[Asterisk-Users] Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual?

[Jamie Carl]

Hi all,
I just picked myself up a Mediatrix FXO SIP gateway to play around with 
and hook into Asterisk but have no documentation.

Are there default passwords or IP's that I need to know if I do a 
factory reset? 

Or better still, would anyone have a User Manual they could send my 
way?  Any help would be appreciated.

TIA.
Jamie
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Re: [Asterisk-Users] GSM codec bandwidth

[steve]

On Thu, 2 Sep 2004, Michael George wrote:

 I've a question about the bandwidth consumed by IAX2/GSM.
 
 According to the wiki page, the GSM codec should run about 13 kilo-bits/sec
 for a voice encoding.
 
 However, watching gkrellm when I initiate a call to Digium, it looks like the
 channel is taking a consistent 5-6 kilo-bytes/sec.  That's a lot more
 bandwidth than it should take.  Is there perhaps a setting I have wrong
 somethere in the conf files?
 
 I have:
 bandwidth=low
 disallow=all
 allow=gsm
 
 so it's surely using GSM and it should be gearing itself for a low-bandwidth
 situation.


The codec itself takes 13kbps, but by the time the codec frames are 
wrapped in all the IP overhead it is a lot more.

If you are sending several concurrent calls to the same place, you can 
reduce the overhead by using trunking - which shares the IP overhead over 
the concurrent calls.

Steve

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[Asterisk-Users] digitnetworks card issues?

[Imran Akbar]

Hi,
   I've purchased two x100p clones, and when I try accessing a  line 
from asterisk with something like this:

exten = _1NXXNXX,1,Dial(Zap/2/{$EXTEN})
(is that only supposed to put you on channel 2 or actually dial the # 
for you?)

but I first hear noise, then a dial tone, but as soon as I start dialing 
numbers I get feedback and noise, and the call doesn't go through.

Any suggestions?
Thanks,
Imran
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[Asterisk-Users] call back on failed transfer?

[shabanip]

hi,
i'm under the impression that this feature is not available in asterisk, 
consider this scenario:
-  you are the operator. you answer a call from outside and you want to
transfer it to one of the extensions. after you transfer, if the person
you transferred the call to, doesn't pick up or if his line is busy, the
call is transfered back to you, you can speak to the caller and tell
him, for example, that the person you want to talk to is not in, and ask
if he would like to talk to leave a message or talk to another person
instead.  now in asterisk, it seems to me that after you transfer a call
to an extension, there's no way to have the caller transfered back to
yourself if the called extension doesn't answer or if it's busy. is this
correct?

thanks,
- shabanip
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RE: [Asterisk-Users] digitnetworks card issues?

[Jay Milk]

Have you contacted digitnetworks for support?  This list is owned and
maintained by Digium, who already gave you Asterisk for free.  Probably
not the best forum to ask for support for a competitive product here.

 -Original Message-
 From: Imran Akbar [mailto:[EMAIL PROTECTED] 
 Sent: Friday, September 03, 2004 1:38 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] digitnetworks card issues?
 
 
 Hi,
 I've purchased two x100p clones, and when I try accessing a  line 
 from asterisk with something like this:
 
 exten = _1NXXNXX,1,Dial(Zap/2/{$EXTEN})
 (is that only supposed to put you on channel 2 or actually dial the # 
 for you?)
 
 but I first hear noise, then a dial tone, but as soon as I 
 start dialing 
 numbers I get feedback and noise, and the call doesn't go through.
 
 Any suggestions?
 
 Thanks,
 Imran

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Re: [Asterisk-Users] digitnetworks card issues?

[Brian Capouch]

Imran Akbar wrote:
Hi,
   I've purchased two x100p clones, and when I try accessing a  line 
from asterisk with something like this:
 . . . .
Any suggestions?
Throw them away and get Digium cards.
B.
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RE: [Asterisk-Users] digitnetworks card issues?

[Peter Childs]

 I have the same hardware (x2)

/etc/zaptel.conf file

fxsks=1-2
loadzone=au
defaultzone=au

/etc/asterisk/zapata.conf file

[channels]
language=en
context=inbound
group=1
musiconhold=default
; need these much shorter than defaults
flash=90
signalling=fxs_ks
threewaycalling=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
;busydetect=no
busydetect=yes
;busycount=6
callprogress=no
channel = 1
channel = 2

from /etc/asterisk/extensions.conf

exten = _X.,1,Dial(Zap/g1/${EXTEN})

I had some noise issues at first, and then I used a decent shielded cable
between the cards and the wall socket and that cleared it up...

YMMV.

Cheers,
  Peter


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Imran Akbar
Sent: Friday, 3 September 2004 4:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] digitnetworks card issues?


Hi,
I've purchased two x100p clones, and when I try accessing a  line
from asterisk with something like this:

exten = _1NXXNXX,1,Dial(Zap/2/{$EXTEN})
(is that only supposed to put you on channel 2 or actually dial the #
for you?)

but I first hear noise, then a dial tone, but as soon as I start dialing
numbers I get feedback and noise, and the call doesn't go through.

Any suggestions?

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[Asterisk-Users] zap barge restrictions

[Asterisk]

I have a couple of questions on the zapbarge:

1) zapbarge asks for a channel - how would a manager know what channel to
enter ? Is there any way of being able to enter an extension number instead
? I know that you can get the information from the manager interface, but I
wouldn't want to give my users access to this, or have to install / write a
system just to get an extension number from a channel

2) is it really all or nothing ? What I mean is that can you restrict a zap
barge to certain extensions only - I wouldn't want one of our operators to
see that the CEO is on the phone and simply barge in without any permission
... I am aware that you can authenticate using a password before you enter
the zapbarge command, but that doesn't fit our requirements.

Am I looking at some custom code / feature request ?

Many thanks.

Julian.


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Re: [Asterisk-Users] digitnetworks card issues?

[Kannaiyan Natesan]

Does it mean that we cannot talk about Cisco or other FXS  products since 
IAXy is released??
I hope this list for every member who uses asterisk not Digium's products 
users alone.


- Original Message - 
From: Jay Milk [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
[EMAIL PROTECTED]
Sent: Friday, September 03, 2004 8:09 AM
Subject: RE: [Asterisk-Users] digitnetworks card issues?


Have you contacted digitnetworks for support?  This list is owned and
maintained by Digium, who already gave you Asterisk for free.  Probably
not the best forum to ask for support for a competitive product here.
-Original Message-
From: Imran Akbar [mailto:[EMAIL PROTECTED]
Sent: Friday, September 03, 2004 1:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] digitnetworks card issues?
Hi,
I've purchased two x100p clones, and when I try accessing a  line
from asterisk with something like this:
exten = _1NXXNXX,1,Dial(Zap/2/{$EXTEN})
(is that only supposed to put you on channel 2 or actually dial the #
for you?)
but I first hear noise, then a dial tone, but as soon as I
start dialing
numbers I get feedback and noise, and the call doesn't go through.
Any suggestions?
Thanks,
Imran
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Re: [Asterisk-Users] digitnetworks card issues?

[William Suffill]

Digitnetworks is profiting off the cards so they should support them.
If it wasn't for Digium there wouldn't be Asterisk anyway. So doesn't
that make it better to support the primary company for software that
many of you use every day at home and work?

On Fri, 3 Sep 2004 08:40:59 +0100, Kannaiyan Natesan
[EMAIL PROTECTED] wrote:
 Does it mean that we cannot talk about Cisco or other FXS  products since
 IAXy is released??
 I hope this list for every member who uses asterisk not Digium's products
 users alone.
 
 
 
 
 - Original Message -
 From: Jay Milk [EMAIL PROTECTED]
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 [EMAIL PROTECTED]
 Sent: Friday, September 03, 2004 8:09 AM
 Subject: RE: [Asterisk-Users] digitnetworks card issues?
 
  Have you contacted digitnetworks for support?  This list is owned and
  maintained by Digium, who already gave you Asterisk for free.  Probably
  not the best forum to ask for support for a competitive product here.
 
  -Original Message-
  From: Imran Akbar [mailto:[EMAIL PROTECTED]
  Sent: Friday, September 03, 2004 1:38 AM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: [Asterisk-Users] digitnetworks card issues?
 
