisk-Users] Zap to SIP transfer problem

2004-03-05 Thread Barton Hodges

I'm having a problem with transferring a call that comes in a Zap
channel and is connected with a SIP channel (on a GS HT-286).

The call is answered automatically, then the user enters an extension.
Dial() is called with both T and t flags.  When the bridge is made
between the channels, the caller on the Zap channel can hit '#' to
transfer, but the caller on the SIP channel cannot.  No messages
whatsoever are displayed on the console when the SIP user hits any
keys.  What am I missing?


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Re: [Asterisk-Users] Internet Phone Concept Question

2004-03-05 Thread Unavailable ID
Hi Greg,

Welcome to * world :-)

Your connection is slow '128k and upload speed of 32k' so you probably need
the G.729 codec ($$$ - $10/channel/call from Digium).

The X100P is only for dial-out from your phones that connect to TDM card.
This should use to dial local number in Costa Rica.

To call to US, use * to connect to the IAX service provider such as
http://connect.voicepulse.com/ or http://www.nufone.net/

It looks like this:

PSTN (USA) -- IAX Provider -- Asterisk -- TDM card -- Analog phones

Hope this help.

Tri Tu

- Original Message - 
From: Greg Kedrovsky [EMAIL PROTECTED]
To: asterisk-user [EMAIL PROTECTED]
Sent: Friday, March 05, 2004 4:59 PM
Subject: [Asterisk-Users] Internet Phone Concept Question


 Hey, all. I'm new to this Asterisk stuff and have a general concept
 question about making calls and whatnot over the net.

 I have a 4-port TDM card and a 1-port x100p card for incoming. All is
 configured and working fine. I have a _very_ simple configuration (start
 simple, add bells and whistles later).

 I have a cable modem hook-up and access the internet with a download
 speed of 128k and upload speed of 32k. My Asterisk server sits on a LAN,
 under a Freesco router running on an Pentium I machine (10BaseT cards
 because).

 I live in Costa Rica and would like to utilize the internet, if
 possible, to call family and friends in the U.S.A. Can I do that with
 Asterisk? Can I do that with standard analog phones through Asterisk?
 Can I do that without having another Asterisk machine State-side?

 If you have a link that would explain the concepts to me, that would be
 fine. Or if you could kinda prime the pump for me so I can get the
 ball rolling on my end - that'd be very much appreciated, too.

 Thanks ahead of time.

 -Greg

 -- 
 Mutt 1.4.1i on Slackware 9.1 Linux
 Curridabat, San Jose, Costa Rica
 http://www.greg-and-sue.com/screenshot.jpg
 Yahoo Instant Messenger ID: gregkedro
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Re: [Asterisk-Users] Sipura SPA 200 Fax

2004-03-05 Thread Panny Malialis
It works on grandstream handytone 286 also, i just tested both directions and it 
worked perfectly first time going into a fax
machine and into a windows xp machine all of this over my flaky wireless link too!

Panny

- Original Message - 
From: Brancaleoni Matteo [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, March 05, 2004 8:33 PM
Subject: Re: [Asterisk-Users] Sipura SPA 200 Fax


 hi
 
  It works, sort of. Basically, about 1 in 4 faxes are going out without
  errors. Of course, that's to an IAX peer, so I'm not sure if it's a
  problem with the IAX peer or with the Siupra.
 check you IAX connection.
 perhaps is using gsm and that could explain the failure
 Faxes must be sent uncompressed, ie with [u-a]law as codecs.

 I have 2 fax machines over SIP here (ulaw) and never missed an hit :)

 -- 
 Brancaleoni Matteo [EMAIL PROTECTED]
 Espia - Emmegi Srl

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Re: [Asterisk-Users] Call roll-over question...

2004-03-05 Thread Chris A. Icide
At 05:04 PM 3/5/2004, you wrote:
I have another question for the group.  I'm trying to make the following
happen on my Cisco phone:
I have two lines configured, 2001 and 3001.  If I'm talking on 2001 and
snip
Try this

exten = 2001,1,Dial(SIP/2001SIP/3001,20)

This will ring them both at the same time for 20 seconds

...any ideas?


