[Asterisk-Users] Asterisk Segmentation Fault - layer3.c/mpg123

[Jason Goecke]

Hello,

I recently switched from using mpg123 to using the
built-in format_mp3 for MusicOnHold.  After playing a
song (which I adapted as perscribed to 8Khz/mono mp3
with lame) for about 30 seconds, I get static in the
earpiece and then:

Jan 11 08:59:08 WARNING[11039]: layer3.c:966
III_dequantize_sample: mpg123: Can't rewind stream by
10 bits!
Segmentation fault

I am testing with another mp3 file now, but I would
never expect a 'Segmentation fault'.  Has anyone seen
this?  If not, should I report as a bug and how best
to do this?

Thank you,

Jason
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Re: [Asterisk-Users] Connecting a Home based worker with An Iaxy

[Wilson Pickett]

 Anyone done this got any comments

If the server is on a dynamic ip, it will need to reprovision every
time it changes. That means it will need to be able to get to the IAXy
thru NAT on its own (=port4569 forwarded to the IAXy)
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Re: [Asterisk-Users] Weir long distance behaviour...

[Wilson Pickett]

 There is a strange behavior, when we do long distance calls, it keeps
 ringing on our end, remote callee answers the call but hear nothing.
Look up callprogress and busydetect

are you in France by any chance?

Look here also 
http://www.voip-info.org/tiki-index.php?page=Asterisk+config+zapata.conf
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RE: [Asterisk-Users] What is acceptablenetworklatencyforvoipconnection?

[Damon Estep]

 How does an ISP provide a Jitter SLA on a Data T1? Jitter  5ms? How
does
 one measure that?

You can get a good feel for delay and jitter just by running a
continuous ping to a core router on your ISPs network during peak
times(or to Google for that matter) and visually monitoring the results.
A good, unsaturated link will have extremely consistent response times
with less than 5ms variations between packets.

There are several ways to measure IP performance;

http://www.cisco.com/warp/public/126/saa.html
http://www.cisco.com/en/US/tech/tk648/tk362/tech_brief0900aecd801752ec.h
tml

Not using Cisco routers? Hunt the web for a Linux equivalent (anyone
know of one?), or;

Use any program capable of sending ICMP echoes and recording the
response times.

Gather ~24 hours of data at a time, every 1 second or more (every 1
second would be 86400 data points per day).

Load the data in a spreadsheet (or create a script to calculate)

Find the absolute value of the difference between every packet and the
packet preceding it, and then average all of the difference values.

The result is your average jitter

Qualify your results by disclosing the parameters you use to measure.

Include the measurement technique and expected results in your SLA so
your curious customers can verify your performance.

Monitor your network continuously, IP networks are very dynamic.

With all of that being said, most ISP customer's measure with their ear
by answering the question is my VoIP working?


 
 We're an ISP, we've been doing T1's for many years. I know that
customers
 can ping any equipment or servers within our network with  10ms
response
 times, we link with 3 large Tier 1 providers, DS3 speeds for 2 of them
and
 a
 90mbs NMLI for the third, we're not over-saturated at all, bandwidth
to
 spare, and our customers generally report 50ms average response times
out
 to
 the internet. To further regions and when going through a couple of
 networks, it can be up to 80ms. The only time I've seen over 100ms is
to
 international destinations, but the ping response times generally stay
 consistent, no dramatic spikes. That's with a standard frame relay T1.
 Point
 to Point T1's are slightly better. However I didn't create or design
nor
 do
 I maintain this company's network, I just do the VOIP thing, so I'm
quite
 curious to see their SLA, I wonder if a misconfiguration somewhere
could
 affect quality and give me a head-ache.
 
 For a few of our heavy VOIP customers, we use 2 Point-To-Point T1's
from
 our
 location to theirs, we then combine traffic to go across both T1's, if
1
 T1
 fails, it'd all automatically go across the remaining T1. They can
ping
 our
 Asterisk server with an average of 10ms latency. We also tend to use
 private
 IP's, to avoid internet DDOS's and worms and such. We've been able to
push
 35-40 calls across this link. Probably more if I used trunking and/or
 switched codecs from ULaw.
 
 


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RE: [Asterisk-Users] Generic modem question

[Derek Conniffe]

 Yes, you can buy a clone.  Yes, it may work currently (although I wouldn't
want to guess for how long).  Also, the ??   
 actual cards that the X100Ps are based on have stopped being produced by
Intel, so you're out of luck as far as a 
 replacement goes in 6 months time.

I though that the X100P were a tigerjet chip?  I'm not looking at one right
now but I've seen the Tigerjet branding on the real X100Ps and also on the
TDM400 board too.  Actually I have a couple of branded tigerjet telephone
gateway (or something) cards that are identical [in appearance] to X100Ps
(although I remember that in zaptel they were identified as generic) - I'm
not using them now but they seemed to work fine. 

 Don't forget that the impedance on the X100P (or clone) is 600Ohms so you
won't be able to use it without echo outside 
 of the United States.

Thats very interesting - I've certinly had echo annoyances (not major
problems - echo canel got rid of it after a second or so) and I put it down
to bad quality telephone lines (probably true too).


Derek

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[Asterisk-Users] AGI Application Hangup when using AGI-getdata

[ALI AL-MUGHRABI]

Before
coming in here , I had a deep dig into Google and couldnt find an
answer, 



Simply
spoken, using agi-getdat in an AGI application , the call disconnects if
digits are entered fast by user.



I'm
certain that others have been though this problem, please pour your experience
here J 



Ali Mughrabi



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Re: [Asterisk-Users] chan_cornet

[richard Coco]

Hi,

we use the oh323 driver (see the post from Joao for installationhttp://www.archivum.info/asterisk-users@lists.digium.com/2005-01/msg00931.html). 

in the oh323.conf
[general]
listenPort= 192.x.x.x /the ip @ of the HG3550
fastStart=yes /*enable fast start
context=voip-h323
codec=G711A

in the extensions.conf

exten = _0.,2,Dial,OH323/h323:[EMAIL PROTECTED],tr /* for outgoing to HiPath
[voip-h323]
include = default /* for incoming from HiPath.
the rest are default settings.
The H4K installation is more difficult. You have to use a HG3550 board.
1. install HG3550 board
2. initial installation of the loadware (use the latest version!)
3. use WebManagment to configure the routing to Asterisk.

marek cervenka [EMAIL PROTECTED] wrote:
 i agree... no H.323 support for endpoint (HG3530) but you still have h.323 (for the moment only version 2.0) support for ip-trunking (HG3550). So what if you have the following setup. [OPTIPOINT400_HFA]--[HIPAT4K][oh.323][ASTERISK]--[OPTIPOINT400_SIP].how can i configure ip-trunking from HI4K to asterisk?any example h323 conf for asterisk?---Marek CervenkaCentrum Vypocetni TechnikyCVT - http://cvt.fpf.slu.czFPF SLU OPAVA - http://www.fpf.slu.czLCNA - http://lcna.slu.cz===___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options
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[Asterisk-Users] asterisk-oh323 and outgoing call

[Alexander Averyanov]

Hello.

I'm try to set up asterisk for making outgoing calls with oh323 channel
driver version 0.7.1 with Asterisk CVS-1-01/09/05-01:41:37.

Our provider uses Mera MVTS softswitch and supports only H.323.

We don't use gatekeeper for connection but provider requires SOURCE PHONE
NUMBER for route out calls and I don't know how I can specify this
number.

Call with this string
exten = _XXX,1,Dial,OH323/[EMAIL PROTECTED]
returns 
-- H.323 call 'ip$localhost/12715' cleared, reason 11 (Gatekeeper could not 
find user)

Please help! How can I supply source phone number for oh323?

-- 
Alexander Averyanov
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Re: [Asterisk-Users] [OFF TOPIC] Voip phone sellers in India

[Sandeep A.S]

check with webtel,Mob:32333033

On Sun, 2005-01-09 at 19:33 +0100, Vikram Rangnekar wrote:
 I am looking for some in India  to buy VOIP phones from. Please get in touch
 with me off the list on [EMAIL PROTECTED]
 
 Sorry for the off topic mail I am just having such a hard time finding any
 voip phones in India that I got desperate and didnt know which list to post
 this on.
 
 
-- 
Sandeep A.S [EMAIL PROTECTED]
Netcontinuum Pvt Ltd 

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Re: [Asterisk-Users] asterisk-oh323 and outgoing call

[Michael Manousos]

Alexander Averyanov wrote:
Hello.
I'm try to set up asterisk for making outgoing calls with oh323 channel
driver version 0.7.1 with Asterisk CVS-1-01/09/05-01:41:37.
Our provider uses Mera MVTS softswitch and supports only H.323.
We don't use gatekeeper for connection but provider requires SOURCE PHONE
NUMBER for route out calls and I don't know how I can specify this
number.
Call with this string
exten = _XXX,1,Dial,OH323/[EMAIL PROTECTED]
returns 
-- H.323 call 'ip$localhost/12715' cleared, reason 11 (Gatekeeper could not find user)

Please help! How can I supply source phone number for oh323?
Use the SetCallerID() app in the dialplan.
Michael.
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[Asterisk-Users] test source for current xorcom rapid

[Tzafrir Cohen]

Hi

I put a snapshot of our current packages updates.xorcom.com . They are
available from the deb source

  deb http://updates.xorcom.com/test sarge main

(this is s/rapid/test/ of the name of the source of the stable version)

Changes include:

* Fixed and simplified zaptel detection
* Support for spandsp: compiled but not yet tested
* IAX extensions
* Mail server configuration
* The script ast-cmd with some useful commands

Xorcom Rapid is a Debian/Asterisk distribution program that features an
auto-install for Debian Linux and pre-configured Asterisk. It quickly
and effortlessly converts any PC to a functioning Asterisk PBX.

http://xorcom.com/rapid/

Should work on a Debian Sarge installation as well. On a Rapid 0.9.0
system it will require an apt-get dist-upgrade .

-- 
Tzafrir Cohen   +---+
http://www.technion.ac.il/~tzafrir/ |vim is a mutt's best friend|
mailto:[EMAIL PROTECTED]   +---+
icq#16849755  +972-50-7952406   tzafrir on freenode
[EMAIL PROTECTED]http://xorcom.com

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[Asterisk-Users] requiring logon for SIP users

[Florian Effenberger]

Hello there,
I am playing around with Asterisk the first time and it really looks 
great. ;-)

However, I have one problem: Any SIP device can connect to my PBX. How 
can I requre logon for SIP users and deny access in the case of wrong or 
missing credentials?

Thanks
Florian
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Re: [Asterisk-Users] requiring logon for SIP users

[Julian J. M.]

You can set the default sip context to a nonexistant, and set the
correct one in the peer definition... Although i guess there must be a
better solution ;)

JulianJM


On Tue, 11 Jan 2005 11:35:36 +0100, Florian Effenberger [EMAIL PROTECTED] 
wrote:
 Hello there,
 
 I am playing around with Asterisk the first time and it really looks
 great. ;-)
 
 However, I have one problem: Any SIP device can connect to my PBX. How
 can I requre logon for SIP users and deny access in the case of wrong or
 missing credentials?
 
 Thanks
 Florian
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Re: [Asterisk-Users] Bristuffed Asterisk 1.0.3 hfc-s card doesn't work

[Remco Barende]

Hi!
There is a link in the wiki on www.voip-info.org
http://www.voip-info.org/wiki-Asterisk+zaphfc
this is the download link:
http://www.junghanns.net/asterisk/downloads/bristuff-0.2.0-RC3.tar.gz
(I don't have a clue why Junghanns would release a new version but not 
update their website)

But it works :)
Remco
On Tue, 11 Jan 2005, Andrew Thrift wrote:
Hi Remco,
just wondering how you got Asterisk 1.0.3 BRI-Stuffed.
On Junghanns.net I can only see 0.1.0-rc4 of bri-stuff and it uses something 
like asterisk 0.8

Your help is much appreciated.
Regards,

Andrew Thrift

Remco Barende wrote:
On Sun, 9 Jan 2005, Remco Barende wrote:
I hva ean HFC-S card in a box that I'm trying to get to work with 
bristuffed Asterisk 1.0.3. The box is an Athlon64 running a RHEL 
rebuild with a plain vanilla 2.6.10 kernel. I tried both APIC and 
NOAPIC mode.

The installation went ok and does give output that seems correct
SPAN 1: CCS/ AMI Buil-out: 399-533 feet (DSX-1)
2 channels and one D-channel
Even though I've configured * to immediately accept calls I do not see 
any calls coming in, when I dial the phone number I get the tone that 
the number is disconnected. When I try zttool to do a loopback test, it 
just displays Looping UP span 1 for about 20 seconds and returns to the 
menu without any message. It doesn't show any alarms.

Any ideas what could be going wrong, how can I test the ISDN card any 
other way?

The line and telephone cord are both OK, when I connect the cord to the 
AB adapter it just works.

Thanks!