 
  Hi,
  I've purchased two x100p clones, and when I try accessing a  line
  from asterisk with something like this:
 
  exten = _1NXXNXX,1,Dial(Zap/2/{$EXTEN})
  (is that only supposed to put you on channel 2 or actually dial the #
  for you?)
 
  but I first hear noise, then a dial tone, but as soon as I
  start dialing
  numbers I get feedback and noise, and the call doesn't go through.
 
  Any suggestions?
 
  Thanks,
  Imran
 
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Re: [Asterisk-Users] digitnetworks card issues?

[Imran Akbar]

Didn't want to start a flamewar here... but anyway, could the issue be 
that both fxo cards are on IRQ 11?  How do I even change that?

Thanks
William Suffill wrote:
Digitnetworks is profiting off the cards so they should support them.
If it wasn't for Digium there wouldn't be Asterisk anyway. So doesn't
that make it better to support the primary company for software that
many of you use every day at home and work?
On Fri, 3 Sep 2004 08:40:59 +0100, Kannaiyan Natesan
[EMAIL PROTECTED] wrote:
 

Does it mean that we cannot talk about Cisco or other FXS  products since
IAXy is released??
I hope this list for every member who uses asterisk not Digium's products
users alone.

- Original Message -
From: Jay Milk [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
[EMAIL PROTECTED]
Sent: Friday, September 03, 2004 8:09 AM
Subject: RE: [Asterisk-Users] digitnetworks card issues?
   

Have you contacted digitnetworks for support?  This list is owned and
maintained by Digium, who already gave you Asterisk for free.  Probably
not the best forum to ask for support for a competitive product here.
 

-Original Message-
From: Imran Akbar [mailto:[EMAIL PROTECTED]
Sent: Friday, September 03, 2004 1:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] digitnetworks card issues?
Hi,
   I've purchased two x100p clones, and when I try accessing a  line
from asterisk with something like this:
exten = _1NXXNXX,1,Dial(Zap/2/{$EXTEN})
(is that only supposed to put you on channel 2 or actually dial the #
for you?)
but I first hear noise, then a dial tone, but as soon as I
start dialing
numbers I get feedback and noise, and the call doesn't go through.
Any suggestions?
Thanks,
Imran
   

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 http://lists.digium.com/mailman/listinfo/asterisk-users
 

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[Asterisk-Users] video

[Altus Snyman]

Good day all
I'm interested in video on asterisk using SIP and windows clients
Now I did my research on http://www.voip-info.org/wiki-Asterisk+video
I have a few question:

*On the page they say you need the H.261 H.263? codecs,are these compiled in 
by default or do I need to do something special and if yes what?
*What windows clients are available?
*What cameras/hardware are the best?

Please advice and comment on this
Thanks
Altus

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Re: [Asterisk-Users] digitnetworks card issues?

[Kannaiyan Natesan]

If you could learn from the previous mails around here, as far i have seen 
the issues were discussed based on the use of asterisk with and without 
devices, not just supporting digium alone. You can see mails from 
broadvoice, voicepulse, iconnecthere. Do they support Digium? never mind 
about it. The issue here is why it is not working with asterisk, how that 
can be resolved and how the users around here solved those problems.

- Original Message - 
From: William Suffill [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
[EMAIL PROTECTED]
Sent: Friday, September 03, 2004 8:45 AM
Subject: Re: [Asterisk-Users] digitnetworks card issues?


Digitnetworks is profiting off the cards so they should support them.
If it wasn't for Digium there wouldn't be Asterisk anyway. So doesn't
that make it better to support the primary company for software that
many of you use every day at home and work?
On Fri, 3 Sep 2004 08:40:59 +0100, Kannaiyan Natesan
[EMAIL PROTECTED] wrote:
Does it mean that we cannot talk about Cisco or other FXS  products since
IAXy is released??
I hope this list for every member who uses asterisk not Digium's products
users alone.

- Original Message -
From: Jay Milk [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
[EMAIL PROTECTED]
Sent: Friday, September 03, 2004 8:09 AM
Subject: RE: [Asterisk-Users] digitnetworks card issues?
 Have you contacted digitnetworks for support?  This list is owned and
 maintained by Digium, who already gave you Asterisk for free.  Probably
 not the best forum to ask for support for a competitive product here.

 -Original Message-
 From: Imran Akbar [mailto:[EMAIL PROTECTED]
 Sent: Friday, September 03, 2004 1:38 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] digitnetworks card issues?


 Hi,
 I've purchased two x100p clones, and when I try accessing a  line
 from asterisk with something like this:

 exten = _1NXXNXX,1,Dial(Zap/2/{$EXTEN})
 (is that only supposed to put you on channel 2 or actually dial the #
 for you?)

 but I first hear noise, then a dial tone, but as soon as I
 start dialing
 numbers I get feedback and noise, and the call doesn't go through.

 Any suggestions?

 Thanks,
 Imran

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[Asterisk-Users] OH323 0.6.3b compilation problem with 1.0 RC2 on RH9

[Vlasis Chatzistayrou]

Hello,

I just tried to compile OH323 0.6.3b on a RH9 machine with Asterisk 1.0 RC2 
installed but failed. I applied the patch to the required OpenH323 library 
according to the instructions, and set the proper directories in the Makefile. 

Here is what I receive after I issue make:


***

g++ -DP_USE_PRAGMA -fno-rtti -ffunction-sections -fdata-sections -D_REENTRANT -
DOPENSSL_NO_KRB5 -Wall -fPIC -I/Downloads/pwlib/v1.6.6/pwlib/include -
DPTRACING -I/Downloads/openh323/v1.13.5/openh323/include -DHAS_OSS -Wall -x 
c++ -Os -DPWLIBVERSION=\1.6.6\ -DOPENH323VERSION=\1.13.5\  -
I/Downloads/pwlib/v1.6.6/pwlib/include/ptlib/unix -
I/Downloads/pwlib/v1.6.6/pwlib/include -
I/Downloads/openh323/v1.13.5/openh323/include -
I/Downloads/openh323/v1.13.5/openh323/include/openh323 -I../asterisk-driver -c 
wrapcaps.cxx -o wrapcaps.o
touch ../asterisk-driver/chan_oh323.c
gcc -shared -Wl,-soname,liboh323wrap.so -o liboh323wrap.so wrapper_misc.o 
asteriskaudio.o wrapendpoint.o wrapconnection.o wrapper.o wrapcaps.o
make[1]: Leaving directory `/Downloads/oh323/oh323-0.6.3b/asterisk-oh323-
0.6.3b/wrapper'
make[1]: Entering directory `/Downloads/oh323/oh323-0.6.3b/asterisk-oh323-
0.6.3b/asterisk-driver'
gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-
declarations -D_REENTRANT -D_GNU_SOURCE -I/usr/include/asterisk -I../wrapper -
g -c -o chan_oh323.o chan_oh323.c
In file included from /usr/include/stdio.h:34,
 from chan_oh323.c:34:
/usr/lib/gcc-lib/i386-redhat-linux/3.2.2/include/stddef.h:213: syntax error 
before typedef
In file included from chan_oh323.c:34:
/usr/include/stdio.h:46: syntax error before typedef
/usr/include/stdio.h:62: syntax error before typedef
In file included from /usr/include/_G_config.h:44,
 from /usr/include/libio.h:32,
 from /usr/include/stdio.h:72,
 from chan_oh323.c:34:
/usr/include/gconv.h:176: parse error before __flexarr
In file included from /usr/include/libio.h:32,
 from /usr/include/stdio.h:72,
 from chan_oh323.c:34:
/usr/include/_G_config.h:47: field `__cd' has incomplete type
/usr/include/_G_config.h:50: field `__cd' has incomplete type
/usr/include/_G_config.h:52: confused by earlier errors, bailing out
make[1]: *** [chan_oh323.o] Error 1
make[1]: Leaving directory `/Downloads/oh323/oh323-0.6.3b/asterisk-oh323-
0.6.3b/asterisk-driver'
make: *** [subdirs_all] Error 1


***

I'm not a very experienced Linux user so I can't really figure out what the 
problem may be in this case. 

Does anyone have any suggestions?