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Re: [Asterisk-Users] Internet Phone Concept Question

2004-03-05 Thread Greg Kedrovsky
On Fri, Mar 05, 2004 at 06:45:32PM -0800, Unavailable ID wrote:
 
 Your connection is slow '128k and upload speed of 32k' so you probably need
 the G.729 codec ($$$ - $10/channel/call from Digium).

Yeah, I know... it's slow. But, I am in a developing country, and I
I'm a tightwad (don't wanna shell out bucks for the wee bit more
bandwidth if I can get by with what I got (which beats dial-up 10 ways
to next Sunday).

 To call to US, use * to connect to the IAX service provider such as
 http://connect.voicepulse.com/ or http://www.nufone.net/

Thank you. 

 It looks like this:
 
 PSTN (USA) -- IAX Provider -- Asterisk -- TDM card -- Analog phones
 
 Hope this help.

Yep. This and the previous post help a lot. Thanks to both! You gave me
just what I needed to start checking things out myself.

-Greg

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Re: [Asterisk-Users] Zap to SIP transfer problem

2004-03-05 Thread Eric Wieling
Maybe you are using inband DTMF with a compressed codec.  DTMF on a call
with any codec other than ulaw or alaw MUST use OOB DTMF like RFC2833 or
INFO.

On Fri, 2004-03-05 at 20:39, Barton Hodges wrote:
 I'm having a problem with transferring a call that comes in a Zap
 channel and is connected with a SIP channel (on a GS HT-286).
 
 The call is answered automatically, then the user enters an extension.
 Dial() is called with both T and t flags.  When the bridge is made
 between the channels, the caller on the Zap channel can hit '#' to
 transfer, but the caller on the SIP channel cannot.  No messages
 whatsoever are displayed on the console when the SIP user hits any
 keys.  What am I missing?
 
 
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http://www.digium.com/index.php?menu=documentation and look at the
Unofficial Links section also see
http://www.voip-info.org/wiki-Asterisk also see my site at
http://www.fnords.org/~eric/asterisk/

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RE: [Asterisk-Users] Zap to SIP transfer problem

2004-03-05 Thread Barton Hodges

I'm using SIP INFO and ulaw.  It seems that the same thing happens
from SIP to SIP as well, regardless of what the DTMF setting is.  The
actual problem is that the calling user can transfer, but the called
user cannot.  I just tried the latest CVS snapshot and the v1.0 stable
branch and they both behave the same way.

[EMAIL PROTECTED] wrote:
 Maybe you are using inband DTMF with a compressed codec. DTMF on a
 call with any codec other than ulaw or alaw MUST use OOB DTMF like
 RFC2833 or INFO.
 
 On Fri, 2004-03-05 at 20:39, Barton Hodges wrote:
 I'm having a problem with transferring a call that comes in a Zap
 channel and is connected with a SIP channel (on a GS HT-286).
 
 The call is answered automatically, then the user enters an
 extension. Dial() is called with both T and t flags.  When the
 bridge is made between the channels, the caller on the Zap channel
 can hit '#' to transfer, but the caller on the SIP channel cannot. 
 No messages whatsoever are displayed on the console when the SIP
 user hits any keys.  What am I missing? 
 
 
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RE: [Asterisk-Users] Zap to SIP transfer problem

2004-03-05 Thread Eric Wieling
What is your ACTUAL Dial line?

On Fri, 2004-03-05 at 21:19, Barton Hodges wrote:
 I'm using SIP INFO and ulaw.  It seems that the same thing happens
 from SIP to SIP as well, regardless of what the DTMF setting is.  The
 actual problem is that the calling user can transfer, but the called
 user cannot.  I just tried the latest CVS snapshot and the v1.0 stable
 branch and they both behave the same way.
 
 [EMAIL PROTECTED] wrote:
  Maybe you are using inband DTMF with a compressed codec. DTMF on a
  call with any codec other than ulaw or alaw MUST use OOB DTMF like
  RFC2833 or INFO.
  
  On Fri, 2004-03-05 at 20:39, Barton Hodges wrote:
  I'm having a problem with transferring a call that comes in a Zap
  channel and is connected with a SIP channel (on a GS HT-286).
  
