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Can Asterisk(*) be installed on this and if so are the setup
instructions any different?
I have a client that's asking.

Specs: 533Mhz, 512MB RAM, 10GB HDD, Cobalt 4 OS

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Re: [Asterisk-Users] Small office setup/using analog lines w/

2005-08-21 Thread karl

Jenny -

I'm glad I'm not the only one! I just installed Asterisk on Friday and I
spent all day trying to de-gremlin my system. I'm glad I'm doing it for
myself and we haven't switched from our legacy system yet, but I have a
potential client that wants to see how well I can implement this system
for myself...

(Speaking of which, don't forget that your supplier and consultant
probably make more off of other phone systems and a lot of old-line
suppliers are kind of afraid of Asterisk. There were some big companies
that lost out when the State of Oregon's court system went to Asterisk for
their teleconferencing needs... 600 MeetMe sessions a day does say
something, I think.)

First, it would help -me- to know precisely what hardware you're using
(including network... switches and cards) and what phones you're using.
You can email me off-list if you want.

I'm personally using a AMD Sempron-based system with a TDM-04B (4x FXO) in
it, routing out through an IntelPro 10/100 card. The data switch is
currently a crappy D-Link, but more on that and the network card in a
second. The other end is one soft phone on Linux and two SPA-841 hardware
phones.

When I first installed, the echo was terrible. Tweaking around with the
echo settings in zapata.conf endlessly didn't do a whole lot. So, using my
go-go-gadget troubleshooting instincts (and my handy-dandy go-go-gadget
credit card) I figured the problem might be network latency ... so I
replaced the crappy realtek network card with a high-quality Intel one,
and borrwed a Dell managed switch from a client for the day. Voila, a few
more tweaks to the echo learning rate and my echo problems went away, and
I spent 45 minutes learning about my sister's love life in the wilds of
downtown San Francisco. (Note to self: Don't use family to test telecom
systems.) The clicking persisted, though. I thought it was the phone at
first, but if you're experiencing it too... well, apparently it's not.

I searched the archives and couldn't find any other reference to clicking
noises, so I'm at a loss but I'm hoping it's wiring-based and heading back
to the office tomorrow to try and solve it. I think you're right that
Digium is focusing on their T-1/E-1 market. I've noticed it when I called
with a question, and that really ticks me off. It might be a decent, sound
business decision based on profits for the different cards, but it's
leaving a lot of us smaller business operators -- those who might take
great experiences with Digium and Asterisk products into potential large
customers -- swinging in the wind.

-Karl Katzke
Streetlamp Software Solutions
http://www.streetlampsoftware.com


 Hi,

 We recently tried installing Asterisk for a small office. We figured the
 safest way to go would be to buy from someone who sold equipment
 specifically for Asterisk and to use a consultant that they
 recommended.  However ... it didn't turn out so great.  Sound quality is
 terrible -- the echo is pretty bad, and there are popping noises, too.
 Callers say that people on the Asterisk end sound very faint, while
 people on the Asterisk end hear people maybe too loundly (might be
 related to the popping noises -- sounds like when you have stereo turned
 up too high).  The reseller and the consultant both say that the most
 likely cause for this is using Digium cards w/ analog phone lines.
 Apparently, they say, sound quality can be pretty bad.

 I called Digium and they gave me some suggestions for settings, but
 nothing has worked well. So I wanted to ask others ... has anyone had
 good luck with using analog phone lines and Asterisk? Especially with
 Digium cards (we use the TDM400P)? Although from reading articles on the
 net it sounds like people do have a lot of echo problems, it also sounds
 like some people are using analog phone lines with some success.

 FYI, what I've mainly done is try changing echotraining, echocancel,
 echocancelwhenbridged, txgain, and rxgain in zapata.conf. I've heard
 from the reseller that what might work better is to trade the Digium
 cards in for VegaStream gateway. It's more expensive, but apparently has
 a DSP built in that should increase voice quality. Of course, they say
 there are no guarantees with this.  They also mentioned (after the fact)
 that Asterisk systems don't necessarily save money. So far, the
 experience has been very frustrating and I'd love to hear some success
 stories from others (or more info on what I can realistically expect
 from an Asterisk system)! And, of course, some ideas on how I can get
 things to work better.

 One of the next tests will be using Asterisk with a VoIP provider to see
 what the sound quality is like with digital on both ends. PRI sounds
 like it'd be even better, but for an office w/ 5 people, it sounds
 pretty expensive. How do other people do this?

 Thanks in advance for any pointers!

 Jen


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[Asterisk-Users] Re: [Asterisk-Dev] IM patch

2005-08-21 Thread Olle E. Johansson

harry gaillac wrote:
 Hello,
 
 I patched asterisk cvs head sources with
 http://juraj.bednar.sk/work/software/asterisk/messaging/
 and  presnce patch without success.
 
 asterisk send 405 method not allowed to sender.
 I use polycom ip300. 
THat is a response to the polycom's PUBLISH request, a method that
is not implemented.

/O
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[Asterisk-Users] Re: [Asterisk-Dev] IM patch

2005-08-21 Thread Olle E. Johansson

harry gaillac wrote:
 Hello,
 
 I patched asterisk cvs head sources with
 http://juraj.bednar.sk/work/software/asterisk/messaging/
 and  presnce patch without success.
 
 asterisk send 405 method not allowed to sender.
 I use polycom ip300. 
THat is a response to the polycom's PUBLISH request, a method that
is not implemented.

/O
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Re: [Asterisk-Users] ISDN BRI voice one way only

2005-08-21 Thread Stefan Gofferje


Hi,

Klemens Kasemaa schrieb:

hi

PSTN -- [Teles ISDN / Asterisk] -- SIP client

When call is made through ISDN, no matter if taken from PSTN or
Asterisk side, person in PSTN side can hear perfectly but in Asterisk
side I only hear a very scrambled or very low quality voice, words
repeated several times. Same is with echo test (call taken from PSTN)


Get a CAPI module for your Teles and try chan_capi-cm from 
http://sourceforge.net/projects/chan-capi/


Regards,
Stefan

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Re: [Asterisk-Users] Small office setup/using analog lines w/ Asterisk

2005-08-21 Thread John Daragon


jennyw wrote:

Hi,

We recently tried installing Asterisk for a small office. We figured the 
safest way to go would be to buy from someone who sold equipment 
specifically for Asterisk and to use a consultant that they 
recommended.  However ... it didn't turn out so great.  Sound quality is 
terrible -- the echo is pretty bad, and there are popping noises, too. 
Callers say that people on the Asterisk end sound very faint, while 
people on the Asterisk end hear people maybe too loundly (might be 
related to the popping noises -- sounds like when you have stereo turned 
up too high).  The reseller and the consultant both say that the most 
likely cause for this is using Digium cards w/ analog phone lines. 
Apparently, they say, sound quality can be pretty bad.


I called Digium and they gave me some suggestions for settings, but 
nothing has worked well. So I wanted to ask others ... has anyone had 
good luck with using analog phone lines and Asterisk? Especially with 
Digium cards (we use the TDM400P)? Although from reading articles on the 
net it sounds like people do have a lot of echo problems, it also sounds 
like some people are using analog phone lines with some success.


FYI, what I've mainly done is try changing echotraining, echocancel, 
echocancelwhenbridged, txgain, and rxgain in zapata.conf. I've heard 
from the reseller that what might work better is to trade the Digium 
cards in for VegaStream gateway. It's more expensive, but apparently has 
a DSP built in that should increase voice quality. Of course, they say 
there are no guarantees with this.  They also mentioned (after the fact) 
that Asterisk systems don't necessarily save money. So far, the 
experience has been very frustrating and I'd love to hear some success 
stories from others (or more info on what I can realistically expect 
from an Asterisk system)! And, of course, some ideas on how I can get 
things to work better.


One of the next tests will be using Asterisk with a VoIP provider to see 
what the sound quality is like with digital on both ends. PRI sounds 
like it'd be even better, but for an office w/ 5 people, it sounds 
pretty expensive. How do other people do this?


I started using Asterisk for my own small business about a year ago.

Externally we have a single analogue PSTN line (it's the house one...), 
an ISDN2e connection and an IAX2 connection (over 20:1 256/512kbps ADSL) 
with a DID in central London. The analogue line comes in to an old 
X100P, and the ISDN into an AVM Fritz! passive card.


Internally, we have a TDM400 which talks to analogue phones in the 
house. In my office (which is in a different building) we have a mixture 
of Snom and ipDialog phones and a Grandstream ATA attached to a fax machine.


We get a little echo on the ipDialog phone (but not enough to be a 
problem) when we talk to people on analogue phones. One of the handsets 
 attached to the TDM400 is a DECT phone, and there's a little flurry of 
training noise at the beginning of an incoming call, but after that the 
quality is good to perfect.


I'm just beginning to sell Asterisk systems. I agree that for some 
installations, it doesn't really make economic sense. In the UK, at 
least, you have to fall into a specific band of numbers-of-users and 
minutes-per-month for IP telephony to show a saving. Some of the small 
3-line-8-extension systems from (say) Panasonic will be cheaper than 
Asterisk once the hardware is bought and the time (or consultancy) 
applied. Of course, these systems don't have much in the way of 
flexibility or features, and I'm talking at the moment to a company that 
has three sites, is using Cisco's Call Manager, and has an Asterisk 
system merely to convert the H.323 from the Cisco to IAX2.  In this 
case, * could replace the CCM system in its entirety.


By the time you have 100 users, * is a no-brainer in economic terms. 
Small users only really save (IMHO) if they a) use an awful lot of 
minutes (or call abroad a lot), b) need flexibility of features, or c) 
need internal control.


Of course there may be local or exceptional circumstances which make 
this all a load of rubbish ! YMMV.


Oh, and on echo; read :

http://lists.digium.com/pipermail/asterisk-users/2005-March/096754.html

jd

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[Asterisk-Users] Warning Unable to allocate socket

2005-08-21 Thread Kamran Ahmad

hello


i m getting follwing messages in asterisk-1.0.9 what
is the reason calls are not going out. can u pls tel
me how to solve this


Aug 20 13:06:09 WARNING[7706]: rtp.c:829 ast_rtcp_new:
Unable to allocate socket: Too many open files
Aug 20 13:06:09 WARNING[7706]: channel.c:311
ast_channel_alloc: Alert pipe creation failed!
Aug 20 13:06:09 WARNING[7706]: chan_sip.c:2081
sip_new: Unable to allocate channel structure
Aug 20 13:06:09 NOTICE[7706]: chan_sip.c:7469
handle_request: Unable to create/find channel
Aug 20 13:06:22 WARNING[7706]: acl.c:216
ast_lookup_iface: Unable to get IP of eth0: Bad file
descriptor


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[Asterisk-Users] TDM11B modprobe wcfxs fails

2005-08-21 Thread Fumiaki Okushi

Hi,

I got my TDM11B and am trying to get it to work on my PC.
However, I'm having difficulty getting the wcfxs driver to
load. I've Googled this problem, and while there are others
who have ran into the same problem, none of the solutions
work for me.

I would very much appreciate it if you could suggest things
I could try to resolve this problem.

Details follow.

The PC is somewhat old - Tyan S2054, which has Intel i810 -
but I don't think this would be a problem. I
clean-installed Debian Sarge and am using 2.4.27 kernel.

The TDM11B has its own IRQ (according to lspci -v).

lspci -v shows the card as

:01:04.0 Communication controller: Tiger Jet Network Inc. Tiger3XX
Modem/ISDN interface

I'm following the instructions in
http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html/book1.html.

I'm using the Asterisk/Zaptel software that's on the CD that
came with the TDM11B. It has version 1.0.7.

The zaptel driver loads fine. The problem I'm having is
with the wcfxs driver.

matsuri:~# modprobe zaptel
matsuri:~# modprobe wcfxs
/lib/modules/2.4.27/misc/wcfxs.o: init_module: No such device
Hint: insmod errors can be caused by incorrect module parameters, including
invalid IO or IRQ parameters.
You may find more information in syslog or the output from dmesg
/lib/modules/2.4.27/misc/wcfxs.o: insmod /lib/modules/2.4.27/misc/wcfxs.o
failed
/lib/modules/2.4.27/zaptel/wcfxs.o: init_module: No such device
Hint: insmod errors can be caused by incorrect module parameters, including
invalid IO or IRQ parameters.
You may find more information in syslog or the output from dmesg
/lib/modules/2.4.27/zaptel/wcfxs.o: insmod /lib/modules/2.4.27/zaptel/wcfxs.o
failed
/lib/modules/2.4.27/zaptel/wcfxs.o: insmod wcfxs failed
matsuri:~#

I've tried all the PCI slots (4 of them). I get the same
error.

The power cable is connected to the card. I've verified
that there is power to the cable and also tried different
power cable.

The kernel used above is built by myself and it's not the
default kernel. (The default kernel that gets installed by
the Debian Installer couldn't even load the zaptel driver -
I get lots of missing references.)

Thanks,
Fumi Okushi

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Re: [Asterisk-Users] Small office setup/using analog lines w/ Asterisk

2005-08-21 Thread Ariel Batista


Jennyw,

I have setup about 8 Asterisk systems with The TDM400p boards in them. Yes 
allot of them had at the beginning some echo and other things. But I have 
been able to work and get them fixed.


1) Make sure your motherboard is able to assign it's own IRQ for the board. 
This is one of the most important things.
2) There is a tool on the system that lets you set the txgain and rxgain. 
It's called ztmonitor which you can use to see how the volume is setup then 
you can make some adjustments.
3) It's important to make sure your system has good network card. I have had 
some problems with Realtech with echo.