Thank you in advance,
Vlasis Hatzistavrou.




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Re: [Asterisk-Users] video

[Vladyslav]

On Fri, 2004-09-03 at 10:56, Altus Snyman wrote:
 Good day all
 I'm interested in video on asterisk using SIP and windows clients
 Now I did my research on http://www.voip-info.org/wiki-Asterisk+video
 I have a few question:
 
 *On the page they say you need the H.261 H.263? codecs,are these compiled in 
 by default or do I need to do something special and if yes what?
They are already in.
All U need to do is just allow them in sip.conf

 *What windows clients are available?
Windows messenger 4.7 (In this version U could specify your * server ip)
 *What cameras/hardware are the best?
 
Have used usb LG PC camera (Flatron) quite good quality 640X480.

 Please advice and comment on this
 Thanks
 Altus
 
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-- 
Best regards
Vlad

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[Asterisk-Users] RC2 with OH323 or H323

[João Amaro]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi All,
I've just finished my upgrade to asterisk RC2.
I need to have H323 support, and in the last months i've been using
the chan-oh323 with good results.
My question is: anyone in the list have made tests with both chans
(oh323 and h323), which is best ?
For this installation i don't need the gatekeeper support, i just want
to receive/place calls to Cisco CallManager.
If anyone tried to install OH323 with asterisk RC2 with success,
please send-me an email. I can compile the driver and the
library, but i can't initialize the driver when i start the asterisk.

Thanks ind advance,
Regards,
João Amaro
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (GNU/Linux)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
iD8DBQFBODD/JUm/Bor63CERAsYrAJ9BUydM1fCRVDZIljpP7efvuARiLgCgp+LO
UCuqUBRPCJMfyAtGZXPhb1c=
=wXUK
-END PGP SIGNATURE-
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Re: [Asterisk-Users] OH323 0.6.3b compilation problem with 1.0 RC2 on RH9

[Joo Amaro]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I Vlasis,
I'm using those versions (Fedora COre 1) and it compiled without
problems, but when i try to initialize asterisk i get the folowwing error:
ERROR [-1084337504]: chanoh323.c:4636 load_module: H.323 listener
creation failed.
Hope someone can help us.

Vlasis Chatzistayrou wrote:
| Hello,
|
| I just tried to compile OH323 0.6.3b on a RH9 machine with Asterisk
| 1.0 RC2 installed but failed. I applied the patch to the required
| OpenH323 library according to the instructions, and set the proper
| directories in the Makefile.
|
| Here is what I receive after I issue make:
|
|
| ***
|
| g++ -DP_USE_PRAGMA -fno-rtti -ffunction-sections -fdata-sections
| -D_REENTRANT - DOPENSSL_NO_KRB5 -Wall -fPIC
| -I/Downloads/pwlib/v1.6.6/pwlib/include - DPTRACING
| -I/Downloads/openh323/v1.13.5/openh323/include -DHAS_OSS -Wall -x
| c++ -Os -DPWLIBVERSION=\1.6.6\ -DOPENH323VERSION=\1.13.5\  -
| I/Downloads/pwlib/v1.6.6/pwlib/include/ptlib/unix -
| I/Downloads/pwlib/v1.6.6/pwlib/include -
| I/Downloads/openh323/v1.13.5/openh323/include -
| I/Downloads/openh323/v1.13.5/openh323/include/openh323
| -I../asterisk-driver -c wrapcaps.cxx -o wrapcaps.o touch
| ../asterisk-driver/chan_oh323.c gcc -shared
| -Wl,-soname,liboh323wrap.so -o liboh323wrap.so wrapper_misc.o
| asteriskaudio.o wrapendpoint.o wrapconnection.o wrapper.o
| wrapcaps.o make[1]: Leaving directory
| `/Downloads/oh323/oh323-0.6.3b/asterisk-oh323- 0.6.3b/wrapper'
| make[1]: Entering directory
| `/Downloads/oh323/oh323-0.6.3b/asterisk-oh323-
| 0.6.3b/asterisk-driver' gcc -Wall -pipe -Wall -Wstrict-prototypes
| -Wmissing-prototypes -Wmissing- declarations -D_REENTRANT
| -D_GNU_SOURCE -I/usr/include/asterisk -I../wrapper - g -c -o
| chan_oh323.o chan_oh323.c In file included from
| /usr/include/stdio.h:34, from chan_oh323.c:34:
| /usr/lib/gcc-lib/i386-redhat-linux/3.2.2/include/stddef.h:213:
| syntax error before typedef In file included from
| chan_oh323.c:34: /usr/include/stdio.h:46: syntax error before
| typedef /usr/include/stdio.h:62: syntax error before typedef In
| file included from /usr/include/_G_config.h:44, from
| /usr/include/libio.h:32, from /usr/include/stdio.h:72, from
| chan_oh323.c:34: /usr/include/gconv.h:176: parse error before
| __flexarr In file included from /usr/include/libio.h:32, from
| /usr/include/stdio.h:72, from chan_oh323.c:34:
| /usr/include/_G_config.h:47: field `__cd' has incomplete type
| /usr/include/_G_config.h:50: field `__cd' has incomplete type
| /usr/include/_G_config.h:52: confused by earlier errors, bailing
| out make[1]: *** [chan_oh323.o] Error 1 make[1]: Leaving directory
| `/Downloads/oh323/oh323-0.6.3b/asterisk-oh323-
| 0.6.3b/asterisk-driver' make: *** [subdirs_all] Error 1
|
|
| ***
|
| I'm not a very experienced Linux user so I can't really figure out
| what the problem may be in this case.
|
| Does anyone have any suggestions?
|
| Thank you in advance, Vlasis Hatzistavrou.
|
|
|
|
| ___ Asterisk-Users
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[Asterisk-Users] Lower cost router suitable for VOIP ?

[Robert Rozman]

Hi,

we're testing Asterisk 1  RC 2 behind ordinary router and NAT. Since we're
sharing network with web server it seems like voip packets are not coming
through fast enough (Digium demo dies after few seconds...). It's the same
if I make direct calls (passing Asterisk) so we conclude it's network
problem - it also work normally outside our router...

I wonder what solutions can we use to give voice packets higher priority.
I'm avare of VOIP routers, but they are pricey. Can some of common routers
help, or maybe implementing router on another simple Linux box?

Any advice, pointers to more info ?
How to trace network and debug Asterisk in convenient way ?

Thanks in advance,

Robert Rozman

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[Asterisk-Users] one doubt

[Murali]

Hi all,

  Im using asterisk. I have one doubt. 

  Im running asterisk in one machine(RedHat9.0)
 running firefly softphone in 3 windows machine

  I hv 3 users in sip.conf like 1001, 2001  3001
 appropriate entry for those users are also include in 
 extensions.conf like
  
 --
 [mainmenu]

  exten = 1001,1,Dial(SIP/1001,20,r)
  exten = 1001,2,Congestion
  exten = 1001,103,Busy

  exten = 2001,1,Dial(SIP/2001,20,r)
  exten = 2001,2,Congestion
  exten = 2001,103,Busy

  exten = 3001,1,Dial(SIP/3001,20,r)
  exten = 3001,2,Congestion
  exten = 3001,103,Busy


  I called 1001 from 2001. 1001 got call from 2001. 
  He attend the call. the call is going on.
  user 3001 try to call 1001. NOW 1001 got call from 3001.
  eventhough he is speaking with user 2001.

  Is it correct?

  When 1001 is talking with 2001. how he will get call from 
  3001 or any other. 

  I think its wrong.

  The user 3001 must get message Busy.

  I need suggestion from any one. please

   
   Thanks in advance


Regards
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Re: [Asterisk-Users] video

[Altus Snyman]

I have my x-lite connected to the server but messanger does not want to log in
It does not even show its trying on the server
I went and seclected sip and adduse the server and username.no 
[EMAIL PROTECTED]
Is there anything special
Thanks Altus


On Friday 03 September 2004 10:38, Vladyslav wrote:


 On Fri, 2004-09-03 at 10:56, Altus Snyman wrote:
  Good day all
  I'm interested in video on asterisk using SIP and windows clients
  Now I did my research on http://www.voip-info.org/wiki-Asterisk+video
  I have a few question:
 
  *On the page they say you need the H.261 H.263? codecs,are these compiled
  in by default or do I need to do something special and if yes what?