Sorry, it was me :(
I loaded zaptel first then zaphfc.
This generates no error message (shouldn't it??) but just doesn't work.
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[Asterisk-Users] asterisk one number service

[Ashling O'Driscoll]

I wonder does anyone have any thoughts or can give me some direction
on the following:

I have an asterisk testbed environment set up. My task is to make a
personal number service available whereby users would be given one
number (perhaps a voip number) and this number would enable them to
be reached via the pstn, pots, gsm etc

Does anyone have ideas where I could start looking at sites to
research this or how asterisk might fit into this?. It would be great
if someone could maybe point me in the right direction.

Thanks,
Aisling.


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Re: [Asterisk-Users] requiring logon for SIP users

[Florian Effenberger]

You can set the default sip context to a nonexistant, and set the
correct one in the peer definition... Although i guess there must be a
better solution ;)
;-)
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Re: [Asterisk-Users] IAX2 keep alive?

[Steve Totaro]

you could also forward port 4569 from the nat router to your asterisk box.
i think a qualify statement in iax.conf will also help if you find out how
quickly the router is shutting down the map and set the qualify statement to
a shorter time frame.


- Original Message - 
From: Dinesh Nair [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, January 11, 2005 12:22 AM
Subject: Re: [Asterisk-Users] IAX2 keep alive?


 On 11/01/2005 04:21 Miguel Ruiz Velasco Sobrino said the following:
  In a setup I've made i have a problem in the two way origination of the
call.
 
  Asterisk 1 == Public internet == NAT == Asterisk 2
  I'm pretty sure it's a NAT loosing state too fast, and i can do nothing
to fix the NAT.
  Is there a way to have a keep-alive between the two * boxes?

 have *2 register (via IAX2) with *1 and all should be fine. IAX2 works
well
 thru NAT, unlike SIP/RTP.

 -- 
 Regards,   /\_/\   All dogs go to heaven.
 [EMAIL PROTECTED](0 0)http://www.alphaque.com/

+==oOO--(_)--OOo==+
 | for a in past present future; do
|
 |   for b in clients employers associates relatives neighbours pets; do
|
 |   echo The opinions here in no way reflect the opinions of my $a $b.
|
 | done; done
|

+=+
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[Asterisk-Users] Analogue RAS Server

[Daniel Niasoff]

Hi,



Does anyone have any idea how to set up Asterisk so
that it can act as an Analogue Remote Access Server. Ive looked around
and as far as I can see it will only act as an ISDN Ras server.



Thanks



Daniel






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Re: [Asterisk-Users] Generic modem question

[Steve Totaro]

 On Mon, 10 Jan 2005 20:09:54 -0600, Rich Adamson [EMAIL PROTECTED]
wrote:
 
   Does asterisk support the intel 537/md3200 chipset?  I don't want to
start
   any flames here, I know all about using generic crap in asterisk,[*]
which
   I really don't approve of other than for testing, but I have a
customer
   demanding a generic chipset for his one backup analog line.  He will
not
   spend the money for a Digium card and says he will find another
   company if I can not provide a generic FXO port.

Sometimes it is prudent to fire a customer.

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[Asterisk-Users] Realtime and include

[Alessio Focardi]

Hi,

I'm testing realtime right now, it does not seem to me that realtime
contexts can be included in normal context, like this

[sip]

include=sip-dial

exten=i,1,Hangup

[sip-dial]

switch=Realtime/sip-dial

Am I getting it wrong ?

Tnx !

-- 
Best regards,
 Alessio  mailto:[EMAIL PROTECTED]

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Re: [Asterisk-Users] asterisk-oh323 and outgoing call

[Alexander Averyanov]

On Tue, Jan 11, 2005 at 12:24:54PM +0200, Michael Manousos wrote:

 Please help! How can I supply source phone number for oh323?

 Use the SetCallerID() app in the dialplan.

Thank you. It works!

-- 
Alexander Averyanov
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[Asterisk-Users] How to mark a user for a conference

[Jagan Mohan]

Hi All,

   I would like to mark a user so that all users other than marked
user hear music-on-hold till the marked user joins the conference.
   I took a look at 
http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe, but could not get
sufficient info.
I'm using meetme for conferencing.
   Could anyone point me to a url which has the configuration details
using meetme.

Thanks,
Jagan
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Re: [Asterisk-Users] fax e-mail spandsp

[Nils Segerdahl]

Matt Riddell wrote:
Brian Dingman wrote:
Anyone care to pass on a makefile that works. This is what my
makefile.rej looks like:
[SNIPPED]
Really it's not that hard.  Open two console windows.  In one open 
that patch.  In the other open the Makefile.

If you look at the patch you can see what lines need to go into the 
Makefile and where.  (the + symbol means add this line, and the lines 
without +'s show what is before and after the section you need to 
change).

If you have any problems, drop me a line off-list and I'll help you 
out (but it's worth your while to at least have a try).
Hi,
Remember that make requires that the indentation in the Makefile is done 
with tab and not spaces.
I you cut and paste there is a risk that the indentation is converted to 
spaces.

/Nils
--
Nils Segerdahl

Upsala Systemkonsult, UPSYS AB  Telefon:(+46) (0)18 56 80 41
Upsala Science Park, 751 83 Upsala  Mobil: (+46) (0)703 55 65 03
http://www.upsys.se Fax: (+46) (0)18 56 80 49

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Re: [Asterisk-Users] Vmail.cgi - Hrm, can't seem to open /var/spool/asterisk/voicemail ....

[Frank Kostin]

Hi, Just doing a "chmod" OK
Halas, not a specialist in cgi and/or perl how to run that automatically into script preferably forspecific box b4 list msg'sAnyone really smart could help ?ThanksMike Dent [EMAIL PROTECTED] wrote:

Yes, its the permissions on the wav/gsm files:--rwx-- 1 root root 330 Nov 16 23:48 msg.gsm-rw-r--r-- 1 root root 231 Nov 16 23:48 msg.txt-rwx-- 1 root root 3244 Nov 16 23:48 msg.wav-rwx-- 1 root root 385 Nov 16 23:48 msg.WAV-rwx-- 1 root root 13794 Nov 16 23:51 msg0001.gsm-rw-r--r-- 1 root root 216 Nov 16 23:51 msg0001.txt-rwx-- 1 root root 133804 Nov 16 23:51 msg0001.wav-rwx-- 1 root root 13646 Nov 16 23:51 msg0001.WAV-rwx-- 1 root root 2310 Nov 17 09:41 msg0002.gsm-rw-r--r-- 1 root root 216 Nov 17 09:41 msg0002.txt-rwx-- 1 root root 22444 Nov 17 09:41 msg0002.wav-rwx-- 1 root root 2336 Nov 17 09:41 msg0002.WAV-rwx-- 1 root root 20460 Nov 18 11:48 msg0003.gsm-rw-r--r-- 1 root root 217 Nov 18 11:48 msg0003.txt-rwx-- 1 root root 198444 Nov 18 11:48
 msg0003.wav-rwx-- 1 root root 20210 Nov 18 11:48 msg0003.WAVthey are not readable by the web process. Anyway I have not fixed ityet, so please let me know if you do.MikeOn Mon, 10 Jan 2005 08:00:13 -0800 (PST), Frank Kostin<[EMAIL PROTECTED]>wrote: Hello everybody, I was trying to install a web interface to my Voice Mail, Vmail.cgi I can log on it, list messages, but no play with the following error msg;   "Hrm, can't seem to open /var/spool/asterisk/voicemail/default/234/INBOX/msg0001.WAV"   Remark: playing the message msg0001.WAV directly OK  Any smart guy up there could help ? Thanks,   Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard.___ Asterisk-Users mailing list
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Re: [Asterisk-Users] Analogue RAS Server

[Niksa Baldun]

I don't think it's possible. Asterisk would have to emulate analog
modem, and I believe that feature is not (at least yet) implemented.

Daniel Niasoff wrote:

  
  
  
  
  Hi,
  
  Does anyone
have any idea how to set up Asterisk so
that it can act as an Analogue Remote Access Server. Ive looked around
and as far as I can see it will only act as an ISDN Ras server.
  
  Thanks
  
  Daniel
  
  

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Re: [Asterisk-Users] Analogue RAS Server

[Paradise Dove]

  I don't think it's possible. Asterisk would have to emulate analog modem,

does anybody know if  there ia any works on emulating analog modems
(not specially to work with asterisk).
something like Steve's spandsp for fax.
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[Asterisk-Users] Zaptel config

[ismaelg]

Hello all,
I am having a lot of problems with zaptel channels,
I have got an TDM02B, and I don't know how setup /etc/zaptel.con and 
/etc/asterisk/zapata.conf for use it on asterisk.

Some one could help me with this configuración?
My problem is about the type of signalling
Thanks,
Regards.
Ismael Gil.
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Re: [Asterisk-Users] Zhone channel bank issues

[James Freire]

Hi Michael,
You might want to check the voltage settings on the FXS side of
things. Also, are you using the correct signalling? (ground start,
loop start, etc.)
In the Zplex users guide, on page 41 you will see 2 sections on TTLP
and RTLP. That might be of some help to you.

Hey... You have caller ID working on that thing??? How did you do that? 
Let me know if you need a PDF copy of the manual

-James


On Mon, 10 Jan 2005 20:55:13 -0500, Michael Lyszczek
[EMAIL PROTECTED] wrote:
 On Mon, 10 Jan 2005 12:51:49 -0500, Michael Lyszczek
 [EMAIL PROTECTED] wrote:
  Anyone have any issues like thisI am fwding broadvoice to zaptel,1
  with my t100p and the t1 goes to a zhone zplex10b.. I can ring
  extension 1, which is pair 1 of the channel bank, but it doesnt
  recognize offhook and it keeps ringing the phone after I pick up.
  Also, its like each ring is like a seperate call as far as the
  callerid history goes.  Anyone have any ideas?
  Michael Lyszczek
 
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Re: [Asterisk-Users] Weir long distance behaviour...

[Francois Meehan]

Hi Wilson,

I had both features enabled in my zapata.conf file, I will try disabling
the  callprogress see if it makes a difference, what troubles me is that I
have no problems with local calls, what could be the difference with long
distance one?

I am from Quebec, Ile-Perrot near Montreal.

Regards,

Francois

 There is a strange behavior, when we do long distance calls, it keeps
 ringing on our end, remote callee answers the call but hear nothing.
 Look up callprogress and busydetect

 are you in France by any chance?

 Look here also
 http://www.voip-info.org/tiki-index.php?page=Asterisk+config+zapata.conf
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Random Thought:
---
Business will be either better or worse.
-- Calvin Coolidge
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RE: [Asterisk-Users] Generic modem question

[Rich Adamson]

  Yes, you can buy a clone.  Yes, it may work currently (although I wouldn't
 want to guess for how long).  Also, the ??   
  actual cards that the X100Ps are based on have stopped being produced by
 Intel, so you're out of luck as far as a 
  replacement goes in 6 months time.
 
 I though that the X100P were a tigerjet chip?  I'm not looking at one right
 now but I've seen the Tigerjet branding on the real X100Ps and also on the
 TDM400 board too.  Actually I have a couple of branded tigerjet telephone
 gateway (or something) cards that are identical [in appearance] to X100Ps
 (although I remember that in zaptel they were identified as generic) - I'm
 not using them now but they seemed to work fine. 
 
  Don't forget that the impedance on the X100P (or clone) is 600Ohms so you
 won't be able to use it without echo outside 
  of the United States.
 
 Thats very interesting - I've certinly had echo annoyances (not major
 problems - echo canel got rid of it after a second or so) and I put it down
 to bad quality telephone lines (probably true too).

I believe one can characterize the TigerJet name as the pci controller
chip, but the card has several other chips as well. My x100p card has
a heatsink glued on top of one of the chips so I can't see the actual
part number; I believe its the Tigerjet chip however.


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Re: [Asterisk-Users] Zaptel config

[Tzafrir Cohen]

On Tue, Jan 11, 2005 at 02:10:11PM +0100, ismaelg wrote:
 Hello all,
 
 I am having a lot of problems with zaptel channels,
 I have got an TDM02B, and I don't know how setup /etc/zaptel.con and 
 /etc/asterisk/zapata.conf for use it on asterisk.
 
 Some one could help me with this configuracin?
 My problem is about the type of signalling

We wrote a simple script to do just that:

  http://updates.xorcom.com/genzaptelconf

Only tested on Rapid and Debian, but should generally work elsewhere

  genzaptelconf -sdv

-- 
Tzafrir Cohen   +---+
http://www.technion.ac.il/~tzafrir/ |vim is a mutt's best friend|
mailto:[EMAIL PROTECTED]   +---+
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RE: [Asterisk-Users] asterisk one number service

[Eric Hall]

I have it setup to dial my sip phone and my cell at the same time. Is
this what you are looking for? 

If so just add  after your dial sip command
(sip/123456789zap/g1/6145551212)

This works for me

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ashling
O'Driscoll
Sent: Tuesday, January 11, 2005 5:47 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] asterisk one number service

I wonder does anyone have any thoughts or can give me some direction on
the following:

I have an asterisk testbed environment set up. My task is to make a
personal number service available whereby users would be given one
number (perhaps a voip number) and this number would enable them to be
reached via the pstn, pots, gsm etc

Does anyone have ideas where I could start looking at sites to research
this or how asterisk might fit into this?. It would be great if someone
could maybe point me in the right direction.

