Re: [Asterisk-Users] App_festival crashing

[Ing. Angel Gomez Garcia]

   Hi.

   I am not using cache, just :

festival.conf
-
[general]
host=localhost
port=1314
festivalcommand=(tts_textasterisk "%s" 'file)(quit)\n
but in extensions.conf when i call the festival app i put the text 
'quoted' like this:
exten => 003,1,Festival('Hello asterisk user, how are you today?')   ; 
<-- note the quotes ...
exten => 003,2,Wait(1)
exten => 003,3,Hangup()

   And everything works like the readme says.
 
   Good luck.

Borut Senicar wrote:

Hi all,

I'm unable to put app_festival to work. I successfully patched,
installed and tested festival (interactive logon and telnet to server
port) which seems to work without problems. 

But when I test it in asterisk I got the following trace in console:

   -- Executing Answer("SIP/bsenicar-850b", "") in new stack
   -- Executing SayDigits("SIP/bsenicar-850b", "123") in new stack
   -- Playing 'digits/1'
   -- Playing 'digits/2'
   -- Playing 'digits/3'
   -- Executing Festival("SIP/bsenicar-850b", "Connect to Festival") in
new stack
 == Parsing '/etc/asterisk/festival.conf': Found
WARNING[147466]: File app_festival.c, Line 304 (festival_exec): Text
passed to festival server : Connect to Festival
WARNING[147466]: File app_festival.c, Line 353 (festival_exec): line
length : 19
WARNING[147466]: File app_festival.c, Line 357 (festival_exec): Seek
position : 23
WARNING[147466]: File app_festival.c, Line 381 (festival_exec): Passing
text to festival...
WARNING[147466]: File app_festival.c, Line 390 (festival_exec): Writing
result to cache...
WARNING[147466]: File app_festival.c, Line 400 (festival_exec): Passing
data to channel...
 == Spawn extension (home-trusted, 1000, 3) exited non-zero on
'SIP/bsenicar-850b'
In festival.conf I enabled all 5 default options and my extensions.conf
looks like this:
[home-trusted]
exten => 1000,1,answer
exten => 1000,2,SayDigits(123)
exten => 1000,3,Festival(Connect to Festival)
exten => 1000,4,Wait(5)
exten => 1000,5,Festival(send the argument)
exten => 1000,6,Hangup
Cache file is created but playback to channel doesn't work correctly.

I'm running Asterisk CVS-09/23/03-23:16:24

I also noticed that parsing of festival.conf in app_festival.c is done
incorrectly for usecache.
On line 281 of app_festival.c 

usecache = ast_true(temp);

value of usecache config entry is tested with ast_true function, which
returns -1 if value is (yes, y, t or 1). For that reason cache is never
used.
Correct line should be:

usecache = ast_true(temp)==-1;

Thanks in advance.
Borut




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Re: [Asterisk-Users] dialing codes..( You can help! )

[WipeOut .]

It looks like its going to be a bigger job than I thought it would be.. I guess I will 
have to start with the countries that we use the most and then add the others as I 
find the details..

Thanks for all the input..

> > I'd be up for setting up some kind of website/database thing for collating
> > all this information, just not sure of the value and if anyone else would
> be
> > up for it/contributing data? Be cool to have though, and nice for customer
> > bill presentation etc?
> 
> Try www.numberingplans.com or www.numberplan.org - they are both commerical
> but have some information for free. Also you can look at
> http://www.wtng.info/ a free site, and the ITU site at
> http://www.itu.int/ITU-T/inr/nnp/index.html
> 
>  its all there if you know where to look!
> 
> Linus
> 
> 
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[Asterisk-Users] festival problem

[Chad Sawyer]

I have loaded festival-1.4.3 patched with the 
1.4.3.diff file.  Festival source is in /usr/src/festival dir.  When I 
try to use it I get this from asterisk:
 
    -- Executing 
Answer("SIP/chad-57a4", "") in new stack    -- Executing 
Festival("SIP/chad-57a4", ""I am talking"") in new stack  == Parsing 
'/etc/asterisk/festival.conf': FoundWARNING[1217602880]: File 
app_festival.c, Line 304 (festival_exec): Text passedto festival server : "I 
am talking"WARNING[1217602880]: File app_festival.c, Line 381 
(festival_exec): Passing text to festival...WARNING[1217602880]: 
File app_festival.c, Line 400 (festival_exec): Passing data to 
channel...WARNING[1217602880]: File app_festival.c, Line 423 
(festival_exec): Festival returned ER  == Spawn extension (local, 
521, 2) exited non-zero on 'SIP/chad-57a4'
 
And this from the festival server:
 
SIOD ERROR: unbound variable : I
 
 
I have done this twice, and came up the same way 
both times.  I followed directions from http://www.marko.net/asterisk/archives/0209/0389.html and 
used the 1.4.3 instead, everything else is the same.
 
Any help is apreciated
 
Chad Sawyer

POE question (was Re: [Asterisk-Users] Skinny)

[Nick]

On a related note, does anyone know if the Compaq/Lucent 6 & 12 port POE
devices provide the correct POE for cisco phones?
Thanks
Nick
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[Asterisk-Users] LIST MOM / DAD PLEASE READ [andersoncbr.sspam@uol.com.br:......]

[John Brown]

List Mom/Dad,

could you please find in the list of subscribers  the person that
is getting mail at 

uol.com.br  or something similar

and smack them for having broken Anti-Spam software ??

I'd be happy to see them just be removed until they fix it.

cheers


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Date: Wed, 24 Sep 2003 02:21:45 -0300 (BRT)
From: AntiSpam UOL <[EMAIL PROTECTED]>
To: [EMAIL PROTECTED]
Subject: RE:[Asterisk-Users] initial review of Grandstream HT-286 ATA device
X-UOL-Srv: T
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[Asterisk-Users] initial review of Grandstream HT-286 ATA device

[John Brown]

Hi List,

Just received a HT-286  Analog Telephone Adapter.

This device is allows the user to take a standard
analog telephone set and connect it to a VoIP/SIP
based gateway.  A low cost way of having PSTN 
devices make use of VoIP/SIP services.  

For example you could take a credit card processing
machine and punch it into a location where you have
broad band access.

You could take a FAX machine and put it someplace
and the number is somewhere else.  

I've tested FAX via the HT-286 and the 3 pages I sent
seemed to come thru just fine.  That path was
PSTN -> ASTERISK -> HT-286 -> Standard Analog FAX machine



Specs:

Color of case:  Black
Size:   About the size of a pack of smokes
Interfaces:
1 Ethernet 10baseT
1 FXS Analog interface
1 DC jack for power

1 Button for selecting the voice
  prompted config menu

1 multi colored LED for states


The UNIT is configurable via a HTTP interface, ala
the GS-101 and 102 models.  

You can also configure the unit via a set of voice
prompts.  It can be a bit of a challenge, but if
you are slow and smooth it works.


The LED states

Off Doing nothing
RED Solid   Off Hook, doing FXS to VOIP
GREEN Solid Voice Prompt Menu
RED Blink   Voice Mail Alert

The Ethernet jack has Link and Traffic LED's like 
most devices today.

In general its simple, easy to config and seems to 
just work.  I'm going to stress test it this weekend
and see how well it holds up.


Time out of the box to a functioning device was less
than 10 minutes.


John Brown
Products avail at:
http://www.chagres.net/products/voip/
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[Asterisk-Users] PROBLEMS WITH IAXATEL AND DIGIUM IAX

[Alvaro Parres]

Hi

 I'm having a extrange problem I cant register with Iaxtel or call to digium...

  But i cant make or recive IAX calls... ( I made some one with irc users )

 Any idea why?


At my logs i have this from iaxtel:

NOTICE[196621]: File chan_iax2.c, Line 2832 (register_verify): No registration
for peer 'xmarts' (from 192.168.0.11)
NOTICE[196621]: File chan_iax2.c, Line 4389 (socket_read): Registration of
'arabe' rejected: Registration Refused

And when i tray to call Digium i get:

-- Called [EMAIL PROTECTED]/[EMAIL PROTECTED]
NOTICE[196621]: File chan_iax2.c, Line 4024 (socket_read): Rejected connect
attempt from 192.168.0.11
WARNING[196621]: File chan_iax2.c, Line 4124 (socket_read): Call rejected by
216.207.245.8: No authority found



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Re: [Asterisk-Users] Very bad echo (appears that...)

[Rich Adamson]

Andrew,

> I must have misread in the documentation that gains could be specified as % 
> or as fractional... i.e. 1.00 or 100.  No warnings from anything for 
> setting them to 100.  :-)

I've seen others mention 100%, etc, however the current sample config's
still show rxgain=0.0, etc, as though these were roughly dB levels. Does
the code actually use either format?  (I've also seen references to using
a % setting.)
 
> So now can someone come clean with me here -- are those X100P cards just 
> some OEM winmodem?   What's up with the telephone jack in addition to the 
> line jack?

Don't know that makes any real difference (and I'm sure you've read the 
many recent postings related to the subject). It's fairly easy to judge
for yourself by looking closely at the cards... no manufacturer name,
no copyright or legal stuff, or anything else on the printed circuit board,
etc. If a US company were to design & manufacturer such boards, you'd see
their name on it somewhere (that's just the nature of US companies).
As I mentioned a couple of weeks ago, take the covers off your Cisco,
3Com, Nortel, or whatever ethernet switches and you'll find some other
company's name etched on many of the boards, etc. No big deal, its
marketing smoke and mirrors. :)



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Re: [Asterisk-Users] Very bad echo (appears that...)

[Andrew Kohlsmith]

> I wouldn't mess with the gains if I were you.

Just a point of order -- asterisk should complain loudly if the gains are 
set outside the allowed ranges...  I had an odd problem where my wcfxo card 
(X100P) was crazy noisy...  I had my gains set to 100 each.

I must have misread in the documentation that gains could be specified as % 
or as fractional... i.e. 1.00 or 100.  No warnings from anything for 
setting them to 100.  :-)

So now can someone come clean with me here -- are those X100P cards just 
some OEM winmodem?   What's up with the telephone jack in addition to the 
line jack?

Regards,
Andrew
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[Asterisk-Users] Example callback/call out AGI script

[Eric Wieling]

I have posted a link to the tarball of my rather simple AGI script that
will call back the number the AGI script receives as your CallerID
number.  Currently only calls back 4 digit extensions (2000 thru 8999)
and IAXTel 700 numbers.

The tarball, other scripts, and various ither info can be found at
http://www.fnords.org/~eric/asterisk/

--Eric
-- 
Sample configs and more: http://www.fnords.org/~eric/asterisk/

BTEL Consulting
+1-850-484-4535 x2111 (Pensacola)
+1-504-595-3916 x2111 (New Orleans)
+1-877-677-9643 x2111 (Toll Free)
This message has been 'sanitized'.  This means that potentially
dangerous content has been rewritten or removed.  The following
log describes which actions were taken.

Sanitizer (start="1064359239"):
  Forcing message to be multipart/mixed, to facilitate logging.
  Writer (pos="734"):
Part (pos="808"):
  Added 1 bytes of scratch space.

Total modifications so far: 1
Added 1 bytes of scratch space.


Anomy 0.0.0 : Sanitizer.pm
$Id: Sanitizer.pm,v 1.79 2003/06/19 19:22:00 bre Exp $

[Asterisk-Users] Grandstream HT-286 review on the way

[John Brown]

Just to tease people

I'll be post a URL to my review of the HT-286 by 
thursday..



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[Asterisk-Users] App_festival crashing

[Borut Senicar]

Hi all,

I'm unable to put app_festival to work. I successfully patched,
installed and tested festival (interactive logon and telnet to server
port) which seems to work without problems. 

But when I test it in asterisk I got the following trace in console:

-- Executing Answer("SIP/bsenicar-850b", "") in new stack
-- Executing SayDigits("SIP/bsenicar-850b", "123") in new stack
-- Playing 'digits/1'
-- Playing 'digits/2'
-- Playing 'digits/3'
-- Executing Festival("SIP/bsenicar-850b", "Connect to Festival") in
new stack
  == Parsing '/etc/asterisk/festival.conf': Found
WARNING[147466]: File app_festival.c, Line 304 (festival_exec): Text
passed to festival server : Connect to Festival
WARNING[147466]: File app_festival.c, Line 353 (festival_exec): line
length : 19
WARNING[147466]: File app_festival.c, Line 357 (festival_exec): Seek
position : 23
WARNING[147466]: File app_festival.c, Line 381 (festival_exec): Passing
text to festival...
WARNING[147466]: File app_festival.c, Line 390 (festival_exec): Writing
result to cache...
WARNING[147466]: File app_festival.c, Line 400 (festival_exec): Passing
data to channel...
  == Spawn extension (home-trusted, 1000, 3) exited non-zero on
'SIP/bsenicar-850b'

In festival.conf I enabled all 5 default options and my extensions.conf
looks like this:

[home-trusted]
exten => 1000,1,answer
exten => 1000,2,SayDigits(123)
exten => 1000,3,Festival(Connect to Festival)
exten => 1000,4,Wait(5)
exten => 1000,5,Festival(send the argument)
exten => 1000,6,Hangup

Cache file is created but playback to channel doesn't work correctly.

I'm running Asterisk CVS-09/23/03-23:16:24

I also noticed that parsing of festival.conf in app_festival.c is done
incorrectly for usecache.

On line 281 of app_festival.c 

usecache = ast_true(temp);

value of usecache config entry is tested with ast_true function, which
returns -1 if value is (yes, y, t or 1). For that reason cache is never
used.

Correct line should be:

usecache = ast_true(temp)==-1;

Thanks in advance.
Borut





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[Asterisk-Users] RE: [Asterisk-Users] can´t call ICH

[Paul Crick]

You've got a whole bunch of numbers you're trying to call there. What is the
full number that you want to call including the country code? It's not clear
if the number you're trying should be 755xxx 55xxx or 055xx ?

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[Asterisk-Users] RE: [Asterisk-Users] can´t call ICH

[Andrew Joakimsen]

Show us your sip.conf file, you can (should) block out the passwords.

Is there a [sipauth.deltathree.com] section in it?

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of listas iPfone
> Sent: Tuesday, September 23, 2003 3:22 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] can´t call ICH
> 
> Hi All
> 
> Asterisk is registered with ICH with no problems, but i can´t make a
call,
> somebody can tell me if that messages from cli are correct or there is
any
> problem?
> 
> Executing Dial("SIP/33-4a71",
"SIP/[EMAIL PROTECTED]")
> in
> new stack
> -- Called [EMAIL PROTECTED]
> 
> *CLI>
>   == Spawn extension (from-sip, 7551136752312, 1) exited non-zero on
> 'SIP/33-4a71'
> -- Executing Dial("SIP/35-74e2",
> "SIP/[EMAIL PROTECTED]") in new stack
> -- Called [EMAIL PROTECTED]
>   == Spawn extension (from-sip, 7551136752312, 1) exited non-zero on
> 'SIP/35-74e2'
> -- Executing Dial("SIP/33-e843",
> "SIP/[EMAIL PROTECTED]") in new stack
> -- Called [EMAIL PROTECTED]
>   == Spawn extension (from-sip, 70551136752312, 1) exited non-zero on
> 'SIP/33-e843'
> 
> thanks
> 
> Miklos
> 
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RE: [Asterisk-Users] Advantage of Cisco 7960 with 5.x firmware

[Paul Vinciguerra]

Yes, without a doubt, this was done to address the loading from TFTP issue,
however, It is my understanding that signed code can be revoked by the issuer,
and also has a limited lifetime based on the lifetime of the certificate it
was signed with. That is what I see as the down side to signed code.