 They are already in.
 All U need to do is just allow them in sip.conf

  *What windows clients are available?

 Windows messenger 4.7 (In this version U could specify your * server ip)

  *What cameras/hardware are the best?

 Have used usb LG PC camera (Flatron) quite good quality 640X480.

  Please advice and comment on this
  Thanks
  Altus
 
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Re: [Asterisk-Users] OH323 0.6.3b compilation problem with 1.0 RC2 on RH9

[Michael Manousos]

Joa~o Amaro wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I Vlasis,
I'm using those versions (Fedora COre 1) and it compiled without
problems, but when i try to initialize asterisk i get the folowwing error:
ERROR [-1084337504]: chanoh323.c:4636 load_module: H.323 listener
creation failed.
There is some other process listening on the TCP port used for
H.323 signaling (default is 1720). This port can be specified in
oh323.conf.
Michael.
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Re: [Asterisk-Users] OH323 0.6.3b compilation problem with 1.0 RC2 on RH9

[Michael Manousos]

It works fine for me on a Slack9.1 laptop.
Michael.
Vlasis Chatzistayrou wrote:
Hello,
I just tried to compile OH323 0.6.3b on a RH9 machine with Asterisk 1.0 RC2 
installed but failed. I applied the patch to the required OpenH323 library 
according to the instructions, and set the proper directories in the Makefile. 

Here is what I receive after I issue make:
***
g++ -DP_USE_PRAGMA -fno-rtti -ffunction-sections -fdata-sections -D_REENTRANT -
DOPENSSL_NO_KRB5 -Wall -fPIC -I/Downloads/pwlib/v1.6.6/pwlib/include -
DPTRACING -I/Downloads/openh323/v1.13.5/openh323/include -DHAS_OSS -Wall -x 
c++ -Os -DPWLIBVERSION=\1.6.6\ -DOPENH323VERSION=\1.13.5\  -
I/Downloads/pwlib/v1.6.6/pwlib/include/ptlib/unix -
I/Downloads/pwlib/v1.6.6/pwlib/include -
I/Downloads/openh323/v1.13.5/openh323/include -
I/Downloads/openh323/v1.13.5/openh323/include/openh323 -I../asterisk-driver -c 
wrapcaps.cxx -o wrapcaps.o
touch ../asterisk-driver/chan_oh323.c
gcc -shared -Wl,-soname,liboh323wrap.so -o liboh323wrap.so wrapper_misc.o 
asteriskaudio.o wrapendpoint.o wrapconnection.o wrapper.o wrapcaps.o
make[1]: Leaving directory `/Downloads/oh323/oh323-0.6.3b/asterisk-oh323-
0.6.3b/wrapper'
make[1]: Entering directory `/Downloads/oh323/oh323-0.6.3b/asterisk-oh323-
0.6.3b/asterisk-driver'
gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-
declarations -D_REENTRANT -D_GNU_SOURCE -I/usr/include/asterisk -I../wrapper -
g -c -o chan_oh323.o chan_oh323.c
In file included from /usr/include/stdio.h:34,
 from chan_oh323.c:34:
/usr/lib/gcc-lib/i386-redhat-linux/3.2.2/include/stddef.h:213: syntax error 
before typedef
In file included from chan_oh323.c:34:
/usr/include/stdio.h:46: syntax error before typedef
/usr/include/stdio.h:62: syntax error before typedef
In file included from /usr/include/_G_config.h:44,
 from /usr/include/libio.h:32,
 from /usr/include/stdio.h:72,
 from chan_oh323.c:34:
/usr/include/gconv.h:176: parse error before __flexarr
In file included from /usr/include/libio.h:32,
 from /usr/include/stdio.h:72,
 from chan_oh323.c:34:
/usr/include/_G_config.h:47: field `__cd' has incomplete type
/usr/include/_G_config.h:50: field `__cd' has incomplete type
/usr/include/_G_config.h:52: confused by earlier errors, bailing out
make[1]: *** [chan_oh323.o] Error 1
make[1]: Leaving directory `/Downloads/oh323/oh323-0.6.3b/asterisk-oh323-
0.6.3b/asterisk-driver'
make: *** [subdirs_all] Error 1

***
I'm not a very experienced Linux user so I can't really figure out what the 
problem may be in this case. 

Does anyone have any suggestions?
Thank you in advance,
Vlasis Hatzistavrou.
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[Asterisk-Users] Digium E100P and PMX in Germany

[Jan Goericke]

Hello ml,

 i need some help on my zaptel configuration. My E100P only shows some 
YELLOW / RED alarm when I load the wct1xxp module and do a 

cat /proc/zaptel/1

Span 1: WCT1/0 Digium Wildcard E100P E1/PRA Card 0 HDB3/CCS YELLOW RED
...
..
.


My /etc/zaptel.conf is: 

span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
loadzone=nl
defaultzone=nl

I tried zaptel-1.0RC2 and the latest CVS version too. So I think it is a 
configuration problem. Can anyone give me a hint how to configure my 
E100P? 

Thank you for your help,
 Jan Goericke
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Re: [Asterisk-Users] one doubt

[Holger Schurig]

 Im using asterisk. I have one doubt

Question, not doubt. I wonder why all people from India have doubts and 
not questions :-)

I guess that because of the hindu language characters you use HTML e-mail?  
However, for english mailing lists it's better to not use HTML, but pure 
text. Then people will flame you less.

 When 1001 is talking with 2001. how he will get call from
 3001 or any other.

He will get calls, because an IP telephone can have more than one lines 
attached to it. If you want only one call per phone, you have to use the 
(deprectaed) incominglimit tag in sip.conf, or better use SetGroup.

http://www.voip-info.org/wiki-Asterisk+cmd+SetGroup

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[Asterisk-Users] busy signalling on PRI doesn't work...

[Roy Sigurd Karlsbakk]

hi all
Attachd is a PRI DEBUG dumped while dialling out to a busy number among 
with zap(ata|tel).conf. asterisk did not flag busy, and I got a busy 
indicator going mep-meep-mep-meep-mep-meep (never heard 
this before)