Thanks,
Aisling.


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[Asterisk-Users] internal caller id on analog phones connected to zap

[Shoval Tomer]

Hi,
We've got IAX softphones, GrandStream VOIP phones and zaptel connected
analog phones.

Caller id, internally, works just fine (as long as I use numeric only
callerids) for IAX and grandstream.

Is there a way to have the analog phones' LCD display show the caller
id?

These are plain old regular analog phone, that if I had callerid from my
telco would show on the screen.

thanks

Shoval Tomer,
IT Manager,
SofTov Advanced Systems, Ltd.
Office: +972-3-9230686 ext. 179
Fax: +972-3-9216642
Mobile: +972-54-8000200


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[Asterisk-Users] sip to h.323

[sai latha]

Hello,
 Happy New Year

where u r downloaded the asterisk server please
tell me.Iam searching the asterisk server site in
google but i dint get this server u please tell me the
site for me 
Is only for sip to sip or sip to h.323 please tell
me
 
Thank u
Bye
Sailatha
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Re: [Asterisk-Users] Vmail.cgi - Hrm, can't seem to open /var/spool/asterisk/voicemail ....

[Jon Radon]

This issue is well documented.

http://www.voip-info.org/tiki-index.php?page=Asterisk%20gui%20vmail.cgi


On Tue, 11 Jan 2005 04:12:53 -0800 (PST), Frank Kostin
[EMAIL PROTECTED] wrote:
 Hi, Just doing a  chmod OK
 
 Halas, not a specialist in cgi and/or perl how to run that automatically
 into script preferably for specific box b4 list msg's
 Anyone really smart could help ?
 Thanks

-- 
Is it something someone said, was it something someone said?
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Re: [Asterisk-Users] Route incoming call on 4 X100P to different Ext. {Scanned}

[C F]

In zapata.conf give each card a different context. In extensions.conf
create 4 different contexts with different s extesnions.


On Mon, 10 Jan 2005 19:47:49 -0800 (PST), David [EMAIL PROTECTED] wrote:
 Hello All,
 
 I have 4 X100P cards. I was hoping to have card (line) go to separate ext.
 
 i.e.
 Card 1 (XXX)555-0001 My Ext
 Card 2 (XXX)555-0002 Wife's Ext
 Card 3 (XXX)555-0003 Fax Ext
 Card 4 (XXX)555-0004 My and Wife Ext.
 
 This is what I have now and all incoming line rings this one extension.
 exten = s,1,Dial(SIP/300,10)
 
 So what is s .
 
 Thanks, David
 
 --
 This message has been scanned for viruses and
 dangerous content by KE6UPI, and is
 believed to be clean.
 KE6UPI thanks MailScanner for their support.
 Please contact [EMAIL PROTECTED] if you have
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[Asterisk-Users] ASTCC - error on call end

[clive]

Hi

I get an error popping up when the call ends as follows:


DBD::mysql::db do failed: Unknown column 'callstart' in 'field list' at 
/var/lib/asterisk/agi-bin/astcc.agi line 90, STDIN line 32.

Does anyone else get this same thing?
Looks as if my database table is wrong, or something else is 
up...not sure

Thanks
Clive

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Re: [Asterisk-Users] internal caller id on analog phones connected to zap

[C F]

How are the analog phones connected to * ? this is where the setting should be.
If you use Digium then you set it in zapata.conf, if you use an ata
the setting should be in the ata.
BTW, how is FC3 working out?
Shalom Ubracha V'Chodesh Tov


On Tue, 11 Jan 2005 16:30:41 +0200, Shoval Tomer [EMAIL PROTECTED] wrote:
 Hi,
 We've got IAX softphones, GrandStream VOIP phones and zaptel connected
 analog phones.
 
 Caller id, internally, works just fine (as long as I use numeric only
 callerids) for IAX and grandstream.
 
 Is there a way to have the analog phones' LCD display show the caller
 id?
 
 These are plain old regular analog phone, that if I had callerid from my
 telco would show on the screen.
 
 thanks
 
 Shoval Tomer,
 IT Manager,
 SofTov Advanced Systems, Ltd.
 Office: +972-3-9230686 ext. 179
 Fax: +972-3-9216642
 Mobile: +972-54-8000200
 
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Re: [Asterisk-Users] sip to h.323

[Scott Stingel]

You can read all about it, and find out where to download at:
http://www.voip-info.org/tiki-index.php?page=Asterisk
Yes, it supports both SIP and H.323
Cheers
Scott Stingel
sai latha wrote:
Hello,
Happy New Year
   where u r downloaded the asterisk server please
tell me.Iam searching the asterisk server site in
google but i dint get this server u please tell me the
site for me 
   Is only for sip to sip or sip to h.323 please tell
me

Thank u
Bye
Sailatha
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.
 

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Re: [Asterisk-Users] Realtime and include

[Matthew Boehm]

What is leading you to believe that this isn't working? You didn't give us
much to work with...

-Matthew
- Original Message - 
From: Alessio Focardi [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, January 11, 2005 5:07 AM
Subject: [Asterisk-Users] Realtime and include


 Hi,

 I'm testing realtime right now, it does not seem to me that realtime
 contexts can be included in normal context, like this

 [sip]

 include=sip-dial

 exten=i,1,Hangup

 [sip-dial]

 switch=Realtime/sip-dial

 Am I getting it wrong ?

 Tnx !

 -- 
 Best regards,
  Alessio  mailto:[EMAIL PROTECTED]

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RE: [Asterisk-Users] ASTCC - error on call end

[Karl H. Putz]

There is a field missing in the admin.cgi CREATE for cdrs.

add: callstart CHAR(24) to the cdrs table

There is a patch to fix the cgi at

http://bugs.digium.com/bug_view_page.php?bug_id=0002796 

It just hasn't made it through to CVS yet.


Karl Putz

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of
 [EMAIL PROTECTED]
 Sent: Tuesday, January 11, 2005 9:45 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] ASTCC - error on call end
 
 
 Hi
 
 I get an error popping up when the call ends as follows:
 
 
 DBD::mysql::db do failed: Unknown column 'callstart' in 'field list' at 
 /var/lib/asterisk/agi-bin/astcc.agi line 90, STDIN line 32.
 
 Does anyone else get this same thing?
 Looks as if my database table is wrong, or something else is 
 up...not sure
 
 Thanks
 Clive
 
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Re: [Asterisk-Users] ASTCC - error on call end

[Darren Wiebe]

There is a bug in the database creation line.  Add a field 'callstart'  
CHAR (24) and it should work.

Darren
[EMAIL PROTECTED] wrote:
Hi
I get an error popping up when the call ends as follows:
DBD::mysql::db do failed: Unknown column 'callstart' in 'field list' at 
/var/lib/asterisk/agi-bin/astcc.agi line 90, STDIN line 32.

Does anyone else get this same thing?
Looks as if my database table is wrong, or something else is 
up...not sure

Thanks
Clive
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[Asterisk-Users] Route incoming call on 4 X100P to different Ext. {Scanned}

[David Shaw]

Hello All,

I have 4 X100P cards. I was hoping to have card (line) go to separate ext.

i.e.
Card 1 (XXX)555-0001 My Ext
Card 2 (XXX)555-0002 Wife's Ext
Card 3 (XXX)555-0003 Fax Ext
Card 4 (XXX)555-0004 My and Wife Ext.

This is what I have now and all incoming line rings this one extension.
exten = s,1,Dial(SIP/300,10)

So what is s .

Thanks, David


-- 
This message has been scanned for viruses and
dangerous content by KE6UPI, and is
believed to be clean.
KE6UPI thanks MailScanner for their support.
Please contact [EMAIL PROTECTED] if you have 
questions about this email.

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Re: [Asterisk-Users] Zhone channel bank issues

[Lyle Giese]

TTLP and RTLP should have no effect on this problem.  Those are for voice
transmission levels through the channel only.

I would check the loop/ground start settings and if possible check the
ringing voltage to make sure it has super imposed DC during the ring cycle
also and if the channel unit is putting out talk battery normally.

Lyle Giese

- Original Message - 
From: James Freire [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Sent: Tuesday, January 11, 2005 7:13 AM
Subject: Re: [Asterisk-Users] Zhone channel bank issues


 Hi Michael,
 You might want to check the voltage settings on the FXS side of
 things. Also, are you using the correct signalling? (ground start,
 loop start, etc.)
 In the Zplex users guide, on page 41 you will see 2 sections on TTLP
 and RTLP. That might be of some help to you.

 Hey... You have caller ID working on that thing??? How did you do that?
 Let me know if you need a PDF copy of the manual

 -James


 On Mon, 10 Jan 2005 20:55:13 -0500, Michael Lyszczek
 [EMAIL PROTECTED] wrote:
  On Mon, 10 Jan 2005 12:51:49 -0500, Michael Lyszczek
  [EMAIL PROTECTED] wrote:
   Anyone have any issues like thisI am fwding broadvoice to zaptel,1
   with my t100p and the t1 goes to a zhone zplex10b.. I can ring
   extension 1, which is pair 1 of the channel bank, but it doesnt
   recognize offhook and it keeps ringing the phone after I pick up.
   Also, its like each ring is like a seperate call as far as the
   callerid history goes.  Anyone have any ideas?
   Michael Lyszczek
  
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[Asterisk-Users] Installing * on fedora 3

[Ferguson, Michael]

G'Day List,

Can someone help me out a bit please.
I just installed Fedora Core 3 on a Dell Power Edge 400 SC and am trying
to install *
I am following
http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation
After running:
   cd /usr/src/asterisk 
   make clean 
   make 
   make install 
   make samples
The instructions says:
Configuring Asterisk 

- Login to your server as user root 
- Right-click on the background and select Open Terminal 
- Run the following commands to download the VoicePulse Connect! public
key (needed for receiving calls): 

   cd /var/lib/asterisk/keys 
   wget http://connect.voicepulse.com/keys/voicepulse01.pub 

However there is NO /var/lib/asterisk/keys directory.
HELP!!

Thanks



Michael E. Ferguson
Manager, Information Systems
Berman Rennert Vogel  Mandler, P.A.
100 SE 2nd., Street, Suite 2900
Miami, FL., 33131
305.423.3408 Direct
305.533.1582 Fax
[EMAIL PROTECTED] 
This message is for the named person's use only. It may contain
confidential, proprietary or legally privileged information. No
confidentiality or privilege is waived or lost by any mistransmission.
If you receive this message in error, please immediately delete it and
all copies of it from your system, destroy any hard copies of it and
notify the sender. You must not, directly or indirectly, use, disclose,
distribute, print, or copy any part of this message if you are not the
intended recipient. BERMAN RENNERT VOGEL  MANDLER, P.A. reserve the
right to monitor all e-mail communications through its networks. Any
views expressed in this message are those of the individual sender,
except where the message states otherwise and the sender is authorized
to state them to be the views of any such entity. 
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RE: [Asterisk-Users] Route incoming call on 4 X100P to different Ext.{Scanned}

[David J Carter]

David,

Try something like this:-

zapata.conf

context=me
signalling=fxs_ks
channel = 1
;
context=her
signalling=fxs_ks
channel = 2
;
context=fax
signalling=fxs_ks
channel = 3
;
context=meandher
signalling=fxs_ks
channel = 4


extensions.conf

[me]
exten = s,1,Dial(SIP/0001,30,t)
exten = s,2,Hangup
;
[her]
exten = s,1,Dial(SIP/0002,30,t)
exten = s,2,Hangup
;


and so on.


Regards


Dave



-Original Message-

Hello All,

I have 4 X100P cards. I was hoping to have card (line) go to separate ext.

i.e.
Card 1 (XXX)555-0001 My Ext
Card 2 (XXX)555-0002 Wife's Ext
Card 3 (XXX)555-0003 Fax Ext
Card 4 (XXX)555-0004 My and Wife Ext.

This is what I have now and all incoming line rings this one extension.
exten = s,1,Dial(SIP/300,10)

So what is s .

Thanks, David


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[Asterisk-Users] RE: Asterisk and InterTel Axxess system?

[Jason Kawakami]

-Original Message-
My office recently purchased an InterTel Axxess system with the IPRC 
card for VoIP.  To our suprise, this card allows the InterTel endpoints 
and MGCP endpoints to work, but not SIP clients.  I was really 
expecting to get a SIP softphone working with this setup, but that 
appears to require our vendor to sell us a SIP gateway and licenses at 
a not yet determined price.

-the sip server comes bundled with an application called Unified
Communicator which is pretty cool but also pretty costly.  Besides that, the
licensing for the SIP endpoints is not so cool.

With this background, I have a few questions:

Is * a proper tool to provide a SIP-MGCP gateway?  Am I even asking 
for something that makes sense?
If so, where's the best from the ground up, assume I don't know the 
lingo guidance on how to get where I want to go?  Like many open 
source projects, the documentation I've seen so far seems to assume the 
reader is not a newbie in this field.  I'm hoping there's introductory 
documentation someone can point me towards.  Asterisk 101, if you will.

-You are asking the wrong question here.  Don't think of designing your *
solution around the limitations of the Inter-Tel.  Besides, if you currently
have * up and running (in any capacity) you can make * do what you want it
to.  Your problem in the end will be making the Axxess work with *.