On Tue, 23 Sep 2003 11:13:58 -0500, Matthew Hardeman wrote
> I've found there are some bugs they don't list (bugs of great 
> severity) that are fixed in the latest release that can cause 
> trouble in certain environments.  The phone's handling of ICMP 
> redirects in a multihomed Ethernet environment (two separate,
>  exclusive subnets running on a single segment) is very flaky, and 
> frequently will result in the phone crashing and rebooting.  The 
> latest release seems to mitigate this in most instances, though I 
> actually found the best solution (sadly) was to prevent my gateway 
> from sending ICMP redirects to the Cisco phones.
> 
> The 5.x+ stuff is annoying, due to the whole code-signing issue. 
>  It's kind of anti-open-architecture...  On the other hand, there 
> aren't any non-Cisco firmware builds for these phones floating 
> around out there... Would you even want one?  Cisco implemented the 
> code signing enforcement as a response to a security analysis of the 
> phones that pointed to the ability to make the phone run arbitrary 
> code via TFTP being a security risk.  That risk is no longer.  I 
> have mixed feelings about it, but have no regrets in having deployed 
> the 5.x solutions in my business.
> 
> Matt Hardeman
> PaperSoft
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Brian 
> West Sent: Tuesday, September 23, 2003 10:23 AM To:
[EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Advantage of Cisco 7960 with 5.x firmware
> 
> their really isn't much fixed between 4.4 and the 5.x stuff but at 
> the time thats all I had.  So I put that on the phone.  So far everything
> works like a champ.  Not one problem.
> 
> 4.4
> http://www.cisco.com/en/US/products/sw/voicesw/ps2156/prod_release_note0
> 9186a008016096f.html#63943
> 
> Resolved Caveats.Release 5.0
> No resolved caveats specific to Cisco IP Phone 7940/7960 Release 5.0
> require documentation in these release notes.
> 
> Resolved Caveats.Release 5.1
> All caveats listed in this section are resolved in Cisco IP Phone
> 7940/7960 Release 5.1. This section lists only severity 1 and 2 caveats
> and select severity 3 caveats
> 
> CSCdz59328: SIPPhone: The UI responsiveness slow, fast fingers cause
> digit
> drop
> CSCdz77783: SIPPhone: Clipping of voice in 7960 SIP phone
> CSCea83100: SIP: Dialing # does not work correctly if dialplan is empty
> CSCea85697: Phone may fail to reset when an Exception occurs
> CSCea93250: SIPPhone: Dialing # does not always work if default rule
> missing
> CSCeb27906: SIPPhone: Null To-tag in REFER causes transfer fail (race
> condition)
> CSCeb29575: SIPPhone: NOTIFY Event header shortform is not supported
> (o:)
> 
> Resolved Caveats.Release 5.2
> All caveats listed in this section are resolved in Cisco IP Phone
> 7940/7960 Release 5.2. This section lists only severity 1 and 2 caveats
> and select severity 3 caveats
> 
> CSCeb41335: DSP mismatch with upgrade failure
> CSCeb44769: Phone removes dots in the IP address when sending ACK
> CSCeb46028: 79x0 Memory leak issues related to DNS query failures
> CSCeb75975: Phone crashes upon any menu exit
> 
> Resolved Caveats.Release 5.3
> All caveats listed in this section are resolved in Cisco IP Phone
> 7940/7960 Release 5.3. This section lists only severity 1 and 2 caveats
> and select severity 3 caveats
> 
> CSCeb85936: SIP phone doesnt use the medium level contact field
> 
> Hope that helps.
> 
> bkw
> 
> On Tue, 23 Sep 2003, Peter Pauly wrote:
> 
> > I'm currently running firmware version 3.2 on my
> > Cisco 7960. I've seen on the list that several
> > people are running the 5.x latest versions.
> >
> > I've avoided going to higher firmware versions
> > because I'm worried about potential problems
> > or issues with the encryption mechanism used
> > in the later firmware versions. (Once you
> > go to an encrypted firmware version, you can't
> > go back, right?)
> >
> > For those of you who have gone to the newer
> > firmware, what features or benefits have
> > you seen by going with the newer firmware?
> > ___
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> >
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Re: [Asterisk-Users] dialing codes..( You can help! )

[Tom Zimnicki]

While not foolproof, you may be able to figure out most of the Canadian Land
vs. Mobile through the following as LNP between LEC's & WSP's isn't an issue
yet. (Other than the potential unique case of Microcell, but for the most
part you can assume wireless.)

http://www.cnac.ca/NPANXX.zip

WSP's:

  Rogers AT&T Wireless

Microcell
[ILEC] Mobility/Mobilite

There may a couple small independents, but I can't remember off the top of
my head.

Regards,

Tom


- Original Message - 
From: "WipeOut ." <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, September 23, 2003 11:32 AM
Subject: [Asterisk-Users] dialing codes..( You can help! )


> Hi,
>
> I am trying to setup some LCR functions on my Asterisk box and have a
cheap call provider that uses various different numbers for landlines and
cell phone numbers in various countrys..
>
> I am finding it difficult to find the various codes..
>
> eg.
> UK Landline - +44[12].
> UK Cell - +44[7].
>
> SA Landline - +27[1-6].
> SA Cell - +27[78].
>
> Please send me your country's dialing rules similar to how I have
represented them in the example and when I have a listing I will be happy to
post it for everyone..
>
> Specific countries I need urgently are US, Canada, France and Germany..
>
> Thanks..
>
>
> -- 
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Re: [Asterisk-Users] IAXTEL

[Steven Critchfield]

On Tue, 2003-09-23 at 13:09, Ariel Batista wrote:
> Well as you know I am very new to this Asterisk.
> I just joined gnophone for a IAX phone number.
> Got my number and password and I have confirmed it.
> 
> So not that I have this how can I put this into my
> Asterisk server and make it get calls to my extension?
> 
> I have read most of the pass post and done a lookup with
> google for this.  But I still am missing where to start. 
> I feel I need to do a register but where? and how? Thanks

google turns up useful information with just iax and register keywords.

Register commands for IAX protocol go in iax.conf in the general
section. iax.conf should have helpful information inside the file
itself.

-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] dialing codes..( You can help! )

[Linus Surguy]

> I'd be up for setting up some kind of website/database thing for collating
> all this information, just not sure of the value and if anyone else would
be
> up for it/contributing data? Be cool to have though, and nice for customer
> bill presentation etc?

Try www.numberingplans.com or www.numberplan.org - they are both commerical
but have some information for free. Also you can look at
http://www.wtng.info/ a free site, and the ITU site at
http://www.itu.int/ITU-T/inr/nnp/index.html

 its all there if you know where to look!

Linus


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Re: [Asterisk-Users] Very bad echo (appears that...)

[Jan Rychter]

> "Mark" == Mark Spencer <[EMAIL PROTECTED]> writes:
 Mark> I wouldn't mess with the gains if I were you.  Mark

What do you mean?

Are the gains an unsupported feature? Aren't we supposed to adjust them?

I have some people who complain that they can't hear me when I dial out
using the X100P adapter. I thought that the gains were precisely for
that purpose?

--J.


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Description: PGP signature

Re: [Asterisk-Users] Calls being interrupted, analog signalling problems

[Jan Rychter]

> "Jan" == Jan Rychter <[EMAIL PROTECTED]> writes:
 Jan> I'm having trouble with a WX100USB adapter and a Siemens Gigaset
 Jan> cordless phone.

 Jan> If I select fxols as a signalling method, calls are being
 Jan> disconnected. Usually after about 4 minutes, and asterisk just
 Jan> says that the phone has hung up.

 Jan> If I choose fxogs, I immediately get a LINE IN USE message on my
 Jan> phone and I can't even get a dialtone.

 Jan> If I choose fxoks, it mostly works, but sometimes after making a
 Jan> call the adapter will get stuck in a LINE IN USE state, too. I
 Jan> don't know of a proper way to correct it, sometimes disconnecting
 Jan> the USB adapter and reloading the drivers and asterisk fixes that,
 Jan> sometimes not.

 Jan> What is the proper signalling method? What do you people use? I'd
 Jan> appreciate any advice.

Hmm. Does the number of responses (zero) indicate that I'm the only one
having such problems?

I've just had to stop asterisk, unload the wcusb module, reload it,
ztcfg it and start asterisk again, because my line was stuck in the LINE
IN USE state (with no dialtone on the phone of course).

I guess I'll report it as a bug, then.

--J.


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Description: PGP signature

[Asterisk-Users] chan_capi in the US

[Justin Huff]

So, I'm using chan_capi with an Eicon Diva Server card here in the US.I
had the telco change some stuff on the line and now I can't do outgoing
calls.

Would someone mine running 
'divactrl dchannel -c 1 -Debug -dmonitor' and making an outgoing ISDN
call for me?  I'd like to see if there's any differences.

thanks!
--Justin


signature.asc
Description: This is a digitally signed message part

[Asterisk-Users] can´t call ICH

[listas iPfone]

Hi All

Asterisk is registered with ICH with no problems, but i can´t make a call,
somebody can tell me if that messages from cli are correct or there is any
problem?

Executing Dial("SIP/33-4a71", "SIP/[EMAIL PROTECTED]") in
new stack
-- Called [EMAIL PROTECTED]

*CLI>
  == Spawn extension (from-sip, 7551136752312, 1) exited non-zero on
'SIP/33-4a71'
-- Executing Dial("SIP/35-74e2",
"SIP/[EMAIL PROTECTED]") in new stack
-- Called [EMAIL PROTECTED]
  == Spawn extension (from-sip, 7551136752312, 1) exited non-zero on
'SIP/35-74e2'
-- Executing Dial("SIP/33-e843",
"SIP/[EMAIL PROTECTED]") in new stack
-- Called [EMAIL PROTECTED]
  == Spawn extension (from-sip, 70551136752312, 1) exited non-zero on
'SIP/33-e843'

thanks

Miklos

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RE: [Asterisk-Users] dialing codes..( You can help! )

[Senad Jordanovic]

here is one phone company listing some of the codes...

i hope this will help you...

www.onetel.co.uk


senad
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[Asterisk-Users] error message

[listas iPfone]

Plese somebody knows what is this message :

*CLI> WARNING[65545]: File chan_sip.c, Line 435 (retrans_pkt): Maximum
retries exceeded on call [EMAIL PROTECTED] for
seqno 102 (Request)

It is happening all time

miklos

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Re: [Asterisk-Users] Segmentation Fault on reload (gdb output included)

[Steven Critchfield]

On Tue, 2003-09-23 at 12:04, jerk face wrote:
> I keep getting segmentation faults when I do a reload.
> 
> Here are the core file outputs from gdb:
> (I have three of them and they produce the same
> output)
> 
> (gdb) core core.6044

you need to run gdb like follows
gdb core.6044 `which asterisk`

This lets gdb load the asterisk binary with all the symbols in it to
understand the core file. Then if you could issue the command 'bt'
inside of gdb it will give a nice listing of functions called to get to
the point the crash happened.

> Core was generated by `asterisk'.
> Program terminated with signal 11, Segmentation fault.
> #0  0x401519fc in ?? ()
> 
> 
> I have no idea what that means, but if somebody could
> point me in the right direction, that would be great.
> 
> Thank you for your time.
> 
> __
> Do you Yahoo!?
> Yahoo! SiteBuilder - Free, easy-to-use web site design software
> http://sitebuilder.yahoo.com
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Re: [Asterisk-Users] Segmentation Fault on reload (gdb output included)

[Martin Pycko]

actually

gdb /usr/sbin/asterisk core.6044, sorry

On Tue, 23 Sep 2003, jerk face wrote:

> I keep getting segmentation faults when I do a reload.
>
> Here are the core file outputs from gdb:
> (I have three of them and they produce the same
> output)
>
> (gdb) core core.6044
> Core was generated by `asterisk'.
> Program terminated with signal 11, Segmentation fault.
> #0  0x401519fc in ?? ()
>
>
> I have no idea what that means, but if somebody could
> point me in the right direction, that would be great.
>
> Thank you for your time.
>
> __
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> Yahoo! SiteBuilder - Free, easy-to-use web site design software
> http://sitebuilder.yahoo.com
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>

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Re: [Asterisk-Users] Segmentation Fault on reload (gdb output included)

[Martin Pycko]

gdb /usr/src/asterisk core.6044

then 'bt'

Martin

On Tue, 23 Sep 2003, jerk face wrote:

> I keep getting segmentation faults when I do a reload.
>
> Here are the core file outputs from gdb:
> (I have three of them and they produce the same
> output)
>
> (gdb) core core.6044
> Core was generated by `asterisk'.
> Program terminated with signal 11, Segmentation fault.
> #0  0x401519fc in ?? ()
>
>
> I have no idea what that means, but if somebody could
> point me in the right direction, that would be great.
>
> Thank you for your time.
>
> __
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> Yahoo! SiteBuilder - Free, easy-to-use web site design software
> http://sitebuilder.yahoo.com
> ___
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>

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RE: [Asterisk-Users] dialing codes..( You can help! )

[Scott Stingel]

Yes, as Linus points out, each country has its own dialing scheme (with a
lot of inconsistencies), and each wholesale provider has its own billing
structure.  Usually the way this is done is to ask your wholesale provider
to give you their rates in digital form, like on a spreadhseet or something.
You can then use this as the basis for your charge structure.  Most of the
bigger wholesalers offer special rates to geographic zones (like big cities)
within larger countries - further complicating your price structure.  You
can, of course, simplify this in your billing, but you have to watch out for
things like premium numbers etc.

Regards
Scott

Scott M. Stingel 
Emerging Voice Technology Inc.

Email:  [EMAIL PROTECTED]    
URL:www.evtmedia.com    



> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut .
> Sent: Tuesday, September 23, 2003 4:32 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] dialing codes..( You can help! )
> 
> 
> Hi,
> 
> I am trying to setup some LCR functions on my Asterisk box 
> and have a cheap call provider that uses various different 
> numbers for landlines and cell phone numbers in various countrys..
> 
> I am finding it difficult to find the various codes..
> 
> eg.
> UK Landline - +44[12].
> UK Cell - +44[7].
> 
> SA Landline - +27[1-6].
> SA Cell - +27[78].
> 
> Please send me your country's dialing rules similar to how I 
> have represented them in the example and when I have a 
> listing I will be happy to post it for everyone..
> 
> Specific countries I need urgently are US, Canada, France and 
> Germany..
> 
> Thanks..
> 
> 
> -- 
> __
> http://www.linuxmail.org/
> Now with e-mail forwarding for only US$5.95/yr
> 
> Powered by Outblaze
> ___
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> 
> 

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Re: [Asterisk-Users] Windows Media Player Error

[Martin Pycko]

make sure the 'format=wav' in voicemail.conf

Martin

On Tue, 23 Sep 2003, Steve Totaro wrote:

> I am getting the following error in Windows Media Player Version 9 when listening to 
> voice mails.
>
> ClassFactory cannot supply requested class  (Error=80040111)
>
> Any ideas?  I tried searching the net but only found references to DivX.
>
> Thanks

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[Asterisk-Users] IAXTEL

[Ariel Batista]

Well as you know I am very new to this Asterisk.
I just joined gnophone for a IAX phone number.
Got my number and password and I have confirmed it.

So not that I have this how can I put this into my
Asterisk server and make it get calls to my extension?

I have read most of the pass post and done a lookup with
google for this.  But I still am missing where to start. 
I feel I need to do a register but where? and how? Thanks
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[Asterisk-Users] Port problem

[Paulo Mannheimer]

Hi All,

I have an equipment loaded with 4 X100P (numbered 1-4)) and one T400P
(numbered 5-8). Everything works fine except that I cannot use one of
the FXS ports (number 5). 

If I configure zapata.conf to recognize it, the whole system voice
quality suffers. I've tried already to switch PCI slots, with no
results.

Below is a snapshot of my /proc/interrupts, maybe this can shed some
light on the problem.
 
  0: 985385  XT-PIC  timer
  1:  3  XT-PIC  keyboard
  2:  0  XT-PIC  cascade
  3:9832048  XT-PIC  wcfxo
  4: 318730  XT-PIC  serial
  5:  0  XT-PIC  usb-uhci, usb-uhci, usb-uhci
  7:9832105  XT-PIC  wcfxo
  8:  1  XT-PIC  rtc
  9: 162893  XT-PIC  eth0
 10:9818599  XT-PIC  wcfxs
 11:   20891396  XT-PIC  wcfxo, wcfxo
 12: 36  XT-PIC  PS/2 Mouse
 14:  26399  XT-PIC  ide0
NMI:  0
LOC:  0
ERR:  0
MIS:  0

Any ideas?

PHM

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Re: [Asterisk-Users] Dial over IAX ahngs up after 3 rings

[Tais M. Hansen]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

On Tuesday 23 September 2003 16:55, Thomas Haeger wrote:
> can somebody explain this ?

Are you using commas (,)or pipes (|) to seperate your dial() arguments? I 
found recently that using commas instead of pipes screwed my IAX calls.

I.e.

exten => 1234,1,Dial(IAX2/user:[EMAIL PROTECTED]/[EMAIL PROTECTED]|30|r)

calls for 30 seconds, but

exten => 1234,1,Dial(IAX2/user:[EMAIL PROTECTED]/[EMAIL PROTECTED],30,r)

only calls for 3-5 seconds and then hangs up as if nobody answered.

- -- 
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.2 (GNU/Linux)

iD8DBQE/cIRT2TEAILET3McRAnwtAKCdCLenITIS84lLuP5VmfeuKFqS7ACdFqzo
ZrM5ycxNZjcrihqxrvU+vRs=
=mS3u
-END PGP SIGNATURE-

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Re: [Asterisk-Users] Call Pickup problem with cisco 7960 (SIP)

[James Sizemore]

Yes this is a known bug.

Manuel Marín García wrote:

Please help! When I try to place a call pickup from a cisco phone 7960
using *8 the call is picked up but the other phone continues ringing. Is
there any problem with call pickup in SIP.
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RE: [Asterisk-Users] dialing codes..( You can help! )

[Senad Jordanovic]

here is one phone company listing some of the codes...

i hope this will help you...

www.onetel.co.uk


senad

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of WipeOut .
Sent: 23 September 2003 16:32
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] dialing codes..( You can help! )


Hi,

I am trying to setup some LCR functions on my Asterisk box and have a cheap
call provider that uses various different numbers for landlines and cell
phone numbers in various countrys..

I am finding it difficult to find the various codes..

eg.
UK Landline - +44[12].
UK Cell - +44[7].

SA Landline - +27[1-6].
SA Cell - +27[78].

Please send me your country's dialing rules similar to how I have
represented them in the example and when I have a listing I will be happy to
post it for everyone..

Specific countries I need urgently are US, Canada, France and Germany..

Thanks..