Can someone help me out here?
thanks
roy


zapata.conf
Description: Binary data


zaptel.conf
Description: Binary data
[EMAIL PROTECTED] root]# asterisk -r
Asterisk CVS-HEAD-07/28/04-14:58:30, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
Connected to Asterisk CVS-HEAD-07/28/04-14:58:30 currently running on pstngw1 (pid = 
1995)
pstngw1*CLI pri debug span 1
Enabled debugging on span 1
-- Accepting AUTHENTICATED call from 213.160.242.5, requested format = 8, actual 
format = 8
-- Executing NoOp(IAX2/[EMAIL PROTECTED]/2, TON: 0) in new stack
-- Executing NoOp(IAX2/[EMAIL PROTECTED]/2, NPI: 0) in new stack
-- Executing NoOp(IAX2/[EMAIL PROTECTED]/2, PRES: 0) in new stack
-- Executing Dial(IAX2/[EMAIL PROTECTED]/2, Zap/g1/22602614|180|t)) in new 
stack
-- Making new call for cr 32830
 Protocol Discriminator: Q.931 (8)  len=39
 Call Ref: len= 2 (reference 62/0x3E) (Originator)
 Message type: SETUP (5)
 [04 03 80 90 a3]
 Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer capability: Speech 
 (0)
  Ext: 1  Trans mode/rate: 64kbps, circuit-mode (16)
  Ext: 1  User information layer 1: A-Law (35)
 [18 03 a9 83 82]
 Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0
ChanSel: Reserved
   Ext: 1  Coding: 0   Number Specified   Channel Type: 3
   Ext: 1  Channel: 2 ]
 [6c 0a 21 80 32 31 39 37 30 30 30 31]
 Calling Number (len=12) [ Ext: 0  TON: National Number (2)  NPI: ISDN/Telephony 
 Numbering Plan (E.164/E.163) (1)
   Presentation: Presentation permitted, user number not 
 screened (0) '21970001' ]
 [70 09 a1 32 32 36 30 32 36 31 34]
 Called Number (len=11) [ Ext: 1  TON: National Number (2)  NPI: ISDN/Telephony 
 Numbering Plan (E.164/E.163) (1) '22602614' ]
 [a1]
 Sending Complete (len= 1)
-- Called g1/22602614
 Protocol Discriminator: Q.931 (8)  len=10
 Call Ref: len= 2 (reference 32830/0x803E) (Terminator)
 Message type: CALL PROCEEDING (2)
 [18 03 a9 83 82]
 Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0
ChanSel: Reserved
   Ext: 1  Coding: 0   Number Specified   Channel Type: 3
   Ext: 1  Channel: 2 ]
-- Processing IE 24 (cs0, Channel Identification)
 Protocol Discriminator: Q.931 (8)  len=9
 Call Ref: len= 2 (reference 32830/0x803E) (Terminator)
 Message type: PROGRESS (3)
 [1e 02 82 88]
 Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   
Location: Public network serving the local user (2)
   Ext: 1  Progress Description: Inband information or 
appropriate pattern now available. (8) ]
-- Processing IE 30 (cs0, Progress Indicator)
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Outgoing call  Proceeding, peerstate 
Incoming Call Proceeding
 Protocol Discriminator: Q.931 (8)  len=9
 Call Ref: len= 2 (reference 62/0x3E) (Originator)
 Message type: DISCONNECT (69)
 [08 02 81 90]
 Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   Location: Private 
 network serving the local user (1)
  Ext: 1  Cause: Normal Clearing (16), class = Normal Event (1) ]
-- Hungup 'Zap/2-1'
  == Spawn extension (iax, 22602614, 4) exited non-zero on 'IAX2/[EMAIL PROTECTED]/2'
-- Hungup 'IAX2/[EMAIL PROTECTED]/2'
 Protocol Discriminator: Q.931 (8)  len=5
 Call Ref: len= 2 (reference 32830/0x803E) (Terminator)
 Message type: RELEASE (77)
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release Request
 Protocol Discriminator: Q.931 (8)  len=9
 Call Ref: len= 2 (reference 62/0x3E) (Originator)
 Message type: RELEASE COMPLETE (90)
 [08 02 81 90]
 Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   Location: Private 
 network serving the local user (1)
  Ext: 1  Cause: Normal Clearing (16), class = Normal Event (1) ]
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null

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Re: [Asterisk-Users] Digium E100P and PMX in Germany

[Steven Critchfield]

On Fri, 2004-09-03 at 05:31, Jan Goericke wrote:
 Hello ml,
 
  i need some help on my zaptel configuration. My E100P only shows some 
 YELLOW / RED alarm when I load the wct1xxp module and do a 
 
 cat /proc/zaptel/1
 
 Span 1: WCT1/0 Digium Wildcard E100P E1/PRA Card 0 HDB3/CCS YELLOW RED
 ...
 ..
 .
 
 
 My /etc/zaptel.conf is: 
 
 span=1,1,0,ccs,hdb3
 bchan=1-15,17-31
 dchan=16
 loadzone=nl
 defaultzone=nl
 
 I tried zaptel-1.0RC2 and the latest CVS version too. So I think it is a 
 configuration problem. Can anyone give me a hint how to configure my 
 E100P? 

Next step is to start asterisk so libpri attaches to your line and
brings up the D channel. 
-- 
Steven Critchfield [EMAIL PROTECTED]

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Re: [Asterisk-Users] OH323 0.6.3b compilation problem with 1.0 RC2 on RH9

[Vlasis Chatzistayrou]

Hello,

Thanks for replying. On a Slackware 9.1 it may compile, but on a RH9 it 
doesn't and I don't think we can install another distro on that machine...  

:-)

I guess I'll have to wait for the new version of OH323 in order to try 
cimpiling again...

Best regards  thanks,
Vlasis.


Michael Manousos [EMAIL PROTECTED]:

 
 It works fine for me on a Slack9.1 laptop.
 
 Michael.
 
 Vlasis Chatzistayrou wrote:
  Hello,
  
  I just tried to compile OH323 0.6.3b on a RH9 machine with Asterisk 1.0 RC2
 
  installed but failed. I applied the patch to the required OpenH323 library
 
  according to the instructions, and set the proper directories in the
 Makefile. 
  
  Here is what I receive after I issue make:
  
  
  ***
  
  g++ -DP_USE_PRAGMA -fno-rtti -ffunction-sections -fdata-sections
 -D_REENTRANT -
  DOPENSSL_NO_KRB5 -Wall -fPIC -I/Downloads/pwlib/v1.6.6/pwlib/include -
  DPTRACING -I/Downloads/openh323/v1.13.5/openh323/include -DHAS_OSS -Wall -x
 
  c++ -Os -DPWLIBVERSION=\1.6.6\ -DOPENH323VERSION=\1.13.5\  -
  I/Downloads/pwlib/v1.6.6/pwlib/include/ptlib/unix -
  I/Downloads/pwlib/v1.6.6/pwlib/include -
  I/Downloads/openh323/v1.13.5/openh323/include -
  I/Downloads/openh323/v1.13.5/openh323/include/openh323 -I../asterisk-driver
 -c 
  wrapcaps.cxx -o wrapcaps.o
  touch ../asterisk-driver/chan_oh323.c
  gcc -shared -Wl,-soname,liboh323wrap.so -o liboh323wrap.so wrapper_misc.o 
  asteriskaudio.o wrapendpoint.o wrapconnection.o wrapper.o wrapcaps.o
  make[1]: Leaving directory `/Downloads/oh323/oh323-0.6.3b/asterisk-oh323-
  0.6.3b/wrapper'
  make[1]: Entering directory `/Downloads/oh323/oh323-0.6.3b/asterisk-oh323-
  0.6.3b/asterisk-driver'
  gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-
  declarations -D_REENTRANT -D_GNU_SOURCE -I/usr/include/asterisk
 -I../wrapper -
  g -c -o chan_oh323.o chan_oh323.c
  In file included from /usr/include/stdio.h:34,
   from chan_oh323.c:34:
  /usr/lib/gcc-lib/i386-redhat-linux/3.2.2/include/stddef.h:213: syntax error
 
  before typedef
  In file included from chan_oh323.c:34:
  /usr/include/stdio.h:46: syntax error before typedef
  /usr/include/stdio.h:62: syntax error before typedef
  In file included from /usr/include/_G_config.h:44,
   from /usr/include/libio.h:32,
   from /usr/include/stdio.h:72,
   from chan_oh323.c:34:
  /usr/include/gconv.h:176: parse error before __flexarr
  In file included from /usr/include/libio.h:32,
   from /usr/include/stdio.h:72,
   from chan_oh323.c:34:
  /usr/include/_G_config.h:47: field `__cd' has incomplete type
  /usr/include/_G_config.h:50: field `__cd' has incomplete type
  /usr/include/_G_config.h:52: confused by earlier errors, bailing out
  make[1]: *** [chan_oh323.o] Error 1
  make[1]: Leaving directory `/Downloads/oh323/oh323-0.6.3b/asterisk-oh323-
  0.6.3b/asterisk-driver'
  make: *** [subdirs_all] Error 1
  
  
  ***
  
  I'm not a very experienced Linux user so I can't really figure out what the
 
  problem may be in this case. 
  
  Does anyone have any suggestions?
  
  Thank you in advance,
  Vlasis Hatzistavrou.
  
 
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Re: [Asterisk-Users] Lower cost router suitable for VOIP ?

[asteriskstuff]

Look at the wrt54g or wrt54gs with sveasoft firmware and wondershaper, allows you to 
QOS VoIP data.

Google for sveasoft forums to find the right forum to search.

P

 -Original Message-
 From: Robert Rozman [mailto:[EMAIL PROTECTED]
 Sent: Friday, September 03, 2004, 2:32 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Lower cost router suitable for VOIP ?
 
 Hi,
 
 we're testing Asterisk 1  RC 2 behind ordinary router and NAT. Since we're
 sharing network with web server it seems like voip packets are not coming
 through fast enough (Digium demo dies after few seconds...). It's the same
 if I make direct calls (passing Asterisk) so we conclude it's network
 problem - it also work normally outside our router...
 