4) You also need to make sure you have the phones on connect to a switch.
5) Asterisk system does save allot of money in the long run.  I am sorry 
that the person you got was not able to help. But I know that the TDM400p 
boards have there well critic's but they do work.
6) For us to give you more help we are going to need to know more about you 
system. What is the server your using? What phones? How is your network 
setup?  If you want you can email me directly. I will try to help you out 
with your setup.


Ariel Batista

jennyw wrote:

Hi,

We recently tried installing Asterisk for a small office. We figured
the safest way to go would be to buy from someone who sold equipment
specifically for Asterisk and to use a consultant that they
recommended.  However ... it didn't turn out so great.  Sound quality
is terrible -- the echo is pretty bad, and there are popping noises,
too. Callers say that people on the Asterisk end sound very faint,
while people on the Asterisk end hear people maybe too loundly (might
be related to the popping noises -- sounds like when you have stereo
turned up too high).  The reseller and the consultant both say that
the most likely cause for this is using Digium cards w/ analog phone
lines. Apparently, they say, sound quality can be pretty bad.

I called Digium and they gave me some suggestions for settings, but
nothing has worked well. So I wanted to ask others ... has anyone had
good luck with using analog phone lines and Asterisk? Especially with
Digium cards (we use the TDM400P)? Although from reading articles on
the net it sounds like people do have a lot of echo problems, it also
sounds like some people are using analog phone lines with some
success.
FYI, what I've mainly done is try changing echotraining, echocancel,
echocancelwhenbridged, txgain, and rxgain in zapata.conf. I've heard
from the reseller that what might work better is to trade the Digium
cards in for VegaStream gateway. It's more expensive, but apparently
has a DSP built in that should increase voice quality. Of course,
they say there are no guarantees with this.  They also mentioned
(after the fact) that Asterisk systems don't necessarily save money.
So far, the experience has been very frustrating and I'd love to hear
some success stories from others (or more info on what I can
realistically expect from an Asterisk system)! And, of course, some
ideas on how I can get things to work better.

One of the next tests will be using Asterisk with a VoIP provider to
see what the sound quality is like with digital on both ends. PRI
sounds like it'd be even better, but for an office w/ 5 people, it
sounds pretty expensive. How do other people do this?

Thanks in advance for any pointers!

Jen


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Re: [Asterisk-Users] Asterisk(*) on a Cobalt RaQ2?

On Sunday 21 August 2005 02:25, Joshua Abbott wrote:
 Can Asterisk(*) be installed on this and if so are the setup
 instructions any different?
 I have a client that's asking.

 Specs: 533Mhz, 512MB RAM, 10GB HDD, Cobalt 4 OS

As long as you don't expect to run more than a couple of calls at a time 
through it, it will work. You don't have much CPU power to go around. I use a 
similar box at my home and it's OK for that, but hardly for a business. 

Pickup this script and you'll have an easy install.

wget szmidt.org/asterisk/asterisk-update.sh

Set the execute bit and run it. To get a specific version you'll modify the 
version variable at the top. 

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Re: [Asterisk-Users] Searching For a Voip Provider

On Thursday 18 August 2005 14:55, chawki hammoud wrote:
 Hi:

 Please advice me of a voip provider with reasonable
 reseller program. I was using voipjet and it has a lot
 of problems.

 Did anyone experienced asteriskout.com service? They
 have good prices.

What you may want to do is to subscribe to asterisk-biz where you can find 
several carriers.

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Re: [Asterisk-Users] Small office setup/using analog lines w/ Asterisk


People, please cut down the original post in your replies. 
It's wasting space, bandwidth and time.

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[Asterisk-Users] Asterisk 1.0.9 on SuSE 9.2 with ISDN BRI zaphfc?

2005-08-21 Thread Lars Dybdahl

I would like to know how to install asterisk 1.0.9 with zaphfc working
on a SuSE 9.2. I tried this:

- The rpms with SuSE 9.2 are asterisk 1.0.6
- bristuff works, except for zaphfc, which doesn't compile.
- The official asterisk download file doesn't contain isdn bri support

Any ideas?

Lars Dybdahl.
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Re: [Asterisk-Users] Re: Asterisk-Users Digest, Vol 13, Issue 131

On Sunday 21 August 2005 02:24, Nguyen Trung Tin wrote:
 Hello

 how to calculator billing exactly when IAX accept the call, my configure

 customer -- telco --- asterisk -- ACD -- IAX

The phone company does not bill for talk time but for use time. You used the 
phone network for 33 seconds. Besides if you make that call you will be 
billed according to their billing system. Which may bill in a number of ways. 
X amount for the first minute, then Y for every N seconds.

That can typically translate to 60 second, 6 second or 1 second billing 
increments. 
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Re: [Asterisk-Users] Warning Unable to allocate socket

2005-08-21 Thread Eric Wieling aka ManxPower


Kamran Ahmad wrote:

i m getting follwing messages in asterisk-1.0.9 what
is the reason calls are not going out. can u pls tel
me how to solve this

Aug 20 13:06:09 WARNING[7706]: rtp.c:829 ast_rtcp_new:
Unable to allocate socket: Too many open files
Aug 20 13:06:09 WARNING[7706]: channel.c:311
ast_channel_alloc: Alert pipe creation failed!
Aug 20 13:06:09 WARNING[7706]: chan_sip.c:2081
sip_new: Unable to allocate channel structure
Aug 20 13:06:09 NOTICE[7706]: chan_sip.c:7469
handle_request: Unable to create/find channel
Aug 20 13:06:22 WARNING[7706]: acl.c:216
ast_lookup_iface: Unable to get IP of eth0: Bad file
descriptor


stop asterisk, then start asterisk.  Asterisk (at least 1.0.9) can leak 
file descriptors in some situations and you should restart Asterisk 
every once in a while.

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Re: [Asterisk-Users] ISDN BRI voice one way only

2005-08-21 Thread Klemens Kasemaa

hi

  PSTN -- [Teles ISDN / Asterisk] -- SIP client
  
  When call is made through ISDN, no matter if taken from PSTN or
  Asterisk side, person in PSTN side can hear perfectly but in Asterisk
  side I only hear a very scrambled or very low quality voice, words
  repeated several times. Same is with echo test (call taken from PSTN)
 
 Get a CAPI module for your Teles and try chan_capi-cm from 
 http://sourceforge.net/projects/chan-capi/

Accordingly capi.org this card does have capi support but not under linux.
Because of it's a ISA card, I can't use zaphfc also. So any help is
appreciated.


with
rgrds,
klem

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Re: [Asterisk-Users] Looking for Provider


I like Broadvoice but there are others.

Do you want SIP or IAX termination? Business or residential?

Mark

Joshua Abbott wrote:

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hello
I currently have internet service through MediaCom (Cable Internet)
and need to find a VOIP provider that is compatible with Asterisk and
Cable Internet.
Any ideas?

I'm in Missouri about 1.5 hours west of St Louis, MO in a town called
Hermann (65041 zip code)


Joshua
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--

Mark, G7LTT/KC2ENI
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http://www.g7ltt.com
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Re: [Asterisk-Users] Looking for Provider

-BEGIN PGP SIGNED MESSAGE-
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SIP and/or IAX would be nice. Residential.


Mark Phillips wrote:

 I like Broadvoice but there are others.

 Do you want SIP or IAX termination? Business or residential?

 Mark

 Joshua Abbott wrote:


 Hello
 I currently have internet service through MediaCom (Cable Internet)
 and need to find a VOIP provider that is compatible with Asterisk and
 Cable Internet.
 Any ideas?

 I'm in Missouri about 1.5 hours west of St Louis, MO in a town called
 Hermann (65041 zip code)


 Joshua



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Re: [Asterisk-Users] Small office setup/using analog lines w/

On the clicking front, it could still be packet loss. I recently (just 
this week in fact) solved a clicking problem my client was having. 
turned out to be interference from his wireless network bridge. Both it 
and his Cisco ATA186 were sitting next to each other. I put about 3 feet 
of space between them and solved the problem.


I think what was happening was that the radio was interfering with the 
data on the cable causing loss/bad checksums/whatever.


That's just my guess. What would I know? I'm only an Extra Class ham 
radio operator.


Mark

[EMAIL PROTECTED] wrote:

Jenny -

I'm glad I'm not the only one! I just installed Asterisk on Friday and I
spent all day trying to de-gremlin my system. I'm glad I'm doing it for
myself and we haven't switched from our legacy system yet, but I have a
potential client that wants to see how well I can implement this system
for myself...

(Speaking of which, don't forget that your supplier and consultant
probably make more off of other phone systems and a lot of old-line
suppliers are kind of afraid of Asterisk. There were some big companies
that lost out when the State of Oregon's court system went to Asterisk for
their teleconferencing needs... 600 MeetMe sessions a day does say
something, I think.)

First, it would help -me- to know precisely what hardware you're using
(including network... switches and cards) and what phones you're using.
You can email me off-list if you want.

I'm personally using a AMD Sempron-based system with a TDM-04B (4x FXO) in
it, routing out through an IntelPro 10/100 card. The data switch is
currently a crappy D-Link, but more on that and the network card in a
second. The other end is one soft phone on Linux and two SPA-841 hardware
phones.

When I first installed, the echo was terrible. Tweaking around with the
echo settings in zapata.conf endlessly didn't do a whole lot. So, using my
go-go-gadget troubleshooting instincts (and my handy-dandy go-go-gadget
credit card) I figured the problem might be network latency ... so I
replaced the crappy realtek network card with a high-quality Intel one,
and borrwed a Dell managed switch from a client for the day. Voila, a few
more tweaks to the echo learning rate and my echo problems went away, and
I spent 45 minutes learning about my sister's love life in the wilds of
downtown San Francisco. (Note to self: Don't use family to test telecom
systems.) The clicking persisted, though. I thought it was the phone at
first, but if you're experiencing it too... well, apparently it's not.

I searched the archives and couldn't find any other reference to clicking
noises, so I'm at a loss but I'm hoping it's wiring-based and heading back
to the office tomorrow to try and solve it. I think you're right that
Digium is focusing on their T-1/E-1 market. I've noticed it when I called
with a question, and that really ticks me off. It might be a decent, sound
business decision based on profits for the different cards, but it's
leaving a lot of us smaller business operators -- those who might take
great experiences with Digium and Asterisk products into potential large
customers -- swinging in the wind.

-Karl Katzke
Streetlamp Software Solutions
http://www.streetlampsoftware.com




Hi,

We recently tried installing Asterisk for a small office. We figured the
safest way to go would be to buy from someone who sold equipment
specifically for Asterisk and to use a consultant that they
recommended.  However ... it didn't turn out so great.  Sound quality is
terrible -- the echo is pretty bad, and there are popping noises, too.
Callers say that people on the Asterisk end sound very faint, while
people on the Asterisk end hear people maybe too loundly (might be
related to the popping noises -- sounds like when you have stereo turned
up too high).  The reseller and the consultant both say that the most
likely cause for this is using Digium cards w/ analog phone lines.
Apparently, they say, sound quality can be pretty bad.

I called Digium and they gave me some suggestions for settings, but
nothing has worked well. So I wanted to ask others ... has anyone had
good luck with using analog phone lines and Asterisk? Especially with
Digium cards (we use the TDM400P)? Although from reading articles on the
net it sounds like people do have a lot of echo problems, it also sounds
like some people are using analog phone lines with some success.

FYI, what I've mainly done is try changing echotraining, echocancel,
echocancelwhenbridged, txgain, and rxgain in zapata.conf. I've heard
from the reseller that what might work better is to trade the Digium
cards in for VegaStream gateway. It's more expensive, but apparently has
a DSP built in that should increase voice quality. Of course, they say
there are no guarantees with this.  They also mentioned (after the fact)
that Asterisk systems don't necessarily save money. So far, the
experience has been very frustrating and I'd love to hear some success
stories from others (or more

RE: [Asterisk-Users] Re: [Asterisk-Dev] IM patch

2005-08-21 Thread harry gaillac

What about asterisk chan_sip and IM +presence !!!
Harry
--- Olle E. Johansson [EMAIL PROTECTED] a écrit :

 harry gaillac wrote:
  Hello,
  
  I patched asterisk cvs head sources with
 

http://juraj.bednar.sk/work/software/asterisk/messaging/
  and  presnce patch without success.
  
  asterisk send 405 method not allowed to sender.
  I use polycom ip300. 
 THat is a response to the polycom's PUBLISH request,
 a method that
 is not implemented.
 
 /O
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Re: [Asterisk-Users] Small office setup/using analog lines w/ Asterisk

2005-08-21 Thread Andrew Kohlsmith

On Sunday 21 August 2005 01:05, jennyw wrote:
 up too high).  The reseller and the consultant both say that the most
 likely cause for this is using Digium cards w/ analog phone lines.
 Apparently, they say, sound quality can be pretty bad.

The reseller/consultant aren't worth the money you paid them, then.

Interfacing to plain-old analog lines can be problematic.  The TDM400P FXO 
modules are tunable but it takes time and testing to get it right, not 
playing about willy-nilly with settings in an attempt to solve the problem 
through entropic little adjustments without a clear idea of what they do and 
how they work.