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[Asterisk-Users] Cisco Callmanager 3.3 Asterisk OpenH323

[Michael Gschwandtner]

Hi,

i'm searching and trying, but can't get it working.

I'm trying to send calls from Cisco Callmanager to Asterisk with oh323 channel
driver. 
Therefor the asterisk is defined as a H323 Gateway in the Cisco Callmanager.

The Call comes from CCM to Asterisk and it works but i didn't get the called
number. This is needed because i want to make Voicemailboxes.

If i connect via gnomemeeting to asterisk (call://[EMAIL PROTECTED]) it works fine. 

Is there any way that i can send the called Number from CCM. I think it is
sent but in a different way.

My oh323.conf is:

[general]
listenAddress=0.0.0.0
listenPort=1720
connectPort=1720
tcpStart=1
tcpEnd=2
udpStart=1
udpEnd=2

fastStart=no
h245Tunnelling=yes
h245inSetup=yes

inBandDTMF=no
silenceSuppression=no
jitterMin=20
jitterMax=100

ipTos=none
outboundMax=10
inboundMax=10

wrapLibTraceLevel=1
libTraceLevel=1
libTraceFile=stdout
gatekeeper=DISABLE
userInputMode=TONE
amaFlags=default
accountCode=H323
context=voip-h323

[register]
alias=3001

[codecs]
codec=G711A
frames=20

Hope anyone can help me.

Thanks in Advance

Mike


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RE: [Asterisk-Users] Question about dialogic hardware

[Paul Crick]

[Insert regular whinge about HTML formatted email here]

> 1. D/120JCT-LS card with 12 ports. This ports are FXS ports?
Depends from which side you're looking? From the telco side, yes - The
D/120JCT-LS card is like 12 regular telephones. You plug 12 loop start phone
lines in to this card and it provides termination.

> 2. It is true that "Dialogic drivers cost of $15 per channel" ?
Yup. That's why we like cards from Digium - they're cheaper and better
supported.

> 3. Can I use this hardware with asterisk (for E1-ISDN
> using Wildcard E400P) ?
The card provides analogue line interfaces - I'm not sure what configuration
you're trying to achieve?

> 4. Anyone with experiance can tell me  how they work and can
> provide a configuration example?
I've used similar Dialogic cards to provide IVR services under Windows but
not Linux. You need some software to make the cards do stuff - you can write
your own or use a scripting language or appgen. Check out CT-ADE from Intel
(previously known as VOS from Parity Software).

> 5. Other posible configuration for 1 E1-ISDN with 24-30 FXS
> ports?
You could use a TE410P card, have your E1 terminate on one port, then have
one or more channel banks connected to the other ports to provide FXS ports.
It's probably cheaper to find T1 channel banks, so you'd need a couple of
you needed 30 FXS.

Cheers
Paul

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RE: [Asterisk-Users] dialing codes..( You can help! )

[Paul Crick]

> Please send me your country's dialing rules similar to how
> I have represented them in the example and when I have a
> listing I will be happy to post it for everyone..
I was thinking about this a while back. Wouldn't it be nice to have all this
stuff online, in one consolidated place, kept up to date? I'm dreaming, I
know, but why wouldn't it be possible? Telcos seem to have some information,
they can tell you that +1604 is Vancouver BC and +441xx is "UNITED KINGDOM",
but why not have all the information available to everyone, so we know that
+441256 is "Basingstoke, UK" for example? Mammoth task? Or not really?

I guess a lot is available online.. I know the North American area code
assignment is available from NANPA and the UK one from Oftel. What about
other countries?

> Specific countries I need urgently are US, Canada, France
> and Germany..
I can tell you that in the US and Canada there is no distinction between
geographic landlines and mobile phone numbers. For example, +1 604 257 
are landlines, and +1 604 339  are mobiles. Locally these are all dialed
as 10 digit numbers and are free calls - called party pays in North America,
as opposed to calling party pays in the rest of the world.

I'd be up for setting up some kind of website/database thing for collating
all this information, just not sure of the value and if anyone else would be
up for it/contributing data? Be cool to have though, and nice for customer
bill presentation etc?

Paul

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Re: [Asterisk-Users] dialing codes..( You can help! )

[Ernest W. Lessenger]

At 08:32 AM 9/23/2003, you wrote:
I am trying to setup some LCR functions on my Asterisk box and have a 
cheap call provider that uses various different numbers for landlines and 
cell phone numbers in various countrys..
First off, you need to know that these codes are proprietary, not 
standardized, but there is a de-facto standard set. That said, the numbers 
you've given so far match the country codes provided to me by NuFone. If 
Jeremy is willing he can provide them for you.

Along those lines, I've created a PERL script that adds a cost-for-call to 
the standard asterisk CDRs. I am interested in whether anyone has come up 
with a better solution. If you want the script, drop me a note.

--Ernest 

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Re: [Asterisk-Users] Cisco 7960 SIP Firmware.

[Sean Figgins]

Hi Jose.

Your University undoubtibly has Cisco gear and a contract with Cisco.  You
should be able to get the SIP firmware from CCO using your existing
account.  If you don't have one, someone athe University should have one.

The 7960G words fine.  I have several here, and they are as functional as
any other phone I have used.

-Sean

On Tue, 23 Sep 2003, Jose Ildefonso Camargo Tolosa wrote:

> Hi!
>
> The university where I work just bought four Cisco 7960G IP phones (they
> didn't ask, just came across the door and gave me a box and told me:
> "Can you make this work with the Asterisk PBX we have?").  According to
> what I read, there is no much hope, because I have not the SIP firmware
> (too bad).  Has anybody succesfully got an answer from cisco?, or does
> anybody happend to have that firmware and can kindly send it to me?
>
> Thanks in advance for your help,
>
> Sincerely,
>
> Ildefonso Camargo
> [EMAIL PROTECTED]
> Network Administrator.
>
>
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Re: [Asterisk-Users] error message playing .mp3

[Steve Totaro]

Thanks from me as well.  I had the same error but just gave up on that mp3
and used another one that worked.  Now I know how to fix it.

As for replies like:
"Learn to create a new thread.

Make sure the file is i the correct directory. The error message says
the file doesn't exist."

They offer no help whatsoever and really start to get on my nerves.


- Original Message - 
From: "Adams, Gavin" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, September 23, 2003 10:02 AM
Subject: RE: [Asterisk-Users] error message playing .mp3


> -Original Message-
> From: Steven Critchfield [mailto:[EMAIL PROTECTED]
> On Tue, 2003-09-23 at 07:31, listas iPfone wrote:
> > Hi All
> >
> > Somebody knows why asterisk gives me that error wile playing .mp3
files?
> >
> > The files play well but the message aperas any way:


> Make sure the file is i the correct directory. The error message says
> the file doesn't exist.

How would * be able to play the file if it didn't exist? MP3 playing
code is touchy on more... "robust" MP3 files. Gavin's helpful hints:

1) Capture or down-convert to <= 160Kbps data rate
2) Mono
3) Remove all ID3 tags
4) Remove extraneous information such as album art, etc.

Basically, use sox and LAME to strip the MP3 file to bare minimum.

Also, I've found that some of the auto-magic Music-on-Hold (MOH) playing
code doesn't always work, or throws exceptions/core dumps in the mohmp3
directory. I use the following bits in my extensions.conf to insure
proper playback:

exten => s,1,Answer
exten => s,2,SetMusicOnHold,random; or whatever MOH defined in
musiconhold.conf
exten => s,3,Dial ; or other commands after that

[snip - top-posting response deleted]

Regards,

--- Gavin
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Re: AW: [Asterisk-Users] Dial over IAX ahngs up after 3 rings

[Eric Wieling]

This is indicating that the telco disconnected your call for some
reason. You can see that Asterisk is dialing "033283077731" and then the
telco disconnects the call.

On Tue, 2003-09-23 at 10:35, Thomas Haeger wrote:
> and here from the other side:
> 
> -- Accepting AUTHENTICATED call from 217.81.111.2, requested format = 8,
> actual format = 8
> -- Executing SetCallerID("[EMAIL PROTECTED]:4569]/1",
> "033283077731") in new stack
> -- Executing Dial("[EMAIL PROTECTED]:4569]/1",
> "Zap/g3/033283077731") in new stack
> -- Called g3/033283077731
> -- Channel 1, span 3 got hangup
> -- Hungup 'Zap/63-1'
>   == No one is available to answer at this time
> -- Executing Hangup("[EMAIL PROTECTED]:4569]/1", "") in new
> stack
>   == Spawn extension (voipout, 99033283077731, 3) exited non-zero on
> '[EMAIL PROTECTED]:4569]/1'
> -- Hungup '[EMAIL PROTECTED]:4569]/1'
> 
> 
> -Ursprüngliche Nachricht-
> Von: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Auftrag von Steven
> Critchfield
> Gesendet: Dienstag, 23. September 2003 17:13
> An: [EMAIL PROTECTED]
> Betreff: Re: [Asterisk-Users] Dial over IAX ahngs up after 3 rings
> 
> 
> On Tue, 2003-09-23 at 09:55, Thomas Haeger wrote:
> > Hi all,
> > can somebody explain this ?
> 
> Do you have something like a |15 in the dial string?
> 
> Do you have logs to show what asterisk did?
> --
> Steven Critchfield <[EMAIL PROTECTED]>
> 
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This message has been 'sanitized'.  This means that potentially
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Sanitizer (start="1064336926"):
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Re: AW: [Asterisk-Users] Dial over IAX ahngs up after 3 rings

[Martin Pycko]

The call does not get compleated on the PRI so you should check the "pri
debug span 1" on your 2nd box.

regards
Martin

On Tue, 23 Sep 2003, Thomas Haeger wrote:

> I have tried it with a timeout and without...
>
> here the * output for the first side:
>
>  -- Starting simple switch on 'Zap/3-1'
> -- Executing Dial("Zap/3-1",
> "IAX2/useranme:[EMAIL PROTECTED]/99033283077731") in new stack
> -- Called thaeger:[EMAIL PROTECTED]/99033283077731
> -- Call accepted by 62.180.50.212 (format ALAW)
> -- Format for call is ALAW
> -- Hungup 'IAX2[62.180.50.212:4569]/2'
>   == No one is available to answer at this time
> -- Executing Hangup("Zap/3-1", "") in new stack
>   == Spawn extension (guersel, 033283077731, 2) exited non-zero on 'Zap/3-1'
> -- Hungup 'Zap/3-1'
>
>
>
> and here from the other side:
>
> -- Accepting AUTHENTICATED call from 217.81.111.2, requested format = 8,
> actual format = 8
> -- Executing SetCallerID("[EMAIL PROTECTED]:4569]/1",
> "033283077731") in new stack
> -- Executing Dial("[EMAIL PROTECTED]:4569]/1",
> "Zap/g3/033283077731") in new stack
> -- Called g3/033283077731
> -- Channel 1, span 3 got hangup
> -- Hungup 'Zap/63-1'
>   == No one is available to answer at this time
> -- Executing Hangup("[EMAIL PROTECTED]:4569]/1", "") in new
> stack
>   == Spawn extension (voipout, 99033283077731, 3) exited non-zero on
> '[EMAIL PROTECTED]:4569]/1'
> -- Hungup '[EMAIL PROTECTED]:4569]/1'
>
>
> -Ursprüngliche Nachricht-
> Von: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Auftrag von Steven
> Critchfield
> Gesendet: Dienstag, 23. September 2003 17:13
> An: [EMAIL PROTECTED]
> Betreff: Re: [Asterisk-Users] Dial over IAX ahngs up after 3 rings
>
>
> On Tue, 2003-09-23 at 09:55, Thomas Haeger wrote:
> > Hi all,
> > can somebody explain this ?
>
> Do you have something like a |15 in the dial string?
>
> Do you have logs to show what asterisk did?
> --
> Steven Critchfield <[EMAIL PROTECTED]>
>
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>
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Re: AW: [Asterisk-Users] Dial over IAX ahngs up after 3 rings

[Steven Critchfield]

On Tue, 2003-09-23 at 10:35, Thomas Haeger wrote:
> I have tried it with a timeout and without...

> -- Executing Dial("[EMAIL PROTECTED]:4569]/1",
> "Zap/g3/033283077731") in new stack
> -- Called g3/033283077731
> -- Channel 1, span 3 got hangup
> -- Hungup 'Zap/63-1'
>   == No one is available to answer at this time

I don't want to beat up on you too much, but it looks like you have a
signaling problem on your zaptel device. When this hangs up withe the
error that no one is available, it terminates the call. If you take the
Zap part of the dial string out, it should not hangup after 3 rings.  

-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] dialing codes..( You can help! )

[Eric Wieling]

In the USA and Canada there are no specific area codes for cell .vs.
landline.  In the USA most places have have flat rate calling so the
caller doesn't really care if they are calling a landline or a cell
phone.  There are cellphone prefixes in each area code, but I'm not
aware that anyone uses them for LCR.

(USA "area code" is sort of like an EU "city code")

On Tue, 2003-09-23 at 10:32, WipeOut . wrote:
> Hi,
> 
> I am trying to setup some LCR functions on my Asterisk box and have a cheap call 
> provider that uses various different numbers for landlines and cell phone numbers in 
> various countrys..
> 
> I am finding it difficult to find the various codes..
> 
> eg.
> UK Landline - +44[12].
> UK Cell - +44[7].
> 
> SA Landline - +27[1-6].
> SA Cell - +27[78].
> 
> Please send me your country's dialing rules similar to how I have represented them 
> in the example and when I have a listing I will be happy to post it for everyone..
> 
> Specific countries I need urgently are US, Canada, France and Germany..
> 
> Thanks..
-- 
Sample configs and more: http://www.fnords.org/~eric/asterisk/

BTEL Consulting
+1-850-484-4535 x2111 (Pensacola)
+1-504-595-3916 x2111 (New Orleans)
+1-877-677-9643 x2111 (Toll Free)
This message has been 'sanitized'.  This means that potentially
dangerous content has been rewritten or removed.  The following
log describes which actions were taken.

Sanitizer (start="1064336799"):
  Forcing message to be multipart/mixed, to facilitate logging.
  Writer (pos="861"):
Part (pos="934"):
  Added 1 bytes of scratch space.

Total modifications so far: 1
Added 1 bytes of scratch space.


Anomy 0.0.0 : Sanitizer.pm
$Id: Sanitizer.pm,v 1.79 2003/06/19 19:22:00 bre Exp $

[Asterisk-Users] Segmentation Fault on reload (gdb output included)

[jerk face]

I keep getting segmentation faults when I do a reload.

Here are the core file outputs from gdb:
(I have three of them and they produce the same
output)

(gdb) core core.6044
Core was generated by `asterisk'.
Program terminated with signal 11, Segmentation fault.
#0  0x401519fc in ?? ()


I have no idea what that means, but if somebody could
point me in the right direction, that would be great.

Thank you for your time.

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FW: [Asterisk-Users] asterisk call waiting X100P -> MGCP ata 186

[Chad Graham]

I am running CVS-09/11/03-14:03 on Redhat 9.0

Trying to get call waiting / call waiting callerid working.
The setup is:
X100P asterisk -> ATA 186 MGCP --> analog phone.

What changes to I need to make to my mgcp.conf and extensions.conf file to
allow answering of Call waiting calls?  And how do you answer call waiting
calls with the system.

I have usecallerid=yes, hidecallerid=no, callwaiting=yes,
callwaitingcallerid=yes, threewaycalling=yes, transfer=yes,
cancallforward=yes, callreturn=yes, echocalcel=yes and
echocancelwhenbridged=yes in the zapta.conf file.

How do you use these features?  Or where can I find some documentation.

I found some information about *0 on the thread but I couldn't find the
corresponding config information.  Also, is thier a list of functions like
*0 and how to use them.

Also, on another note.  Using this setup to make normal calls I am getting
significant echo for the first 15-20 seconds of a call and then it goes
away.  I assume this is the echo canceller but is their any way to fix this.

Thanks for your help.
Chad Graham





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[Asterisk-Users] Windows Media Player Error

[Steve Totaro]

I am getting the following error in Windows Media 
Player Version 9 when listening to voice mails.
 
ClassFactory cannot supply requested class  
(Error=80040111)
 
Any ideas?  I tried searching the net but only 
found references to DivX.
 
Thanks

Re: [Asterisk-Users] dialing codes..( You can help! )

[Linus Surguy]

> I am trying to setup some LCR functions on my Asterisk box and have a
cheap call provider that uses various different numbers for landlines and
cell phone numbers in various countrys..
>
> I am finding it difficult to find the various codes..
>
> eg.
> UK Landline - +44[12].
> UK Cell - +44[7].

You have to be careful to be sure exactly what you are trying to obtain, the
UK for example is:

441,442 - Geographic numbering
443,444 - not in use
44500 - Freephone
445 - 'multimedia/corporate' numbering, not much in use.
446 - not in use
4470 - UK personal 'follow me' numbering
4476 - UK Paging
4477/78/79 - UK Mobile
44800,44808 - Freephone
448 - Other, general non-geographic numbering, often used for company main
numbering, eg. 0870
449 - UK Premium rate

However, this doesnt necessarily tie in with what your carrier(s) will be
charging or routing on.