 I wonder what solutions can we use to give voice packets higher priority.
 I'm avare of VOIP routers, but they are pricey. Can some of common routers
 help, or maybe implementing router on another simple Linux box?
 
 Any advice, pointers to more info ?
 How to trace network and debug Asterisk in convenient way ?
 
 Thanks in advance,
 
 Robert Rozman
 
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[Asterisk-Users] Zaprtc help

[David Davies]

 
Hi,

Having no digium hardware in my box and two cpus and a ohci usb bus im
forced to use zaprtc.

I have recompiled the kernel and removed enhanced rtc support.
When I attempt to compile zaprtc I get the following error.

zaprtc.c:1077: warning: implicit declaration of function `barrier'
zaprtc.c:1078: warning: implicit declaration of function `cpu_relax'
zaprtc.c: At top level:
zaprtc.c:109: storage size of `rtc_irq_timer' isn't known
zaprtc.c:719: storage size of `rtc_fops' isn't known
zaprtc.c:107: warning: `DECLARE_WAIT_QUEUE_HEAD' declared `static' but never
defined
make: *** [zaprtc.o] Error 1

Can anyone offer advice on where to start .

Thanks

David

Counting the days to astricon.

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RE: [Asterisk-Users] Polycom SIP INFO Changing Ringers

[Matthew Marlowe]

Well thanks for trying to help, mod=0 didn't fix that problem.

I'll check out the sequential problem later, didn't notice that.
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Baker
Sent: Thursday, September 02, 2004 11:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom SIP INFO  Changing Ringers

Don't know.  Try setting se.rt.modification.enabled=0
and se.rt.1.mod=0 in ipmid.cfg

John

Matthew Marlowe wrote:

 Ok, so I'm blind. That worked.
 
 Do you know why setting the ringtype though doesn't change the DEFAULT

 ring? Not that I can't do it via alert info now... It's just odd that 
 won't work. And I definitely had se.rt.1. for all of the settings.
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of John 
 Baker
 Sent: Thursday, September 02, 2004 10:29 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Polycom SIP INFO  Changing Ringers
 
 Matthew Marlowe wrote:
 
 In ipmid.cfg, try
 
 G3INTERCOM se.rt.10.name=G3INTERCOM se.rt.10.type=ring-answer 
 se.rt.10.timeout=1000 se.rt.10.ringer=7/
 
 (note rt.10 instead of rt.4)
 
 John
 
 
In ipmid.cfg I have:

  G3INTERCOM se.rt.10.name=G3INTERCOM
 
 se.rt.4.type=ring-answer
 
se.rt.4.timeout=1000 se.rt.10.ringer=7/

In sip.cfg I have:

alertInfo voIpProt.SIP.alertInfo.1.value=G3INTERCOM
voIpProt.SIP.alertInfo.1.class=10/

I set up a test extension:

exten = 8614,1,SetVar(ALERT_INFO=G3INTERCOM)
exten = 8614,2,Dial(SIP/614p)

Ringer isn't changed..

In addition I have tried using the ringType option to change ringer 
type on boot, and it doesn't accept it.

I've tried everything I think of

Any help would be greatly appreciated
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[Asterisk-Users] Re: AVM B1, chan_capi, Kernel 2.6

[Stefan Tichy]

On Tue, Aug 10, 2004 at 10:00:58AM +0200, Stefan Tichy wrote:
 Using active AVM cards in connection with kernel 2.6 seems to be a
 bad idea.


http://listserv.isdn4linux.de/pipermail/i4ldeveloper/2004-August/000630.html

This patch should be interesting if you are using AVM B1 cards and
kernel 2.6.


-- 
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Re: [Asterisk-Users] GSM codec bandwidth

[Michael George]

On Fri, Sep 03, 2004 at 08:26:28AM +0200, [EMAIL PROTECTED] wrote:
 On Thu, 2 Sep 2004, Michael George wrote:
  I've a question about the bandwidth consumed by IAX2/GSM.
  
  According to the wiki page, the GSM codec should run about 13 kilo-bits/sec
  for a voice encoding.
  
  However, watching gkrellm when I initiate a call to Digium, it looks like the
  channel is taking a consistent 5-6 kilo-bytes/sec.  That's a lot more
  bandwidth than it should take.  Is there perhaps a setting I have wrong
  somethere in the conf files?
  
  I have:
  bandwidth=low
  disallow=all
  allow=gsm
  
  so it's surely using GSM and it should be gearing itself for a low-bandwidth
  situation.
 
 
 The codec itself takes 13kbps, but by the time the codec frames are 
 wrapped in all the IP overhead it is a lot more.

Yes, I understand about overhead, but this is 4x the bandwidth usage.  Even if
that is 13kbps for each stream of audio (23kbps total), that is doubled by
(TCP/UDP)/IP overhead.  That struck me as a lot of overhead.  I guess, though,
that since the packets need to be sent quite frequently, that could happen.

If that is what others are experiencing, then I accept it.

 If you are sending several concurrent calls to the same place, you can 
 reduce the overhead by using trunking - which shares the IP overhead over 
 the concurrent calls.

That makes sense, and I've read that trunking pays off with even 2
conversations.

-- 
-M

There are 10 kinds of people in this world:
Those who can count in binary and those who cannot.
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Re: [Asterisk-Users] Leaving messages on answering machines (no its notspam)

[Areski]

Hello Clayton,

Is there chances that you share your work with the list :)
I am planning to create an Asterisk testing tool, 
- Generate call to an other Asterisk Box 
- Check if the Asterisk answer correctly
- Check if the application is well played, etc...

I guess your application would be a good starting point ;)
Ideas would be really appreciated!

Cheers,
Areski


On Thu, 2004-09-02 at 20:33, Clayton Smith wrote:
 Hey thanks, thats a great idea too
 
 Basicly just check for a pause, if i don't get one quickly, then its an 
 answering machine
 
 And both ideas are compatible, so i could do both at the same time
 
 
 Chears
 
 
 Scott Stingel wrote:
 
 Answering machine detection is usually accomplished by analysing the timing
 of the voice energy in the initial answer period.  People usually answer by
 saying: Hello, Frank Giwerski, Pencil sharpening department, or
 something fairly short, whereas answering messages are usually longer.
 
 So, I think the usualy method is to have the software listen to the voice
 energy for some initial period until there's a pause, and decide based on
 the duration of this energy whether it's a human or machine.
 
 But listening for a beep, although less efficient maybe, might work too!
 
 Regards
 Scott
 
 
 Scott M. Stingel
 President,
 Emerging Voice Technology, Inc.
 Palo Alto California  London England
 www.evtmedia.com
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Clayton Smith
 Sent: Thursday, September 02, 2004 11:07 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] Leaving messages on answering machines (no its
 notspam)
 
 Hey there
 
 I'm trying to get asterisk to leave messages on answering machines So i have
 a pretty cool php notifying script (it notifys, it doesn't
 spam!!) to phones and cellphones
 
 Now all is fine if a human picks up, but if an answering machine picks up,
 well the script plays, but only the ending is recorded
 
 So really, the tricky part is knowing WHEN to leave a message
 
 Now to the best of my knowledge, there is no way to tell when an answering
 machine picks it (be it the sprint cellphone operator, or  a home owned
 cellphone), but i was thinking...
 
 I could play my script using an EAGI script So i get extensions to run an
 EAGI script, that then manages everything, So when the call is picked up,
 relay the message, but if a high pitched beep is detected (via the EAGI
 script), repeat the message from scratch
 
 Now I'm no expert on asterisk, and i can see that this method could be a
 little buggy, so I'm wondering if there are any suggestions or if there is a
 better way to leaving messages on answering machines
 
 Any help or suggestions will be greatly appreciated Thanks
 
 
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[Asterisk-Users] Re: asterisk config and root

[Stefan Tichy]

On Thu, Sep 02, 2004 at 01:30:05PM +0300, Tzafrir Cohen wrote:
 Another beginner's question:
 
 Can I gain root if I have write access to asterisk's config files?