  The call is answered automatically, then the user enters an
  extension. Dial() is called with both T and t flags.  When the
  bridge is made between the channels, the caller on the Zap channel
  can hit '#' to transfer, but the caller on the SIP channel cannot. 
  No messages whatsoever are displayed on the console when the SIP
  user hits any keys.  What am I missing? 
  
  
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http://www.digium.com/index.php?menu=documentation and look at the
Unofficial Links section also see
http://www.voip-info.org/wiki-Asterisk also see my site at
http://www.fnords.org/~eric/asterisk/

BTEL Consulting

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RE: [Asterisk-Users] Zap to SIP transfer problem

2004-03-05 Thread Barton Hodges

exten = s,10,Dial(${ARG1}/${DIALED},19,Ttm) which translates to
Dial(SIP/210-80f2, SIP/280|19|Ttm) 

I believe the problem is related to the Grandstream HandyTone-286.  A
caller can transfer, but a callee cannot.  The problem does not exist
with a BT101 (1.0.4.23).  I just tried all of the firmware on their
BETA site, from 1.0.4.35 through 1.0.4.50 and the problem was never
solved.

Can anyone confirm this for me?

I am SO SICK of dealing with HT-286 firmware bugs!

[EMAIL PROTECTED] wrote:
 What is your ACTUAL Dial line?
 
 On Fri, 2004-03-05 at 21:19, Barton Hodges wrote:
 I'm using SIP INFO and ulaw.  It seems that the same thing happens
 from SIP to SIP as well, regardless of what the DTMF setting is.
The
 actual problem is that the calling user can transfer, but the
called
 user cannot.  I just tried the latest CVS snapshot and the v1.0
 stable branch and they both behave the same way.
 
 [EMAIL PROTECTED] wrote:
 Maybe you are using inband DTMF with a compressed codec. DTMF on a
 call with any codec other than ulaw or alaw MUST use OOB DTMF like
 RFC2833 or INFO. 
 
 On Fri, 2004-03-05 at 20:39, Barton Hodges wrote:
 I'm having a problem with transferring a call that comes in a Zap
 channel and is connected with a SIP channel (on a GS HT-286).
 
 The call is answered automatically, then the user enters an
 extension. Dial() is called with both T and t flags.  When the
 bridge is made between the channels, the caller on the Zap
channel
 can hit '#' to transfer, but the caller on the SIP channel
cannot.
 No messages whatsoever are displayed on the console when the SIP
 user hits any keys.  What am I missing?
 
 
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Re: [Asterisk-Users] Dialogic supported well?

2004-03-05 Thread Steve Underwood
Scott Stingel wrote:

If you'll be running commercial apps, I would recommend that you do a lot of
testing, especially load testing, with the types of applications you'll be
running.  Dialogic boards, although incredibly expensive, do have lots of
horsepower built in for the purposes of encoding and decoding the voice
streams, decoding single and DTMF codes, voice energy and cadence detection,
etc.  Digium boards rely 100% on the processor that they run in to perform
these functions.  I was a little disappointed to find that I can (so far)
reliably only handle 4 E1's in a (very) high-volume IVR app.  In the past
I've run Dialogic-based systems which handled much more load (but also which
cost 4 times as much in hardware!)  Although Digium's newer TE410P board is
capable of bus mastering, I found that it made little difference in the
number of channels I could run.
 

Actually, there isn't that much processing power on the Dialogic boards. 
They have rather limited DSP and MCU resources. That is why they only 
handle codecs of trivial complexity. The reason they get better high 
load results for IVRs is they have huge latency, which helps enormously 
with the response times the applications level code needs to achieve. 
Such high latency kills phone calls, but nobody notices for IVR use. 
Most Dialogic cards are incapable of doing anything other than IVR or 
call switching through their mezzanine buses, as they are not full 
duplex. Even so, the throughput you can achieve in pure IVR applications 
is not that great. When you hear of people with a large bunch of T1s or 
E1s into a Dialogic box, it is normally some limited IVR work plus a lot 
of call switching. That call switched data passes across the mezzanine 
bus, and has no impact on the main processor.

Some of the JCT cards from Dialogic can be set to a lower latency, and 
are full duplex. This is aimed at TTS + ASR use, rather than VoIP calls. 
The lowest latency is still much higher than the Digium cards, but even 
with this the number of channels you can handle reliably is a lot lower 
than when you use the traditional high latency Dialogic modes.