Asterisk is a very difficult application on a system.  Interfacing to anything 
outside the computer, whether it be an analog telephone line, a local SIP 
phone or a remote VOIP provider requires that the system's ability to access 
its resources reliably and with repeatable access times.  As simple as this 
sounds it is a very difficult problem and the #1 reason why VOIP is so 
difficult to roll out on commodity hardware.  You simply can't use any old 
system and any old network card and any old network gear (router/firewall, 
switches, etc.) and get good results.

 nothing has worked well. So I wanted to ask others ... has anyone had
 good luck with using analog phone lines and Asterisk? Especially with
 Digium cards (we use the TDM400P)? Although from reading articles on the
 net it sounds like people do have a lot of echo problems, it also sounds
 like some people are using analog phone lines with some success.

The echo problems are almost always due to one of two things: poor line tuning 
or crappy base hardware (computer).

Now the older version of the TDM cards and FXO modules specifically had 
issues, but they have, to my knowledge, all been resolved.  I used to 
recommend a T1 card + channel bank (Adit600) even for a couple channels, but 
nowadays I have no compunctions in recommending the TDM400 and FXO modules.

 echocancelwhenbridged, txgain, and rxgain in zapata.conf. I've heard
 from the reseller that what might work better is to trade the Digium
 cards in for VegaStream gateway. It's more expensive, but apparently has
 a DSP built in that should increase voice quality. Of course, they say
 there are no guarantees with this.  They also mentioned (after the fact)
 that Asterisk systems don't necessarily save money. So far, the

Find a new reseller, and post their name here so we can all avoid them.

I've rolled out numerous asterisk installations with good success.  As I 
mentioned earlier, the trick is measured, controlled tests and methodical  
experimentation.  

http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html

is the method I use to tune the line, but there is more to it now since there 
is the fxotune application which is used to set up the FIR filter on the FXO 
card's DAA.

If you're using a SIP phone to test with, put a second network card in the 
Asterisk box and plug the phone directly into it (with a crossover cable) and 
adjust the Asterisk settings to see and use the second card.  This will 
determine if your network is causing issues by eliminating it from the 
equation.  Make sure you use a decent network card (I love the Intel 
eepro/100 cards myself).

Also, if you're going out through a VOIP provider, make damn sure you either 
use a dedicated DSL link (my personal recommendation) or make really sure 
that your router can properly tag and prioritize outgoing traffic, and that 
it is also doing its best to prevent the other side of the link from 
flooding your incoming pipe.  http://www.mixdown.ca/~andrew/dump/rc.tc is the 
script I use with good success.

As far as Asterisk being more expensive than other systems...  doubtful.  You 
can get a cheap Nortel 3x8 for cheap, sure, but then its limitations will 
have you buying a small MICS... Now add their $4000 voicemail system, $500 
trunk cards for four FXO channels... oh wait, you want caller-id on those?  
$600 then...  oh wait, you want VOIP on it?  $2500 here, $500 there, $1000 
the other place...  

Make sure you're comparing apples to apples.  I feel that Asterisk runs *very* 
well on most hardware I've thrown it at, and it is far far far more 
configurable than any proprietary KSU or PBX, and a damn sight cheaper than 
*ANY* PBX out there.

 One of the next tests will be using Asterisk with a VoIP provider to see
 what the sound quality is like with digital on both ends. PRI sounds
 like it'd be even better, but for an office w/ 5 people, it sounds
 pretty expensive. How do other people do this?

Yes, an office with 5 people (probably only two POTS lines I am guessing) is 
not really a good choice for PRI.  ISDN BRI if you can get it would be 
better, or simply finding a trusted VOIP provider and getting a DID from them 
would be easiest, but I would then recommend two DSL providers on the same 
POTS line (it's all PPPoE anyway) for failover.

-A.

[Asterisk-Users] Not-Registered Problem

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hello,
What are some common reasons why a phone would report not registered
even when the extension has been setup through Asterisk(*) AND phone
username/password is correct?

Joshua

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Re: [Asterisk-Users] ZT_CHANCONFIG failed on channel 1 - WAS WORKING!!

2005-08-21 Thread Scott Brown


Hi Paul:

Thanks very much for the suggestion.  I don't understand why this just  
began to happnen.  I never
had problems before.  Your suggestion has shed a lot of light on the  
problem.  Because wcfxs disagrees about the version of the symbols  
listed below, I get the following unknown symbol problems for:


zt_receive
zt_qevent_lock
zt_ec_chunk
zt_transmit
zt_unregister
zt_hooksig
zt_register

With: wcfxs: disagrees about version of symbol...

BTW, I also did a full checkout (v1-0) and rebuild (deleting modules and  
includes beforehand) of

astrisk zaptel and libpri.  That didn't change anything.

Below is the resultant output from the -v modprobe and dmesg output.  If  
you or anyone else can help with this it would really be appreciated,  
thanks:


# modprobe -v -n wctdm
insmod /lib/modules/2.6.11.4-21.8-default/kernel/lib/crc-ccitt.ko
insmod /lib/modules/2.6.11.4-21.8-default/misc/zaptel.ko
install /sbin/modprobe --ignore-install wcfxs  /sbin/ztcfg
# modprobe -v wctdm
insmod /lib/modules/2.6.11.4-21.8-default/kernel/lib/crc-ccitt.ko
insmod /lib/modules/2.6.11.4-21.8-default/misc/zaptel.ko
install /sbin/modprobe --ignore-install wcfxs  /sbin/ztcfg
insmod /lib/modules/2.6.11.4-21.8-default/extra/wcfxs.ko
FATAL: Error inserting wcfxs  
(/lib/modules/2.6.11.4-21.8-default/extra/wcfxs.ko): Unknown symbol

in module, or unknown parameter (see dmesg)
FATAL: Error running install command for wcfxs

pertinent dmesg output:
zaptel: module not supported by Novell, setting U taint flag.
Zapata Telephony Interface Registered on major 196
wcfxs: module not supported by Novell, setting U taint flag.
Zapata Telephony Interface Unloaded
zaptel: module not supported by Novell, setting U taint flag.
Zapata Telephony Interface Registered on major 196
wcfxs: module not supported by Novell, setting U taint flag.
wcfxs: disagrees about version of symbol zt_receive
wcfxs: Unknown symbol zt_receive, st_info == 0x1
wcfxs: disagrees about version of symbol zt_qevent_lock
wcfxs: Unknown symbol zt_qevent_lock, st_info == 0x1
wcfxs: disagrees about version of symbol zt_ec_chunk
wcfxs: Unknown symbol zt_ec_chunk, st_info == 0x1
wcfxs: disagrees about version of symbol zt_transmit
wcfxs: Unknown symbol zt_transmit, st_info == 0x1
wcfxs: disagrees about version of symbol zt_unregister
wcfxs: Unknown symbol zt_unregister, st_info == 0x1
wcfxs: disagrees about version of symbol zt_hooksig
wcfxs: Unknown symbol zt_hooksig, st_info == 0x1
wcfxs: disagrees about version of symbol zt_register
wcfxs: Unknown symbol zt_register, st_info == 0x1
load_module: err 0xfffe (dont worry)
wcfxs: module not supported by Novell, setting U taint flag.
wcfxs: disagrees about version of symbol zt_receive
wcfxs: Unknown symbol zt_receive, st_info == 0x1
wcfxs: disagrees about version of symbol zt_qevent_lock
wcfxs: Unknown symbol zt_qevent_lock, st_info == 0x1
wcfxs: disagrees about version of symbol zt_ec_chunk
wcfxs: Unknown symbol zt_ec_chunk, st_info == 0x1
wcfxs: disagrees about version of symbol zt_transmit
wcfxs: Unknown symbol zt_transmit, st_info == 0x1
wcfxs: disagrees about version of symbol zt_unregister
wcfxs: Unknown symbol zt_unregister, st_info == 0x1
wcfxs: disagrees about version of symbol zt_hooksig
wcfxs: Unknown symbol zt_hooksig, st_info == 0x1
wcfxs: disagrees about version of symbol zt_register
wcfxs: Unknown symbol zt_register, st_info == 0x1
load_module: err 0xfffe (dont worry)

Thanks for the help.

Scott
 On Sat, 20 Aug 2005 03:58:18 -0600, Paul Hewlett  
[EMAIL PROTECTED] wrote:



On Saturday 20 August 2005 09:58, Scott Brown wrote:

Hi Matt:

That suggestion is possibly on the right track.  It made me remember  
that -

although I'm not using Fedora, but SuSE 9.3, that it went through an
automatic network update just recently.  After that, I tried updating  
the

Zaptel files from CVS and recompiling everything, but to no avail.  The
same error still occured.  I eliminated hardware by swapping out a  
working

TDM400 with the same FXS/FXO configuration.  The same error occurs.  The
SuSE update may have moved some of the required files, although there  
are
no complaints during the build and I can't determine what may have  
moved.


   Currently I am doing the following on SuSE :

   First reboot the PC with asterisk disabled. This will force the  
creation of

the devices during boot from the /etc/udev/rules.d files.

   Try modprobing :

modprobe -v -n wctdm

   This does nothing but tells you what would happen. If your
modprobe.d/zaptel file is correct the the output from this command will  
be
loading of zaptel,wcfxs and an execution of ztcfg. In other words you do  
not

have to modprobe more than one module - dependencies are sorted by the
modprobe.d/zaptel file. If you want -vv on the ztcfg file edit
modprobe.d/zaptel. I remember from the wiki somewhere that one must not
execute ztcfg more than once and this will happen if you modprobe zaptel  
and

then wctdm and then execute

Re: [Asterisk-Users] ZT_CHANCONFIG failed on channel 1 - WAS WORKING!!

2005-08-21 Thread Scott Brown


Hi Rich:

Does the new distro tree issue explain the unresolved symbol references  
noted in my last post?  I don't know if SuSE's 9.3 network autoupdate  
would have changed the tree structure.  The unresolved symbols are  (from  
dmesg):


zt_receive
zt_qevent_lock
zt_ec_chunk
zt_transmit
zt_unregister
zt_hooksig
zt_register

With: wcfxs: disagrees about version of symbol...

Scott

On Sat, 20 Aug 2005 06:11:53 -0600, Rich Adamson [EMAIL PROTECTED]  
wrote:


Seems the latest distro's have changed the layout of the linux source  
tree
needed to compile the zaptel stuff. I'm using FC3, upgraded the kernel,  
and

have the same issue. Was able to install new sources, but they too are
completely different tree layout compared to earlier stuff. The same is
apparently happening with other distro's as well.

There has been a bug item open for last several weeks relative to  
reworking

the make files for these items.




That suggestion is possibly on the right track.  It made me remember  
that -

although I'm not using Fedora, but SuSE 9.3, that it went through an
automatic network update just recently.  After that, I tried updating  
the

Zaptel files from CVS and recompiling everything, but to no avail.  The
same error still occured.  I eliminated hardware by swapping out a  
working

TDM400 with the same FXS/FXO configuration.  The same error occurs.  The
SuSE update may have moved some of the required files, although there  
are
no complaints during the build and I can't determine what may have  
moved.


I have still present and installed Bison 1.875, OpenSSL and zlib-devel,  
and
of course Linux source for this SuSE disto.  I'm completely faklempt!   
Can
someone shed light on this delima??   Thanks so much if you can.  I  
want my
As-terisk back!!!  It was working, damnit.   Thanks, Matt for your  
suggestion.


Scott

At 09:30 PM 8/19/2005, you wrote:
[EMAIL PROTECTED] wrote:
 
  Hi:
  I hope that someone can help with this problem that came up  
suddenly. I


Did you upgrade Fedora Core?

Check if the udev files still contain the required entries (normally  
fedora

copies the old ones to 50-udev-rules.old and makes new ones).



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Re: [Asterisk-Users] ISDN BRI voice one way only

2005-08-21 Thread Remco Barende

The easiest thing is probably to get a card that is more widely supported. 
Any cheap pci HFC-S card will do, they are sold for anthything between 9 
and 15 eur.


With an hfc-s card you can then use bristuff or chan_capi

On Sun, 21 Aug 2005, Klemens Kasemaa wrote:


hi


PSTN -- [Teles ISDN / Asterisk] -- SIP client

When call is made through ISDN, no matter if taken from PSTN or
Asterisk side, person in PSTN side can hear perfectly but in Asterisk
side I only hear a very scrambled or very low quality voice, words
repeated several times. Same is with echo test (call taken from PSTN)


Get a CAPI module for your Teles and try chan_capi-cm from
http://sourceforge.net/projects/chan-capi/


Accordingly capi.org this card does have capi support but not under linux.
Because of it's a ISA card, I can't use zaphfc also. So any help is
appreciated.


with
rgrds,
klem

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Re: [Asterisk-Users] Asterisk(*) on a Cobalt RaQ2?

Sure, this can work. I have an Asterisk install in my home office
running about the same hardware. Our is a newer mini-itx system booting
Astlinux from a compact flash card.

Just don't bother with any form of transcoding, especially G.729a...you
don't have the cpu power for more than 2 channels of transcode with out
other processes suffering badly. Stay G.711 all the way and you can
have multiple calls ongoing, VM, even conferencing without any trouble.

The wiki has good info on running * on small systems.

Michael

On Sun, 21 Aug 2005 01:25:32 -0500, Joshua Abbott wrote:

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Can Asterisk(*) be installed on this and if so are the setup
instructions any different?
I have a client that's asking.

Specs: 533Mhz, 512MB RAM, 10GB HDD, Cobalt 4 OS

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Re: [Asterisk-Users] Small office setup/using analog lines w/ Asterisk

I too have had trouble with FXO interfaces. I tried the Sipura SPA-3000
FXS/FXO device , X101p cards and a TDM11B card. None were satisfactory
for my small office with 6 extensions and 3 lines.