If the asterisk process has root priviledges only root should be
allowed to modify its config files. But root priviledges are not
mandatory for a running asterisk process.

http://www.voip-info.org/wiki-Asterisk+non-root


Asterisk can be started by root (init script) using a command like
this:

/usr/sbin/asterisk -p -Uasterisk -Gdialout

(options are explained in the man page asterisk(8))

The user asterisk has been created as described in the wiki.
Probably you have to choose a different group and it might even be
necessary to change the permissions of some device file.


-- 
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Re: [Asterisk-Users] Digium E100P and PMX in Germany

[Michael Bielicki]

did you tried it with crc4 as well ?
span=1,1,0,ccs,hdb3,crc4 ?

On Fri, 2004-09-03 at 13:00, Steven Critchfield wrote:
 On Fri, 2004-09-03 at 05:31, Jan Goericke wrote:
  Hello ml,
  
   i need some help on my zaptel configuration. My E100P only shows some 
  YELLOW / RED alarm when I load the wct1xxp module and do a 
  
  cat /proc/zaptel/1
  
  Span 1: WCT1/0 Digium Wildcard E100P E1/PRA Card 0 HDB3/CCS YELLOW RED
  ...
  ..
  .
  
  
  My /etc/zaptel.conf is: 
  
  span=1,1,0,ccs,hdb3
  bchan=1-15,17-31
  dchan=16
  loadzone=nl
  defaultzone=nl
  
  I tried zaptel-1.0RC2 and the latest CVS version too. So I think it is a 
  configuration problem. Can anyone give me a hint how to configure my 
  E100P? 
 
 Next step is to start asterisk so libpri attaches to your line and
 brings up the D channel. 

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[Asterisk-Users] G729 license

[Sergey Lapin]

Hi, all!
Will asterisk use G729 license if both ends have support for G729 and no 
transcoding needed? So, the scheme:

remote phone G729Asterisk with G729 codeclocal phone G729
As I understand, in this situation everything can be passed through, and 
is so on default asterisk installation, but we require transcoding, 
because most our phones are X-lites on LAN and use GSM codec. Remote 
link is voipexchange.ru and prefer G729 since it's cheapest there and 
there are many directions where other codecs are not available or too 
expensive.
So, another question is will all this transcoding between GSM and G729 
work for 30 calls at the same time on Celeron 1700 (PIV) ?

All the best,
S.
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Re: [Asterisk-Users] Digium E100P and PMX in Germany

[Jan Goericke]

Thanks for the hint.

I did it and zap show channels shows me the 31 channel. But when I check 
/proc/zaptel/1, i still get the same error as before. 


On Fri, 3 Sep 2004, Steven Critchfield wrote:

 On Fri, 2004-09-03 at 05:31, Jan Goericke wrote:
  Hello ml,
  
   i need some help on my zaptel configuration. My E100P only shows some 
  YELLOW / RED alarm when I load the wct1xxp module and do a 
  
  cat /proc/zaptel/1
  
  Span 1: WCT1/0 Digium Wildcard E100P E1/PRA Card 0 HDB3/CCS YELLOW RED
  ...
  ..
  .
  
  
  My /etc/zaptel.conf is: 
  
  span=1,1,0,ccs,hdb3
  bchan=1-15,17-31
  dchan=16
  loadzone=nl
  defaultzone=nl
  
  I tried zaptel-1.0RC2 and the latest CVS version too. So I think it is a 
  configuration problem. Can anyone give me a hint how to configure my 
  E100P? 
 
 Next step is to start asterisk so libpri attaches to your line and
 brings up the D channel. 
 -- 
 Steven Critchfield [EMAIL PROTECTED]
 
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Re: [Asterisk-Users] GSM codec bandwidth

[Rich Adamson]

   I've a question about the bandwidth consumed by IAX2/GSM.
   
   According to the wiki page, the GSM codec should run about 13 kilo-bits/sec
   for a voice encoding.
   
   However, watching gkrellm when I initiate a call to Digium, it looks like the
   channel is taking a consistent 5-6 kilo-bytes/sec.  That's a lot more
   bandwidth than it should take.  Is there perhaps a setting I have wrong
   somethere in the conf files?
   
   I have:
   bandwidth=low
   disallow=all
   allow=gsm
   
   so it's surely using GSM and it should be gearing itself for a low-bandwidth
   situation.
  
  
  The codec itself takes 13kbps, but by the time the codec frames are 
  wrapped in all the IP overhead it is a lot more.
 
 Yes, I understand about overhead, but this is 4x the bandwidth usage.  Even if
 that is 13kbps for each stream of audio (23kbps total), that is doubled by
 (TCP/UDP)/IP overhead.  That struck me as a lot of overhead.  I guess, though,
 that since the packets need to be sent quite frequently, that could happen.

Just a guess here and I've not use gkrellm at all, but is it possible
gkrellm is adding incoming  outgoing traffic together?

If I take your numbers, divide by two, the result is roughly 
equivalent to the actual codec bandwidth plus the pkt overhead for
data moving in each direction.



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Re: [Asterisk-Users] zap barge restrictions

[Steve Maroney]

Try using Authenticate() to permit zapbarge access to others. With
Zapbarge you may also supply the channel number.  You can also implment the
secruity that you want by using the simple features of extensions.conf.
For example:

exten = 100,1,Zapbarge()
- OR -
exten = 100/5002,1,Zapbarge()

the second line will match when extension 100 is dialed but only from
extension 5002


Thank you,
Steve Maroney

On Fri, 3 Sep 2004, Asterisk wrote:

 I have a couple of questions on the zapbarge:

 1) zapbarge asks for a channel - how would a manager know what channel to
 enter ? Is there any way of being able to enter an extension number instead
 ? I know that you can get the information from the manager interface, but I
 wouldn't want to give my users access to this, or have to install / write a
 system just to get an extension number from a channel

 2) is it really all or nothing ? What I mean is that can you restrict a zap
 barge to certain extensions only - I wouldn't want one of our operators to
 see that the CEO is on the phone and simply barge in without any permission
 ... I am aware that you can authenticate using a password before you enter
 the zapbarge command, but that doesn't fit our requirements.

 Am I looking at some custom code / feature request ?

 Many thanks.

 Julian.


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Re: [Asterisk-Users] Going to voicemail instead of queue if no agent is logged in ?

[Kurt Bauer]

Hi,
I did this the following way:
-) define a global variable - AGENTS_AVAIL=0
-) when agent logs in increment - 
SetGlobalVar(AGENTS_AVAIL=$[${AGENTS_AVAIL} + 1]);
-) when agent logs off decrement - 
SetGlobalVar(AGENTS_AVAIL=$[${AGENTS_AVAIL} - 1]);
-) when queue is called evaluate and goto label - 
gotoif,$[${AGENTS_AVAIL}]?${Q}:${NO_Q)

Hope that helps and if there is an easier way of doing this please show me 
how.

br,
Kurt

--On Tuesday, August 31, 2004 09:57:29 PM +0200 Robert Rozman 
[EMAIL PROTECTED] wrote:

Hi,
I'd like to implement scenario to send user to operator's queue by default
(if doesn't dial any extension) but only if there is operator agent
logged, so user could get response. If not I'd like to send it to
voicemail...
Any quick advice ?
Thanks in advance,
Robert.
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[Asterisk-Users] SIP / Keep alive...

[Jefferson Carvalho]

Hello list,
Is there some parameter on sip.conf to always let the client reachable ?
I'm trying to avoid  this situation :
Sep  3 09:49:29 NOTICE[135442432]: chan_sip.c:7653 sip_poke_noanswer: 
Peer '1264' is now UNREACHABLE!
Sep  3 09:49:39 NOTICE[135442432]: chan_sip.c:6408 handle_response: Peer 
'1264' is now REACHABLE!

Regards,
-Jefferson Carvalho
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Re: [Asterisk-Users] Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual?

[Rich Adamson]

 I just picked myself up a Mediatrix FXO SIP gateway to play around with 
 and hook into Asterisk but have no documentation.

I spent a substantial amount of time evaluating the 1204 box back in
the January timeframe, and then returned it to the reseller. I can 
answer some of your questions but not all.

The Mediatrix products are not bad at all, but they can only be configured
via a Windows SNMP application that comes with each firmware version
on the 1204. There is no telnet or web interface. Without that app,
getting the box to work with asterisk will not be possible.