I don't know what version of Axxess you are running but the very recently
released version 9.0 supports SIP trunking (I think) via the IPRC card).  It
will require a couple of licenses but I think it is a better idea then doing
MGCP-SIP stuff.  I haven't played with MGCP in * but the Inter-Tel uses an
OEM'd AudioCodes box as an FXO gateway and I have had nothing but problems
with them.  Can't tell if it is gateway related or Axxess related but I
ended up putting a PRI into the Axxess and connecting to * via that PRI,
then doing all of my IP stuff (via SIP) in *.

I have access to an axxess for testing so I will play with it a bit and see
if I can figure out a better way.

Jason Kawakami
www.optellabs.com
Salt Lake City, UT  


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RE: [Asterisk-Users] Installing * on fedora 3

[David Ishmael]

I run * on FC3 and I have a /var/lib/asterisk/keys directory.  Did the make
of the * software have any errors.

-Dave

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ferguson,
Michael
Sent: Tuesday, January 11, 2005 9:26 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Installing * on fedora 3

G'Day List,

Can someone help me out a bit please.
I just installed Fedora Core 3 on a Dell Power Edge 400 SC and am trying
to install *
I am following
http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation
After running:
   cd /usr/src/asterisk 
   make clean 
   make 
   make install 
   make samples
The instructions says:
Configuring Asterisk 

- Login to your server as user root 
- Right-click on the background and select Open Terminal 
- Run the following commands to download the VoicePulse Connect! public
key (needed for receiving calls): 

   cd /var/lib/asterisk/keys 
   wget http://connect.voicepulse.com/keys/voicepulse01.pub 

However there is NO /var/lib/asterisk/keys directory.
HELP!!

Thanks



Michael E. Ferguson
Manager, Information Systems
Berman Rennert Vogel  Mandler, P.A.
100 SE 2nd., Street, Suite 2900
Miami, FL., 33131
305.423.3408 Direct
305.533.1582 Fax
[EMAIL PROTECTED] 
This message is for the named person's use only. It may contain
confidential, proprietary or legally privileged information. No
confidentiality or privilege is waived or lost by any mistransmission.
If you receive this message in error, please immediately delete it and
all copies of it from your system, destroy any hard copies of it and
notify the sender. You must not, directly or indirectly, use, disclose,
distribute, print, or copy any part of this message if you are not the
intended recipient. BERMAN RENNERT VOGEL  MANDLER, P.A. reserve the
right to monitor all e-mail communications through its networks. Any
views expressed in this message are those of the individual sender,
except where the message states otherwise and the sender is authorized
to state them to be the views of any such entity. 
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[Asterisk-Users] Newbie question: call routing

[Andreas Pelzner]

Hello,

is it possible to route a phone call by Asterisk to a Skype user? 

Scenario:
-

Incoming phone call
 |
My telephone system
---
| | 
---
 |
Internal call routing 
to extension # with a modem 
connected to Asterisk Linux Box
 |
---
| | 
---
   |
Asterisk routes call 
to Skype User   

If so, does anyone has configuration samples and best practice experience?

Thanks.

Andreas Pelzner, AixVision GmbH

°°°
Andreas Pelzner| Wasserburg Haus Heyden  | www.aixvision.net
Geschäftsführer| Heyder Feldweg 50   | [EMAIL PROTECTED]
AixVision Gesellschaft | 52072 Aachen| Tel: +49024075684970
fuer Neue Medien mbH   | Germany | Fax: +49024075684972
°°°
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RE: [Asterisk-Users] Installing * on fedora 3

[David Ishmael]

Not sure if this helps, but here's the instructions I followed for setting
up * on FC3:

http://www.automated.it/guidetoasterisk.htm

See if that helps, perhaps there's a step you missed along the way.

-Dave

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ferguson,
Michael
Sent: Tuesday, January 11, 2005 9:26 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Installing * on fedora 3

G'Day List,

Can someone help me out a bit please.
I just installed Fedora Core 3 on a Dell Power Edge 400 SC and am trying
to install *
I am following
http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation
After running:
   cd /usr/src/asterisk 
   make clean 
   make 
   make install 
   make samples
The instructions says:
Configuring Asterisk 

- Login to your server as user root 
- Right-click on the background and select Open Terminal 
- Run the following commands to download the VoicePulse Connect! public
key (needed for receiving calls): 

   cd /var/lib/asterisk/keys 
   wget http://connect.voicepulse.com/keys/voicepulse01.pub 

However there is NO /var/lib/asterisk/keys directory.
HELP!!

Thanks



Michael E. Ferguson
Manager, Information Systems
Berman Rennert Vogel  Mandler, P.A.
100 SE 2nd., Street, Suite 2900
Miami, FL., 33131
305.423.3408 Direct
305.533.1582 Fax
[EMAIL PROTECTED] 
This message is for the named person's use only. It may contain
confidential, proprietary or legally privileged information. No
confidentiality or privilege is waived or lost by any mistransmission.
If you receive this message in error, please immediately delete it and
all copies of it from your system, destroy any hard copies of it and
notify the sender. You must not, directly or indirectly, use, disclose,
distribute, print, or copy any part of this message if you are not the
intended recipient. BERMAN RENNERT VOGEL  MANDLER, P.A. reserve the
right to monitor all e-mail communications through its networks. Any
views expressed in this message are those of the individual sender,
except where the message states otherwise and the sender is authorized
to state them to be the views of any such entity. 
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RE: [Asterisk-Users] Installing * on fedora 3

[Ferguson, Michael]

I did see an Errors 1.
I will go back and check. Thanks

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
Ishmael
Sent: Tuesday, January 11, 2005 10:34 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Installing * on fedora 3


I run * on FC3 and I have a /var/lib/asterisk/keys directory.  Did the
make of the * software have any errors.

-Dave

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ferguson,
Michael
Sent: Tuesday, January 11, 2005 9:26 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Installing * on fedora 3

G'Day List,

Can someone help me out a bit please.
I just installed Fedora Core 3 on a Dell Power Edge 400 SC and am trying
to install * I am following
http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation
After running:
   cd /usr/src/asterisk 
   make clean 
   make 
   make install 
   make samples
The instructions says:
Configuring Asterisk 

- Login to your server as user root 
- Right-click on the background and select Open Terminal 
- Run the following commands to download the VoicePulse Connect! public
key (needed for receiving calls): 

   cd /var/lib/asterisk/keys 
   wget http://connect.voicepulse.com/keys/voicepulse01.pub 

However there is NO /var/lib/asterisk/keys directory.
HELP!!

Thanks



Michael E. Ferguson
Manager, Information Systems
Berman Rennert Vogel  Mandler, P.A.
100 SE 2nd., Street, Suite 2900
Miami, FL., 33131
305.423.3408 Direct
305.533.1582 Fax
[EMAIL PROTECTED] 
This message is for the named person's use only. It may contain
confidential, proprietary or legally privileged information. No
confidentiality or privilege is waived or lost by any mistransmission.
If you receive this message in error, please immediately delete it and
all copies of it from your system, destroy any hard copies of it and
notify the sender. You must not, directly or indirectly, use, disclose,
distribute, print, or copy any part of this message if you are not the
intended recipient. BERMAN RENNERT VOGEL  MANDLER, P.A. reserve the
right to monitor all e-mail communications through its networks. Any
views expressed in this message are those of the individual sender,
except where the message states otherwise and the sender is authorized
to state them to be the views of any such entity. 
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[Asterisk-Users] BroadVoice

[Vitalie Apostu]

Did somebody connect Asterisk to BroadVoice provider? If so, can you share
instruction with me?

Thanks.

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Re: [Asterisk-Users] BroadVoice

[skamp]

Broadvoice has instructions on their site on how to configure asterisk
with their service, and it works i use broadvoice with asterisk 

On Tue, 2005-01-11 at 10:43 -0500, Vitalie Apostu wrote:
 Did somebody connect Asterisk to BroadVoice provider? If so, can you share
 instruction with me?
 
 Thanks.
 
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-- 
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Re: [Asterisk-Users] Unicall errors

[Sam Njenga]

Hi Steve

Thanks. I got past the errors by specifying the prefix. I have now a new
error

loader.c:248 ast_load_resource: /usr/lib/asterisk/modules/chan_unicall.so:
undefined symbol: get_supervisory_tone_set

Jan 11 18:52:46 WARNING[1076216448]: loader.c:380 load_modules: Loading
module chan_unicall.so failed!

Sam Njenga



- Original Message - 
From: Steve Underwood [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, January 11, 2005 3:17 AM
Subject: Re: [Asterisk-Users] Unicall errors


 Hi Sam,

 Did you build libunicall with

 ./configure
 make
 make install

 If so, the library will be in /ustr/local/lib. Is this in your search
 path? Wither add this directory to /etc/ld.so.conf, or build with:

 ./configure --prefix=/usr
 make
 make install

 This is an issue common to most packages which use ./configure to set
 them up.

 Regards,
 Steve


 Sam Njenga wrote:

 Hi Steve
 
 I have compiled everything now without errors. Problem is loading the
 unicall module when starting asterisk. This is the error ..
 
 [chan_unicall.so]Jan 10 18:37:16 WARNING[1076216448]: loader.c:248
 ast_load_resource: /usr/lib/asterisk/modules/chan_unicall.so: undefined
 symbol: uc_channel_write
 Jan 10 18:37:16 WARNING[1076216448]: loader.c:380 load_modules: Loading
 module chan_unicall.so failed!
 [EMAIL PROTECTED] asterisk]#
 
 Sam Njenga
 
 
 
 
 - Original Message - 
 From: Steve Underwood [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Monday, January 10, 2005 6:05 PM
 Subject: Re: [Asterisk-Users] Unicall errors
 
 
 
 
 Hi Sam,
 
 Sorry about that. The copy of libsupertone on the FTP site appeared to
 be faulty. I have just replaced it. Please try again.
 
 Strange. You should have had this same problem when testing 0.0.2pre1.
 
 Regards,
 Steve
 
 Sam Njenga wrote:
 
 
 
 Steve
 
 I have stated below that I tried to install Libsupertone and gote
errors.
 
 
 
 
 
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RE: [Asterisk-Users] BroadVoice

[Vitalie Apostu]

Can you give me example of sip.conf and extention.conf which work with
broadvoice? I want users who registered with Messenger through sip to be
able to make a call thought broadvoice.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of skamp
Sent: Tuesday, January 11, 2005 10:55 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users] BroadVoice

Broadvoice has instructions on their site on how to configure asterisk with
their service, and it works i use broadvoice with asterisk 

On Tue, 2005-01-11 at 10:43 -0500, Vitalie Apostu wrote:
 Did somebody connect Asterisk to BroadVoice provider? If so, can you 
 share instruction with me?
 
 Thanks.
 
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--
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RE: [Asterisk-Users] BroadVoice

[Nabeel Jafferali]

 Can you give me example of sip.conf and extention.conf which
 work with broadvoice? I want users who registered with
 Messenger through sip to be able to make a call thought broadvoice.

I posted this just a few days ago:
http://lists.digium.com/pipermail/asterisk-users/2005-January/081534.htm
l

-- 
Nabeel Jafferali
Tel: +1 (416) 628-9342  Toronto
 +1 (646) 225-7426  New York
FWD: 46990
Email/MSN: nabeelatjafferali.net
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RE: [Asterisk-Users] Asterisk to PSTN

[Walid Azab]

Thanks.

Any 
tips on a dial plan example to route from Asterisk to CCM and vice 
versa?

Also 
with H323 between * and CCM can I still use SIP phones behind 
Asterisk.

ThanksWalid


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Jo?o 
AmaroSent: Monday, January 10, 2005 5:04 PMTo: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: Re: 
[Asterisk-Users] Asterisk to PSTN
-BEGIN PGP SIGNED MESSAGE-Hash: 
SHA1HelloYou can use H323 to connect to Cisco 
CallManager.Add asterisk as an h323 gateway on cisco callmanager.Then 
you can send  receive call from asterisk.TIP: Use OH323 instead off 
asterisk h323 native driver.RegardsJoão 
AmaroWalid Azab wrote:| I have installed [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] on a PC here| 
and need to have it forward calls to the PSTN. We have Cisco| CallManager 
3.3.4. However I found out that this version doesn't| support configuring 
SIP Trunks. Is there an alternative solution.| Thanks|| 
Walid||| 
--||| 
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RE: [Asterisk-Users] Ring Voltage Supplied by Wildcard TDM400P REV E/F AUTO FXS/DPO

[Jim Van Meggelen]

[EMAIL PROTECTED] wrote:
 Hi;
 
 I'm trying to connect a TDM400P with an FXS module to a Valcom V-9940
 Paging adaptor. This port on the TDM400P was connected to a 2500 Set
 and was working I just re-connected it to the Valcom (which
 is known to
 work on a Telco POTS line) and its not picking up.  The
 Valcom docs say
 it need a minimum of 75 Volts at 20-30 Hz to recognize a
 call... So the
 question is what ring voltage does the FXS modules on a TDM400P put
 out? 