My longer term workaround was something that I setup just to bridge a
perioud when I was taking down a test server and putting a poduction
server online. I put provider based call forwarding on my POTS lines
(only 2) and forwarded them to my IP based 800 number. This turned out
so reliable that I have remained operating this way ever since, about 4
months now. People have remarked that the call quality is superb.

I ever there is a problem with my DSL line or * server I simply turn
off the call forwarding using a pair of analogue phones that I leave on
the POTS lines for just such emergencies.

This only addreses incomming calls. All outgoing calls are handled via
IP through another termination provider.

Michael

On Sat, 20 Aug 2005 22:05:31 -0700, jennyw wrote:

Hi,

We recently tried installing Asterisk for a small office. We figured the 
safest way to go would be to buy from someone who sold equipment 
specifically for Asterisk and to use a consultant that they 
recommended.  However ... it didn't turn out so great.  Sound quality is 
terrible -- the echo is pretty bad, and there are popping noises, too. 
Callers say that people on the Asterisk end sound very faint, while 

,snip

One of the next tests will be using Asterisk with a VoIP provider to see 
what the sound quality is like with digital on both ends. PRI sounds 
like it'd be even better, but for an office w/ 5 people, it sounds 
pretty expensive. How do other people do this?

Thanks in advance for any pointers!

Jen


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Re: [Asterisk-Users] TDM11B modprobe wcfxs fails

I would very much appreciate it if you could suggest things
I could try to resolve this problem.

Details follow.

The PC is somewhat old - Tyan S2054, which has Intel i810 -
but I don't think this would be a problem. I
clean-installed Debian Sarge and am using 2.4.27 kernel.

The TDM11B has its own IRQ (according to lspci -v).

lspci -v shows the card as

:01:04.0 Communication controller: Tiger Jet Network Inc. Tiger3XX
Modem/ISDN
interface

I'm following the instructions in

http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html/book1.html
.

I'm using the Asterisk/Zaptel software that's on the CD that
came with the TDM11B. It has version 1.0.7.

The zaptel driver loads fine. The problem I'm having is
with the wcfxs driver.

matsuri:~# modprobe zaptel
matsuri:~# modprobe wcfxs
/lib/modules/2.4.27/misc/wcfxs.o: init_module: No such device
Hint: insmod errors can be caused by incorrect module parameters, including
invalid IO or
IRQ parameters.
You may find more information in syslog or the output from dmesg
/lib/modules/2.4.27/misc/wcfxs.o: insmod /lib/modules/2.4.27/misc/wcfxs.o
failed
/lib/modules/2.4.27/zaptel/wcfxs.o: init_module: No such device
Hint: insmod errors can be caused by incorrect module parameters, including
invalid IO or
IRQ parameters.
You may find more information in syslog or the output from dmesg
/lib/modules/2.4.27/zaptel/wcfxs.o: insmod
/lib/modules/2.4.27/zaptel/wcfxs.o failed
/lib/modules/2.4.27/zaptel/wcfxs.o: insmod wcfxs failed
matsuri:~#

I've tried all the PCI slots (4 of them). I get the same
error.

The power cable is connected to the card. I've verified
that there is power to the cable and also tried different
power cable.

Two possible issues

1. The TDM card has several different revisions (rev e through h,
I believe). If you have one of the later revisions, you may need
zaptel software later then stable v1.07.

2. I think the driver you want is wctdm (not wcfxs). I don't use
the fxs modules, but the fxo modules use wctdm. Seems to me there
was a change some time ago where the fxs modules are now supported
from with wctdm. I'm not 100% sure though.

Also after doing a modprobe zaptel, do a lsmod to see what is
loaded. Your likely to find all the appropriate drivers were loaded
and you don't need to modprobe anything else. Look for something
like this:
wctdm 33728 4
wcfxo 13088 0
zaptel209028 19
wcusb,wctdm,wcfxo,wcte11xp,wct1xxp,wct4xxp,tor2
in the listing.

Re: [Asterisk-Users] TDM11B modprobe wcfxs fails

2005-08-21 Thread Eric Wieling aka ManxPower


Rich Adamson wrote:
  2. I think the driver you want is wctdm (not wcfxs). I don't use

the fxs modules, but the fxo modules use wctdm. Seems to me there
was a change some time ago where the fxs modules are now supported
from with wctdm. I'm not 100% sure though.


In 1.0.x it's called wcfxs in post 1.0.x it is called wctdm.  It's 
pretty easy to see what card goes with which driver.  The info is in the 
README in the zaptel source directory.


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Re: [Asterisk-Users] Looking for Provider

www.teliax.com has treated me very well for about six months, and have
lots of choices for DID (and 800) numbers.

www.nufone.com handles calls very nicely and provides 800 numbers. But,
their web site leaves a little to be desired, and they only use paypal.



 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1
 
 SIP and/or IAX would be nice. Residential.
 
 
 Mark Phillips wrote:
 
  I like Broadvoice but there are others.
 
  Do you want SIP or IAX termination? Business or residential?
 
  Mark
 
  Joshua Abbott wrote:
 
 
  Hello
  I currently have internet service through MediaCom (Cable Internet)
  and need to find a VOIP provider that is compatible with Asterisk and
  Cable Internet.
  Any ideas?
 
  I'm in Missouri about 1.5 hours west of St Louis, MO in a town called
  Hermann (65041 zip code)
 
 
  Joshua
 
 
 
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---End of Original Message-


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[Asterisk-Users] Vtech/Vonage Wireless Phone system

2005-08-21 Thread Andre Normandin

Hi,

I just stopped in at Best Buy here in CT, USA. I found an interesting
offering from Vtech there. It states it's a VOIP wireless phone system made
for the Vonage service.

Here it is at their website:
http://www.vtechphones.com/vtechui/guide/dsp_voip_guide.cfm

For $149, it comes with 2 (supposidly) VOIP 802.11b wireless phones, and a
'broadband/Wireless' router.

Does anyone know anything about these? Are they SIP phones? If so, I wonder
if it's possible to modify them for use with Asterisk?  Do they need their
'base' to talk, or is that nothing more than an 802.11b wireless router?

I'm thinking of picking up a kit just to see if I can get them to work
nativly with Asterisk, but if anyone has any experience with them before I
do, I'd appreciate it..

 - Andre

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Re: [Asterisk-Users] ZT_CHANCONFIG failed on channel 1 - WAS WORKING!!

 Hi Rich:
 
 Does the new distro tree issue explain the unresolved symbol references  
 noted in my last post?  I don't know if SuSE's 9.3 network autoupdate  
 would have changed the tree structure.  The unresolved symbols are  (from  
 dmesg):
 
 zt_receive
 zt_qevent_lock
 zt_ec_chunk
 zt_transmit
 zt_unregister
 zt_hooksig
 zt_register
 
 With: wcfxs: disagrees about version of symbol...

I don't use SuSE, so not 100% sure. Those sysbols are all related to
the zaptel drivers, so I'd have to guess that something is messed up
in that (as opposed to the kernel stuff).

I don't recall if you mentioned using Stable or Head, but right now
the cvs head appears to be a much better choice then Stable as a lot
of effort has focused on getting Head cleaned up for a 1.2 release.
I'm using cvs head with no problems right now; you might try that.


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[Asterisk-Users] Problem with auto-attendant config, I think..

2005-08-21 Thread Howard Leadmon

  I never heard anything on the AMP list, so figured maybe someone here might
be able to help me sort this one out..

  I was making some updates to my attendant config, which is really very
basic, and now incoming call processing stopped.  Not sure exactly what the
heck happened, but figured maybe someone could help me with a clue as to what
broke.   Now incoming calls are not being answered at all, no matter what I
do, which is quite annoying to say the least.

 Here is some log output from the server:

Aug 20 11:16:46 VERBOSE[1617]: -- Starting simple switch on 'Zap/4-1'
Aug 20 11:16:47 DEBUG[1617]: Expression is '1'
Aug 20 11:16:47 VERBOSE[1617]: -- Executing GotoIf(Zap/4-1,
1?from-pstn-reghours|s|1:) in new stack
Aug 20 11:16:47 VERBOSE[1617]: -- Goto (from-pstn-reghours,s,1)
Aug 20 11:16:47 DEBUG[1617]: Expression is '0'
Aug 20 11:16:47 VERBOSE[1617]: -- Executing GotoIf(Zap/4-1,
0?from-pstn-reghours-nofax|s|1:2) in new stack
Aug 20 11:16:47 VERBOSE[1617]: -- Goto (from-pstn-reghours,s,2)
Aug 20 11:16:47 VERBOSE[1617]: -- Executing Answer(Zap/4-1, ) in new
stack
Aug 20 11:16:47 DEBUG[1617]: Took Zap/4-1 off hook
Aug 20 11:16:47 DEBUG[1617]: Enabled echo cancellation on channel 4
Aug 20 11:16:47 DEBUG[1617]: Engaged echo training on channel 4
Aug 20 11:16:47 VERBOSE[1617]: -- Executing Wait(Zap/4-1, 1) in new
stack
Aug 20 11:16:48 VERBOSE[1617]: -- Executing SetVar(Zap/4-1,
intype=aa_2) in new stack
Aug 20 11:16:48 VERBOSE[1617]: -- Executing Cut(Zap/4-1,
intype=intype|-|1) in new stack
Aug 20 11:16:48 DEBUG[1617]: Expression is '0'
Aug 20 11:16:48 VERBOSE[1617]: -- Executing GotoIf(Zap/4-1, 0?7:9) in
new stack
Aug 20 11:16:48 VERBOSE[1617]: -- Goto (from-pstn-reghours,s,9)
Aug 20 11:16:48 DEBUG[1617]: Expression is '0'
Aug 20 11:16:48 VERBOSE[1617]: -- Executing GotoIf(Zap/4-1, 0?10:12)
in new stack
Aug 20 11:16:48 VERBOSE[1617]: -- Goto (from-pstn-reghours,s,12)
Aug 20 11:16:48 DEBUG[1617]: Expression is '0'
Aug 20 11:16:48 VERBOSE[1617]: -- Executing GotoIf(Zap/4-1, 0?13:15)
in new stack
Aug 20 11:16:48 VERBOSE[1617]: -- Goto (from-pstn-reghours,s,15)
Aug 20 11:16:48 VERBOSE[1617]: -- Executing Goto(Zap/4-1, aa_2|s|1) in
new stack
Aug 20 11:16:48 VERBOSE[1617]: -- Goto (aa_2,s,1)
Aug 20 11:16:48 WARNING[1617]: ast_yyerror(): syntax error: syntax error;
Input:
 = ANSWER
^^
   ^
Aug 20 11:16:48 DEBUG[1617]: Expression is '0'
Aug 20 11:16:48 VERBOSE[1617]: -- Executing GotoIf(Zap/4-1, 0?4) in
new stack
Aug 20 11:16:48 DEBUG[1617]: Not taking any branch
Aug 20 11:16:48 VERBOSE[1617]: -- Executing Answer(Zap/4-1, ) in new
stack
Aug 20 11:16:48 VERBOSE[1617]: -- Executing Wait(Zap/4-1, 1) in new
stack
Aug 20 11:16:49 VERBOSE[1617]: -- Executing SetVar(Zap/4-1,
DIR-CONTEXT=general) in new stack
Aug 20 11:16:49 VERBOSE[1617]: -- Executing DigitTimeout(Zap/4-1, 3)
in new stack
Aug 20 11:16:49 VERBOSE[1617]: -- Set Digit Timeout to 3
Aug 20 11:16:49 VERBOSE[1617]: -- Executing ResponseTimeout(Zap/4-1,
7) in new stack
Aug 20 11:16:49 VERBOSE[1617]: -- Set Response Timeout to 7
Aug 20 11:16:49 VERBOSE[1617]: -- Executing BackGround(Zap/4-1,
custom/aa_2) in new stack
Aug 20 11:16:49 DEBUG[1617]: Scheduling timer at 160 sample intervals
Aug 20 11:16:49 VERBOSE[1617]: -- Playing 'custom/aa_2' (language 'en')




Sure looks like it is trying to answer, but I keep hearing ringing on the
incoming side.   If I look at my config for aa_2 I see:

[aa_2]
include = aa_2-custom
exten = fax,1,Goto(ext-fax,in_fax,1)   ; 
exten = h,1,Hangup()   ; 
exten = i,1,Playback(invalid)  ; 
exten = i,2,Goto(s,7)  ; 
include = ext-local
include = app-messagecenter
include = app-directory
exten = s,1,GotoIf($[${DIALSTATUS} = ANSWER]?4); 
exten = s,2,Answer()   ; 
exten = s,3,Wait(1); 
exten = s,4,SetVar(DIR-CONTEXT=general); 
exten = s,5,DigitTimeout(3); 
exten = s,6,ResponseTimeout(7) ; 
exten = s,7,Background(custom/aa_2); 



Heck and I thought I had this thing running well, and then it all breaks.  If
anyone can point me in the right direction it would be most appreciated..






---
Howard Leadmon
http://www.leadmon.net



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Re: [Asterisk-Users] Vtech/Vonage Wireless Phone system

 I just stopped in at Best Buy here in CT, USA. I found an interesting
 offering from Vtech there. It states it's a VOIP wireless phone system made
 for the Vonage service.
 
 Here it is at their website:
 http://www.vtechphones.com/vtechui/guide/dsp_voip_guide.cfm
 
 For $149, it comes with 2 (supposidly) VOIP 802.11b wireless phones, and a
 'broadband/Wireless' router.
 