Dialogic support was great 10 years ago, but is now almost non-existant. 
Their drivers are buggy, and not keeping up with the times. Their 
previously active forums seem to be in serious decline. Most of their 
long term customers find it hard to say nice things about them. They are 
expensive. On the other hand, they do have broad approvals across the 
globe. I think that is their biggest asset.

Regards,
Steve
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[Asterisk-Users] gnophone and sip phone

2004-03-05 Thread Zen Kato
Hi,

I installed 'gnophone' on my notePC(RHL9 linux-2.4.20-30.9)
with asterisk CVS-02/05/04. I have three unsolved problems:
(1)call from gnophone to sip phone is OK, but gnophone's 
   speaker volume is very low even though setting highest 
   volume with gmix, the speaker volume is very high. 
   The sip hardphone side: my voice returns back to 
   earphone of handset(echo?).

(2)can not make a call from sip hardphone to gnophone
   *CLI says as follows;
Mar  6 11:27:55 NOTICE[81926]: chan_iax.c:4098 socket_read: Rejected connect attempt 
from 192.168.0.11, request '[EMAIL PROTECTED]' does not exist
Urgent handler
Mar  6 11:27:55 WARNING[81926]: chan_iax.c:3951 socket_read: Call rejected by 
192.168.0.11: No such context/extension
Mar  6 11:27:55 NOTICE[81926]: chan_iax.c:1050 iax_destroy: Avoiding IAX destroy 
deadlock
-- Called [EMAIL PROTECTED]
Urgent handler
-- Nobody picked up in 5000 ms
-- Hungup 'IAX[192.168.0.11:5036]/7'
Mar  6 11:28:00 WARNING[294929]: channel.c:517 ast_channel_free: Channel 
'IAX[192.168.0.11:5036]/7' may not have been hung up properly
Urgent handler
Mar  6 11:28:10 WARNING[294929]: pbx.c:1834 ast_pbx_run: Timeout, but no rule 't' in 
context 'sip'
- end of *CLI -

my iax.conf is;
[916]
type=friend
host=dynamic
defaultip=192.168.0.11
port=5036
secret=916
context=default

my extensions.conf is;
[default]
..
exten = 916,1,Dial(IAX/[EMAIL PROTECTED],5,r)
.

What is the meaning of '[EMAIL PROTECTED]' above *CLI?
Do I miss something in 'iax.conf'?

(3)When I start 'gnophone', I have to do the following
sequence;
1.start mpg123 some.mp3
2.start 'asterisk'
3.stop mpg123
4.start 'gnophone'

Because, asterisk graps sound device and the others can not
use sound device after asterisk started. How can I release
'sound device' after asterisk started?


Zen

 
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[Asterisk-Users] 7960 conference ?

2004-03-05 Thread Chris Clifton
Anyone been able to get the conference feature on the 7960's to work without
using meetme ?

I get  - warning,  chan_sip.c:2103 process_sdp: No compatible codecs!

Thanks,
Chris

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[Asterisk-Users] E1 Red Alarm

2004-03-05 Thread Nicholas Bachmann
Howdy -

I'm trying to get a Malaysian PRI E1 up on a TE410P, with no luck.  
Right now, the setup is

Telco - HDSL - WorldDSL UTU801- 2 BNC E1 - balun - crossover - TE410P

Right now, the CSU/DSU-ish WorldDSL box has a green light indicating E1 
sync, but the TE410P shows a red alarm.  I checked the card by plugging 
the crossover from port 1 to port 2 on the 410 (it worked fine).  It I 
change any of the cabling (i.e. swap things around), the green light 
goes off.

I have my suspicions about the balun 
(http://www.ctcu.com/catalog/datacom/balun.pdf).  Would a DB15F-RJ45 
converter be better the the BNC-balun-RJ45 arrangement we have now?

Here's my zaptel.conf:
span=1,1,0,ccs,hdb3
bchan=1-15
dchan=16
bchan=17-31
The telco line IS working; it was tested and put in a couple of days 
ago.  Any ideas why this isn't working?

Nick

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