Ultimately, that depends on how much current is being drawn, but my
multimeter reports about 70V on a 6 foot line cord. Generally, I'd
expect an FXS to put out more like 90-110V (that's what my TalkSwitch
supplies).





-- 
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 265.6.10 - Release Date: 10/01/2005
 

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[Asterisk-Users] Cisco ATA 186 for PSTN dialing

[Walid Azab]

Hi all.. can I 
configure Cisco ATA 186 to dial out to PSTN? I need a quick and easy to set up 
scenario to have SIP phones dial PSTN numbers.

Thanks

Walid

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RE: [Asterisk-Users] BroadVoice

[Vitalie Apostu]

Following links says: HTTP 404 - File not found . Is it a right link
http://lists.digium.com/pipermail/asterisk-users/2005-January/081534.htm ?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nabeel
Jafferali
Sent: Tuesday, January 11, 2005 11:09 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users] BroadVoice

 Can you give me example of sip.conf and extention.conf which work with 
 broadvoice? I want users who registered with Messenger through sip to 
 be able to make a call thought broadvoice.

I posted this just a few days ago:
http://lists.digium.com/pipermail/asterisk-users/2005-January/081534.htm
l

--
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Tel: +1 (416) 628-9342  Toronto
 +1 (646) 225-7426  New York
FWD: 46990
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Re: [Asterisk-Users] Analogue RAS Server

[Peter Svensson]

On Tue, 11 Jan 2005, Paradise Dove wrote:

   I don't think it's possible. Asterisk would have to emulate analog modem,
 
 does anybody know if  there ia any works on emulating analog modems
 (not specially to work with asterisk).
 something like Steve's spandsp for fax.

There are a few projects, none completed. One of the more complete is 

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


Peter


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[Asterisk-Users] RE: Asterisk-Users Digest, Vol 6, Issue 142

[Luis Mata]

 root root 2310 Nov 17 09:41 msg0002.gsm
-rw-r--r-- 1 root root 216 Nov 17 09:41 msg0002.txt
-rwx-- 1 root root 22444 Nov 17 09:41 msg0002.wav
-rwx-- 1 root root 2336 Nov 17 09:41 msg0002.WAV
-rwx-- 1 root root 20460 Nov 18 11:48 msg0003.gsm
-rw-r--r-- 1 root root 217 Nov 18 11:48 msg0003.txt
-rwx-- 1 root root 198444 Nov 18 11:48 msg0003.wav
-rwx-- 1 root root 20210 Nov 18 11:48 msg0003.WAV

they are not readable by the web process. Anyway I have not fixed it
yet, so please let me know if you do.

Mike



On Mon, 10 Jan 2005 08:00:13 -0800 (PST), Frank Kostin
wrote:
 Hello everybody,
 I was trying to install a web interface to my Voice Mail, Vmail.cgi
 I can log on it, list messages, but no play with the following error msg; 
 
 Hrm, can't seem to open
 /var/spool/asterisk/voicemail/default/234/INBOX/msg0001.WAV 
 
 Remark: playing the message msg0001.WAV directly OK 
 Any smart guy up there could help ?
 Thanks,
 
 
 Do you Yahoo!?
 Read only the mail you want - Yahoo! Mail SpamGuard. 
 
 
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-
Do you Yahoo!?
 Yahoo! Mail - 250MB free storage. Do more. Manage less.
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Message: 4
Date: Tue, 11 Jan 2005 13:32:58 +
From: Niksa Baldun [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Analogue RAS Server
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=iso-8859-1

I don't think it's possible. Asterisk would have to emulate analog
modem, and I believe that feature is not (at least yet) implemented.

Daniel Niasoff wrote:

 Hi,

  

 Does anyone have any idea how to set up Asterisk so that it can act as
 an Analogue Remote Access Server. I've looked around and as far as I
 can see it will only act as an ISDN Ras server.

  

 Thanks

  

 Daniel



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Message: 5
Date: Tue, 11 Jan 2005 16:38:25 +0330
From: Paradise Dove [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Analogue RAS Server
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=US-ASCII

  I don't think it's possible. Asterisk would have to emulate analog modem,

does anybody know if  there ia any works on emulating analog modems
(not specially to work with asterisk).
something like Steve's spandsp for fax.


--

Message: 6
Date: Tue, 11 Jan 2005 14:10:11 +0100
From: ismaelg [EMAIL PROTECTED]
Subject: [Asterisk-Users] Zaptel config
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Hello all,

I am having a lot of problems with zaptel channels,
I have got an TDM02B, and I don't know how setup /etc/zaptel.con and 
/etc/asterisk/zapata.conf for use it on asterisk.

Some one could help me with this configuracisn?
My problem is about the type of signalling

Thanks,

Regards.

Ismael Gil.



--

Message: 7
Date: Tue, 11 Jan 2005 08:13:03 -0500
From: James Freire [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Zhone channel bank issues
To: [EMAIL PROTECTED],  Asterisk Users Mailing List - Non-Commercial
Discussion  asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=US-ASCII

Hi Michael,
You might want to check the voltage settings on the FXS side of
things. Also, are you using the correct signalling? (ground start,
loop start, etc.)
In the Zplex users guide, on page 41 you will see 2 sections on TTLP
and RTLP. That might be of some help to you.

Hey... You have caller ID working

Re: [Asterisk-Users] How to mark a user for a conference

[Peter Svensson]

On Tue, 11 Jan 2005, Jagan Mohan wrote:

 Hi All,
 
I would like to mark a user so that all users other than marked
 user hear music-on-hold till the marked user joins the conference.
I took a look at 
 http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe, but could not get
 sufficient info.
 I'm using meetme for conferencing.
Could anyone point me to a url which has the configuration details
 using meetme.

The easiest way to find out the options to MeetMe (which you will find is 
whay you need in this case) is to run show application meetme in the 
asterisk console. The options you are interested in are probably 'w' and 
'A'.

Peter

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Re: [Asterisk-Users] Request to schedule in the past?!?!

[Andrei (MPI)]

Michael Greb wrote:
On Mon, Jan 10, 2005 at 03:26:04PM -, Paul Brock wrote:
 

On Mon, Jan 10, 2005 at 15:18, Paradise Dove said:
   

On Mon, 10 Jan 2005 06:45:54 -0800 (PST), Jason Goecke
[EMAIL PROTECTED] wrote:
 

Hello,
Ever since I started using Asterisk I always get this
error:
Jan 10 15:39:26 NOTICE[4501]: res_musiconhold.c:463
monmp3thread: Request to schedule in the past?!?!
I have a dedicated system system that really runs only
Asterisk:
- Pentium III 500Mhz
- 128MB of RAM
- 10GB of Disk Space
   

it's clear that your processor is overloaded. 
recommend you to use rawplayer instead of mpg123 for moh
by converting your mp3 files to raw using sox (with mp3 support)
take a look at cvs head.
 

I would disagree, purely because I'm getting the same message on an xp2100,
with just OS and asterisk running - and that's with approx 98% free time
Paul
   

And I on a dual xenon and have received the message since installation
with 0.01 load average so clearly you know not what you speak of.
 

Sync up your clock, guys: on the * server, PCs and the phones. And the 
problem will go away.

Andrei
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RE: [Asterisk-Users] Ring Voltage Supplied by Wildcard TDM400P REV E/F AUTO FXS/DPO

[Peter Svensson]

On Tue, 11 Jan 2005, Jim Van Meggelen wrote:

 [EMAIL PROTECTED] wrote:
  I'm trying to connect a TDM400P with an FXS module to a Valcom V-9940
  Paging adaptor. This port on the TDM400P was connected to a 2500 Set
  and was working I just re-connected it to the Valcom (which
  is known to
  work on a Telco POTS line) and its not picking up.  The
  Valcom docs say
  it need a minimum of 75 Volts at 20-30 Hz to recognize a
  call... So the
  question is what ring voltage does the FXS modules on a TDM400P put
  out? 
 
 Ultimately, that depends on how much current is being drawn, but my
 multimeter reports about 70V on a 6 foot line cord. Generally, I'd
 expect an FXS to put out more like 90-110V (that's what my TalkSwitch
 supplies).

The ring voltage can be configured. Try setting the module parameter 
boostringer when the wctdm module is insmod / modprobed.

Peter

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RE: [Asterisk-Users] BroadVoice

[Paul Brock]

Add an 'l' on the end of the link... i.e. 081534.html

Then it'll work :)

Paul

-Original Message-
From: [EMAIL PROTECTED]
[mailto:asterisk-users-[EMAIL PROTECTED] On Behalf Of Vitalie
Apostu
Sent: 11 January 2005 16:12
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] BroadVoice

Following links says: HTTP 404 - File not found . Is it a right link
http://lists.digium.com/pipermail/asterisk-users/2005-January/081534.htm ?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nabeel
Jafferali
Sent: Tuesday, January 11, 2005 11:09 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users] BroadVoice

 Can you give me example of sip.conf and extention.conf which work with 
 broadvoice? I want users who registered with Messenger through sip to 
 be able to make a call thought broadvoice.

I posted this just a few days ago:
http://lists.digium.com/pipermail/asterisk-users/2005-January/081534.htm
l

--
Nabeel Jafferali
Tel: +1 (416) 628-9342  Toronto
 +1 (646) 225-7426  New York
FWD: 46990
Email/MSN: nabeelatjafferali.net
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RE: [Asterisk-Users] BroadVoice

[David Ishmael]

I got the same error.

-Dave

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vitalie Apostu
Sent: Tuesday, January 11, 2005 10:12 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] BroadVoice

Following links says: HTTP 404 - File not found . Is it a right link
http://lists.digium.com/pipermail/asterisk-users/2005-January/081534.htm ?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nabeel
Jafferali
Sent: Tuesday, January 11, 2005 11:09 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users] BroadVoice

 Can you give me example of sip.conf and extention.conf which work with 
 broadvoice? I want users who registered with Messenger through sip to 
 be able to make a call thought broadvoice.

I posted this just a few days ago:
http://lists.digium.com/pipermail/asterisk-users/2005-January/081534.htm
l

--
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Tel: +1 (416) 628-9342  Toronto
 +1 (646) 225-7426  New York
FWD: 46990
Email/MSN: nabeelatjafferali.net
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RE: [Asterisk-Users] requiring logon for SIP users

[Race Vanderdecken]

Greeting Florian,

You can do it a couple of ways.

Under the SIP config put md5secret= or secret= flags.

This will require the phone itself to answer and give authorization
information. You will need access to the phone to set the memory in the
phone to the correct answers.

You can also use RADIUS to check that the Originate and the Answer leg
DID's are allowed.

You can create dialplans that look that the dialed numbers and the
caller and make a descision.

All depends on your call volume and your needs as to how deep you do the
work.


Race The Tyrant Vanderdecken



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Florian
Effenberger
Sent: 11 January 2005 05:36
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] requiring logon for SIP users

Hello there,

I am playing around with Asterisk the first time and it really looks 
great. ;-)

However, I have one problem: Any SIP device can connect to my PBX. How 
can I requre logon for SIP users and deny access in the case of wrong or

missing credentials?

Thanks
Florian
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RE: [Asterisk-Users] BroadVoice

[Paul Rodan]

Curious, has anybody ever been charged the 3 cents a minute or whatever when
they've had more than 1 or 2 simul calls? 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nabeel
Jafferali
Sent: Tuesday, January 11, 2005 11:09 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users] BroadVoice

 Can you give me example of sip.conf and extention.conf which
 work with broadvoice? I want users who registered with
 Messenger through sip to be able to make a call thought broadvoice.

I posted this just a few days ago:
http://lists.digium.com/pipermail/asterisk-users/2005-January/081534.htm
l

-- 
Nabeel Jafferali
Tel: +1 (416) 628-9342  Toronto
 +1 (646) 225-7426  New York
FWD: 46990
Email/MSN: nabeelatjafferali.net
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RE: [Asterisk-Users] sip to h.323

[Kanuri, Seshu (Company IT)]

http://www.voip-info.org/tiki-index.php?page=Asterisk 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of sai latha
Sent: Tuesday, January 11, 2005 9:27 AM
To: asterisk-users@lists.digium.com
Cc: [EMAIL PROTECTED]; [EMAIL PROTECTED]
Subject: [Asterisk-Users] sip to h.323

Hello,
 Happy New Year

where u r downloaded the asterisk server please tell me.Iam
searching the asterisk server site in google but i dint get this server
u please tell me the site for me 
Is only for sip to sip or sip to h.323 please tell me
 
Thank u
Bye
Sailatha
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NOTICE: If received in error, please destroy and notify sender.  Sender does 
not waive confidentiality or privilege, and use is prohibited. 
 
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[Asterisk-Users] howto dump binary data on zap channel?

[Klaus Darilion]

Hi!
I'm using a PRI card. When a call arrives, I want to answer the call and 
dump the binary data received on the B-channel into a file or stdout or 
to the console (for debugging the B-Channels).

Is this possible?
regards,
klaus
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RE: [Asterisk-Users] BroadVoice

[Greg Hill]

Did you try BroadVoice's site yet? www.broadvoice.com, click Support,
click Installation, click Asterisk, follow instructions there.