 Does anyone know anything about these? Are they SIP phones? If so, I wonder
 if it's possible to modify them for use with Asterisk?  Do they need their
 'base' to talk, or is that nothing more than an 802.11b wireless router?
 
 I'm thinking of picking up a kit just to see if I can get them to work
 nativly with Asterisk, but if anyone has any experience with them before I
 do, I'd appreciate it..

I seen their ad in the paper this morning. It appeared the phones were
the standard cordless (not 802.11b), and the base unit had pc wireless
(802.11b) integrated with the linksys/sipura broadband stuff. You might
want to check that out a little closer; but, fairly good return policy
if the ads are a little misleading. ;)


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Re: [Asterisk-Users] Ring more than two isdn phones simultaneously

2005-08-21 Thread Arik Funke

If this is a limitation of asterisk, where is it located? In the 
chan_zap module? In zaphfc? I.e. would it help if I switched to mISDN?


Best regards,
Arik


Peter Svensson [EMAIL PROTECTED] wrote:

On Sat, 20 Aug 2005, Nico Giefing wrote:


 how many connection do you have from your asterisk to the old pbx?

 i think on 1 ISDN connection its only possible to let 2 phones ring, 
because

 1 ISDN 2 channels...


This is a limitation in Asterisk, not ISDN. Asterisk reserves a B-channel
for each destination at the time of the CONNECT message. In the isdn world
it is common to not actually allocate a B-channel until it is needed to
carry audio. This also prevents Asterisk from letting the upstream switch
select the B-channel on outgoing calls to the pstn.

Asterisk is written this way since it uses the audio channel as the
fundamental unit, with the D-channel as carrier of signalling for the
individual B-channels. Another way to view ISDN is to consider the
D-channel the fundamental unit, which can carry several audio streams as a
side effect of the signalling. The first viewpoint resembles the
traditional view of telephony as individual circuits, the second resembles
the ISDN/SS7 view of the world.

Changing Asterisk to be more ISDN-like is quite a lot of work.

Peter



 - Original Message -
 From: Arik Funke [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Sent: Saturday, August 20, 2005 7:44 PM
 Subject: [Asterisk-Users] Ring more than two isdn phones simultaneously



  I am using a HFC-S card in nt mode with zaphfc driver to connect an
  internal isdn bus. I would like to signal an incoming call on, let's
  say, 4 phones. Right now I use:
 
  Dial(Zap/g1/21Zap/g1/22Zap/g1/24Zap/g1/23Zap/g1/29,,t)
 
  where g1 are my two isdn channels provided by HFC-S card an the
  21,22,etc my internal numbers.
 
  When the command is executed however, only the first two specified
  phones ring. Etc. with the first channel 21 ist called, with the 
second

  22. How can I get asterisk to signal to all phones with just one isdn
  channel? I am trying to duplicate the setup I had with my old 
isdn pbx
  with did above trick just fine... Maybe somebody can help me 
configure

  asterisk appropriately?
 
  Cheers,
  Arik
 
 
  PS: I gave following a try but without success:
  Dial(Zap/g1/21-29,,t)
  Dial(Zap/g1/21+29,,t)
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-- Peter Svensson ! Pgp key available by finger, fingerprint: 
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Remember, Luke, your source will be with you... always...

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[Asterisk-Users] Re: Asterisk 1.0.9 on SuSE 9.2 with ISDN BRI zaphfc?

2005-08-21 Thread Lars Dybdahl

I would like to know how to install asterisk 1.0.9 with zaphfc working
on a SuSE 9.2. I tried this:

- The rpms with SuSE 9.2 are asterisk 1.0.6
- bristuff works, except for zaphfc, which doesn't compile.
- The official asterisk download file doesn't contain isdn bri support

Any ideas?

Lars Dybdahl.
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RE: [Asterisk-Users] ViaTalk Down?

2005-08-21 Thread Sherwood McGowan

Is this a CVS-HEAD that was released AFTER 8/13? I ask because we're using
that release and it's still deadlocking.

Thanks,
Sherwood McGowan 

--Original Message-
-From: [EMAIL PROTECTED] 
-[mailto:[EMAIL PROTECTED] On Behalf Of 
-Kevin P. Fleming
-Sent: Saturday, August 20, 2005 3:25 PM
-To: Asterisk Users Mailing List - Non-Commercial Discussion
-Subject: Re: [Asterisk-Users] ViaTalk Down?
-
-Sherwood McGowan wrote:
-
- Anyone have a quickie answer as to why asterisk would suddenly just 
- stop responding? I was able to issue the restart command but I 
- couldn't do sip show peer num and couldn't show channels, etc 
- This is very disconcerting
-
-Your SIP channel driver was deadlocked. This can happen for a 
-number of reasons, but all of them are bad, and need to be fixed.
-
-Depending on the Asterisk version you are running, there may 
-be some known situations under which they can occur; as best 
-we can tell, there are none left in CVS HEAD related to chan_sip.
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Re: [Asterisk-Users] Not-Registered Problem

2005-08-21 Thread karl

Yes, and no. It would help to know what phones you're working with, what
you've got in sip.conf, and what you've got in extensions.conf.

-K


 Hello,
 What are some common reasons why a phone would report not registered
 even when the extension has been setup through Asterisk(*) AND phone
 username/password is correct?

 Joshua

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[Asterisk-Users] Broadvoice Issue

2005-08-21 Thread Tressler, Joshua Adam









I did a quick google search of the lists site and couldnt
find a definitive answer, so if its there, I apologize for asking again.



Starting about noon yesterday, I am no longer able to
send/receive calls via Broadvoice. When calling in, I get a fast busy, and when
calling out I get the following error:



 -- Executing Dial(SIP/112-572a, [EMAIL PROTECTED])
in new stack

 -- Called [EMAIL PROTECTED]

Aug 21 13:34:47 NOTICE[20742]: chan_sip.c:8648
handle_response: Failed to authenticate on INVITE to 'Mobile sip:[EMAIL PROTECTED];tag=as124e3440'

 == Spawn extension (agents, 78126631234, 1) exited
non-zero on 'SIP/112-572a'



I have the following in sip.conf:





register = [EMAIL PROTECTED]:password:[EMAIL PROTECTED]/



[sip.broadvoice.com]

type=peer

user=phone

host=sip.broadvoice.com

fromdomain=sip.broadvoice.com

fromuser=XX

secret=password

insecure=very

context=incoming

authname=XX

dtmfmode=inband

dtmf=inband

canreinvite=no





Does anyone know what Im missing here? Everything was
working fine yesterday morning.





JT






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RE: [Asterisk-Users] Vtech/Vonage Wireless Phone system

2005-08-21 Thread Andre Normandin

Thanks,

I just read all the literature on the Vtech website, and, I think you are
exactly correct!

Oh well, so much for cheap SIP wireless phones :-)

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rich
Adamson
Sent: Sunday, August 21, 2005 2:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Vtech/Vonage Wireless Phone system


 I just stopped in at Best Buy here in CT, USA. I found an interesting
 offering from Vtech there. It states it's a VOIP wireless phone system
made
 for the Vonage service.

 Here it is at their website:
 http://www.vtechphones.com/vtechui/guide/dsp_voip_guide.cfm

 For $149, it comes with 2 (supposidly) VOIP 802.11b wireless phones, and a
 'broadband/Wireless' router.

 Does anyone know anything about these? Are they SIP phones? If so, I
wonder
 if it's possible to modify them for use with Asterisk?  Do they need their
 'base' to talk, or is that nothing more than an 802.11b wireless router?

 I'm thinking of picking up a kit just to see if I can get them to work
 nativly with Asterisk, but if anyone has any experience with them before I
 do, I'd appreciate it..

I seen their ad in the paper this morning. It appeared the phones were
the standard cordless (not 802.11b), and the base unit had pc wireless
(802.11b) integrated with the linksys/sipura broadband stuff. You might
want to check that out a little closer; but, fairly good return policy
if the ads are a little misleading. ;)


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Re: [Asterisk-Users] Ring more than two isdn phones simultaneously

2005-08-21 Thread Peter Svensson

On Sun, 21 Aug 2005, Arik Funke wrote:

 If this is a limitation of asterisk, where is it located? In the 
 chan_zap module? In zaphfc? I.e. would it help if I switched to mISDN?

It is inherent in the channel-based structure of Asterisk. An audio 
channel is the basic measure used by applications such as Dial etc. This 
is shared by all channels as far as I know. 

One can imagine a special version of chan_zap that decouples the Asterisk 
channel entities from the actual B-channels. It would always generate a 
new fictitious asterisk channel structure and only link it to a real 
B-channel once the signaling indicated that a B-channel was required. 

I would be interested in how the commercial SS7 implementation for Asterisk 
works. SS7 would normally allow the audio paths to change in mid-call to 
potentially follow an altogether different route.

Peter


 
 Peter Svensson [EMAIL PROTECTED] wrote:
 
 On Sat, 20 Aug 2005, Nico Giefing wrote:
 
   how many connection do you have from your asterisk to the old pbx?
  
   i think on 1 ISDN connection its only possible to let 2 phones ring, 
 because
   1 ISDN 2 channels...
 
 
 This is a limitation in Asterisk, not ISDN. Asterisk reserves a B-channel
 for each destination at the time of the CONNECT message. In the isdn world
 it is common to not actually allocate a B-channel until it is needed to
 carry audio. This also prevents Asterisk from letting the upstream switch
 select the B-channel on outgoing calls to the pstn.
 
 Asterisk is written this way since it uses the audio channel as the
 fundamental unit, with the D-channel as carrier of signalling for the
 individual B-channels. Another way to view ISDN is to consider the
 D-channel the fundamental unit, which can carry several audio streams as a
 side effect of the signalling. The first viewpoint resembles the
 traditional view of telephony as individual circuits, the second resembles
 the ISDN/SS7 view of the world.
 
 Changing Asterisk to be more ISDN-like is quite a lot of work.
 
 Peter
 
 
  
   - Original Message -
   From: Arik Funke [EMAIL PROTECTED]
   To: [EMAIL PROTECTED]
   Sent: Saturday, August 20, 2005 7:44 PM
   Subject: [Asterisk-Users] Ring more than two isdn phones simultaneously
  
  
  
I am using a HFC-S card in nt mode with zaphfc driver to connect an
internal isdn bus. I would like to signal an incoming call on, let's
say, 4 phones. Right now I use:
   
Dial(Zap/g1/21Zap/g1/22Zap/g1/24Zap/g1/23Zap/g1/29,,t)
   
where g1 are my two isdn channels provided by HFC-S card an the
21,22,etc my internal numbers.
   
When the command is executed however, only the first two specified
phones ring. Etc. with the first channel 21 ist called, with the 
 second
22. How can I get asterisk to signal to all phones with just one isdn
channel? I am trying to duplicate the setup I had with my old 
 isdn pbx
with did above trick just fine... Maybe somebody can help me 
 configure
asterisk appropriately?
   
Cheers,
Arik
   
   
PS: I gave following a try but without success:
Dial(Zap/g1/21-29,,t)
Dial(Zap/g1/21+29,,t)
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[Asterisk-Users] On Network Usage from the CDR

2005-08-21 Thread Michael Blood

Title: Message



We want touse 
the accountcode in the CDR for billing and tracking total usage. 

We wanted to set the 
accountcode for calls coming into our network so we know which of our users to 
assign the usage to.
But then when we 
receive an "on network" call we run into a problem with which accountcode to use 
for the call (our caller or our callee).

It seems that it would be best to have two separate CDR 
entries for each "on network" call. I can imagine some ways of routing 
calls out and back into the box without using a valuable Zap 
channel.
Or we could use the 
dst channel to track things but I am curious how have others dealt with 
this?

Thanks 


Michael
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Re: [Asterisk-Users] Ring more than two isdn phones simultaneously

2005-08-21 Thread Arik Funke

In your opinion, is there any chance that this situation will change in 
forseable future? Is anybody already working on such a pseudo-channel 
structure required for decoupling asterisk channels from the physical 
channels? If not, is there a significant interest in the asterisk 
community to do this?


Arik


Peter Svensson wrote:

On Sun, 21 Aug 2005, Arik Funke wrote:


If this is a limitation of asterisk, where is it located? In the 
chan_zap module? In zaphfc? I.e. would it help if I switched to mISDN?



It is inherent in the channel-based structure of Asterisk. An audio 
channel is the basic measure used by applications such as Dial etc. This 
is shared by all channels as far as I know. 

One can imagine a special version of chan_zap that decouples the Asterisk 
channel entities from the actual B-channels. It would always generate a 
new fictitious asterisk channel structure and only link it to a real 
B-channel once the signaling indicated that a B-channel was required. 

I would be interested in how the commercial SS7 implementation for Asterisk 
works. SS7 would normally allow the audio paths to change in mid-call to 
potentially follow an altogether different route.


Peter




Peter Svensson [EMAIL PROTECTED] wrote:

On Sat, 20 Aug 2005, Nico Giefing wrote:

 how many connection do you have from your asterisk to the old pbx?

 i think on 1 ISDN connection its only possible to let 2 phones ring, 
because

 1 ISDN 2 channels...


This is a limitation in Asterisk, not ISDN. Asterisk reserves a B-channel
for each destination at the time of the CONNECT message. In the isdn world
it is common to not actually allocate a B-channel until it is needed to
carry audio. This also prevents Asterisk from letting the upstream switch
select the B-channel on outgoing calls to the pstn.