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RE: [Asterisk-Users] Dial over IAX ahngs up after 3 rings

[Thomas Haeger]

I have tried it with a timeout and without...

here the * output for the first side:

 -- Starting simple switch on 'Zap/3-1'
-- Executing Dial("Zap/3-1",
"IAX2/useranme:[EMAIL PROTECTED]/99033283077731") in new stack
-- Called thaeger:[EMAIL PROTECTED]/99033283077731
-- Call accepted by 62.180.50.212 (format ALAW)
-- Format for call is ALAW
-- Hungup 'IAX2[62.180.50.212:4569]/2'
  == No one is available to answer at this time
-- Executing Hangup("Zap/3-1", "") in new stack
  == Spawn extension (guersel, 033283077731, 2) exited non-zero on 'Zap/3-1'
-- Hungup 'Zap/3-1'



and here from the other side:

-- Accepting AUTHENTICATED call from 217.81.111.2, requested format = 8,
actual format = 8
-- Executing SetCallerID("[EMAIL PROTECTED]:4569]/1",
"033283077731") in new stack
-- Executing Dial("[EMAIL PROTECTED]:4569]/1",
"Zap/g3/033283077731") in new stack
-- Called g3/033283077731
-- Channel 1, span 3 got hangup
-- Hungup 'Zap/63-1'
  == No one is available to answer at this time
-- Executing Hangup("[EMAIL PROTECTED]:4569]/1", "") in new
stack
  == Spawn extension (voipout, 99033283077731, 3) exited non-zero on
'[EMAIL PROTECTED]:4569]/1'
-- Hungup '[EMAIL PROTECTED]:4569]/1'

-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Steven
Critchfield
Gesendet: Dienstag, 23. September 2003 17:13
An: [EMAIL PROTECTED]
Betreff: Re: [Asterisk-Users] Dial over IAX ahngs up after 3 rings


On Tue, 2003-09-23 at 09:55, Thomas Haeger wrote:
> Hi all,
> can somebody explain this ?

Do you have something like a |15 in the dial string?

Do you have logs to show what asterisk did?
--
Steven Critchfield <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Dial over IAX ahngs up after 3 rings

[Thomas Haeger]

I have tried it with a timeout and without...

here the * output for the first side:

 -- Starting simple switch on 'Zap/3-1'
-- Executing Dial("Zap/3-1",
"IAX2/useranme:[EMAIL PROTECTED]/99033283077731") in new stack
-- Called thaeger:[EMAIL PROTECTED]/99033283077731
-- Call accepted by 62.180.50.212 (format ALAW)
-- Format for call is ALAW
-- Hungup 'IAX2[62.180.50.212:4569]/2'
  == No one is available to answer at this time
-- Executing Hangup("Zap/3-1", "") in new stack
  == Spawn extension (guersel, 033283077731, 2) exited non-zero on 'Zap/3-1'
-- Hungup 'Zap/3-1'



and here from the other side:

-- Accepting AUTHENTICATED call from 217.81.111.2, requested format = 8,
actual format = 8
-- Executing SetCallerID("[EMAIL PROTECTED]:4569]/1",
"033283077731") in new stack
-- Executing Dial("[EMAIL PROTECTED]:4569]/1",
"Zap/g3/033283077731") in new stack
-- Called g3/033283077731
-- Channel 1, span 3 got hangup
-- Hungup 'Zap/63-1'
  == No one is available to answer at this time
-- Executing Hangup("[EMAIL PROTECTED]:4569]/1", "") in new
stack
  == Spawn extension (voipout, 99033283077731, 3) exited non-zero on
'[EMAIL PROTECTED]:4569]/1'
-- Hungup '[EMAIL PROTECTED]:4569]/1'

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RE: [Asterisk-Users] Advantage of Cisco 7960 with 5.x firmware

[Matthew Hardeman]

I've found there are some bugs they don't list (bugs of great severity)
that are fixed in the latest release that can cause trouble in certain
environments.  The phone's handling of ICMP redirects in a multihomed
Ethernet environment (two separate, exclusive subnets running on a
single segment) is very flaky, and frequently will result in the phone
crashing and rebooting.  The latest release seems to mitigate this in
most instances, though I actually found the best solution (sadly) was to
prevent my gateway from sending ICMP redirects to the Cisco phones.

The 5.x+ stuff is annoying, due to the whole code-signing issue.  It's
kind of anti-open-architecture...  On the other hand, there aren't any
non-Cisco firmware builds for these phones floating around out there...
Would you even want one?  Cisco implemented the code signing enforcement
as a response to a security analysis of the phones that pointed to the
ability to make the phone run arbitrary code via TFTP being a security
risk.  That risk is no longer.  I have mixed feelings about it, but have
no regrets in having deployed the 5.x solutions in my business.

Matt Hardeman
PaperSoft


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Tuesday, September 23, 2003 10:23 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Advantage of Cisco 7960 with 5.x firmware

their really isn't much fixed between 4.4 and the 5.x stuff but at the
time thats all I had.  So I put that on the phone.  So far everything
works like a champ.  Not one problem.

4.4
http://www.cisco.com/en/US/products/sw/voicesw/ps2156/prod_release_note0
9186a008016096f.html#63943

Resolved Caveats.Release 5.0
No resolved caveats specific to Cisco IP Phone 7940/7960 Release 5.0
require documentation in these release notes.

Resolved Caveats.Release 5.1
All caveats listed in this section are resolved in Cisco IP Phone
7940/7960 Release 5.1. This section lists only severity 1 and 2 caveats
and select severity 3 caveats

CSCdz59328: SIPPhone: The UI responsiveness slow, fast fingers cause
digit
drop
CSCdz77783: SIPPhone: Clipping of voice in 7960 SIP phone
CSCea83100: SIP: Dialing # does not work correctly if dialplan is empty
CSCea85697: Phone may fail to reset when an Exception occurs
CSCea93250: SIPPhone: Dialing # does not always work if default rule
missing
CSCeb27906: SIPPhone: Null To-tag in REFER causes transfer fail (race
condition)
CSCeb29575: SIPPhone: NOTIFY Event header shortform is not supported
(o:)


Resolved Caveats.Release 5.2
All caveats listed in this section are resolved in Cisco IP Phone
7940/7960 Release 5.2. This section lists only severity 1 and 2 caveats
and select severity 3 caveats

CSCeb41335: DSP mismatch with upgrade failure
CSCeb44769: Phone removes dots in the IP address when sending ACK
CSCeb46028: 79x0 Memory leak issues related to DNS query failures
CSCeb75975: Phone crashes upon any menu exit


Resolved Caveats.Release 5.3
All caveats listed in this section are resolved in Cisco IP Phone
7940/7960 Release 5.3. This section lists only severity 1 and 2 caveats
and select severity 3 caveats

CSCeb85936: SIP phone doesnt use the medium level contact field

Hope that helps.

bkw

On Tue, 23 Sep 2003, Peter Pauly wrote:

> I'm currently running firmware version 3.2 on my
> Cisco 7960. I've seen on the list that several
> people are running the 5.x latest versions.
>
> I've avoided going to higher firmware versions
> because I'm worried about potential problems
> or issues with the encryption mechanism used
> in the later firmware versions. (Once you
> go to an encrypted firmware version, you can't
> go back, right?)
>
> For those of you who have gone to the newer
> firmware, what features or benefits have
> you seen by going with the newer firmware?
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Re: [Asterisk-Users] RE: Asterisk stops responding

[Thomas Haeger]

I have tried it with a timeout and without...

here the * output for the first side:

 -- Starting simple switch on 'Zap/3-1'
-- Executing Dial("Zap/3-1",
"IAX2/useranme:[EMAIL PROTECTED]/99033283077731") in new stack
-- Called thaeger:[EMAIL PROTECTED]/99033283077731
-- Call accepted by 62.180.50.212 (format ALAW)
-- Format for call is ALAW
-- Hungup 'IAX2[62.180.50.212:4569]/2'
  == No one is available to answer at this time
-- Executing Hangup("Zap/3-1", "") in new stack
  == Spawn extension (guersel, 033283077731, 2) exited non-zero on 'Zap/3-1'
-- Hungup 'Zap/3-1'



and here from the other side:

-- Accepting AUTHENTICATED call from 217.81.111.2, requested format = 8,
actual format = 8
-- Executing SetCallerID("[EMAIL PROTECTED]:4569]/1",
"033283077731") in new stack
-- Executing Dial("[EMAIL PROTECTED]:4569]/1",
"Zap/g3/033283077731") in new stack
-- Called g3/033283077731
-- Channel 1, span 3 got hangup
-- Hungup 'Zap/63-1'
  == No one is available to answer at this time
-- Executing Hangup("[EMAIL PROTECTED]:4569]/1", "") in new
stack
  == Spawn extension (voipout, 99033283077731, 3) exited non-zero on
'[EMAIL PROTECTED]:4569]/1'
-- Hungup '[EMAIL PROTECTED]:4569]/1'

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RE: [Asterisk-Users] Dial over IAX ahngs up after 3 rings

[Thomas Haeger]

I have tried it with a timeout and without...

here the * output for the first side:

 -- Starting simple switch on 'Zap/3-1'
-- Executing Dial("Zap/3-1", "IAX2/useranme:[EMAIL PROTECTED]/99033283077731") in 
new stack
-- Called thaeger:[EMAIL PROTECTED]/99033283077731
-- Call accepted by 62.180.50.212 (format ALAW)
-- Format for call is ALAW
-- Hungup 'IAX2[62.180.50.212:4569]/2'
  == No one is available to answer at this time
-- Executing Hangup("Zap/3-1", "") in new stack
  == Spawn extension (guersel, 033283077731, 2) exited non-zero on 'Zap/3-1'
-- Hungup 'Zap/3-1'



and here from the other side:

-- Accepting AUTHENTICATED call from 217.81.111.2, requested format = 8, actual format 
= 8
-- Executing SetCallerID("[EMAIL PROTECTED]:4569]/1", "033283077731") in new stack
-- Executing Dial("[EMAIL PROTECTED]:4569]/1", "Zap/g3/033283077731") in new stack
-- Called g3/033283077731
-- Channel 1, span 3 got hangup
-- Hungup 'Zap/63-1'
  == No one is available to answer at this time
-- Executing Hangup("[EMAIL PROTECTED]:4569]/1", "") in new stack
  == Spawn extension (voipout, 99033283077731, 3) exited non-zero on '[EMAIL 
PROTECTED]:4569]/1'
-- Hungup '[EMAIL PROTECTED]:4569]/1'

-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Eric
Wieling
Gesendet: Dienstag, 23. September 2003 17:12
An: [EMAIL PROTECTED]
Betreff: Re: [Asterisk-Users] Dial over IAX ahngs up after 3 rings


It's not a general problem.  Can you give us more info like the Dial()
command and any output on the Asterisk console.  As good as some people
on this list are at solving problems, I'm not aware of any mindreaders
on the list.

On Tue, 2003-09-23 at 09:55, Thomas Haeger wrote:
> Hi all,
> can somebody explain this ?
> 
> Thanks,
> 
> Thomas.
> 
> ***
> beroNet technologies GmbH
> Dipl.- Ing. Thomas Häger
> Potsdamer Str. 18 A
> 14513 Teltow
> 
> FON:+49 (0) 3328 3077731
> FAX:+49 (0) 3328 334779
> Email:  [EMAIL PROTECTED]
> ***
> 
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Re: [Asterisk-Users] New Cisco "Color" Phone

[WipeOut .]

> I thought you guys would be interested to know:
> 
> eWeek has a short article about Cisco bringing out
> a new IP phone:  7970G. It has a high resolution
> color touch-screen display with support for XML and
> can act as a mini-browser to allow the development
> of vertical applications. 
> 
> But get this: the price will be $995. I don't think
> I'll be getting one any time soon. 

I guess they are still catering for the young, cash rich dot com ventures..

Has no one told them about the dot bomb..
or that there is currently a major slump in the economies of the world?? ;)

I guess they still work in the "Call it Cisco and people will buy" philosophy.. :)

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[Asterisk-Users] TDM400P port does not operate

[Jim Paraschou]

Hello

I have 3 TDM400p on a PC but one port on a certain
card does not operate and the light is off. I have two
power supplies on the pc one 300W and the other 200W.
Is it a hardware problem of the card or poor power
problem?
Thanks



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AW: [Asterisk-Users] Dial over IAX ahngs up after 3 rings

[Thomas Haeger]

I have tried it with a timeout and without...

here the * output for the first side:

 -- Starting simple switch on 'Zap/3-1'
-- Executing Dial("Zap/3-1",
"IAX2/useranme:[EMAIL PROTECTED]/99033283077731") in new stack
-- Called thaeger:[EMAIL PROTECTED]/99033283077731
-- Call accepted by 62.180.50.212 (format ALAW)
-- Format for call is ALAW
-- Hungup 'IAX2[62.180.50.212:4569]/2'
  == No one is available to answer at this time
-- Executing Hangup("Zap/3-1", "") in new stack
  == Spawn extension (guersel, 033283077731, 2) exited non-zero on 'Zap/3-1'
-- Hungup 'Zap/3-1'



and here from the other side:

-- Accepting AUTHENTICATED call from 217.81.111.2, requested format = 8,
actual format = 8
-- Executing SetCallerID("[EMAIL PROTECTED]:4569]/1",
"033283077731") in new stack
-- Executing Dial("[EMAIL PROTECTED]:4569]/1",
"Zap/g3/033283077731") in new stack
-- Called g3/033283077731
-- Channel 1, span 3 got hangup
-- Hungup 'Zap/63-1'
  == No one is available to answer at this time
-- Executing Hangup("[EMAIL PROTECTED]:4569]/1", "") in new
stack
  == Spawn extension (voipout, 99033283077731, 3) exited non-zero on
'[EMAIL PROTECTED]:4569]/1'
-- Hungup '[EMAIL PROTECTED]:4569]/1'


-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Steven
Critchfield
Gesendet: Dienstag, 23. September 2003 17:13
An: [EMAIL PROTECTED]
Betreff: Re: [Asterisk-Users] Dial over IAX ahngs up after 3 rings


On Tue, 2003-09-23 at 09:55, Thomas Haeger wrote:
> Hi all,
> can somebody explain this ?

Do you have something like a |15 in the dial string?

Do you have logs to show what asterisk did?
--
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[Asterisk-Users] Cisco Sip Gateway Register with *

[David Harris]

Is there anyway to get a cisco sip
gateway (3660) to register with asterisk?

 

Thanks,

/davidh

Re: [Asterisk-Users] Advantage of Cisco 7960 with 5.x firmware?

[Jamie Neil]

Peter Pauly wrote:

I'm currently running firmware version 3.2 on my
Cisco 7960. I've seen on the list that several 
people are running the 5.x latest versions. 

I've avoided going to higher firmware versions
because I'm worried about potential problems
or issues with the encryption mechanism used
in the later firmware versions. (Once you 
go to an encrypted firmware version, you can't
go back, right?)  

For those of you who have gone to the newer
firmware, what features or benefits have
you seen by going with the newer firmware?
I'm in the same boat - running 4.4 which is the last release before the 
signed code.

Nobody I've asked has mentioned any bugs that are not present in the pre 
5 code, but then again there don't seem to be any advantages either.

I would be interested to hear if there were some useful new features in 5.x.

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[Asterisk-Users] dialing codes..( You can help! )

[WipeOut .]

Hi,

I am trying to setup some LCR functions on my Asterisk box and have a cheap call 
provider that uses various different numbers for landlines and cell phone numbers in 
various countrys..

I am finding it difficult to find the various codes..

eg.
UK Landline - +44[12].
UK Cell - +44[7].

SA Landline - +27[1-6].
SA Cell - +27[78].

Please send me your country's dialing rules similar to how I have represented them in 
the example and when I have a listing I will be happy to post it for everyone..

Specific countries I need urgently are US, Canada, France and Germany..

Thanks..


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[Asterisk-Users] ix66 and asterisk domain

[listas iPfone]

Hi

I have an ix66 from intertex and use it with asterisk..it have a dyndns
custom domain registered and resolving.

My question is about setting up a domain for asterisk, how can i do it, i
can´t find info about. I have to install a dns server in my machine runing
redhat 8?

If someone have an ix66 please share info.

thanks!

Miklos

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[Asterisk-Users] Cisco 7960 SIP Firmware.

[Jose Ildefonso Camargo Tolosa]

Hi!

The university where I work just bought four Cisco 7960G IP phones (they 
didn't ask, just came across the door and gave me a box and told me: 
"Can you make this work with the Asterisk PBX we have?").  According to 
what I read, there is no much hope, because I have not the SIP firmware 
(too bad).  Has anybody succesfully got an answer from cisco?, or does 
anybody happend to have that firmware and can kindly send it to me?

Thanks in advance for your help,

Sincerely,

Ildefonso Camargo
[EMAIL PROTECTED]
Network Administrator.
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Re: [Asterisk-Users] Advantage of Cisco 7960 with 5.x firmware

[Brian West]

their really isn't much fixed between 4.4 and the 5.x stuff but at the
time thats all I had.  So I put that on the phone.  So far everything
works like a champ.  Not one problem.

4.4
http://www.cisco.com/en/US/products/sw/voicesw/ps2156/prod_release_note09186a008016096f.html#63943

Resolved Caveats.Release 5.0
No resolved caveats specific to Cisco IP Phone 7940/7960 Release 5.0
require documentation in these release notes.