Greg


On Tue, 11 Jan 2005, Vitalie Apostu wrote:

 Following links says: HTTP 404 - File not found . Is it a right link
 http://lists.digium.com/pipermail/asterisk-users/2005-January/081534.htm ?

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Nabeel
 Jafferali
 Sent: Tuesday, January 11, 2005 11:09 AM
 To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
 Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] BroadVoice

  Can you give me example of sip.conf and extention.conf which work with
  broadvoice? I want users who registered with Messenger through sip to
  be able to make a call thought broadvoice.

 I posted this just a few days ago:
 http://lists.digium.com/pipermail/asterisk-users/2005-January/081534.htm
 l

 --
 Nabeel Jafferali
 Tel: +1 (416) 628-9342  Toronto
  +1 (646) 225-7426  New York
 FWD: 46990
 Email/MSN: nabeelatjafferali.net
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Re: [Asterisk-Users] Generic modem question

[Andrew Kohlsmith]

On January 10, 2005 07:50 pm, Henry Devito wrote:
 Does asterisk support the intel 537/md3200 chipset?  I don't want to start
 any flames here, I know all about using generic crap in asterisk,[*]  which
 I really don't approve of other than for testing, but I have a customer
 demanding a generic chipset for his one backup analog line.  He will not
 spend the money for a Digium card and says he will find another
 company if I can not provide a generic FXO port.

Give him a Sipura then or find another customer.  It sounds like he doesn't 
have Clue One about what he really wants.

-A.
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[Asterisk-Users] Re: Asterisk-Users Digest, Vol 6, Issue 144

[John Voss]

I am running on Core 3 also with a voicepulse account.
I found this document quite helpfulwww.voip-info.org/tiki-print.php?page=Asterisk+Fedora+Core+3
I did deviate in that I ran my make of Asterisk itself as follows
cd /usr/src/asterisk
make clean make linux26 make install make samples
Hope it helps
JV- Original Message -  Message: 1  Date: Tue, 11 Jan 2005 10:41:42 -0500  From: "Ferguson, Michael" <[EMAIL PROTECTED]> Subject: RE: [Asterisk-Users] Installing * on fedora 3  To: <[EMAIL PROTECTED]>, "Asterisk Users Mailing List -  Non-Commercial Discussion"  Message-ID: [EMAIL PROTECTED]  Content-Type: text/plain; charset="us-ascii"   I did see an Errors 1.  I will go back and check. Thanks   -Original Message-  From: [EMAIL PROTECTED]  [mailto:[EMAIL PROTECTED] On Behalf Of David  Ishmael  Sent: Tuesday, January 11, 2005 10:34 AM  To: 'Asterisk Users Mailing List - Non-Commercial Discussion'  Subject: RE: [Asterisk-Users] Installing * on fedora 3I run * on FC3 and I have a /var/lib/asterisk/keys directory. Did the  make of the * software have any errors.   -Dave   -Original Message-  From: [EMAIL PROTECTED]  [mailto:[EMAIL PROTECTED] On Behalf Of Ferguson,  Michael  Sent: Tuesday, January 11, 2005 9:26 AM  To: asterisk-users@lists.digium.com  Subject: [Asterisk-Users] Installing * on fedora 3   G'Day List,   Can someone help me out a bit please.  I just installed Fedora Core 3 on a Dell Power Edge 400 SC and am trying  to install * I am following  http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation  After running:  cd /usr/src/asterisk  make clean  make  make install  make samples  The instructions says:  Configuring Asterisk   - Login to your server as user "root"  - Right-click on the background and select Open Terminal  - Run the following commands to download the VoicePulse Connect! public  key (needed for receiving calls):   cd /var/lib/asterisk/keys  wget http://connect.voicepulse.com/keys/voicepulse01.pub   However there is NO /var/lib/asterisk/keys directory.  HELP!!   Thanks Michael E. Ferguson  Manager, Information Systems  Berman Rennert Vogel  Mandler, P.A.  100 SE 2nd., Street, Suite 2900  Miami, FL., 33131  305.423.3408 Direct  305.533.1582 Fax  [EMAIL PROTECTED]  This message is for the named person's use only. It may contain  confidential, proprietary or legally privileged information. No  confidentiality or privilege is waived or lost by any mistransmission.  If you receive this message in error, please immediately delete it and  all copies of it from your system, destroy any hard copies of it and  notify the sender. You must not, directly or indirectly, use, disclose,  distribute, print, or copy any part of this message if you are not the  intended recipient. BERMAN RENNERT VOGEL  MANDLER, P.A. reserve the  right to monitor all e-mail communications through its networks. Any  views expressed in this message are those of the individual sender,  except where the message states otherwise and the sender is authorized  to state them to be the views of any such entity. 
-- 
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RE: [Asterisk-Users] BroadVoice

[skamp]

not that ive ever seen occur :)

On Tue, 2005-01-11 at 11:28 -0500, Paul Rodan wrote:
 Curious, has anybody ever been charged the 3 cents a minute or whatever when
 they've had more than 1 or 2 simul calls? 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Nabeel
 Jafferali
 Sent: Tuesday, January 11, 2005 11:09 AM
 To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
 Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] BroadVoice
 
  Can you give me example of sip.conf and extention.conf which
  work with broadvoice? I want users who registered with
  Messenger through sip to be able to make a call thought broadvoice.
 
 I posted this just a few days ago:
 http://lists.digium.com/pipermail/asterisk-users/2005-January/081534.htm
 l
 
-- 
skamp [EMAIL PROTECTED]

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RE: [Asterisk-Users] BroadVoice

[skamp]

Guys
http://lists.digium.com/pipermail/asterisk-users/2005-January/081534.html


On Tue, 2005-01-11 at 10:21 -0600, David Ishmael wrote:
 I got the same error.
 
 -Dave
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Vitalie Apostu
 Sent: Tuesday, January 11, 2005 10:12 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] BroadVoice
 
 Following links says: HTTP 404 - File not found . Is it a right link
 http://lists.digium.com/pipermail/asterisk-users/2005-January/081534.htm ?
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Nabeel
 Jafferali
 Sent: Tuesday, January 11, 2005 11:09 AM
 To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
 Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] BroadVoice
 
  Can you give me example of sip.conf and extention.conf which work with 
  broadvoice? I want users who registered with Messenger through sip to 
  be able to make a call thought broadvoice.
 
 I posted this just a few days ago:
 http://lists.digium.com/pipermail/asterisk-users/2005-January/081534.htm
 l
 
 --
 Nabeel Jafferali
 Tel: +1 (416) 628-9342  Toronto
  +1 (646) 225-7426  New York
 FWD: 46990
 Email/MSN: nabeelatjafferali.net
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[Asterisk-Users] operator says that dial 1 or 0

[hak atil]

Hi,

I realized that I have a problem with my asterisk. I am using 1 TDM400P 4
FXS card from digium. My problem is I can't make calls one after another.
When I dial long distance number from my SIP phone, my phone company's
operator says that dial 1 or 0, then hangs up. After a couple of seconds I
dial the same phone number, this time it works with no problem. Weird part
is that Asterisk is working properly; CLI does not display any errors when I
got the operator. I have to dial a few times on certain phone numbers. 


Is there anything I can change on the configuration?

HAKAN


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[Asterisk-Users] Unicall errors

[Sam Njenga]

 Hi Steve

 Thanks. I got past the errors by specifying the prefix. I have now a new
 error

 loader.c:248 ast_load_resource: /usr/lib/asterisk/modules/chan_unicall.so:
 undefined symbol: get_supervisory_tone_set

 Jan 11 18:52:46 WARNING[1076216448]: loader.c:380 load_modules: Loading
 module chan_unicall.so failed!

 Sam Njenga



 - Original Message - 
 From: Steve Underwood [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Tuesday, January 11, 2005 3:17 AM
 Subject: Re: [Asterisk-Users] Unicall errors


  Hi Sam,
 
  Did you build libunicall with
 
  ./configure
  make
  make install
 
  If so, the library will be in /ustr/local/lib. Is this in your search
  path? Wither add this directory to /etc/ld.so.conf, or build with:
 
  ./configure --prefix=/usr
  make
  make install
 
  This is an issue common to most packages which use ./configure to set
  them up.
 
  Regards,
  Steve
 
 
  Sam Njenga wrote:
 
  Hi Steve
  
  I have compiled everything now without errors. Problem is loading the
  unicall module when starting asterisk. This is the error ..
  
  [chan_unicall.so]Jan 10 18:37:16 WARNING[1076216448]: loader.c:248
  ast_load_resource: /usr/lib/asterisk/modules/chan_unicall.so: undefined
  symbol: uc_channel_write
  Jan 10 18:37:16 WARNING[1076216448]: loader.c:380 load_modules: Loading
  module chan_unicall.so failed!
  [EMAIL PROTECTED] asterisk]#
  
  Sam Njenga
  
  
  
  
  - Original Message - 
  From: Steve Underwood [EMAIL PROTECTED]
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  asterisk-users@lists.digium.com
  Sent: Monday, January 10, 2005 6:05 PM
  Subject: Re: [Asterisk-Users] Unicall errors
  
  
  
  
  Hi Sam,
  
  Sorry about that. The copy of libsupertone on the FTP site appeared to
  be faulty. I have just replaced it. Please try again.
  
  Strange. You should have had this same problem when testing 0.0.2pre1.
  
  Regards,
  Steve
  
  Sam Njenga wrote:
  
  
  
  Steve
  
  I have stated below that I tried to install Libsupertone and gote
 errors.
  
  
  
  
  
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[Asterisk-Users] No ring tone while calling from h323 to SIP

[Jalil BOUREKBA]

I set up asterisk since a while, I had no problem using it, but now I
just added H323 module, it´s working fine, but when I place calls from
H323 to pstn via Asterisk ; idon´t hear the ring tone, it goes silent
until the other part pickup the phone and we can talk.

Could someone help me please.

Thank you and good luck to everybody.
Jalil.

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[Asterisk-Users] RealTime Configuration Doubts

[Oswaldo Arratia]

 Hi there,
I've been running * for some time now and thanks God no problem so far,
everything is configured using text files, and I'd like to move everything
to realtime database configuration to ease management using a GUI
application.

I've read about Realtime function of * and I see something that confuses me
and it's the following (text taken from WIKI): The database peers/users are
not kept in memory. These are only loaded when we have a call and then
deleted, so there's no support for NAT keep-alives (qualify=) or voicemail
indications for these peers.

Does this mean that current users that get a different dialtone when they
pick up their extension and  have a new voicemail won't get that indication
anymore??   Another doubt, my * server has a public IP and most of the
extensions are behing different NATs, so I use the nat=yes option in
sip.conf for each context...if I move my setup to realtime database, can I
still do the nat support? I am not sure what that NAT keep-alive does and
exactly what WIKI means by no support for NAT keep-alives.

I'd like to know if any person out there is using this type of setup and how
good it works. I have not been able to find any comments to this matter in
my searches.

Many thanks in advance to anyone who shares their experience and clarify
these doubts.

Oz


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RE : [Asterisk-Users] sound problem

[Jalil BOUREKBA]

Check the used codecs, if it´s according to the codec used by your sip
users.

Welcome

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Muhammad
Rizwan Khan
Envoyé : lundi 10 janvier 2005 18:53
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [Asterisk-Users] sound problem


Is there any config file related to voice, which should be change in
order to 
hear the sound in dialer?

On Monday 10 January 2005 21:23, you wrote:
 I have configured asterisk, but when i calls from my dialler, it
connects
 successfully, but did not give any voice at both ends.
 What should i need to do?

 Thanks

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[Asterisk-Users] ACD Queues Agent Status

[Ronald Hartmann]

Good Day List,

I am finalizing my research on the ACD Ques and have (what I
hope to be) one last hurdle.

Is there any feasible way to determine if a que has agents
currently available to take a call.

I have looked at the Show Queues, show queue quename, show
agents and they all give me pieces, 
However, for example lets say that Agent-1, and Agent-2 belong
to Group1.

When I perform show queue myqueue it only shows me that
Agent/@1 is logged in... it does not tell me
Which agents are in that group, nor does it tell me if they are
accepting calls.

Further, is there a way to determine if an agent is in the
WrapUptime Mode?

Any url links would be helpful.

Ultimately what I am looking to do is as follows.

In my dialplan, I want to be able to check for agent
availability, and then play a your being placed in que, 
If there are no agents immediately available to take a call.
Then pass the call to the Queue() application.
However, if there is an agent immediately available to take a
call I will not play a file.  

Example:

[que-test]
exten = s,1,Answer
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout,5
exten = s,4,ResponseTimeout,10
exten = s,5,   IF AGENTS ARE AVAILBLE TO TAKE CALL
GOTO6 ELSE GOTO5
exten = s,5,Background(your call is being placed in
the Support Que please hold for an agent)
exten = s,6,Queue(techsupport)
exten = s,7,Playback(All Agents are busy, please leave
your name and callback number)
exten = s,8,VoiceMail(SUPPORT_MAILBOX) ; 

Final Thought, 

Is it possible to Set the Que up such that it plays a special
recording every X interval.

Currently it will play musiconhold,
Then at specific intervals it can tell the caller where they are
in the que and hold times.