Asterisk is written this way since it uses the audio channel as the
fundamental unit, with the D-channel as carrier of signalling for the
individual B-channels. Another way to view ISDN is to consider the
D-channel the fundamental unit, which can carry several audio streams as a
side effect of the signalling. The first viewpoint resembles the
traditional view of telephony as individual circuits, the second resembles
the ISDN/SS7 view of the world.

Changing Asterisk to be more ISDN-like is quite a lot of work.

Peter



 - Original Message -
 From: Arik Funke [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Sent: Saturday, August 20, 2005 7:44 PM
 Subject: [Asterisk-Users] Ring more than two isdn phones simultaneously



  I am using a HFC-S card in nt mode with zaphfc driver to connect an
  internal isdn bus. I would like to signal an incoming call on, let's
  say, 4 phones. Right now I use:
 
  Dial(Zap/g1/21Zap/g1/22Zap/g1/24Zap/g1/23Zap/g1/29,,t)
 
  where g1 are my two isdn channels provided by HFC-S card an the
  21,22,etc my internal numbers.
 
  When the command is executed however, only the first two specified
  phones ring. Etc. with the first channel 21 ist called, with the 
second

  22. How can I get asterisk to signal to all phones with just one isdn
  channel? I am trying to duplicate the setup I had with my old 
isdn pbx
  with did above trick just fine... Maybe somebody can help me 
configure

  asterisk appropriately?
 
  Cheers,
  Arik
 
 
  PS: I gave following a try but without success:
  Dial(Zap/g1/21-29,,t)
  Dial(Zap/g1/21+29,,t)
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Peter





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[Asterisk-Users] PrivacyManager not working Asterisk 1.0.9-BRIstuffed-0.2.0-RC8n

2005-08-21 Thread Remco Barende


Hi list!

I'm trying to get PrivacyManager working but for some reason it always 
thinks that CallerID is present (when it isn't). I get this on the 
console:


  == Primary D-Channel on span 1 up
-- Accepting voice call from '' to '0711234567' on channel 0/2, span 1
-- Executing Ringing(Zap/2-1, ) in new stack
-- Executing Zapateller(Zap/2-1, answer|nocallerid) in new stack
-- Executing PrivacyManager(Zap/2-1, ) in new stack
-- CallerID Present: Skipping
-- Executing SetCIDNum(Zap/2-1, ) in new stack
-- Executing LookupCIDName(Zap/2-1, ) in new stack


I have this in my extensions.conf:
exten = 0711234567,1,Ringing
exten = 0711234567,2,Zapateller(answer|nocallerid)
exten = 0711234567,3,PrivacyManager
exten = 0711234567,4,SetCIDNum(${PRI_NETWORK_CID})
exten = 0711234567,5,LookupCIDName
exten = 0711234567,6,Dial(${EVERYONE},45,t)
exten = 0711234567,7,Answer

I need SetCIDNum(${PRI_NETWORK_CID}) to get CallerID working but I guess 
it needn't be before Zapateller or PrivacyManager to trigger the 
PrivacyManager.


Other question, would it be possible to use my own (custom recorded) 
message instead of a standard prompt?


Thanks!!

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Re: [Asterisk-Users] ZT_CHANCONFIG failed on channel 1 - WAS WORKING!!

2005-08-21 Thread Scott Brown


Hi Everyone:

Problem Solved.  Thanks to Matt, Paul and Rich for their excellent help!   
It is always appreciated.


Here's the solution, for thsoe interested:

SuSE distributes zaptel drivers and the auto update referenced puts  
drivers that may not be compatible with the current release on CVS (head  
or stable) or whateve version you may have compiled and installed apart  
from the disto.


Thus the incompatible symbols in loadable modules with no compile errors.   
One set from the disto in the extra directory and the compiled set in  
the misc directory under /lib/modules/{kernelname-version}/


It seems to me that this could conceivably cause Zaptel drivers to be  
flaky as reported by Paul, when they aren't sufficiently out of sync  
with symbol version error to fail in loading.  For the complete story on  
SuSE-Zaptel issues, please see:


http://www.voip-info.org/wiki-Asterisk+Linux+SuSE

Scott

On Sun, 21 Aug 2005 08:51:19 -0600, Scott Brown [EMAIL PROTECTED]  
wrote:



Hi Paul:

Thanks very much for the suggestion.  I don't understand why this just  
began to happnen.  I never
had problems before.  Your suggestion has shed a lot of light on the  
problem.  Because wcfxs disagrees about the version of the symbols  
listed below, I get the following unknown symbol problems for:


zt_receive
zt_qevent_lock
zt_ec_chunk
zt_transmit
zt_unregister
zt_hooksig
zt_register

With: wcfxs: disagrees about version of symbol...

BTW, I also did a full checkout (v1-0) and rebuild (deleting modules and  
includes beforehand) of

astrisk zaptel and libpri.  That didn't change anything.

Below is the resultant output from the -v modprobe and dmesg output.  If  
you or anyone else can help with this it would really be appreciated,  
thanks:


# modprobe -v -n wctdm
insmod /lib/modules/2.6.11.4-21.8-default/kernel/lib/crc-ccitt.ko
insmod /lib/modules/2.6.11.4-21.8-default/misc/zaptel.ko
install /sbin/modprobe --ignore-install wcfxs  /sbin/ztcfg
# modprobe -v wctdm
insmod /lib/modules/2.6.11.4-21.8-default/kernel/lib/crc-ccitt.ko
insmod /lib/modules/2.6.11.4-21.8-default/misc/zaptel.ko
install /sbin/modprobe --ignore-install wcfxs  /sbin/ztcfg
insmod /lib/modules/2.6.11.4-21.8-default/extra/wcfxs.ko
FATAL: Error inserting wcfxs  
(/lib/modules/2.6.11.4-21.8-default/extra/wcfxs.ko): Unknown symbol

in module, or unknown parameter (see dmesg)
FATAL: Error running install command for wcfxs

pertinent dmesg output:
zaptel: module not supported by Novell, setting U taint flag.
Zapata Telephony Interface Registered on major 196
wcfxs: module not supported by Novell, setting U taint flag.
Zapata Telephony Interface Unloaded
zaptel: module not supported by Novell, setting U taint flag.
Zapata Telephony Interface Registered on major 196
wcfxs: module not supported by Novell, setting U taint flag.
wcfxs: disagrees about version of symbol zt_receive
wcfxs: Unknown symbol zt_receive, st_info == 0x1
wcfxs: disagrees about version of symbol zt_qevent_lock
wcfxs: Unknown symbol zt_qevent_lock, st_info == 0x1
wcfxs: disagrees about version of symbol zt_ec_chunk
wcfxs: Unknown symbol zt_ec_chunk, st_info == 0x1
wcfxs: disagrees about version of symbol zt_transmit
wcfxs: Unknown symbol zt_transmit, st_info == 0x1
wcfxs: disagrees about version of symbol zt_unregister
wcfxs: Unknown symbol zt_unregister, st_info == 0x1
wcfxs: disagrees about version of symbol zt_hooksig
wcfxs: Unknown symbol zt_hooksig, st_info == 0x1
wcfxs: disagrees about version of symbol zt_register
wcfxs: Unknown symbol zt_register, st_info == 0x1
load_module: err 0xfffe (dont worry)
wcfxs: module not supported by Novell, setting U taint flag.
wcfxs: disagrees about version of symbol zt_receive
wcfxs: Unknown symbol zt_receive, st_info == 0x1
wcfxs: disagrees about version of symbol zt_qevent_lock
wcfxs: Unknown symbol zt_qevent_lock, st_info == 0x1
wcfxs: disagrees about version of symbol zt_ec_chunk
wcfxs: Unknown symbol zt_ec_chunk, st_info == 0x1
wcfxs: disagrees about version of symbol zt_transmit
wcfxs: Unknown symbol zt_transmit, st_info == 0x1
wcfxs: disagrees about version of symbol zt_unregister
wcfxs: Unknown symbol zt_unregister, st_info == 0x1
wcfxs: disagrees about version of symbol zt_hooksig
wcfxs: Unknown symbol zt_hooksig, st_info == 0x1
wcfxs: disagrees about version of symbol zt_register
wcfxs: Unknown symbol zt_register, st_info == 0x1
load_module: err 0xfffe (dont worry)

Thanks for the help.

Scott
  On Sat, 20 Aug 2005 03:58:18 -0600, Paul Hewlett  
[EMAIL PROTECTED] wrote:



On Saturday 20 August 2005 09:58, Scott Brown wrote:

Hi Matt:

That suggestion is possibly on the right track.  It made me remember  
that -

although I'm not using Fedora, but SuSE 9.3, that it went through an
automatic network update just recently.  After that, I tried updating  
the

Zaptel files from CVS and recompiling everything, but to no avail.  The
same error still occured.  I eliminated hardware by swapping out a  
working

RE: [Asterisk-Users] any ISDN/PRI signaling experts out there?

2005-08-21 Thread Damon Estep

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Paul Belanger
 Sent: Friday, August 19, 2005 4:26 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] any ISDN/PRI signaling experts out
there?

 See comments inline!

 Damon Estep wrote:
  I have officially engaged in a pissing contest with the local Telco
over
  PRI calling name delivery.

 Welcome to my world, I deal with theses guys daily!  Errgiant arn't
 they.  We have a saying around work 'The telco is always wrong!'.

  The telco publishes their calling name delivery over PRI feature as
  being bellcore gr-1367-core compliant.

  The gr-1367-core spec states that the calling name is to be included
as
  a facility IE in the setup message, or sent in a subsequent facility
IE
  message with an indicator in the setup message that the CNAM will
  follow.

  Extensive testing and ISDN/PRI protocol analysis shows that the
facility
  IE they are sending out with the CNAM in it comes only after we have
  sent back PROGRESS and ALERTING in response to the SETUP. If we
block
  the PROGRESS and ALERTING and sit and WAIT for the FACILITY we never
get
  it, the call will time out, so we know they are actually waiting for
the
  call to progress before sending the facility IE CNAM.

 This sounds a little fishy, Orgination Number is usually transmitted
in
 the SETUP message.  Your are almost correct in your messaging:

 Network  User(Switch)
 Setup
  CALL PROCEEDING
  ALERTING
  CALL CONNECT
 CALL CONNECT ACKNOWLEDGE

 There is about a 4sec timeout allow after SETUP is initially sent, if
 CALL PROCEEDING is not transmitted by that time, the Network side will
 terminiate the call.

  As far as I can tell the GR-1367-CORE spec does not define a maximum
  delay in sending the facility IE or whether it is acceptable to wait
for
  PROGRESS and ALERT before sending it.

  The setup is; Telco PRI Lucent 5ESS  Lucent MAX TNT  Asterisk

 Here is an ISDN trace from a Dialogic board attached to a 5ESS switch
 with framing/coding ESF/B8ZS:

 SETUP(0x05)
   1:   BEARER CAPABILITY(0x04)
   2:   IE Length(0x03)
   3:  1--- Extension Bit
   -00- Coding Standard
   ---0 Info. Transfer Cap.
   4:  1--- Extension Bit
   -00- Transfer Mode
   ---1 Info. Transfer Rate
   5:  1--- Extension Bit
   -01- Layer 1 Indent
   ---00010 User Info. Layer 1
   1:   CHANNEL ID(0x18)
   2:   IE Length(0x03)
   3:  1--- Extension Bit
   -0-- Interface ID Present
   --1- Interface Type
   ---0 Spare
   1--- Preferred/Exclusive
   -0-- D-Channel Indicator
   --01 Info. Channel Sel.
 3.2:  1--- Extension Bit
   -00- Coding Standard
   ---0 Number Map
   0011 Channel/Map Element
   4:  1--- Extension Bit
   -001 Channel Number/Slot Map
   1:   CALLING PARTY NUM(0x6c)
   2:   IE Length(0x0b)
   3:  1--- Extension Bit
   -010 Type Of Number
   0001 Numbering Plan ID
   949459  Number Digit(s)-- Here is the ANI
   1:   CALLED PARTY NUM(0x70)
   2:   IE Length(0x04)
   3:  1--- Extension Bit
   -100 Type of Number
   0001 Numbering plan ID
   200  Number Digit(s)   -- Here is the DNIS

 Notice my comments on where ANI and DNIS arrive in the SETUP message.

  The MAX TNT responds to the Facility IE with ISDN error 98, invalid
  message for call state.

 This is an actual CAUSE CODE from Q.931:

 Cause No. 98 - Message not compatible

 This cause indicates that the message received is not compatible with
 the call state or the message type is non-existent or not implemented.

 In short it is a protocol error.  Check out
 http://www.telos-systems.com/?/techtalk/cause.htm for a complete lists
 of causes and there meaning.

  The SIP INVITE from the TNT to Asterisk contains no Caller Name
  information.

  It seems really odd to me that a Lucent TNT can not translate the
caller
  ID Name info delivered by a Lucent 5ESS switch.

  On the same setup, if I connect another PRI device to it that
emulates
  switch side signaling and includes the CNAM as a Display IE in the
  setup, the SIP invite is properly formatted and * receives the
calling
  party name.

  Does anyone here have enough experience with ISDN PRI signaling to
  comment with some level of authority on this?

 Can you set a ISDN trace from your telco to your switch?  I would be
 curious to see what it looks like.