Resolved Caveats.Release 5.1
All caveats listed in this section are resolved in Cisco IP Phone
7940/7960 Release 5.1. This section lists only severity 1 and 2 caveats
and select severity 3 caveats

CSCdz59328: SIPPhone: The UI responsiveness slow, fast fingers cause digit
drop
CSCdz77783: SIPPhone: Clipping of voice in 7960 SIP phone
CSCea83100: SIP: Dialing # does not work correctly if dialplan is empty
CSCea85697: Phone may fail to reset when an Exception occurs
CSCea93250: SIPPhone: Dialing # does not always work if default rule
missing
CSCeb27906: SIPPhone: Null To-tag in REFER causes transfer fail (race
condition)
CSCeb29575: SIPPhone: NOTIFY Event header shortform is not supported (o:)


Resolved Caveats.Release 5.2
All caveats listed in this section are resolved in Cisco IP Phone
7940/7960 Release 5.2. This section lists only severity 1 and 2 caveats
and select severity 3 caveats

CSCeb41335: DSP mismatch with upgrade failure
CSCeb44769: Phone removes dots in the IP address when sending ACK
CSCeb46028: 79x0 Memory leak issues related to DNS query failures
CSCeb75975: Phone crashes upon any menu exit


Resolved Caveats.Release 5.3
All caveats listed in this section are resolved in Cisco IP Phone
7940/7960 Release 5.3. This section lists only severity 1 and 2 caveats
and select severity 3 caveats

CSCeb85936: SIP phone doesnt use the medium level contact field

Hope that helps.

bkw

On Tue, 23 Sep 2003, Peter Pauly wrote:

> I'm currently running firmware version 3.2 on my
> Cisco 7960. I've seen on the list that several
> people are running the 5.x latest versions.
>
> I've avoided going to higher firmware versions
> because I'm worried about potential problems
> or issues with the encryption mechanism used
> in the later firmware versions. (Once you
> go to an encrypted firmware version, you can't
> go back, right?)
>
> For those of you who have gone to the newer
> firmware, what features or benefits have
> you seen by going with the newer firmware?
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[Asterisk-Users] Interesting product

[Steve Totaro]

http://www.multitech.com/PRODUCTS/Families/SocketSLIC/

Re: [Asterisk-Users] RE: Asterisk stops responding

[David C. Troy]

All --

I am starting to see a similar problem which appeared out of nowhere 
yesterday.  SIP registrations begin to fail and 'reload' does nothing from 
console.  I have not been able to identify a specific event that triggers 
it.  Only thing I can see when this condition is happening is many "UNKN" 
format SIP channels sitting dormant.

I am running CVS-09/04/03.  Has anyone identified a firm cause/fix for 
this problem?

Dave

=
David C. Troy   [EMAIL PROTECTED]   410-384-2500 Sales
ToadNet - Want to go fast?410-544-1329 FAX
570 Ritchie Highway, Severna Park, MD 21146-2925  www.toad.net

On Fri, 5 Sep 2003, Paul Cheng wrote:

> Update to latest CVS and check the bug report that I filed re:DTMF. 
> Your problem could be related. Latest CVS seems to fix the blocking 
> problem for me.
> 
> On Friday, September 5, 2003, at 01:15  AM, Andres wrote:
> 
> > It happened once again here.  This time I called an IVR (SIP to SIP) 
> > and upon
> > sending the 1st DTMF tone, * bombed out.  The console got filled with 
> > these
> > messages (and they wouldn't stop):
> >
> > DEBUG[4101]: File chan_sip.c, Line 5001 (sipsock_read): Failed to grab 
> > lock,
> > trying again...
> > DEBUG[4101]: File chan_sip.c, Line 5001 (sipsock_read): Failed to grab 
> > lock,
> > trying again...
> > DEBUG[4101]: File chan_sip.c, Line 5001 (sipsock_read): Failed to grab 
> > lock,
> > trying again...
> > DEBUG[4101]: File chan_sip.c, Line 5001 (sipsock_read): Failed to grab 
> > lock,
> > trying again...
> > DEBUG[4101]: File chan_sip.c, Line 5001 (sipsock_read): Failed to grab 
> > lock,
> > trying again..
> >
> > * stopped responding and I had to kill the process manually.
> > *CLI> show version
> > Asterisk CVS-08/22/03-22:24:05 built by [EMAIL PROTECTED] on a i686 running 
> > Linux
> >
> > Has anybody else seen this message?
> > Regards,
> > Andres
> >
> >
> > On Thursday 28 August 2003 13:37, Andres wrote:
> >> We run Iptel's SER as our SIP Server.  All subs register with our SIP
> >> Server, but if anyone needs to call the PSTN then the call gets 
> >> forwared to
> >> *.
> >>
> >> The "Request to schedule in the past"  messages have to do with MOH 
> >> and I
> >> was told it was due to a slow PC.  I don't think it is related with
> >> Asterisk hanging up.
> >>
> >> Regards,
> >> Andres
> >>
> >> On Thursday 28 August 2003 13:27, David Harris wrote:
>  Gazing at the console I was able to determine the exact time 
>  Asterisk
>  froze.
>  Even with DEBGUG on it did not show anything important.   The 
>  moment it
>  freezes is when a call from Phone1 tries to connect to a SIP 
>  Provider
> >>>
> >>> like
> >>>
>  Iconnect:
> >>>
> >>> I have not been able to pin point exactly what event causes the
> >>> freeze-up but I have been on the console when it has happened.  It
> >>> didn't print out anything interesting.  The call I was on cut off.
> >>>
>  Phone1Our SIP Server---Our AsteriskSIP Provider
> 
> 
>  It was by no means 100% reproducible.  Maybe 1 out of 10 calls 
>  caused
> >>>
> >>> the
> >>>
>  trouble.
> >>>
> >>> Same here except I would say more like 1 out of 100 calls.
> >>>
>  A bad symptom would be that the command "show sip channels"
>  would show several calls, even though they had hungup a long time 
>  ago.
> >>>
> >>> I definitely have this problem.
> >>>
>  Troubleshooting revealed that the BYE message was not being sent by 
>  our
> >>>
> >>> SIP
> >>>
>  Server to the Asterisk server upon hangup.  We rectified this and 
>  we no
>  longer see those phantom SIP Channels and Aterisk has not froze for
> >>>
> >>> about a >week.
> >>>
> >>> What is your "SIP Server" what does it do?  Maybe I have the same 
> >>> issue
> >>> with my Cisco Voice Gateway not sending the BYE message sometimes.  
> >>> But
> >>> would this cause asterisk to freeze?
> >>>
> >>>
> >>> Other "symptoms" I have are these errors in the asterisk messages log
> >>> file
> >>>
> >>> Aug 27 09:21:00 NOTICE[1081364]: File sched.c, Line 209 
> >>> (sched_settime):
> >>> Request to schedule in the past?!?!
> >>> Aug 27 09:21:24 NOTICE[1081364]: File sched.c, Line 209 
> >>> (sched_settime):
> >>> Request to schedule in the past?!?!
> >>> Aug 27 09:21:29 NOTICE[1081364]: File sched.c, Line 209 
> >>> (sched_settime):
> >>> Request to schedule in the past?!?!
> >>> Aug 27 09:21:35 NOTICE[1081364]: File sched.c, Line 209 
> >>> (sched_settime):
> >>> Request to schedule in the past?!?!
> >>> Aug 27 09:22:05 NOTICE[1081364]: File sched.c, Line 209 
> >>> (sched_settime):
> >>> Request to schedule in the past?!?!
> >>>
> >>> Thanks,
> >>> David Harris
> >>>
> >>>
> >>>
> >>> ___
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> >>> [EMAIL PROTECTED]
> >>> http://lists.digium.com/mailman/listinfo/asterisk-use

Re: [Asterisk-Users] Dial over IAX ahngs up after 3 rings

[Eric Wieling]

It's not a general problem.  Can you give us more info like the Dial()
command and any output on the Asterisk console.  As good as some people
on this list are at solving problems, I'm not aware of any mindreaders
on the list.

On Tue, 2003-09-23 at 09:55, Thomas Haeger wrote:
> Hi all,
> can somebody explain this ?
> 
> Thanks,
> 
> Thomas.
> 
> ***
> beroNet technologies GmbH
> Dipl.- Ing. Thomas Häger
> Potsdamer Str. 18 A
> 14513 Teltow
> 
> FON:+49 (0) 3328 3077731
> FAX:+49 (0) 3328 334779
> Email:  [EMAIL PROTECTED]
> ***
> 
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-- 
Sample configs and more: http://www.fnords.org/~eric/asterisk/

BTEL Consulting
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Re: [Asterisk-Users] Dial over IAX ahngs up after 3 rings

[Steven Critchfield]

On Tue, 2003-09-23 at 09:55, Thomas Haeger wrote:
> Hi all,
> can somebody explain this ?

Do you have something like a |15 in the dial string?

Do you have logs to show what asterisk did?
-- 
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[Asterisk-Users] New Cisco "Color" Phone

[Peter Pauly]

I thought you guys would be interested to know:

eWeek has a short article about Cisco bringing out
a new IP phone:  7970G. It has a high resolution
color touch-screen display with support for XML and
can act as a mini-browser to allow the development
of vertical applications. 

But get this: the price will be $995. I don't think
I'll be getting one any time soon. 
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Re: [Asterisk-Users] New kid on block

[PJ Welsh]

On Tue, Sep 23, 2003 at 08:54:50AM -0400, costas  wrote:
> Hi,
> 
> I am an experienced developer with Windows and familiar with Linux. I am looking for 
> a SIP solution.
> 
> 1) How does Asterisk compare to VOCAL in terms of support.

Sorry, don't know

> 
> 2) Is Asterisk free? 

yes

> 
> 3) Where are the docs? Or even better. Where do I start? 

Take a long read at the sites below. The www.voip-forum.org is VERY useful! Read both 
of the * Handbooks for asterisk.org. Spend a little time with the google search below 
with you qutestion.

http://www.voip-forum.org/
http://www.asterisk.org/index.php?menu=support
http://www.fnords.org/~eric/asterisk/
http://asterisk.gnuinter.net/
http://megaglobal.net/docs/asterisk/html/
http://home.cogeco.ca/~camstuff/
http://www.wwworks-inc.com/asterisk/
http://www.google.com/custom?q=&sa=Google+Search&cof=LW%3A40%3BL%3Ahttp%3A%2F%2Fwww.asterisk.org%2Fimages%2Ftopics%2Fasterisk.png%3BLH%3A40%3B%0D%0AAH%3Acenter%3BGL%3A0%3BS%3Ahttp%3A%2F%2Fwww.AsteriskPBX.org%3BAWFID%3Ad7bc203313616854%3B&domains=www.marko.net&sitesearch=www.marko.net


> 
> 4) Will it run on RH9?

My AGI's seem to go defunct on RH9

> 
> Thanks in advance.
> 
> Costas
> 
> --
> Costas Menico
> Meezon Software Corp
> 201-224-8111
> [EMAIL PROTECTED]
> 
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[Asterisk-Users] Advantage of Cisco 7960 with 5.x firmware?

[Peter Pauly]

I'm currently running firmware version 3.2 on my
Cisco 7960. I've seen on the list that several 
people are running the 5.x latest versions. 

I've avoided going to higher firmware versions
because I'm worried about potential problems
or issues with the encryption mechanism used
in the later firmware versions. (Once you 
go to an encrypted firmware version, you can't
go back, right?)  

For those of you who have gone to the newer
firmware, what features or benefits have
you seen by going with the newer firmware?
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[Asterisk-Users] Dial over IAX ahngs up after 3 rings

[Thomas Haeger]

Hi all,
can somebody explain this ?

Thanks,

Thomas.

***
beroNet technologies GmbH
Dipl.- Ing. Thomas Häger
Potsdamer Str. 18 A
14513 Teltow

FON:+49 (0) 3328 3077731
FAX:+49 (0) 3328 334779
Email:  [EMAIL PROTECTED]
***

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Re: [Asterisk-Users] Also CR Spam filters

[Steven Critchfield]

On Tue, 2003-09-23 at 09:13, Steven Critchfield wrote:
> Granted this is a brain dead simple script and most of you could have
> written it yourself. Please use this to make a point to this user, or at
> least their admin. I have used the address of this annoying user as both
> the to and from address. This should help it get through his C-R system
> so he will get the idea how broken it is. If it doesn't get through,
> maybe the admin will get annoyed enough at the logs/bounces to fix the
> problem. 

I realized after I sent this that the email address is not correct. Fix
this if you plan to use this app.

> #!/usr/bin/perl
> 
> use Mail::Sendmail;
> 
> my %fields = (
> 'To' => '[EMAIL PROTECTED]',
> 'From' => '[EMAIL PROTECTED]',
> 'Subject' => "Fix your broken challenge response spam annoyance",
> 'Message' => "Fix your challenge response system. You will receive
> more of these everytime I have to deal with one of your messages.  "
> );
> 
> for($i=0;$i<100;$i++){
> Mail::Sendmail::sendmail(%fields);
> }
> 
>  
-- 
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Re: [Asterisk-Users] dlink104S->asterisk->OH323->OpenPhone

[Michael Bielicki]

the codec in asterisk is g729a.

On Tuesday 23 September 2003 3:41 pm, Max Speransky wrote:
> On Tue, Sep 23, 2003 at 04:27:50PM +0300, Michael Manousos wrote:
> >Max Speransky wrote:
> >>Hi
> >>
> >>A have a setup like this:
> >>
> >>dlink104S->asterisk->OH323->OpenPhone
> >>
> >>And I have a codec problem with Dlink. When I make call from OpenPhone to
> >>dlink using G711 codec all works fine, but when I set g729 coded in
> >>OpenPhone, Asterisk tell me that cannot convert audio. So I need to setup
> >
> >What are the exact messages of Asterisk?
>
> Here it is:
>
> *CLI> WrapH323Connection::WrapH323Connection: WrapH323Connection created.
> -- Executing SetMusicOnHold("H323:20221", "default") in new stack
> -- Executing ResponseTimeout("H323:20221", "20") in new stack
> -- Set Response Timeout to 20
> -- Executing Dial("H323:20221", "MGCP/aaln/[EMAIL PROTECTED]|20|c") in new
> stack -- Called aaln/[EMAIL PROTECTED]
> -- MGCP/aaln/[EMAIL PROTECTED] is ringing
> -- MGCP/aaln/[EMAIL PROTECTED] answered H323:20221
> PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
> PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
> PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized.
> PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
> PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
> PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized.
> WARNING[1163287]: File channel.c, Line 1631 (ast_channel_make_compatible):
> No pa
> th to translate from H323:20221(256) to MGCP/aaln/[EMAIL PROTECTED](4)
> WARNING[1163287]: File channel.c, Line 1982 (ast_channel_bridge): Can't
> make H323:20221 and MGCP/aaln/[EMAIL PROTECTED] compatible
> WARNING[1163287]: File res_parking.c, Line 209 (ast_bridge_call): Bridge
> failed
> on channels H323:20221 and MGCP/aaln/[EMAIL PROTECTED]
>   == Spawn extension (mgcp, 103, 3) exited non-zero on 'H323:20221'
> ClearCallThread::ClearCallThread: Object initialized.
> -- Hungup 'H323:20221'
> 2:01:29.782 H225 Answer:8105018 H225Read error (0):
> PAsteriskSoundChannel::PAsteriskSoundChannel: Object deleted.
> PAsteriskAudioDelay::PAsteriskAudioDelay: Object deleted.
> PAsteriskAudioDelay::PAsteriskAudioDelay: Object deleted.
> PAsteriskSoundChannel::PAsteriskSoundChannel: Object deleted.
> PAsteriskAudioDelay::PAsteriskAudioDelay: Object deleted.
> PAsteriskAudioDelay::PAsteriskAudioDelay: Object deleted.
> 2:01:29.853H323 Cleaner H323Connection
> ip$62.244.62.23:1274/
> 20221 terminated.
> WrapH323Connection::WrapH323Connection: WrapH323Connection deleted.
> ClearCallThread::ClearCallThread: Object deleted.
>
> >>g729 codec in dlink. I type
> >>
> >>set coding 2 codec_type g729ab (is it similar to g729a ? )
> >>set coding 2 usage voice on
> >>set coding 2 encap rtp
> >>
> >>and nothing works ... May be somebody just configured dlink for using
> >> g729 codec ? Please, help me ;)
> >
> >Michael.
> >
> >___
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> >[EMAIL PROTECTED]
> >http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
Michael Bielicki
Managing Director
TAAN Consultants Ltd
http://www.global-gateway.net/

--

This correspondence is for the named person's use only. It may contain
confidential or legally privileged information or both. No confidentiality
or privilege is waived or lost by any mistransmission. If you receive this
correspondence in error, please immediately delete it from your system and
notify the sender. You must not disclose, copy or rely on any part of this
correspondence if you are not the intended recipient.

Any opinions expressed in this message are those of the individual sender.

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Re: [Asterisk-Users] dlink104S->asterisk->OH323->OpenPhone

[Eric Wieling]

4 = G.711 u-law
256 = G.729A audio

So what Asterisk is saying is that the H323 device is using G729 and the
MGCP device is using ulaw/G711.