I would like to be able to Also at certain intervals play a
recording stating.

If you would like to leave a call back number please press X
now.

Etc.

Thanks for your time in reading this note. I am sure that as soon as
I send this it will turn into a RTFM dummy

But I have googled and wikied my fingers to the bone and now require
some outside direction.

Ron.


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[Asterisk-Users] Is this good packet latency/jitter ? (ping resultsfor BabyTel...)

[Kim Lux]

I'm about to order an account with BabyTel.  They are based in Montreal
and have line access in most Canadian centers.  Does this look good
enough for VOIP ?

$ ping sip.babytel.com
PING sip.babytel.com (64.40.102.42) 56(84) bytes of data.
64 bytes from cluster1.domaindeluxe.com (64.40.102.42): icmp_seq=0
ttl=56 time=24.4 ms
64 bytes from cluster1.domaindeluxe.com (64.40.102.42): icmp_seq=1
ttl=56 time=22.5 ms
64 bytes from cluster1.domaindeluxe.com (64.40.102.42): icmp_seq=2
ttl=56 time=23.2 ms
64 bytes from cluster1.domaindeluxe.com (64.40.102.42): icmp_seq=3
ttl=56 time=22.5 ms
64 bytes from cluster1.domaindeluxe.com (64.40.102.42): icmp_seq=4
ttl=56 time=21.6 ms
64 bytes from cluster1.domaindeluxe.com (64.40.102.42): icmp_seq=5
ttl=56 time=23.0 ms
64 bytes from cluster1.domaindeluxe.com (64.40.102.42): icmp_seq=6
ttl=56 time=22.0 ms
64 bytes from cluster1.domaindeluxe.com (64.40.102.42): icmp_seq=7
ttl=56 time=23.4 ms
64 bytes from cluster1.domaindeluxe.com (64.40.102.42): icmp_seq=8
ttl=56 time=22.4 ms
64 bytes from cluster1.domaindeluxe.com (64.40.102.42): icmp_seq=9
ttl=56 time=26.8 ms
64 bytes from cluster1.domaindeluxe.com (64.40.102.42): icmp_seq=10
ttl=56 time=28.4 ms
64 bytes from cluster1.domaindeluxe.com (64.40.102.42): icmp_seq=11
ttl=56 time=22.4 ms
64 bytes from cluster1.domaindeluxe.com (64.40.102.42): icmp_seq=12
ttl=56 time=23.7 ms
64 bytes from cluster1.domaindeluxe.com (64.40.102.42): icmp_seq=13
ttl=56 time=26.5 ms
64 bytes from cluster1.domaindeluxe.com (64.40.102.42): icmp_seq=14
ttl=56 time=21.5 ms
64 bytes from cluster1.domaindeluxe.com (64.40.102.42): icmp_seq=15
ttl=56 time=22.8 ms
64 bytes from cluster1.domaindeluxe.com (64.40.102.42): icmp_seq=16
ttl=56 time=31.8 ms
64 bytes from cluster1.domaindeluxe.com (64.40.102.42): icmp_seq=17
ttl=56 time=22.8 ms
64 bytes from cluster1.domaindeluxe.com (64.40.102.42): icmp_seq=18
ttl=56 time=23.9 ms
64 bytes from cluster1.domaindeluxe.com (64.40.102.42): icmp_seq=19
ttl=56 time=21.7 ms
64 bytes from cluster1.domaindeluxe.com (64.40.102.42): icmp_seq=20
ttl=56 time=22.5 ms
64 bytes from cluster1.domaindeluxe.com (64.40.102.42): icmp_seq=21
ttl=56 time=21.9 ms
64 bytes from cluster1.domaindeluxe.com (64.40.102.42): icmp_seq=22
ttl=56 time=40.0 ms
64 bytes from cluster1.domaindeluxe.com (64.40.102.42): icmp_seq=23
ttl=56 time=31.9 ms
64 bytes from cluster1.domaindeluxe.com (64.40.102.42): icmp_seq=24
ttl=56 time=24.7 ms
64 bytes from cluster1.domaindeluxe.com (64.40.102.42): icmp_seq=25
ttl=56 time=25.6 ms
64 bytes from cluster1.domaindeluxe.com (64.40.102.42): icmp_seq=26
ttl=56 time=22.0 ms
64 bytes from cluster1.domaindeluxe.com (64.40.102.42): icmp_seq=27
ttl=56 time=34.1 ms
64 bytes from cluster1.domaindeluxe.com (64.40.102.42): icmp_seq=28
ttl=56 time=23.0 ms
64 bytes from cluster1.domaindeluxe.com (64.40.102.42): icmp_seq=29
ttl=56 time=33.1 ms
64 bytes from cluster1.domaindeluxe.com (64.40.102.42): icmp_seq=30
ttl=56 time=24.0 ms
64 bytes from cluster1.domaindeluxe.com (64.40.102.42): icmp_seq=31
ttl=56 time=22.8 ms
64 bytes from cluster1.domaindeluxe.com (64.40.102.42): icmp_seq=32
ttl=56 time=26.0 ms
64 bytes from cluster1.domaindeluxe.com (64.40.102.42): icmp_seq=33
ttl=56 time=22.0 ms
64 bytes from cluster1.domaindeluxe.com (64.40.102.42): icmp_seq=34
ttl=56 time=23.4 ms
64 bytes from cluster1.domaindeluxe.com (64.40.102.42): icmp_seq=35
ttl=56 time=23.2 ms
64 bytes from cluster1.domaindeluxe.com (64.40.102.42): icmp_seq=36
ttl=56 time=22.7 ms

--- sip.babytel.com ping statistics ---
37 packets transmitted, 37 received, 0% packet loss, time 36030ms
rtt min/avg/max/mdev = 21.586/24.910/40.073/4.134 ms, pipe 2


-- 
Kim Lux,  Diesel Research Inc.


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RE: [Asterisk-Users] BroadVoice

[Vitalie Apostu]

What about extention.conf? Can you share with us?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of skamp
Sent: Tuesday, January 11, 2005 11:42 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] BroadVoice


Guys
http://lists.digium.com/pipermail/asterisk-users/2005-January/081534.html


On Tue, 2005-01-11 at 10:21 -0600, David Ishmael wrote:
 I got the same error.
 
 -Dave
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Vitalie 
 Apostu
 Sent: Tuesday, January 11, 2005 10:12 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] BroadVoice
 
 Following links says: HTTP 404 - File not found . Is it a right link 
 http://lists.digium.com/pipermail/asterisk-users/2005-January/081534.htm ?
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Nabeel 
 Jafferali
 Sent: Tuesday, January 11, 2005 11:09 AM
 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - 
 Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] BroadVoice
 
  Can you give me example of sip.conf and extention.conf which work 
  with broadvoice? I want users who registered with Messenger through 
  sip to be able to make a call thought broadvoice.
 
 I posted this just a few days ago:
 http://lists.digium.com/pipermail/asterisk-users/2005-January/081534.h
 tm
 l
 
 --
 Nabeel Jafferali
 Tel: +1 (416) 628-9342  Toronto
  +1 (646) 225-7426  New York
 FWD: 46990
 Email/MSN: nabeelatjafferali.net
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RE: [Asterisk-Users] internal caller id on analog phones connected tozap

[Shoval Tomer]

 -Original Message-
 From: C F [mailto:[EMAIL PROTECTED]
 Sent: Tuesday, January 11, 2005 4:38 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] internal caller id on analog phones
 connected tozap
 
 How are the analog phones connected to * ? this is where the setting
 should be.

They're connected to Digium cards.
How do I set it in Zapata.conf? everything we tried didn't work.
Here's an excerpt:

signalling = fxo_ks
context = internal
threewaycalling = yes
transfer = yes
group = 1
pickupgroup = 1
mailbox = 202
channel = 2

this channel is connected to an analog phone with an LCD display.

How do I make it show caller id?


 If you use Digium then you set it in zapata.conf, if you use an ata
 the setting should be in the ata.
 BTW, how is FC3 working out?

We decided to go with FC 2 as it supports SATA and seems more stable.
Asterisk compiled just fine on FC 2.
We had to use modprobe for the wcfxs drive instead of insmod (we're
using STABLE 1.0.3, not CVS-HEAD)
During the time it took to download it there, I downloaded FC 3 here,
and asterisk compiled just fine. So did zaptel, but I had no Digium
hardware to test with.
As soon as I will I'll post to the list.

 Shalom Ubracha V'Chodesh Tov
 
 


Thanks (TODA RABA)


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Re: [Asterisk-Users] Is this good packet latency/jitter ? (ping resultsfor BabyTel...)

[Raymond McKay]

On Jan 11, 2005, at 12:15 PM, Kim Lux wrote:
I'm about to order an account with BabyTel.  They are based in Montreal
and have line access in most Canadian centers.  Does this look good
enough for VOIP ?
Easily.  I have connections where the latency is up to 300ms but a 
consistent 300ms without loss.  The key to clear VoIP isn't always the 
latency but more of an issue of packet loss and ordering.  As long as 
your packets arrive constantly and in order, most times you are going 
to find that the connection is good enough for VoIP.  Mind you higher 
latency will equal more delay in communications and echo problems, but 
those can be dealt with.

Raymond McKay
President
RAYNET Technologies LLC
http://www.raynettech.com
Phone: (860) 693-2226 x 31
Toll Free: (877) 693-2226
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[Asterisk-Users] Sounds cut out problem - HFC-S card, zaphfc, Xlite

[Rob Scott]

Hello Asteriskians!

I have an Asterisk box with a simple HFC card in it and a bunch of
people using the Xlite software to connect. The HFC card is connected to
an internal extension on our legacy PBX.

So far so good. The Xlite clients can call each other, and the internal
extensions on the PBX and the Xlites can call each other, no problem.

The problem is when using an Xlite to dial an external number through
the legacy PBX.
What seems to happen is that there is some kind of noise suppression so
that unless the remote party is speaking very loudly the sound cuts out.

Now, I don't know if it is the ISDN connection, Asterisk or the Xlite
client that is causing the problem. I've tried different settings on
everything I can think of and trawled the web for days but so far
nothing useful. I've turned off silence suppression on all the Xlites.
I've turned up the rxgain on the ISDN channel in case it is too quiet. 

Nothing so far has helped.

Calling directly through the PBX from a normal extension phone doesn't
seem to have any problems.

Anyone have any idea what I should look at?

Thanks

Rob Scott
EPAM Systems Ltd.

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[Asterisk-Users] SIP, * and clients behind NAT

[John Huang]

I am new to VOIP, Linux and Asterisk.  Through a lot of reading (this
list, voip-info.org, documentation, etc.), I successfully installed FC3
and * on a new Dell SC420 with two X100P connecting to two PSTN lines at
my office.  I've also installed AMP to help me configure IVRs, call
groups, extensions, etc.

I use a Handytone-286 ATA and x-lite clients on the internal network and
all works fine.

I would like to connect to * as an extension from home, from client
sites, from hotels, etc.  Most of these places will be behind some type
of NAT and/or firewall.  At my home, for example, I have a consumer
grade firewall/NAT.  I cannot get the Handytone-286 to work properly
from there.  I connect to the * server and register, I can call out and
incoming calls ring in, but there is no audio sent nor received
regardless of whether dialing out or calling in.

I suspect this has to do with RTP and how my home firewall/NAT handles
RTP.  Is my thinking correct here?  What's frustrating is that I can't
get it to work even if I put the Handytone-286 in a DMZ.  Maybe the
firewall/NAT is still processing and malforming the RTP packets?

Even if I do get the ATA working fine behind my home NAT, I would have
to do some reconfiguration most likely anywhere else I try plugging it
in, right?  And, if I wanted to add another ATA at home connected to the
same remote * server, it's most like not going to work without custom
RTP port forwards, etc., right?

Thanks,

John

John Huang


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RE: [Asterisk-Users] Installing * on fedora 3

[Alexander Lopez]

 
Do you need my help???



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ferguson,
Michael
Sent: Tuesday, January 11, 2005 10:26 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Installing * on fedora 3

G'Day List,

Can someone help me out a bit please.
I just installed Fedora Core 3 on a Dell Power Edge 400 SC and am trying
to install * I am following
http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation
After running:
   cd /usr/src/asterisk 
   make clean 
   make 
   make install 
   make samples
The instructions says:
Configuring Asterisk 

- Login to your server as user root 
- Right-click on the background and select Open Terminal
- Run the following commands to download the VoicePulse Connect! public
key (needed for receiving calls): 

   cd /var/lib/asterisk/keys 
   wget http://connect.voicepulse.com/keys/voicepulse01.pub 

However there is NO /var/lib/asterisk/keys directory.
HELP!!