 Again, it looks like your telco's problem.  Your best to ask them to
 through a ThunderBird (T-Bird) on your circuit at your demarc and ask
 them if they see the CallerID, chances are they don't

  Damon

Peter,

Keep in mind it is CALLER ID NAME

RE: [Asterisk-Users] Vtech/Vonage Wireless Phone system

All these things are is just an ATA built into a plain vanilla analog
cordless phone system...ok, a 5.8 Ghz systemvanilla with sprinkles
;-)

Nothing special at all. You'd be better off with a Sipura device and a
Panasonic cordless phone.

Michael

On Sun, 21 Aug 2005 15:12:49 -0400, Andre Normandin wrote:

Thanks,

I just read all the literature on the Vtech website, and, I think you are
exactly correct!

Oh well, so much for cheap SIP wireless phones :-)

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rich
Adamson
Sent: Sunday, August 21, 2005 2:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Vtech/Vonage Wireless Phone system


 I just stopped in at Best Buy here in CT, USA. I found an interesting
 offering from Vtech there. It states it's a VOIP wireless phone system
made
 for the Vonage service.

 Here it is at their website:
 http://www.vtechphones.com/vtechui/guide/dsp_voip_guide.cfm

 For $149, it comes with 2 (supposidly) VOIP 802.11b wireless phones, and a
 'broadband/Wireless' router.

 Does anyone know anything about these? Are they SIP phones? If so, I
wonder
 if it's possible to modify them for use with Asterisk?  Do they need their
 'base' to talk, or is that nothing more than an 802.11b wireless router?

 I'm thinking of picking up a kit just to see if I can get them to work
 nativly with Asterisk, but if anyone has any experience with them before I
 do, I'd appreciate it..

I seen their ad in the paper this morning. It appeared the phones were
the standard cordless (not 802.11b), and the base unit had pc wireless
(802.11b) integrated with the linksys/sipura broadband stuff. You might
want to check that out a little closer; but, fairly good return policy
if the ads are a little misleading. ;)


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[Asterisk-Users] Nortel Meridian-1 Line Side E1

2005-08-21 Thread Leo Ann Boon

Anyone has the settings to connect a TE405 to Meridian-1 line side E1? I 
saw T1 on the voip-info.org, but no E1. Is Nortel's E1 a variant of MFC/R2?

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Re: [Asterisk-Users] Broadvoice Issue

There seems to be a random thread of BV issues this last week all 
amounting to the same proble - no calls.


Do a sip debug peer sip.broadvoice.com and see what happens. I found 
that BV were sending calls to my number and for some odd reason my * 
server wasn't dumping them into the exten=s,1,blah logic that it 
previously had been. The only way I could fix it was to do 
exten=phonenumber,1,blah and that works fine now.


As for outgoing, are you sure you are registered?

In the shoirt term, log in to your bv account and set up your VM. At 
least you won;t lose calls that way.


Mark

Tressler, Joshua Adam wrote:
I did a quick google search of the lists site and couldn’t find a 
definitive answer, so if it’s there, I apologize for asking again.


 

Starting about noon yesterday, I am no longer able to send/receive calls 
via Broadvoice. When calling in, I get a fast busy, and when calling out 
I get the following error:


 

-- Executing Dial(SIP/112-572a, 
[EMAIL PROTECTED]) in new stack


-- Called [EMAIL PROTECTED]

Aug 21 13:34:47 NOTICE[20742]: chan_sip.c:8648 handle_response: Failed 
to authenticate on INVITE to 'Mobile 
sip:[EMAIL PROTECTED];tag=as124e3440'


  == Spawn extension (agents, 78126631234, 1) exited non-zero on 
'SIP/112-572a'


 


I have the following in sip.conf:

 

 

register = 
[EMAIL PROTECTED]:password:[EMAIL PROTECTED]/


 


[sip.broadvoice.com]

type=peer

user=phone

host=sip.broadvoice.com

fromdomain=sip.broadvoice.com

fromuser=XX

secret=password

insecure=very

context=incoming

authname=XX

dtmfmode=inband

dtmf=inband

canreinvite=no

 

 

Does anyone know what I’m missing here? Everything was working fine 
yesterday morning.


 

 


JT




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[Asterisk-Users] Call duration limits not working

2005-08-21 Thread tim

Hello everybody.  Recently I've been trying to limit the duration of some
calls for a simple application I'm writing.  Unfortunately all of the
documented methods are failing and I'm not sure what else to try.  Here
are some samples of what I've done:

1) The AbsoluteTimeout application.
   - exten = 1,1,AbsoluteTimeout (30)

2) The new version of AbsoluteTimeout.
   - exten = 1,1,Set(TIMEOUT(absolute)=30)

3) The S option to the Dial application.
   - exten = 1,1,Dial(${PROVIDER}/${NUMBER}|30|S(30))

4) The L option to the Dial application.
   - exten = 1,1,Dial(${PROVIDER}/${NUMBER}|30|L(3))

5) A combination of all of the above.
   - exten = 1,1,Set(TIMEOUT(absolute)=30) OR exten =
1,1,AbsoluteTimeout (30)
   - exten = 1,2,Dial(${PROVIDER}/${NUMBER}|30|S(30)L(3))

Nothing seems to work.  When using S I get a message on the console that
says the call is going to be limited to 30 seconds but it just keeps
going.  For completeness I'm dialing an IAX provider and I'm using the
Asterisk CVS HEAD.  I also am sure that did not forget to reload (and
sometimes restart entirely) after I made changes to extensions.conf.  The
timeout |30 works when calls don't supervise but the time limit |S(30)
or |L(3) lets calls continue forever when calls do supervise.

What could I be doing wrong?  What should I try to remedy this situation?

Thank you,
Tim Mattison
[EMAIL PROTECTED]

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[Asterisk-Users] call waiting beep on PSTN and TDM400P FXO line hook flash

2005-08-21 Thread Jeff Otterson

 I have been looking for the answer to this question for a 
while.  Google-ing and reading the archives of Asterisk-Users has not 
enlightened me.


  It seems that this question has been asked many times, and many times it 
has gone unanswered.


 I have call waiting and three way calling on my PSTN line from Verizon 
(the local telco).  This is connected to a FXO port on a TDM400P.  I also 
have two FXS ports on the TDM400P.


  So my problem is, how do I flash the Verizon PSTN line when I hear the 
call waiting beep?  How can I send a hook flash to the Verizon trunk to 
activate their 3-way calling feature.


  I have seen some stuff like hook flash then send *0 to get the bridged 
Zap trunk to flash but I can't get it to work.  I get the reorder 
signal.  I need something my wife and kid can do.


  Can anybody help?

  Thanks,

  Jeff

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RE: [Asterisk-Users] Vtech/Vonage Wireless Phone system

2005-08-21 Thread Andre Normandin

I already have a Sipura and a panasonic cordless, I was just hoping that I
could find a true SIP wireless phone at a reasonable price  :-)

Oh well, one can drea laugh

Thanks,
  - Andre

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michael
Graves
Sent: Sunday, August 21, 2005 4:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Vtech/Vonage Wireless Phone system


All these things are is just an ATA built into a plain vanilla analog
cordless phone system...ok, a 5.8 Ghz systemvanilla with sprinkles
;-)

Nothing special at all. You'd be better off with a Sipura device and a
Panasonic cordless phone.

Michael

On Sun, 21 Aug 2005 15:12:49 -0400, Andre Normandin wrote:

Thanks,

I just read all the literature on the Vtech website, and, I think you are
exactly correct!

Oh well, so much for cheap SIP wireless phones :-)

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rich
Adamson
Sent: Sunday, August 21, 2005 2:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Vtech/Vonage Wireless Phone system


 I just stopped in at Best Buy here in CT, USA. I found an interesting
 offering from Vtech there. It states it's a VOIP wireless phone system
made
 for the Vonage service.

 Here it is at their website:
 http://www.vtechphones.com/vtechui/guide/dsp_voip_guide.cfm

 For $149, it comes with 2 (supposidly) VOIP 802.11b wireless phones, and
a
 'broadband/Wireless' router.

 Does anyone know anything about these? Are they SIP phones? If so, I
wonder
 if it's possible to modify them for use with Asterisk?  Do they need
their
 'base' to talk, or is that nothing more than an 802.11b wireless router?

 I'm thinking of picking up a kit just to see if I can get them to work
 nativly with Asterisk, but if anyone has any experience with them before
I
 do, I'd appreciate it..

I seen their ad in the paper this morning. It appeared the phones were
the standard cordless (not 802.11b), and the base unit had pc wireless
(802.11b) integrated with the linksys/sipura broadband stuff. You might
want to check that out a little closer; but, fairly good return policy
if the ads are a little misleading. ;)


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Re: [Asterisk-Users] Call duration limits not working

2005-08-21 Thread Eric Wieling aka ManxPower


[EMAIL PROTECTED] wrote:

Hello everybody.  Recently I've been trying to limit the duration of some
calls for a simple application I'm writing.  Unfortunately all of the
documented methods are failing and I'm not sure what else to try.  Here
are some samples of what I've done:


I believe this was fixed in CVS-HEAD a couple of weeks ago.  There was a 
big announcement on the mailing list about it.

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RE: [Asterisk-Users] Broadvoice Issue

2005-08-21 Thread Tressler, Joshua Adam

Mark,

Thanks for the tips. After adding the exten = XX,1,BLAH, i am able to 
received calls, however I still get the same error when dialing out, and now, 
there is an additional error on the end. I am beginning to think this is a 
Broadvoice issue and will try to contact them after sending this message. The 
new error is as follows:

===

-- Executing Dial(SIP/202-7ea7, SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
Aug 21 17:33:23 NOTICE[20742]: chan_sip.c:8648 handle_response: Failed to 
authenticate on INVITE to 'Cisco 02 sip:[EMAIL PROTECTED];tag=as6a8b6a73'
  == Spawn extension (agents, number, 1) exited non-zero on 'SIP/202-7ea7'
-- Got SIP response 481 Call Leg/Transaction Does Not Exist back from 
147.135.20.128

===

Can anyone tell from this message if their service is trying to reinvite? I 
have that set to no for the devices i'm using as well as for the 
[sip.broadvoice.com].

Doing a sip show registry shows me as registered, however, I still cannot make 
calls. Any other suggestions?

Thanks,

Josh

-Original Message-
From: [EMAIL PROTECTED] on behalf of Mark Phillips
Sent: Sun 8/21/2005 4:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Broadvoice Issue
 
There seems to be a random thread of BV issues this last week all 
amounting to the same proble - no calls.

Do a sip debug peer sip.broadvoice.com and see what happens. I found 
that BV were sending calls to my number and for some odd reason my * 
server wasn't dumping them into the exten=s,1,blah logic that it 
previously had been. The only way I could fix it was to do 
exten=phonenumber,1,blah and that works fine now.

As for outgoing, are you sure you are registered?

In the shoirt term, log in to your bv account and set up your VM. At 
least you won;t lose calls that way.

Mark

Tressler, Joshua Adam wrote:
 I did a quick google search of the lists site and couldn't find a 
 definitive answer, so if it's there, I apologize for asking again.
 
  
 
 Starting about noon yesterday, I am no longer able to send/receive calls 
 via Broadvoice. When calling in, I get a fast busy, and when calling out 
 I get the following error:
 
  
 
 -- Executing Dial(SIP/112-572a, 
 [EMAIL PROTECTED]) in new stack
 
 -- Called [EMAIL PROTECTED]
 
 Aug 21 13:34:47 NOTICE[20742]: chan_sip.c:8648 handle_response: Failed 
 to authenticate on INVITE to 'Mobile 
 sip:[EMAIL PROTECTED];tag=as124e3440'
 
   == Spawn extension (agents, 78126631234, 1) exited non-zero on 
 'SIP/112-572a'
 
  
 
 I have the following in sip.conf:
 
  
 
  
 
 register = 
 [EMAIL PROTECTED]:password:[EMAIL PROTECTED]/
 
  
 
 [sip.broadvoice.com]
 
 type=peer
 
 user=phone
 
 host=sip.broadvoice.com
 
 fromdomain=sip.broadvoice.com
 
 fromuser=XX
 
 secret=password
 
 insecure=very
 
 context=incoming
 
 authname=XX
 
 dtmfmode=inband
 
 dtmf=inband
 
 canreinvite=no
 
  
 
  
 
 Does anyone know what I'm missing here? Everything was working fine 
 yesterday morning.
 
  
 
  
 
 JT
 
 
 
 
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[Asterisk-Users] Voice Problem ...asterisk on lan not woriking well...!!

2005-08-21 Thread Ing. Marlo R. Beltran G

Hi

 

 i just implemented asterisk and is such a grate solution...i am using 

 polycom 301 and 501 phoneson lan a iam using g.711 and i have a 

 16 port linksys switch...

 

 the problem come when somebody inside the network is making  a call to 

 other extension (in the same network) and is sending an e 

 mail through  internet the quality goes down...it hears rally bad...

 

 i am on a 10/100 network with cat5e on wire, and switches...what can 

 i do to have an excelent voice quality inside my network???

 

 The e mail doesn't go trough the asterisk computer.

 

Marlo

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Re: [Asterisk-Users] CVS-HEAD Compile Problem

2005-08-21 Thread Laurent Foulonneau


Nico,

Same problem for me, did you find a way to compile latest CVS-HEAD ?

And by the way, you're right realtime do need HEAD version.