Since Asterisk can't translate between those two codecs unless you have
a G729 license installed the call fails.

Do you specify the allowed codecs in your mgcp.conf file?

If the H323 device and the MGCP device both talk G729 I can't see why
Asterisk would need to transcode the CODEC part, but it might be an
issue because Asterisk is converting from H323 to MGCP.


On Tue, 2003-09-23 at 08:41, Max Speransky wrote:
> On Tue, Sep 23, 2003 at 04:27:50PM +0300, Michael Manousos wrote:
> >Max Speransky wrote:
> >>Hi
> >>
> >>A have a setup like this:
> >>
> >>dlink104S->asterisk->OH323->OpenPhone
> >>
> >>And I have a codec problem with Dlink. When I make call from OpenPhone to
> >>dlink using G711 codec all works fine, but when I set g729 coded in
> >>OpenPhone, Asterisk tell me that cannot convert audio. So I need to setup
> >
> >What are the exact messages of Asterisk?
> 
> Here it is:
> 
> *CLI> WrapH323Connection::WrapH323Connection: WrapH323Connection created.
> -- Executing SetMusicOnHold("H323:20221", "default") in new stack
> -- Executing ResponseTimeout("H323:20221", "20") in new stack
> -- Set Response Timeout to 20
> -- Executing Dial("H323:20221", "MGCP/aaln/[EMAIL PROTECTED]|20|c") in new stack
> -- Called aaln/[EMAIL PROTECTED]
> -- MGCP/aaln/[EMAIL PROTECTED] is ringing
> -- MGCP/aaln/[EMAIL PROTECTED] answered H323:20221
> PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
> PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
> PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized.
> PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
> PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
> PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized.
> WARNING[1163287]: File channel.c, Line 1631 (ast_channel_make_compatible):
> No pa
> th to translate from H323:20221(256) to MGCP/aaln/[EMAIL PROTECTED](4)
> WARNING[1163287]: File channel.c, Line 1982 (ast_channel_bridge): Can't make
> H323:20221 and MGCP/aaln/[EMAIL PROTECTED] compatible
> WARNING[1163287]: File res_parking.c, Line 209 (ast_bridge_call): Bridge
> failed
> on channels H323:20221 and MGCP/aaln/[EMAIL PROTECTED]
>   == Spawn extension (mgcp, 103, 3) exited non-zero on 'H323:20221'
> ClearCallThread::ClearCallThread: Object initialized.
> -- Hungup 'H323:20221'
> 2:01:29.782 H225 Answer:8105018 H225Read error (0):
> PAsteriskSoundChannel::PAsteriskSoundChannel: Object deleted.
> PAsteriskAudioDelay::PAsteriskAudioDelay: Object deleted.
> PAsteriskAudioDelay::PAsteriskAudioDelay: Object deleted.
> PAsteriskSoundChannel::PAsteriskSoundChannel: Object deleted.
> PAsteriskAudioDelay::PAsteriskAudioDelay: Object deleted.
> PAsteriskAudioDelay::PAsteriskAudioDelay: Object deleted.
> 2:01:29.853H323 Cleaner H323Connection
> ip$62.244.62.23:1274/
> 20221 terminated.
> WrapH323Connection::WrapH323Connection: WrapH323Connection deleted.
> ClearCallThread::ClearCallThread: Object deleted.
> 
> 
> 
> >
> >>g729 codec in dlink. I type 
> >>
> >>set coding 2 codec_type g729ab (is it similar to g729a ? )
> >>set coding 2 usage voice on
> >>set coding 2 encap rtp
> >>
> >>and nothing works ... May be somebody just configured dlink for using g729
> >>codec ? Please, help me ;)
> >>
> >
> >
> >Michael.
> >
> >___
> >Asterisk-Users mailing list
> >[EMAIL PROTECTED]
> >http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Sample configs and more: http://www.fnords.org/~eric/asterisk/

BTEL Consulting
+1-850-484-4535 x2111 (Pensacola)
+1-504-595-3916 x2111 (New Orleans)
+1-877-677-9643 x2111 (Toll Free)
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Re: [Asterisk-Users] Recommended OS

[Sean Figgins]

On Tue, 23 Sep 2003, Michael A. Miller wrote:

> Is there a recommended OS that Asterisk should be used with? I have been
> trying to get Asterisk running on Red Hat 9.0 with little success.

I have * running on Redhat 9.0.  It seems to work fine with SIP and my
Cisco 7960 phones running 5.x firmware.  I have been having a very
difficult time getting h.323 to work so I can connect to a remote Cisco
Callmanager, though.  Specifically, I don't seem to get h.323 to compile
correctly.

I woudl prefer to use FreeBSD, and there was some talk a couple weeks ago
about getting * ported for use there.  I haven't seen an update for a
while, though.

-Sean



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Re: [Asterisk-Users] Also CR Spam filters

[Steven Critchfield]

On Mon, 2003-09-22 at 13:49, Brian West wrote:
> [EMAIL PROTECTED] needs to fix their spam filter.  Please stop using it or
> learn to configure it.
> 
> +  1 Sep 22 AntiSpam UOL  (6828) RE:Re: [Asterisk-Users] MS Outlook
> 

Granted this is a brain dead simple script and most of you could have
written it yourself. Please use this to make a point to this user, or at
least their admin. I have used the address of this annoying user as both
the to and from address. This should help it get through his C-R system
so he will get the idea how broken it is. If it doesn't get through,
maybe the admin will get annoyed enough at the logs/bounces to fix the
problem. 


#!/usr/bin/perl

use Mail::Sendmail;

my %fields = (
'To' => '[EMAIL PROTECTED]',
'From' => '[EMAIL PROTECTED]',
'Subject' => "Fix your broken challenge response spam annoyance",
'Message' => "Fix your challenge response system. You will receive
more of these everytime I have to deal with one of your messages.  "
);

for($i=0;$i<100;$i++){
Mail::Sendmail::sendmail(%fields);
}

 

-- 
Steven Critchfield <[EMAIL PROTECTED]>

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RE: [Asterisk-Users] error message playing .mp3

[Adams, Gavin]

> -Original Message-
> From: Steven Critchfield [mailto:[EMAIL PROTECTED]
> On Tue, 2003-09-23 at 07:31, listas iPfone wrote:
> > Hi All
> >
> > Somebody knows why asterisk gives me that error wile playing .mp3
files?
> >
> > The files play well but the message aperas any way:


> Make sure the file is i the correct directory. The error message says
> the file doesn't exist.

How would * be able to play the file if it didn't exist? MP3 playing
code is touchy on more... "robust" MP3 files. Gavin's helpful hints:

1) Capture or down-convert to <= 160Kbps data rate
2) Mono
3) Remove all ID3 tags
4) Remove extraneous information such as album art, etc.

Basically, use sox and LAME to strip the MP3 file to bare minimum.

Also, I've found that some of the auto-magic Music-on-Hold (MOH) playing
code doesn't always work, or throws exceptions/core dumps in the mohmp3
directory. I use the following bits in my extensions.conf to insure
proper playback:

exten => s,1,Answer
exten => s,2,SetMusicOnHold,random; or whatever MOH defined in
musiconhold.conf
exten => s,3,Dial ; or other commands after that

[snip - top-posting response deleted]

Regards,

--- Gavin
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Re: [Asterisk-Users] dlink104S->asterisk->OH323->OpenPhone

[Michael Manousos]

Max Speransky wrote:
On Tue, Sep 23, 2003 at 04:27:50PM +0300, Michael Manousos wrote:

Max Speransky wrote:

Hi

A have a setup like this:

dlink104S->asterisk->OH323->OpenPhone

And I have a codec problem with Dlink. When I make call from OpenPhone to
dlink using G711 codec all works fine, but when I set g729 coded in
OpenPhone, Asterisk tell me that cannot convert audio. So I need to setup
What are the exact messages of Asterisk?


Here it is:

*CLI> WrapH323Connection::WrapH323Connection: WrapH323Connection created.
-- Executing SetMusicOnHold("H323:20221", "default") in new stack
-- Executing ResponseTimeout("H323:20221", "20") in new stack
-- Set Response Timeout to 20
-- Executing Dial("H323:20221", "MGCP/aaln/[EMAIL PROTECTED]|20|c") in new stack
-- Called aaln/[EMAIL PROTECTED]
-- MGCP/aaln/[EMAIL PROTECTED] is ringing
-- MGCP/aaln/[EMAIL PROTECTED] answered H323:20221
PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized.
PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized.
WARNING[1163287]: File channel.c, Line 1631 (ast_channel_make_compatible):
No pa
th to translate from H323:20221(256) to MGCP/aaln/[EMAIL PROTECTED](4)
WARNING[1163287]: File channel.c, Line 1982 (ast_channel_bridge): Can't make
H323:20221 and MGCP/aaln/[EMAIL PROTECTED] compatible
WARNING[1163287]: File res_parking.c, Line 209 (ast_bridge_call): Bridge
failed
on channels H323:20221 and MGCP/aaln/[EMAIL PROTECTED]
[snip]

It seems that Asterisk doesn't have the G.729 codec.
If you want the G.729 codec, you will have to buy it from Digium.



g729 codec in dlink. I type 

set coding 2 codec_type g729ab (is it similar to g729a ? )
set coding 2 usage voice on
set coding 2 encap rtp
and nothing works ... May be somebody just configured dlink for using g729
codec ? Please, help me ;)


Michael.



Michael.

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Re: [Asterisk-Users] THIS IS STRANGE

[Bartosz Jozwiak]

Right now it works great!
Thanks so much.

Could you tell me what is that: 
'canreinvite=no' in sip.conf ?

-- Bart

- Original Message - 
From: "Stephen Varga" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, September 23, 2003 8:53 AM
Subject: Re: [Asterisk-Users] THIS IS STRANGE


> On Mon, 2003-09-22 at 13:51, Bartosz Jozwiak wrote:
> >  
> > When I call directly to  X-Lite from ATA it doesn't work but when I
> > call to X-lie with queue from ATA is works...
> > It is really strange.
> 
> Try adding 'canreinvite=no' to the sip definition for both the X-lite
> phone and the ATA.
> 
> ___
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> http://lists.digium.com/mailman/listinfo/asterisk-users
> 

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Re: [Asterisk-Users] error message playing .mp3

[Steven Critchfield]

On Tue, 2003-09-23 at 07:31, listas iPfone wrote:
> Hi All
> 
> Somebody knows why asterisk gives me that error wile playing .mp3 files?
> 
> The files play well but the message aperas any way:

Learn to create a new thread.

Make sure the file is i the correct directory. The error message says
the file doesn't exist. 


> *CLI> -- Starting simple switch on 'Zap/1-1'
> NOTICE[131089]: File chan_zap.c, Line 4277 (ss_thread): Got event 2
> (Ring/Answered)...
> -- Executing Wait("Zap/1-1", "2") in new stack
> -- Executing Answer("Zap/1-1", "") in new stack
> -- Executing Playback("Zap/1-1", "pop") in new stack
> -- Playing 'pop'
> WARNING[131089]: File format_mp3.c, Line 120 (mp3_read): Short read (0 of 4
> bytes) (No such file or directory)!
> -- Executing Wait("Zap/1-1", "2") in new stack
> -- Executing Playback("Zap/1-1", "bemvindo") in new stack
> -- Playing 'bemvindo'
> WARNING[131089]: File format_mp3.c, Line 120 (mp3_read): Short read (0 of 4
> bytes) (No such file or directory)!
> -- Executing Wait("Zap/1-1", "2") in new stack
> -- Executing Playback("Zap/1-1", "acesse") in new stack
> -- Playing 'acesse'
> WARNING[131089]: File format_mp3.c, Line 120 (mp3_read): Short read (0 of 4
> bytes) (No such file or directory)!
> -- Executing Dial("Zap/1-1", "SIP/33&SIP/34&SIP/35|10") in new stack
> -- Called 33
> -- Called 34
> -- Called 35
> -- SIP/34-f869 is ringing
> -- SIP/33-47ac is ringing
> -- SIP/35-53cd is ringing
> -- SIP/34-f869 is ringing
> -- SIP/33-47ac is ringing
> -- SIP/34-f869 is ringing
> -- SIP/33-47ac is ringing
> -- SIP/34-f869 is ringing
> -- SIP/33-47ac is ringing
> -- SIP/33-47ac answered Zap/1-1
>   == Spawn extension (default, s, 8) exited non-zero on 'Zap/1-1'
> -- Hungup 'Zap/1-1'
> 
> 
> miklos
> 
> ___
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-- 
Steven Critchfield <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] dlink104S->asterisk->OH323->OpenPhone

[Max Speransky]

On Tue, Sep 23, 2003 at 04:27:50PM +0300, Michael Manousos wrote:
>Max Speransky wrote:
>>Hi
>>
>>A have a setup like this:
>>
>>dlink104S->asterisk->OH323->OpenPhone
>>
>>And I have a codec problem with Dlink. When I make call from OpenPhone to
>>dlink using G711 codec all works fine, but when I set g729 coded in
>>OpenPhone, Asterisk tell me that cannot convert audio. So I need to setup
>
>What are the exact messages of Asterisk?

Here it is:

*CLI> WrapH323Connection::WrapH323Connection: WrapH323Connection created.
-- Executing SetMusicOnHold("H323:20221", "default") in new stack
-- Executing ResponseTimeout("H323:20221", "20") in new stack
-- Set Response Timeout to 20
-- Executing Dial("H323:20221", "MGCP/aaln/[EMAIL PROTECTED]|20|c") in new stack
-- Called aaln/[EMAIL PROTECTED]
-- MGCP/aaln/[EMAIL PROTECTED] is ringing
-- MGCP/aaln/[EMAIL PROTECTED] answered H323:20221
PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized.
PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized.
WARNING[1163287]: File channel.c, Line 1631 (ast_channel_make_compatible):
No pa
th to translate from H323:20221(256) to MGCP/aaln/[EMAIL PROTECTED](4)
WARNING[1163287]: File channel.c, Line 1982 (ast_channel_bridge): Can't make
H323:20221 and MGCP/aaln/[EMAIL PROTECTED] compatible
WARNING[1163287]: File res_parking.c, Line 209 (ast_bridge_call): Bridge
failed
on channels H323:20221 and MGCP/aaln/[EMAIL PROTECTED]
  == Spawn extension (mgcp, 103, 3) exited non-zero on 'H323:20221'
ClearCallThread::ClearCallThread: Object initialized.
-- Hungup 'H323:20221'
2:01:29.782 H225 Answer:8105018 H225Read error (0):
PAsteriskSoundChannel::PAsteriskSoundChannel: Object deleted.
PAsteriskAudioDelay::PAsteriskAudioDelay: Object deleted.
PAsteriskAudioDelay::PAsteriskAudioDelay: Object deleted.
PAsteriskSoundChannel::PAsteriskSoundChannel: Object deleted.
PAsteriskAudioDelay::PAsteriskAudioDelay: Object deleted.
PAsteriskAudioDelay::PAsteriskAudioDelay: Object deleted.
2:01:29.853H323 Cleaner H323Connection
ip$62.244.62.23:1274/
20221 terminated.
WrapH323Connection::WrapH323Connection: WrapH323Connection deleted.
ClearCallThread::ClearCallThread: Object deleted.



>
>>g729 codec in dlink. I type 
>>
>>set coding 2 codec_type g729ab (is it similar to g729a ? )
>>set coding 2 usage voice on
>>set coding 2 encap rtp
>>
>>and nothing works ... May be somebody just configured dlink for using g729
>>codec ? Please, help me ;)
>>
>
>
>Michael.
>
>___
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-- 
... All opinions expressed are mine and not those of my employer.

Yours, Max   [Msg N 2266]
---
mailto: [EMAIL PROTECTED] phone: +380-44-2054455
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[Asterisk-Users] Question about dialogic hardware

[Cristian Vasiliu]

1. D/120JCT-LS card with 12 ports. This ports are FXS ports? 
2. It is true that "Dialogic drivers cost of $15 per channel" ?
3. Can I use this hardware with asterisk (for E1-ISDN using Wildcard
E400P) ?
4. Anyone with experiance can tell me  how they work and can provide a
configuration example? (2 DSP Motorola procesors  - I think it means 
something!!!) 
5. Other posible configuration for 1 E1-ISDN with 24-30 FXS ports?

P.S. Thanks for all answers!

Cristian Vasiliu
mail to:<[EMAIL PROTECTED]>
www :

Re: [Asterisk-Users] dlink104S->asterisk->OH323->OpenPhone

[Michael Manousos]

Max Speransky wrote:
Hi

A have a setup like this:

dlink104S->asterisk->OH323->OpenPhone

And I have a codec problem with Dlink. When I make call from OpenPhone to
dlink using G711 codec all works fine, but when I set g729 coded in
OpenPhone, Asterisk tell me that cannot convert audio. So I need to setup
What are the exact messages of Asterisk?

g729 codec in dlink. I type 

set coding 2 codec_type g729ab (is it similar to g729a ? )
set coding 2 usage voice on
set coding 2 encap rtp
and nothing works ... May be somebody just configured dlink for using g729
codec ? Please, help me ;)


Michael.