Thanks



Michael E. Ferguson
Manager, Information Systems
Berman Rennert Vogel  Mandler, P.A.
100 SE 2nd., Street, Suite 2900
Miami, FL., 33131
305.423.3408 Direct
305.533.1582 Fax
[EMAIL PROTECTED]
This message is for the named person's use only. It may contain
confidential, proprietary or legally privileged information. No
confidentiality or privilege is waived or lost by any mistransmission.
If you receive this message in error, please immediately delete it and
all copies of it from your system, destroy any hard copies of it and
notify the sender. You must not, directly or indirectly, use, disclose,
distribute, print, or copy any part of this message if you are not the
intended recipient. BERMAN RENNERT VOGEL  MANDLER, P.A. reserve the
right to monitor all e-mail communications through its networks. Any
views expressed in this message are those of the individual sender,
except where the message states otherwise and the sender is authorized
to state them to be the views of any such entity. 
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RE: [Asterisk-Users] Ring Voltage Supplied by Wildcard TDM400P REV E/F AUTO FXS/DPO

[Jim Van Meggelen]

Peter Svensson wrote:
 On Tue, 11 Jan 2005, Jim Van Meggelen wrote:
 
 [EMAIL PROTECTED] wrote:
 I'm trying to connect a TDM400P with an FXS module to a Valcom
 V-9940 Paging adaptor. This port on the TDM400P was connected to a
 2500 Set and was working I just re-connected it to the Valcom (which
 is known to work on a Telco POTS line) and its not picking up.  The
 Valcom docs say it need a minimum of 75 Volts at 20-30 Hz to
 recognize a call... So the question is what ring voltage does the
 FXS modules on a TDM400P put out?
 
 Ultimately, that depends on how much current is being drawn, but my
 multimeter reports about 70V on a 6 foot line cord. Generally, I'd
 expect an FXS to put out more like 90-110V (that's what my TalkSwitch
 supplies).
 
 The ring voltage can be configured. Try setting the module parameter
 boostringer when the wctdm module is insmod / modprobed.

Which is accomplished in 1.0.x by the following:

# modprobe wcfxs boostringer=1

And in CVS HEAD (I assume, as I haven't tested this) by:

# modprobe wctdm boostringer=1

Who, exactly, Boo Stringer is has not yet been determined.

Boo Radley? NO, Boo Stringer.

(Sorry for the pun, folks, but I will now never forget this parameter
name).

-- 
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 265.6.10 - Release Date: 10/01/2005
 

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RE: [Asterisk-Users] internal caller id on analog phones connectedtozap

[John Bohman]

I just noticed the same thing, I just pulled the latest version from CVS and
what was working perfectly before is now dead..
I'm running a 4port card with 3 FXS modules and 1 FXO..
Running: Asterisk CVS-v1-0-01/10/05-17:32:27
Did something change, someone add a new switch to turn it on..
I'm running a basic stock Zapata.conf file that was working great before..
John B.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shoval Tomer
Sent: Tuesday, January 11, 2005 12:25 PM
To: C F; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] internal caller id on analog phones
connectedtozap



 -Original Message-
 From: C F [mailto:[EMAIL PROTECTED]
 Sent: Tuesday, January 11, 2005 4:38 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] internal caller id on analog phones
 connected tozap
 
 How are the analog phones connected to * ? this is where the setting
 should be.

They're connected to Digium cards.
How do I set it in Zapata.conf? everything we tried didn't work.
Here's an excerpt:

signalling = fxo_ks
context = internal
threewaycalling = yes
transfer = yes
group = 1
pickupgroup = 1
mailbox = 202
channel = 2

this channel is connected to an analog phone with an LCD display.

How do I make it show caller id?


 If you use Digium then you set it in zapata.conf, if you use an ata
 the setting should be in the ata.
 BTW, how is FC3 working out?

We decided to go with FC 2 as it supports SATA and seems more stable.
Asterisk compiled just fine on FC 2.
We had to use modprobe for the wcfxs drive instead of insmod (we're
using STABLE 1.0.3, not CVS-HEAD)
During the time it took to download it there, I downloaded FC 3 here,
and asterisk compiled just fine. So did zaptel, but I had no Digium
hardware to test with.
As soon as I will I'll post to the list.

 Shalom Ubracha V'Chodesh Tov
 
 


Thanks (TODA RABA)


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Re: [Asterisk-Users] dialing into * then forwarded out gets choppy audio

[rsenykoff]

snip
Hello all!

If I place a call to our number, the call is routed to our Asterisk box
from teliax -- IAX2 -- firewall w/ port forwarding -- *

If that caller dials an extension that rings an outside line, where our *
box makes an outbound connection to teliax to terminate the call, we get
choppy audio. Internal extensions have been dialing outbound calls no
problem for over a week.

What could be causing this? IAX2 debug is not showing any errors.

To be more specific about the audio:
The originating caller can hear the called user fine.
The called user may only intermittently hear the caller, and usually only
if the caller talks extremely loud or close to the mic.

I've tested this in the middle of the night with no other network traffic
happening. Even then we've got a fat pipe with IAX2 ports QoSd at top
priority. We're only 3 hops (~30 ms) from teliax.

Any ideas are appreciated...

-Ron
/snip

Sorry for the bump... catching flak for this though, if someone can help
it's much appreciated.

Best regards,
-Ron

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[Asterisk-Users] operator says that dial 1 or 0

[hak atil]

Hi,

I realized that I have a problem with my asterisk. I am using 1 TDM400P 4
FXS card from digium. My problem is I can't make calls one after another.
When I dial long distance number from my SIP phone, my phone company's
operator says that dial 1 or 0, then hangs up. After a couple of seconds I
dial the same phone number, this time it works with no problem. Weird part
is that Asterisk is working properly; CLI does not display any errors when I
got the operator. I have to dial a few times on certain phone numbers. 


Is there anything I can change on the configuration?

HAKAN


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Re: [Asterisk-Users] I need your feedback related to the DIAX 0.9.9f stability

[Denis Galvão - iSolve]

Dan, I'm using DIAX(Portuguese Language) for one week too... Without any 
problems.

I've tested with all suported CODECs... But Im using with a-law and u-law 
for now.

If you need some help to translate to Brazillian Portuguese, call me!

I like the incoming calls ring... ;)

Denis.

Em Seg 10 Jan 2005 05:46, Dan escreveu:
 Hi all,

 I kindly ask DIAX users to send me a feedback related to the stability of
 the new version (0.9.9f),
 comparing with the older versions (especially 0.9.8).
 I ask this because I have DIAX runing for one week now without any crash.
 It is used mainly to control some X10 devices through a regular phone.

 Thank you and best regards,
 Dan


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RE: [Asterisk-Users] BroadVoice

[Paul Rodan]

:-)

I'm still using my config from before BroadVoice supported Asterisk. Had to
change the useragent to look like a Cisco phone, and change the realm, they
were actively filtering out the word Asterisk. Had to change chan_sip at the
time and finagle the SIP user ID/Password and server out of one of their
techs. Was a tricky config but it worked. Tempted to upgrade now and go with
their new standard Asterisk configuration. But for now, they still think
they're talking to a Cisco.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of skamp
Sent: Tuesday, January 11, 2005 11:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] BroadVoice

not that ive ever seen occur :)

On Tue, 2005-01-11 at 11:28 -0500, Paul Rodan wrote:
 Curious, has anybody ever been charged the 3 cents a minute or whatever
when
 they've had more than 1 or 2 simul calls? 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Nabeel
 Jafferali
 Sent: Tuesday, January 11, 2005 11:09 AM
 To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
 Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] BroadVoice
 
  Can you give me example of sip.conf and extention.conf which
  work with broadvoice? I want users who registered with
  Messenger through sip to be able to make a call thought broadvoice.
 
 I posted this just a few days ago:
 http://lists.digium.com/pipermail/asterisk-users/2005-January/081534.htm
 l
 
-- 
skamp [EMAIL PROTECTED]

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Re: [Asterisk-Users] Ring Voltage Supplied by Wildcard TDM400P REV E/F AUTO FXS/DPO

[Timothy Costello]

On Jan 11, 2005, at 11:42 AM, Jim Van Meggelen wrote:
Peter Svensson wrote:
On Tue, 11 Jan 2005, Jim Van Meggelen wrote:
[EMAIL PROTECTED] wrote:
I'm trying to connect a TDM400P with an FXS module to a Valcom
V-9940 Paging adaptor. This port on the TDM400P was connected to a
2500 Set and was working I just re-connected it to the Valcom (which
is known to work on a Telco POTS line) and its not picking up.  The
Valcom docs say it need a minimum of 75 Volts at 20-30 Hz to
recognize a call... So the question is what ring voltage does the
FXS modules on a TDM400P put out?
Ultimately, that depends on how much current is being drawn, but my
multimeter reports about 70V on a 6 foot line cord. Generally, I'd
expect an FXS to put out more like 90-110V (that's what my TalkSwitch
supplies).
The ring voltage can be configured. Try setting the module parameter
boostringer when the wctdm module is insmod / modprobed.
Which is accomplished in 1.0.x by the following:
# modprobe wcfxs boostringer=1
And in CVS HEAD (I assume, as I haven't tested this) by:
# modprobe wctdm boostringer=1
Who, exactly, Boo Stringer is has not yet been determined.
Boo Radley? NO, Boo Stringer.
(Sorry for the pun, folks, but I will now never forget this parameter
name).
Or adding the line:
options wctdm boostringer=1
to the file /etc/modules.conf
Thank you to all who replied, I will be trying this out as soon as I 
can down the system to reload the modules.

Later;
Tim
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Re: [Asterisk-Users] operator says that dial 1 or 0

[C F]

looks like you will need to put in a pause to wait for dial tone b4 dialing.
do something like this
exten = _9.,1,Dial(Zap/g1/ww${EXTEN:1})
the w puts in half a second pause.


On Tue, 11 Jan 2005 10:50:42 -0600, hak atil [EMAIL PROTECTED] wrote:
 Hi,
 
 I realized that I have a problem with my asterisk. I am using 1 TDM400P 4
 FXS card from digium. My problem is I can't make calls one after another.
 When I dial long distance number from my SIP phone, my phone company's
 operator says that dial 1 or 0, then hangs up. After a couple of seconds I
 dial the same phone number, this time it works with no problem. Weird part
 is that Asterisk is working properly; CLI does not display any errors when I
 got the operator. I have to dial a few times on certain phone numbers.
 
 Is there anything I can change on the configuration?
 
 HAKAN
 
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[Asterisk-Users] TDM box Hardware

[ismaelg]

Hello all,
Recently I bought a TDM02B digium card to conect to the PSTN.
We pluged it on a Pentium IV 2,8 Ghz, Asus Motherboard, but when we try 
to start asterisk, the box hangs.

Someone have the same card running with asterisk in a similar machine?
Could you tell me your box hardware details?
Thanks for your time,
Ismael Gil.
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Re: [Asterisk-Users] internal caller id on analog phones connectedtozap

[C F]

You should put in zapata.conf
usecallerid=yes


On Tue, 11 Jan 2005 12:48:40 -0500, John Bohman [EMAIL PROTECTED] wrote:
 I just noticed the same thing, I just pulled the latest version from CVS and
 what was working perfectly before is now dead..
 I'm running a 4port card with 3 FXS modules and 1 FXO..
 Running: Asterisk CVS-v1-0-01/10/05-17:32:27
 Did something change, someone add a new switch to turn it on..
 I'm running a basic stock Zapata.conf file that was working great before..
 John B.
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Shoval Tomer
 Sent: Tuesday, January 11, 2005 12:25 PM
 To: C F; Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] internal caller id on analog phones
 connectedtozap
 
  -Original Message-
  From: C F [mailto:[EMAIL PROTECTED]
  Sent: Tuesday, January 11, 2005 4:38 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [Asterisk-Users] internal caller id on analog phones
  connected tozap
 
  How are the analog phones connected to * ? this is where the setting
  should be.
 
 They're connected to Digium cards.
 How do I set it in Zapata.conf? everything we tried didn't work.
 Here's an excerpt:
 
 signalling = fxo_ks
 context = internal
 threewaycalling = yes
 transfer = yes
 group = 1
 pickupgroup = 1
 mailbox = 202
 channel = 2
 
 this channel is connected to an analog phone with an LCD display.
 
 How do I make it show caller id?
 
  If you use Digium then you set it in zapata.conf, if you use an ata
  the setting should be in the ata.
  BTW, how is FC3 working out?
 
 We decided to go with FC 2 as it supports SATA and seems more stable.
 Asterisk compiled just fine on FC 2.
 We had to use modprobe for the wcfxs drive instead of insmod (we're
 using STABLE 1.0.3, not CVS-HEAD)
 During the time it took to download it there, I downloaded FC 3 here,
 and asterisk compiled just fine. So did zaptel, but I had no Digium
 hardware to test with.
 As soon as I will I'll post to the list.
 
  Shalom Ubracha V'Chodesh Tov
 
 
 
 Thanks (TODA RABA)
 
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Re: [Asterisk-Users] Dialplan variables

[Diego Ercolani]

Il 01:20, mercoledì 29 dicembre 2004, Norman Zhang ha scritto:
 Hi,

 May I ask what does

 exten = s,1,Answer
 exten = s,2,ResponseTimeout(5)

 exten = i,1,Playback(pbx-invalid)

 s, t, i stands for? It says it is someexten but I still don't get it.

s: start  is the extension invoked when there is the option immediate=yes in 
the channel

t: timeout, is the extension where asterisk goes when a user doesn't respond 
in time to a directory request

i: invalid, is the extension to go when it's digited a wrong extension.
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