Thanks

Laurent

At 00:04 20/08/2005 +0200, Nico Giefing wrote:

but the non head version is not working with realtime configuration?

hm, i think its a problem with app_expr.c but i will try now to copy the
app_expr.c from cvs-version

i will let you know

Nico

- Original Message -
From: Trey Scarborough [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Friday, August 19, 2005 8:31 PM
Subject: Re: [Asterisk-Users] CVS-HEAD Compile Problem


 I ran into the same problem the other day and just went back to non head
 version It would be nice to figure out why it does this.

 - Original Message -
 From: Nico Giefing
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Sent: Friday, August 19, 2005 9:20 AM
 Subject: [Asterisk-Users] CVS-HEAD Compile Problem


 I have a little Problem,

 I will compile asterisk CVS-HEAD but after  20 second of compiling i get
the
 message as shown at http://pastebin.com/340654 about 1000 times.

 Do anybody know a solution for this?

 Thanks a lot

 Nico



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Re: [Asterisk-Users] Small office setup/using analog lines w/ Asterisk

2005-08-21 Thread jennyw

Thanks, everyone, for your suggestions. I'm going to stop by the office 
tomorrow to try some of these out.


Here's more info on the setup: We bought a brand new computer for this 
-- I don't have the specifics right now, but will look that up in the 
office tomorrow. We have two Digium cards -- a TDM04B and a TDM01B.  
This system supports two organizations sharing the same office space: 3 
lines go to one org., the other 2 go to the other.


The Digium cards actually are sharing IRQs with other devices -- the 
installer mentioned it could be an issue initially, but when he saw that 
the devices that the cards were sharing with were the network card and 
the video card, he said to just try and see if it works first. Sounds 
like some of the problems that we're having could be related to this, 
and it's probably the first thing I should try changing.


I have tried using ztmonitor to adjust the txgain and rxgain. It was 
very frustrating ... changing the tx to higher or lower values did not 
necessarily change it in the direction you'd think. Also, changing rx 
sometimes affected tx and vice versa. In the end, I gave up because 
nothing I did increased sound quality -- although it wasn't so hard to 
make things worse. Any hints here?


We're using Sipura phones. We realize these aren't has high quality as 
Polycoms, but this was what the manger decided to get based on the 
budget. We're spending more than originally anticipated now because of 
the problems we're having, but it seems like the Digium cards are a more 
likley culprit than the Sipura phones.  However, if there's a way to 
isolate problems, that'd be great. I guess one thing is we could try 
more experiments with one extension calling another -- if the sound 
quality is a lot better, then it's unlikely a problem with the Sipura 
phones. If the sound quality is poor, I'll try hooking up the phones to 
a new network card.


As for interference ... we don't have wireless devices, and the main 
phone we're testing with is about 10' away from the switch. I suppose 
there's still some possibility with interference, but I think it's 
something I'll prioritize a little lower.


One additional question -- are VoIP lines generally easier to get going 
w/ good sound quality than POTS lines? One reason we went with POTS was 
for sound quality. Of course, we also figured it'd be more reliable than 
DSL, which was the main reason for going with POTS lines. Right now we 
use 5 POTS lines ... if VoIP sounds better, we were thinking of dropping 
three of those lines and getting VoIP lines, keeping the POTS lines as 
backup. Any thoughts on that plan? One thing I'm not sure about is how 
to select the POTS lines from the phone sets. I suppose we could assign 
a second extension to the phones (they support 2) and have the second 
one be POTS and the first be VoIP. Another thing I was wondering is 
whether we could get hunting to work properly with a mix of VoIP and 
POTS. I'll call the phone company tomorrow, but if anyone has tried 
anything like this, I'd like to hear about it.


Also, are there any VoIP vendors that work particularly well with 
Asterisk? I've seen Broadvoice mentioned a bit ... are they a good 
company for this?  I'd love a company that could do automatic failover 
-- if the VoIP line cuts out, transfer to an analog phone line (I know 
Vonage has a feature like this, but they're expensive and we don't need 
a lot of the features).


Thanks again for all the suggestions!

Jen


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Re: [Asterisk-Users] Voice Problem ...asterisk on lan not woriking well...!!

Does your linksys support traffic shaping in any way? If so you should 
mark your VoIP data as a high priority.


Alternatively, use a slimer codec like G729. I'm not sure what the 
Polycom supports but It'll probably do 729. You'll have to buy some 
licences for your * box too.


Mark

Ing. Marlo R. Beltran G wrote:

Hi

 

 i just implemented asterisk and is such a grate solution...i am using 

 polycom 301 and 501 phoneson lan a iam using g.711 and i have a 


 16 port linksys switch...

 

 the problem come when somebody inside the network is making  a call to 

 other extension (in the same network) and is sending an e 


 mail through  internet the quality goes down...it hears rally bad...

 

 i am on a 10/100 network with cat5e on wire, and switches...what can 


 i do to have an excelent voice quality inside my network???

 


 The e mail doesn't go trough the asterisk computer.

 


Marlo

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[Asterisk-Users] dtmf tones during conversation

2005-08-21 Thread Leonardo F. Bauchwitz


Help for this use case:

I need detect a dtmf tone during conversation.
For example:
A call B
B answer A
A and B talk
B (while talk) press *111

and I defined an action en extensions.conf to
exten = *111,1,Application()

and while the application is executed A and B continues talk

---
What application would have to use?

Thanks and regards

Leonardo






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Re: [Asterisk-Users] Call duration limits not working

2005-08-21 Thread tim

I have CVS-HEAD as of yesterday and it's still not working for me.  Maybe
I'll try updating again and post the results.

Thanks,
Tim

 [EMAIL PROTECTED] wrote:
 Hello everybody.  Recently I've been trying to limit the duration of
 some
 calls for a simple application I'm writing.  Unfortunately all of the
 documented methods are failing and I'm not sure what else to try.  Here
 are some samples of what I've done:

 I believe this was fixed in CVS-HEAD a couple of weeks ago.  There was a
 big announcement on the mailing list about it.
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[Asterisk-Users] Dial Zero to get outside line?

2005-08-21 Thread Michael Felder

Hello,

My asterisk currently will dial an outside number after I dial the
number and press send on the phone.
How can I get it setup so I have to press '0' for an outside line.
 
Kind regards
 
Michael Felder
IT Medic Australia Pty. Ltd.
P: 03 9557 2213
F: 03 9557 2214
M: 0419 568 217 
E: [EMAIL PROTECTED]
http://www.ITMedic.com.au

Keeping your computer systems healthy.
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[Asterisk-Users] SDP media attribute

2005-08-21 Thread ggtel

Hello,

Any use of SDP media attribute in conjunction with SIP /Asterisk ?
I would appreciate any insight!

George







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RE: [Asterisk-Users] Vtech/Vonage Wireless Phone system

I currently have one of the Hitachi WIP-5000 SIP phones. I've been
using it off and on for about four months. Just one as an initial test.
While it's pretty good it does have some minor issue, or they could be
issues related to my wlan access points. Whatever the case, it drops
off lan periodically causing a silent period in the current call. I've
had to extend the qualify setting to 3000 or it gets reported as not
available now and again.

My next test will be the Aastra/Sayson 480 cti SIP deskphone with
cordless handset. It costs about the same as the Hitachi phone on its
own. I don't think that the cordless handset is truly SIP/Wifi based.
But that may not matter if its well integrated into the phone.

Michael

On Sun, 21 Aug 2005 18:47:50 -0400, Andre Normandin wrote:

I already have a Sipura and a panasonic cordless, I was just hoping that I
could find a true SIP wireless phone at a reasonable price  :-)

Oh well, one can drea laugh

Thanks,
  - Andre

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michael
Graves
Sent: Sunday, August 21, 2005 4:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Vtech/Vonage Wireless Phone system


All these things are is just an ATA built into a plain vanilla analog
cordless phone system...ok, a 5.8 Ghz systemvanilla with sprinkles
;-)

Nothing special at all. You'd be better off with a Sipura device and a
Panasonic cordless phone.

Michael

On Sun, 21 Aug 2005 15:12:49 -0400, Andre Normandin wrote:

Thanks,

I just read all the literature on the Vtech website, and, I think you are
exactly correct!

Oh well, so much for cheap SIP wireless phones :-)

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rich
Adamson
Sent: Sunday, August 21, 2005 2:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Vtech/Vonage Wireless Phone system


 I just stopped in at Best Buy here in CT, USA. I found an interesting
 offering from Vtech there. It states it's a VOIP wireless phone system
made
 for the Vonage service.

 Here it is at their website:
 http://www.vtechphones.com/vtechui/guide/dsp_voip_guide.cfm

 For $149, it comes with 2 (supposidly) VOIP 802.11b wireless phones, and
a
 'broadband/Wireless' router.

 Does anyone know anything about these? Are they SIP phones? If so, I
wonder
 if it's possible to modify them for use with Asterisk?  Do they need
their
 'base' to talk, or is that nothing more than an 802.11b wireless router?

 I'm thinking of picking up a kit just to see if I can get them to work
 nativly with Asterisk, but if anyone has any experience with them before
I
 do, I'd appreciate it..

I seen their ad in the paper this morning. It appeared the phones were
the standard cordless (not 802.11b), and the base unit had pc wireless
(802.11b) integrated with the linksys/sipura broadband stuff. You might
want to check that out a little closer; but, fairly good return policy
if the ads are a little misleading. ;)


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

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



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

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



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[Asterisk-Users] perl-cpan

2005-08-21 Thread Tommy Denton

Dear list,

I was installing Asterisk via the AMP method off the AMP website.

There is a portion in there where they want you to use perl-cpan to
install telnet.

The first time I installed I had no problem.  I messed up and trashed
the box further down in the  install.. This time I made a mistake and
put a -jp3 switch for multi processors.  This has caused the install
of telnet tank.  I found the cpan config file and removed it.  This
allowed me to get past the first mistake but now when I try to complie
telnet it fails and I do not get any error's on the make file.

How can I remove perl-cpan and start again, or am I facing another
trip to the colo to splash the box and start again?

Thank you for your time,

Tommy
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Re: [Asterisk-Users] All Page ??

2005-08-21 Thread Matthew T. O'Connor


Steve Maroney wrote:


Does anyone know of any plans to add an intercom/all-page feature in *?

The few SIP phones that have auto-answer capability would be better if
Asterisk could broadcast one leg of a channel to many legs at one time.



I'm looking for an answer to this problem also.  I am putting an 
Asterisk system into our new office.  In our old office we used the old 
phone system to act as an intercom, you hit page all and your voice 
comes out of the speaker on several handsets throughout the office.  
This allows you to announce information or to the whole office, simply 
announcing to someones desk doesn't work since our people move around a 
lot and are not always at their desk.


Anyway, I have some Polycom phones, and I have Autoanswer working with 
Asterisk, but which ever phone happens to answer the call first is the 
only one who's speaker my voice comes out of.


Anyone have an answer to this problem? 


Thanks,

Matt
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Re: [Asterisk-Users] TDM11B modprobe wcfxs fails

2005-08-21 Thread Fumiaki Okushi

  I got my TDM11B and am trying to get it to work on my PC.
  However, I'm having difficulty getting the wcfxs driver to
  load.  I've Googled this problem, and while there are others
...
 1. The TDM card has several different revisions (rev e through h,
 I believe). If you have one of the later revisions, you may need
 zaptel software later then stable v1.07.

Thanks!

It turned out to be this case.
My card was a rev I and needed 1.0.9 or later.
I now have my card working and so is Asterisk!

Thanks again!

Fumi Okushi
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[Asterisk-Users] Re: call waiting beep on PSTN and TDM400P FXO linehook flash

2005-08-21 Thread Larry Shields


I had a similar issue with sending a flash to the PSTN for call waiting.  I
found that my dial plan in the extensions.conf file was not allowing me to
dial *xx. Once I corrected my dial plan I was able to dial *0, *69, *78,
*79, etc. Training the wife on how to actually use it was an entirely
different issue. I have not tried to enable 3-way calling via the PSTN.

Jeff Otterson [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]...
   I have been looking for the answer to this question for a 
 while.  Google-ing and reading the archives of Asterisk-Users has not 
 enlightened me.
 
It seems that this question has been asked many times, and many times
it 
 has gone unanswered.
 
   I have call waiting and three way calling on my PSTN line from Verizon 
 (the local telco).  This is connected to a FXO port on a TDM400P.  I also 
 have two FXS ports on the TDM400P.
 
So my problem is, how do I flash the Verizon PSTN line when I hear the 
 call waiting beep?  How can I send a hook flash to the Verizon trunk to 
 activate their 3-way calling feature.
 
I have seen some stuff like hook flash then send *0 to get the bridged

 Zap trunk to flash but I can't get it to work.  I get the reorder 
 signal.  I need something my wife and kid can do.
 
Can anybody help?
 
Thanks,
 
Jeff
 
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[Asterisk-Users] Unable to create channel of type 'Zap'

2005-08-21 Thread root linux

My zaptel.conf config: -

# Below setting is for E1
span=1,1,0,cas,hdb3
bchan=1-15 
bchan=17-31 
dchan=16 

loadzone = us
defaultzone=us


My zapata.conf config: -

# Below setting is for E1
switchtype = national
signalling = pri_cpe
group = 1
channel = 1-15
channel = 17-31

My extension.conf config: -

[default]
exten = 181,1,Dial(Zap/1/181)

When I perform a dailing from my SIP Phone, I got the
error message as below: -

-- Executing Dial(SIP/118301-6f4e,
Zap/1/181) in new stack
Aug 22 11:03:02 NOTICE[9023]: app_dial.c:764
dial_exec: Unable to create channel of type 'Zap'
  == Everyone is busy/congested at this time


I am beginner...How to solve this?






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