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Re: [Asterisk-Users] error message playing .mp3

[listas iPfone]

Thanks Gavin!

It works  now.

Miklos
- Original Message - 
From: "Adams, Gavin" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, September 23, 2003 9:51 AM
Subject: RE: [Asterisk-Users] error message playing .mp3


> -Original Message-
> From: listas iPfone [mailto:[EMAIL PROTECTED]
> 
> Somebody knows why asterisk gives me that error wile playing .mp3
files?
> 
> The files play well but the message aperas any way:
> WARNING[131089]: File format_mp3.c, Line 120 (mp3_read): Short read (0
of
> 4
> bytes) (No such file or directory)!

Listas,

You might try down-sampling the MP3 files to 160Kb/sec, mono through
LAME or some other MP3 encoder. Prior to converting some royalty-free
music from 320Kbs joint-stereo, mpg123/asterisk would barf on the file.
Assumeably due to the original encoding (EAC under Windows/LAME).

HTH,

--- Gavin

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Re: [Asterisk-Users] New kid on block

[WipeOut .]

Hi,

> Hi,
> 
> I am an experienced developer with Windows and familiar with Linux. I am looking for 
> a SIP solution.
> 
> 1) How does Asterisk compare to VOCAL in terms of support.
Digium are the best people to ask for issues related to commercial support, otherwise 
this mailing list or the IRC channel are good options for community support..

> 
> 2) Is Asterisk free? 
Yes..

> 
> 3) Where are the docs? Or even better. Where do I start? 
Look at www.digium.com on the documentation page.. I would recommend reading bothe the 
old and new handbooks..

> 
> 4) Will it run on RH9?
Yes, no problem.. you can take a look at my install guide for RH9 if you want a quick 
and easy installation..
members.lycos.co.uk/wipe_out/asterisk/

> 
> Thanks in advance.
No Problem..

Later..
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RE: [Asterisk-Users] Recommended OS

[Michael A. Miller]

Thanks for the link.

I have followed the installation instructions for the Netjet but I get a
number of errors when I recompile the kernel with ISDN support. I am sure
that it is something simple and is most likely not related to the Traverse
drivers. I have picked up a book to learn a bit more about RH9 and Linux in
general. Either way it can't hurt.

Regards,

Michael

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alastair Maw
Sent: Tuesday, September 23, 2003 9:08 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Recommended OS



Michael A. Miller wrote:
> Other than the generalized tools, most drivers and
> such are specific to the company that released that version as well as the
> hardware.

This is quite incorrect. But here is not really the place to discuss 
such things. Suffice to say that the vast majority of hardware support 
is provided by the Linux kernel. You'll only use distribution specific 
drivers if RedHat or SuSE or whoever deem them necessary (support for 
802.11g springs to mind).

What makes the distributions different is merely how they package their 
files (if at all) and how they update things, find patches 
automatically, what extra tools they have, what logo they've put on the 
start screen, etc. It's all academic. Provided you have recent enough 
versions of the libraries and compiler tools Asterisk needs installed, 
it will all work, no matter what distribution you're using. That's all 
that matters.

> I have gotten Asterisk to work. It was a knowledge issue on my end. My
main
> issue is getting a Netjet ISDN and voice modem installed correctly. Both
of
> which I am having issues with the drivers at the OS level and have not
> attempted any use with Asterisk at this point. If someone has experience
> with a Netjet install under RH9 or Lucent/Agere linmodem, I would
sincerely
> appreciate a bit of help.

This page may be of use:
  - http://www.traverse.com.au/downloads/drivers/

Welcome to the Asterisk community. :)

-- 
Alastair Maw
MX Telecom - Systems Analyst
http://www.mxtelecom.com

-- 
Alastair Maw <[EMAIL PROTECTED]>
MX Telecom - Systems Analyst
http://www.mxtelecom.com

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Re: [Asterisk-Users] New kid on block

[andrewg]

On Tue, Sep 23, 2003 at 08:54:50AM -0400, costas  wrote:
> Hi,
> 
> I am an experienced developer with Windows and familiar with Linux. I am looking for 
> a SIP solution.
> 
> 1) How does Asterisk compare to VOCAL in terms of support.

*shrug*

> 
> 2) Is Asterisk free? 
>

yeah
 
> 3) Where are the docs? Or even better. Where do I start? 

look for the handbook on the asterisk.org website

> 
> 4) Will it run on RH9?

yes. its easy, when you've got the required previous packages installed.

> 
> Thanks in advance.

no problems.

> 
> Costas
> 
> --
> Costas Menico
> Meezon Software Corp
> 201-224-8111
> [EMAIL PROTECTED]
> 
> --
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[Asterisk-Users] New kid on block

[costas]

Hi,

I am an experienced developer with Windows and familiar with Linux. I am looking for a 
SIP solution.

1) How does Asterisk compare to VOCAL in terms of support.

2) Is Asterisk free? 

3) Where are the docs? Or even better. Where do I start? 

4) Will it run on RH9?

Thanks in advance.

Costas

--
Costas Menico
Meezon Software Corp
201-224-8111
[EMAIL PROTECTED]

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RE: [Asterisk-Users] error message playing .mp3

[Adams, Gavin]

> -Original Message-
> From: listas iPfone [mailto:[EMAIL PROTECTED]
> 
> Somebody knows why asterisk gives me that error wile playing .mp3
files?
> 
> The files play well but the message aperas any way:
> WARNING[131089]: File format_mp3.c, Line 120 (mp3_read): Short read (0
of
> 4
> bytes) (No such file or directory)!

Listas,

You might try down-sampling the MP3 files to 160Kb/sec, mono through
LAME or some other MP3 encoder. Prior to converting some royalty-free
music from 320Kbs joint-stereo, mpg123/asterisk would barf on the file.
Assumeably due to the original encoding (EAC under Windows/LAME).

HTH,

--- Gavin

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[Asterisk-Users] dlink104S->asterisk->OH323->OpenPhone

[Max Speransky]

Hi

A have a setup like this:

dlink104S->asterisk->OH323->OpenPhone

And I have a codec problem with Dlink. When I make call from OpenPhone to
dlink using G711 codec all works fine, but when I set g729 coded in
OpenPhone, Asterisk tell me that cannot convert audio. So I need to setup
g729 codec in dlink. I type 

set coding 2 codec_type g729ab (is it similar to g729a ? )
set coding 2 usage voice on
set coding 2 encap rtp

and nothing works ... May be somebody just configured dlink for using g729
codec ? Please, help me ;)

-- 
Yours, Max   [Msg N 2265]
---
mailto: [EMAIL PROTECTED] phone: +380-44-2054455
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[Asterisk-Users] error message playing .mp3

[listas iPfone]

Hi All

Somebody knows why asterisk gives me that error wile playing .mp3 files?

The files play well but the message aperas any way:

*CLI> -- Starting simple switch on 'Zap/1-1'
NOTICE[131089]: File chan_zap.c, Line 4277 (ss_thread): Got event 2
(Ring/Answered)...
-- Executing Wait("Zap/1-1", "2") in new stack
-- Executing Answer("Zap/1-1", "") in new stack
-- Executing Playback("Zap/1-1", "pop") in new stack
-- Playing 'pop'
WARNING[131089]: File format_mp3.c, Line 120 (mp3_read): Short read (0 of 4
bytes) (No such file or directory)!
-- Executing Wait("Zap/1-1", "2") in new stack
-- Executing Playback("Zap/1-1", "bemvindo") in new stack
-- Playing 'bemvindo'
WARNING[131089]: File format_mp3.c, Line 120 (mp3_read): Short read (0 of 4
bytes) (No such file or directory)!
-- Executing Wait("Zap/1-1", "2") in new stack
-- Executing Playback("Zap/1-1", "acesse") in new stack
-- Playing 'acesse'
WARNING[131089]: File format_mp3.c, Line 120 (mp3_read): Short read (0 of 4
bytes) (No such file or directory)!
-- Executing Dial("Zap/1-1", "SIP/33&SIP/34&SIP/35|10") in new stack
-- Called 33
-- Called 34
-- Called 35
-- SIP/34-f869 is ringing
-- SIP/33-47ac is ringing
-- SIP/35-53cd is ringing
-- SIP/34-f869 is ringing
-- SIP/33-47ac is ringing
-- SIP/34-f869 is ringing
-- SIP/33-47ac is ringing
-- SIP/34-f869 is ringing
-- SIP/33-47ac is ringing
-- SIP/33-47ac answered Zap/1-1
  == Spawn extension (default, s, 8) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'


miklos

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Re: [Asterisk-Users] THIS IS STRANGE

[Stephen Varga]

On Mon, 2003-09-22 at 13:51, Bartosz Jozwiak wrote:
>  
> When I call directly to  X-Lite from ATA it doesn't work but when I
> call to X-lie with queue from ATA is works...
> It is really strange.

Try adding 'canreinvite=no' to the sip definition for both the X-lite
phone and the ATA.

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Re: [Asterisk-Users] Status of shipdate on the 4 port FX0 card?

[Bartosz Jozwiak]

I am also waiting for that card


- Original Message - 
From: "James Sizemore" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>; <[EMAIL PROTECTED]>
Sent: Monday, September 22, 2003 4:44 PM
Subject: [Asterisk-Users] Status of shipdate on the 4 port FX0 card?


> Does any-e-one know if the 4 port FX0 cards will
> be shipping anytime soon?
> 
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Re: [Asterisk-Users] Recommended OS

[Alastair Maw]

Michael A. Miller wrote:
Other than the generalized tools, most drivers and
such are specific to the company that released that version as well as the
hardware.
This is quite incorrect. But here is not really the place to discuss 
such things. Suffice to say that the vast majority of hardware support 
is provided by the Linux kernel. You'll only use distribution specific 
drivers if RedHat or SuSE or whoever deem them necessary (support for 
802.11g springs to mind).

What makes the distributions different is merely how they package their 
files (if at all) and how they update things, find patches 
automatically, what extra tools they have, what logo they've put on the 
start screen, etc. It's all academic. Provided you have recent enough 
versions of the libraries and compiler tools Asterisk needs installed, 
it will all work, no matter what distribution you're using. That's all 
that matters.

I have gotten Asterisk to work. It was a knowledge issue on my end. My main
issue is getting a Netjet ISDN and voice modem installed correctly. Both of
which I am having issues with the drivers at the OS level and have not
attempted any use with Asterisk at this point. If someone has experience
with a Netjet install under RH9 or Lucent/Agere linmodem, I would sincerely
appreciate a bit of help.
This page may be of use:
 - http://www.traverse.com.au/downloads/drivers/
Welcome to the Asterisk community. :)

--
Alastair Maw
MX Telecom - Systems Analyst
http://www.mxtelecom.com
--
Alastair Maw <[EMAIL PROTECTED]>
MX Telecom - Systems Analyst
http://www.mxtelecom.com
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Re: AW: [Asterisk-Users] how to dial a h323 destination ?

[Michael Manousos]

Thomas Haeger wrote:
OK. This is what i know too...
But this don't work. The gatekeeper tells me everytime "caller not
registered".
If i start *, the registration at the gatekeeper is ok.
If i make i call  it is not ok. Is there any other info that i have to
send with ?
Ask your provider about it.
Normally, you just use:
Dial(OH323/)

like : Dial(OH323/[EMAIL PROTECTED]/H323ID or similar like this ?

Thanks for help,

Thomas.


Michael.


-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Sergio
Serrano Revuelto
Gesendet: Dienstag, 23. September 2003 11:15
An: [EMAIL PROTECTED]
Betreff: RE: [Asterisk-Users] how to dial a h323 destination ?
exten=>XXX,1,Dial(h323/3|17|tTm)

srsergio



-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Thomas
Haeger
Enviado el: martes, 23 de septiembre de 2003 11:07
Para: [EMAIL PROTECTED]
Asunto: RE: [Asterisk-Users] how to dial a h323 destination ?
Please, can somebody tell me how do a h323 call correctly with the dial
app ?
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Thomas
Haeger
Gesendet: Montag, 22. September 2003 18:26
An: Asterisk User
Betreff: [Asterisk-Users] how to dial a h323 destination ?
Hi all,

i have big problems to make a h323 call over the gatekeeper from my
provider. The provider demanded following account data:
H323 ID:XXX-XXX-XX-X
DetinationNumer: XXX
I have configured the oh323.conf following:

gatekeeper=XX.XX.XXX.XXX
alias=XXX-XXX-XX-X
Isx the alias equal to the h323id ?

And how i have to make a call with the dial app ?

I have following config:

exten => _01099X.,1,Dial,OH323/${EXTEN:7}
exten => _01099X.,2,Hangup
I thought it would be enough when i give the destination number if i
registered at the gk, isn't it ? Or is  a ip and something like a
userbname necessary ? And if how can i dial so?
Can somebody help please ?

Thanks,

Thomas.

***
beroNet technologies GmbH
Dipl.- Ing. Thomas Häger
Potsdamer Str. 18 A
14513 Teltow
FON:+49 (0) 3328 3077731
FAX:+49 (0) 3328 334779
Email:  [EMAIL PROTECTED]
***
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RE: [Asterisk-Users] how to dial a h323 destination ?

[Sergio Serrano Revuelto]

When you register your * gw in gatekeeper you must say to gatekeeper
which are the number that you must redirect to your * gw. For example,
if you dial 555xx, you input in your oh323.conf must be like this:

[register]
alias=BER-BER-GW-1
Gwprefix=555

Regards,
srsergio



-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Thomas
Haeger
Enviado el: martes, 23 de septiembre de 2003 12:36
Para: [EMAIL PROTECTED]
Asunto: AW: [Asterisk-Users] how to dial a h323 destination ?


What is the gwprefix ? I try to connect the gk directly from our *
gw

-Ursprungliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Sergio
Serrano Revuelto
Gesendet: Dienstag, 23. September 2003 12:27
An: [EMAIL PROTECTED]
Betreff: RE: [Asterisk-Users] how to dial a h323 destination ?


Try to add gwprefix in oh323.conf after your alias. You must know that
you can configure * gw in gnugk.ini or in oh323.conf. I recommend you
put in your oh323.conf.

srsergio



-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Thomas
Haeger Enviado el: martes, 23 de septiembre de 2003 12:07
Para: [EMAIL PROTECTED]
Asunto: RE: [Asterisk-Users] how to dial a h323 destination ?


Here is my oh323.conf ...


;
; Configuration file of OpenH323 channel driver
;

;-
; General configuration options
; (ports, jitter, GK, ...)
;-
[general]
;
; Address to bind to for incoming connections.
; Default is ALL.
;
listenAddress=0.0.0.0
;
; Port to listen to.
; Default value is 1720.
;
listenPort=1720
;
; Port to connect to.
; (Used only when we don't have a gatekeeper)
; Default value is 1720.
;
connectPort=1720
;
; Enable fast start (yes,no).
;
fastStart=yes
;
; Enable H.245 tunnelling (yes,no).
;
h245Tunnelling=yes
;
; Enable early H.245 messages in call SETUP message.
;
h245inSetup=yes
;
; Enable in-band-DTMF detection. 
; (Note: Netmeeting uses in-band DTMFs)
;
inBandDTMF=no
;
; Enable silence suppression.
;
silenceSuppression=no
;
; Set jitter buffer (in milliseconds, 20...1).
;
jitterMin=20
jitterMax=100
;
; Set IP Type-of-Service byte for RTP channels.
; Valid values for this option are:
;   lowdelay, throughput, reliability, mincost, none
;
ipTos=none
;
; Set the maximum number of inbound/outbound H.323 connections. ;
outboundMax=50 inboundMax=10 ; ; Set tracing options for the H.323
wrapper library. ; libTraceFile can be 'stdout' or a full path name to a
logfile ; libTraceLevel=3 ;libTraceFile=stdout
libTraceFile=/var/log/asterisk/oh323.log
;
; Disable gatekeeper or specify a gatekeeper.
; Valid values for this option are:
;   DISABLE,
;   DISCOVER,
;   ,
;   ,
;   GKID:
;
gatekeeper=80.86.166.196
;gatekeeper=DISABLE
;
; Set the gatekeeper password
;
;gatekeeperPassword=secret
;
; Set the mode for sending user-input
; Valid values for this option are:
;   Q931-   Q.931 Keypad Information Element
;   STRING  -   H.245 string
;   TONE-   H.245 tone
;   RFC2833 -   RFC2833
;
userInputMode=Q931
;
; AMA flags (default, omit, billing, documentation)
;
amaFlags=default
;
; Account code
;
accountCode=H323
;
; Set the default context of H.323 calls.
;
context=voipout
;-
; Configure H.323 aliases, prefixes and
; related ASTERISK's contexts
;-
[register]
alias=BER-BER-GW-1
;
; Aliases/prefixes associated with the default context
; defined in section [general].
;
;alias=asterisk
;alias=123
;
; Aliases/prefixes routed in "all-aliases" context.
;
;context=all-aliases
;alias=ASTERISK
;alias=666
;
; Aliases/prefixes routed in "more-aliases" context.
;
;context=more-aliases
;alias=665
;
; Aliases/prefixes routed in "all-prefixes" context.
;
;context=all-prefixes
;gwprefix=00
;gwprefix=01
;
; Aliases/prefixes routed in "more-stuff" context.
;
;context=more-stuff
;alias=664
;gwprefix=02

;-
; Specify and configure CODEC related
; options
;-
[codecs]
;
; Define the codec list of the channel driver.
; Every "codec" option may have a "frames" option
; associated with it.
; Valid values for the "codec" option are:
;   G711U   -   G.711 u-Law
;   G711A   -   G.711 A-Law
;   G7231   -   G.723.1(6.3k)
;   G72316K3-   G.723.1(6.3k)
;   G72315K3-   G.723.1(5.3k)
;   G7231A6K3   -   G.723.1A(6.3k)
;   G7231A6K3   -   G.723.1A(6.3k)
;   G728-   G.728
;   G729-   G.729
;   G729A   -   G.729A
;   G729B   -   G.729B
;   G729AB  -   G.729AB
;   GSM0610 -   GSM 0610
;   MSGSM   -   Microsoft GSM Audio Capability
;   LPC10

AW: [Asterisk-Users] how to dial a h323 destination ?

[Thomas Haeger]

What is the gwprefix ? I try to connect the gk directly from our * gw

-Ursprungliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Sergio
Serrano Revuelto
Gesendet: Dienstag, 23. September 2003 12:27
An: [EMAIL PROTECTED]
Betreff: RE: [Asterisk-Users] how to dial a h323 destination ?


Try to add gwprefix in oh323.conf after your alias. You must know that
you can configure * gw in gnugk.ini or in oh323.conf. I recommend you
put in your oh323.conf.

srsergio



-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Thomas
Haeger
Enviado el: martes, 23 de septiembre de 2003 12:07
Para: [EMAIL PROTECTED]
Asunto: RE: [Asterisk-Users] how to dial a h323 destination ?


Here is my oh323.conf ...


;
; Configuration file of OpenH323 channel driver
;

;-
; General configuration options
; (ports, jitter, GK, ...)
;-
[general]
;
; Address to bind to for incoming connections.
; Default is ALL.
;
listenAddress=0.0.0.0
;
; Port to listen to.
; Default value is 1720.
;
listenPort=1720
;
; Port to connect to.
; (Used only when we don't have a gatekeeper)
; Default value is 1720.
;
connectPort=1720
;
; Enable fast start (yes,no).
;
fastStart=yes
;
; Enable H.245 tunnelling (yes,no).
;
h245Tunnelling=yes
;
; Enable early H.245 messages in call SETUP message.
;
h245inSetup=yes
;
; Enable in-band-DTMF detection. 
; (Note: Netmeeting uses in-band DTMFs)
;
inBandDTMF=no
;
; Enable silence suppression.
;
silenceSuppression=no
;
; Set jitter buffer (in milliseconds, 20...1).
;
jitterMin=20
jitterMax=100
;
; Set IP Type-of-Service byte for RTP channels.
; Valid values for this option are:
;   lowdelay, throughput, reliability, mincost, none
;
ipTos=none
;
; Set the maximum number of inbound/outbound H.323 connections. ;
outboundMax=50 inboundMax=10 ; ; Set tracing options for the H.323
wrapper library. ; libTraceFile can be 'stdout' or a full path name to a
logfile ; libTraceLevel=3 ;libTraceFile=stdout
libTraceFile=/var/log/asterisk/oh323.log
;
; Disable gatekeeper or specify a gatekeeper.
; Valid values for this option are:
;   DISABLE,
;   DISCOVER,
;   ,
;   ,
;   GKID:
;
gatekeeper=80.86.166.196
;gatekeeper=DISABLE
;
; Set the gatekeeper password
;
;gatekeeperPassword=secret
;
; Set the mode for sending user-input
; Valid values for this option are:
;   Q931-   Q.931 Keypad Information Element
;   STRING  -   H.245 string
;   TONE-   H.245 tone
;   RFC2833 -   RFC2833
;
userInputMode=Q931
;
; AMA flags (default, omit, billing, documentation)
;
amaFlags=default
;
; Account code
;
accountCode=H323
;
; Set the default context of H.323 calls.
;
context=voipout
;-
; Configure H.323 aliases, prefixes and
; related ASTERISK's contexts
;-
[register]
alias=BER-BER-GW-1
;
; Aliases/prefixes associated with the default context
; defined in section [general].
;
;alias=asterisk
;alias=123
;
; Aliases/prefixes routed in "all-aliases" context.
;
;context=all-aliases
;alias=ASTERISK
;alias=666
;
; Aliases/prefixes routed in "more-aliases" context.
;
;context=more-aliases
;alias=665
;
; Aliases/prefixes routed in "all-prefixes" context.
;
;context=all-prefixes
;gwprefix=00
;gwprefix=01
;
; Aliases/prefixes routed in "more-stuff" context.
;
;context=more-stuff
;alias=664
;gwprefix=02

;-
; Specify and configure CODEC related
; options
;-
[codecs]
;
; Define the codec list of the channel driver.
; Every "codec" option may have a "frames" option
; associated with it.
; Valid values for the "codec" option are:
;   G711U   -   G.711 u-Law
;   G711A   -   G.711 A-Law
;   G7231   -   G.723.1(6.3k)
;   G72316K3-   G.723.1(6.3k)
;   G72315K3-   G.723.1(5.3k)
;   G7231A6K3   -   G.723.1A(6.3k)
;   G7231A6K3   -   G.723.1A(6.3k)
;   G728-   G.728
;   G729-   G.729
;   G729A   -   G.729A
;   G729B   -   G.729B
;   G729AB  -   G.729AB
;   GSM0610 -   GSM 0610
;   MSGSM   -   Microsoft GSM Audio Capability
;   LPC10   -   LPC-10
; Number of frames in RTP packet (if not specified) is 1.
;   
;codec=G729A
;frames=20
;codec=GSM0610
;frames=4
;codec=G7231
;frames=20
;codec=G72316K3
;codec=G72315K3
;codec=G7231A6K3
;codec=G7231A5K3
codec=G711A
;codec=G711U
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RE: [Asterisk-Users] Recommended OS

[Michael A. Miller]

Thanks for the tip on HTML.

I will quickly admit that my background is mostly Microsoft. I do realize
that Linux, as general group, is considered the OS. However, from a Linux
newbie's point of view, I believe that there is danger in thinking that all
flavors are the same OS. Other than the generalized tools, most drivers and
such are specific to the company that released that version as well as the
hardware. Again, this is from my limited experience with Linux at this point
in time.

I have gotten Asterisk to work. It was a knowledge issue on my end. My main
issue is getting a Netjet ISDN and voice modem installed correctly. Both of
which I am having issues with the drivers at the OS level and have not
attempted any use with Asterisk at this point. If someone has experience
with a Netjet install under RH9 or Lucent/Agere linmodem, I would sincerely
appreciate a bit of help.

Thanks for the tips! I promise to start getting the nomenclature correct as
I learn. I the mean time, I do appreciate the patients and guidance.

Regards,

Michael


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Tuesday, September 23, 2003 12:57 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Recommended OS

On Mon, 2003-09-22 at 21:39, Michael A. Miller wrote:
> Is there a recommended OS that Asterisk should be used with? I have
> been trying to get Asterisk running on Red Hat 9.0 with little
> success.

You hit 2 of my pet peeves at once. Fist, please understand that HTML
has no business in normal email communication. Turn it off or you will
start getting ignored(hopefully for you).

Linux is a OS. Asterisk runs on Linux with only limited success on some
*bsd system. If you search the archives you will find someone who has
had some problem or another with just about every distribution. RH has a
higher number of support problems, but I'll grant that more newbies pick
RH and that contributes to it's problem count. There are quite a few
people here that us RH9 though. So if you wish to post some of your
problems we can them help you get over some of these bumps in the road. 
 
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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RE: [Asterisk-Users] how to dial a h323 destination ?

[Sergio Serrano Revuelto]

Try to add gwprefix in oh323.conf after your alias. You must know that
you can configure * gw in gnugk.ini or in oh323.conf. I recommend you
put in your oh323.conf.

srsergio



-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Thomas
Haeger
Enviado el: martes, 23 de septiembre de 2003 12:07
Para: [EMAIL PROTECTED]
Asunto: RE: [Asterisk-Users] how to dial a h323 destination ?


Here is my oh323.conf ...


;
; Configuration file of OpenH323 channel driver
;

;-
; General configuration options
; (ports, jitter, GK, ...)
;-
[general]
;
; Address to bind to for incoming connections.
; Default is ALL.
;
listenAddress=0.0.0.0
;
; Port to listen to.
; Default value is 1720.
;
listenPort=1720
;
; Port to connect to.
; (Used only when we don't have a gatekeeper)
; Default value is 1720.
;
connectPort=1720
;
; Enable fast start (yes,no).
;
fastStart=yes
;
; Enable H.245 tunnelling (yes,no).
;
h245Tunnelling=yes
;
; Enable early H.245 messages in call SETUP message.
;
h245inSetup=yes
;
; Enable in-band-DTMF detection. 
; (Note: Netmeeting uses in-band DTMFs)
;
inBandDTMF=no
;
; Enable silence suppression.
;
silenceSuppression=no
;
; Set jitter buffer (in milliseconds, 20...1).
;
jitterMin=20
jitterMax=100
;
; Set IP Type-of-Service byte for RTP channels.
; Valid values for this option are:
;   lowdelay, throughput, reliability, mincost, none
;
ipTos=none
;
; Set the maximum number of inbound/outbound H.323 connections. ;
outboundMax=50 inboundMax=10 ; ; Set tracing options for the H.323
wrapper library. ; libTraceFile can be 'stdout' or a full path name to a
logfile ; libTraceLevel=3 ;libTraceFile=stdout
libTraceFile=/var/log/asterisk/oh323.log
;
; Disable gatekeeper or specify a gatekeeper.
; Valid values for this option are:
;   DISABLE,
;   DISCOVER,
;   ,
;   ,
;   GKID:
;
gatekeeper=80.86.166.196
;gatekeeper=DISABLE
;
; Set the gatekeeper password
;
;gatekeeperPassword=secret
;
; Set the mode for sending user-input
; Valid values for this option are:
;   Q931-   Q.931 Keypad Information Element
;   STRING  -   H.245 string
;   TONE-   H.245 tone
;   RFC2833 -   RFC2833
;
userInputMode=Q931
;
; AMA flags (default, omit, billing, documentation)
;
amaFlags=default
;
; Account code
;
accountCode=H323
;
; Set the default context of H.323 calls.
;
context=voipout
;-
; Configure H.323 aliases, prefixes and
; related ASTERISK's contexts
;-
[register]
alias=BER-BER-GW-1
;
; Aliases/prefixes associated with the default context
; defined in section [general].
;
;alias=asterisk
;alias=123
;
; Aliases/prefixes routed in "all-aliases" context.
;
;context=all-aliases
;alias=ASTERISK
;alias=666
;
; Aliases/prefixes routed in "more-aliases" context.
;
;context=more-aliases
;alias=665
;
; Aliases/prefixes routed in "all-prefixes" context.
;
;context=all-prefixes
;gwprefix=00
;gwprefix=01
;
; Aliases/prefixes routed in "more-stuff" context.
;
;context=more-stuff
;alias=664
;gwprefix=02

;-
; Specify and configure CODEC related
; options
;-
[codecs]
;
; Define the codec list of the channel driver.
; Every "codec" option may have a "frames" option
; associated with it.
; Valid values for the "codec" option are:
;   G711U   -   G.711 u-Law
;   G711A   -   G.711 A-Law
;   G7231   -   G.723.1(6.3k)
;   G72316K3-   G.723.1(6.3k)
;   G72315K3-   G.723.1(5.3k)
;   G7231A6K3   -   G.723.1A(6.3k)
;   G7231A6K3   -   G.723.1A(6.3k)
;   G728-   G.728
;   G729-   G.729
;   G729A   -   G.729A
;   G729B   -   G.729B
;   G729AB  -   G.729AB
;   GSM0610 -   GSM 0610
;   MSGSM   -   Microsoft GSM Audio Capability
;   LPC10   -   LPC-10
; Number of frames in RTP packet (if not specified) is 1.
;   
;codec=G729A
;frames=20
;codec=GSM0610
;frames=4
;codec=G7231
;frames=20
;codec=G72316K3
;codec=G72315K3
;codec=G7231A6K3
;codec=G7231A5K3
codec=G711A
;codec=G711U
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FW: RE: [Asterisk-Users] iaxtel and iax.conf (HTML CONTENT, FYI)

[Josh Roberson]

I hate to post HTML to the list, but I
refuse to respond to this, and I would like to say that whomever
is using this service is kinda stupid for subscribing
an email address to the list using this service.

 

I hope they learn a lesson by us, the list
users, NOT responding to this, and eventually, after not receiving any list
mail, they’ll wonder “hmm… why is this list so dead?”

 

Just my .02.   

 

Again, sorry for the html post.

 

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Subject: RE:RE: [Asterisk-Users]
iaxtel and iax.conf

 


 
  
  
  
  
  
  
  
  
  
 
 
  
  
   

 


 
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RE: [Asterisk-Users] how to dial a h323 destination ?

[Thomas Haeger]

Here is my oh323.conf ...


;
; Configuration file of OpenH323 channel driver
;

;-
; General configuration options
; (ports, jitter, GK, ...)
;-
[general]
;
; Address to bind to for incoming connections.
; Default is ALL.
;
listenAddress=0.0.0.0
;
; Port to listen to.
; Default value is 1720.
;
listenPort=1720
;
; Port to connect to.
; (Used only when we don't have a gatekeeper)
; Default value is 1720.
;
connectPort=1720
;
; Enable fast start (yes,no).
;
fastStart=yes
;
; Enable H.245 tunnelling (yes,no).
;
h245Tunnelling=yes
;
; Enable early H.245 messages in call SETUP message.
;
h245inSetup=yes
;
; Enable in-band-DTMF detection. 
; (Note: Netmeeting uses in-band DTMFs)
;
inBandDTMF=no
;
; Enable silence suppression.
;
silenceSuppression=no
;
; Set jitter buffer (in milliseconds, 20...1).
;
jitterMin=20
jitterMax=100
;
; Set IP Type-of-Service byte for RTP channels.
; Valid values for this option are:
;   lowdelay, throughput, reliability, mincost, none
;
ipTos=none
;
; Set the maximum number of inbound/outbound H.323 connections.
;
outboundMax=50
inboundMax=10
;
; Set tracing options for the H.323 wrapper library.
; libTraceFile can be 'stdout' or a full path name to a logfile
;
libTraceLevel=3
;libTraceFile=stdout
libTraceFile=/var/log/asterisk/oh323.log
;
; Disable gatekeeper or specify a gatekeeper.
; Valid values for this option are:
;   DISABLE,
;   DISCOVER,
;   ,
;   ,
;   GKID:
;
gatekeeper=80.86.166.196
;gatekeeper=DISABLE
;
; Set the gatekeeper password
;
;gatekeeperPassword=secret
;
; Set the mode for sending user-input
; Valid values for this option are:
;   Q931-   Q.931 Keypad Information Element
;   STRING  -   H.245 string
;   TONE-   H.245 tone
;   RFC2833 -   RFC2833
;
userInputMode=Q931
;
; AMA flags (default, omit, billing, documentation)
;
amaFlags=default
;
; Account code
;
accountCode=H323
;
; Set the default context of H.323 calls.
;
context=voipout
;-
; Configure H.323 aliases, prefixes and
; related ASTERISK's contexts
;-
[register]
alias=BER-BER-GW-1
;
; Aliases/prefixes associated with the default context
; defined in section [general].
;
;alias=asterisk
;alias=123
;
; Aliases/prefixes routed in "all-aliases" context.
;
;context=all-aliases
;alias=ASTERISK
;alias=666
;
; Aliases/prefixes routed in "more-aliases" context.
;
;context=more-aliases
;alias=665
;
; Aliases/prefixes routed in "all-prefixes" context.
;
;context=all-prefixes
;gwprefix=00
;gwprefix=01
;
; Aliases/prefixes routed in "more-stuff" context.
;
;context=more-stuff
;alias=664
;gwprefix=02

;-
; Specify and configure CODEC related
; options
;-
[codecs]
;
; Define the codec list of the channel driver.
; Every "codec" option may have a "frames" option
; associated with it.
; Valid values for the "codec" option are:
;   G711U   -   G.711 u-Law
;   G711A   -   G.711 A-Law
;   G7231   -   G.723.1(6.3k)
;   G72316K3-   G.723.1(6.3k)
;   G72315K3-   G.723.1(5.3k)
;   G7231A6K3   -   G.723.1A(6.3k)
;   G7231A6K3   -   G.723.1A(6.3k)
;   G728-   G.728
;   G729-   G.729
;   G729A   -   G.729A
;   G729B   -   G.729B
;   G729AB  -   G.729AB
;   GSM0610 -   GSM 0610
;   MSGSM   -   Microsoft GSM Audio Capability
;   LPC10   -   LPC-10
; Number of frames in RTP packet (if not specified) is 1.
;   
;codec=G729A
;frames=20
;codec=GSM0610
;frames=4
;codec=G7231
;frames=20
;codec=G72316K3
;codec=G72315K3
;codec=G7231A6K3
;codec=G7231A5K3
codec=G711A
;codec=G711U
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