Mediatrix does not have any direct support; they expect their resellers
to support the user, and they expect the reseller to invoice you
for each software upgrade, etc.

The box is shipped from Mediatrix with both H.323 and SIP software,
however the reseller is suppose to only give you one or the other.
(There are different model numbers for those two, but its the same
box, just a different software load.)

The software required to configure the box _must_ match the firware
running in the box. When I was testing, they were at v1.4.6.20, and
each firmware release required a deinstall and reinstall of the 
configuration software. I tried two or three different SIP firmware
versions to address different problems, and had to go through the
process multiple times.

The firmware upgrade process actually forces you to start the process
with the old configuration software (on Windows), initiate the
upgrade, and sometime prior to rebooting the 1204, deinstall and
reinstall the new configuration software so you can interact with
the new firmware. Its a real pain.

Given where you're at with the box, you'll probably need to get the
latest sip firmware, the manual that goes with that version, and the
configuration software that matches that firmware.

Since they rely on the use of SNMP to configure the box, you'll spend
a fair amount of time working with the MIBs within the configuration
software trying to find the parameters necessary to accomplish some
task. The admin manual is pretty good, but finding the words (and
appropriate MIB variable) to match an asterisk function is far less 
then ideal. (The more you know about SNMP, the easier it is.)

 Are there default passwords or IP's that I need to know if I do a 
 factory reset? 

A factory reset will but the box into dhcp mode, and will obtain an
IP address on subsequent reboots. The SNMP community string (password)
defaults to public, and in January 2004, could not be changed to
anything else period. Again, without their SNMP configuration software
you'll not be able to get the box configured properly.
 
 Or better still, would anyone have a User Manual they could send my 
 way?  Any help would be appreciated.

Mediatrix still seems to be focused on the toll bypass business, and
intended the 1204 (fxo) to be used in conjunction with the 1104 (fxs)
box. As a result, there are a fair number of non-sip-compliant
protocol 'enhancements' in their firmware, however the box can be 
made to work with *. There are a few users on this list that are
using the 1204 successfully.

The box does some strange things that made it unusable for me. Like
it detects ring cadance on the first incoming call following a reboot
and applies that same cadance to all four lines. In my case, I had
one pstn line (of four) with a different cadance which caused the box 
to never answer incoming calls on that port. :(

There's also no nice way to pick a specific pstn port number when
making outgoing calls via the box. You'll need to muck around with
setting a 'callerid' in *, and then set a matching parameter within
the 1204 to recognize that callerid on a per-port basis. The box 
will then use that port for the call. It's default config is to use 
'silence suppression' which will cause very choppy sound with asterisk, 
so that config parameter will need to change as well.

To get the box to work (and be legal), you'll need to contact a reseller
and order the current software from them. That cdrom will include the
user manual (*.pdf), the configuration software (for Windows only),
and the binary image needed to upgrade it. (Be sure to specify either
H.323 or SIP as they won't ship both.) You'll also need a tftp server
to complete the process. And, be constantly aware that if you discover
what you believe to be a firmware problem, they will want to charge
you again for the next version.

There are lots of different reasons for not using that box in a 
production business environment (mostly revolving around support,
enhancements, bug fixes, cost of ownership, potential bankrupcy again)
but for the home or small office it functions rather well.

Good luck...

Rich


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Re: [Asterisk-Users] SIP / Keep alive...

[Eric Wieling]

Jefferson Carvalho wrote:
Hello list,
Is there some parameter on sip.conf to always let the client reachable ?
I'm trying to avoid  this situation :
Sep  3 09:49:29 NOTICE[135442432]: chan_sip.c:7653 sip_poke_noanswer: 
Peer '1264' is now UNREACHABLE!
Sep  3 09:49:39 NOTICE[135442432]: chan_sip.c:6408 handle_response: Peer 
'1264' is now REACHABLE!
Get a better ISP.  Asterisk is not getting any response from the SIP 
peer.  This is bad.  You CAN set qualify=no.  This will prevent 
Asterisk from ever trying to reach the peer when it doesn not have to.

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[Asterisk-Users] RES: Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual

[miguel]

I have the user manual, I'll send it to your email tonight when I'll be in
my home.

I have an APA III-4FXO too, until today I can't put it to work with
asterisk.
 
Kind regards,

Miguel

Date: Fri, 03 Sep 2004 16:07:59 +1000
From: Jamie Carl [EMAIL PROTECTED]
Subject: [Asterisk-Users] Mediatrix APA III-4FXO (or 1204) help.
Anyone with user manual?
To: [EMAIL PROTECTED]
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Hi all,

I just picked myself up a Mediatrix FXO SIP gateway to play around with 
and hook into Asterisk but have no documentation.

Are there default passwords or IP's that I need to know if I do a 
factory reset? 

Or better still, would anyone have a User Manual they could send my 
way?  Any help would be appreciated.

TIA.

Jamie



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Re: [Asterisk-Users] Digium E100P and PMX in Germany

[Jan Goericke]

Yes I tried this too. But the problem is the same. 

On Fri, 3 Sep 2004, Michael Bielicki wrote:

 did you tried it with crc4 as well ?
 span=1,1,0,ccs,hdb3,crc4 ?
 
 On Fri, 2004-09-03 at 13:00, Steven Critchfield wrote:
  On Fri, 2004-09-03 at 05:31, Jan Goericke wrote:
   Hello ml,
   
i need some help on my zaptel configuration. My E100P only shows some 
   YELLOW / RED alarm when I load the wct1xxp module and do a 
   
   cat /proc/zaptel/1
   
   Span 1: WCT1/0 Digium Wildcard E100P E1/PRA Card 0 HDB3/CCS YELLOW RED
   ...
   ..
   .
   
   
   My /etc/zaptel.conf is: 
   
   span=1,1,0,ccs,hdb3
   bchan=1-15,17-31
   dchan=16
   loadzone=nl
   defaultzone=nl
   
   I tried zaptel-1.0RC2 and the latest CVS version too. So I think it is a 
   configuration problem. Can anyone give me a hint how to configure my 
   E100P? 
  
  Next step is to start asterisk so libpri attaches to your line and
  brings up the D channel. 
 
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[Asterisk-Users] I forgot to add my email please contact me offline we have around 300, 000 to 1/2 million minutes per month for India and Pakistan .. can ztdummy help trunk mode?

[Maxim Litnitsky]

Hi all, did not find much info in lists about subj.
I have ztdummy working properly because I can use conferences without
any errors.
But when I try to use trunk=yes, I get the following:

Sep  2 21:20:51 WARNING[1137720112]: chan_iax2.c:6422 build_user:
Unable to support trunking on user home' without zaptel timing
Sep  2 21:20:51 WARNING[1137720112]: chan_iax2.c:6246 build_peer:
Unable to support trunking on peer 'home' without zaptel timing

Can something be done??

Thx in advance for your replies.
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[Asterisk-Users] mpg123 - multiple instances, taxing CPU

[Matthew Boehm]

Is there any reason why there should ever be more than 1 instance of mpg123
running on a * server?

I just did an 'uptime' and noticed all 3 of my loads where over 3.00.

'top' showed 8 mpg123 processes all processing the same 3 songs (our
background music).

I tried to kill one of them but another one spawned in its place.

Any ideas?

Thanks,
Matthew

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Re: [Asterisk-Users] digitnetworks card issues?

[Lyle Giese]

Absolutely the IRQ issue is probably the root cause.

How do you change that?  Move the cards around on the PCI slots until they
are on seperate and unique IRQ's.

Lyle


- Original Message - 
From: Imran Akbar [EMAIL PROTECTED]
To: William Suffill [EMAIL PROTECTED]; Asterisk Users Mailing
List - Non-Commercial Discussion [EMAIL PROTECTED]
Sent: Friday, September 03, 2004 2:55 AM
Subject: Re: [Asterisk-Users] digitnetworks card issues?


 Didn't want to start a flamewar here... but anyway, could the issue be
 that both fxo cards are on IRQ 11?  How do I even change that?


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RE: [Asterisk-Users] mpg123 - multiple instances, taxing CPU

[Steve Hanselman]

 
check your musiconhold.conf, for each one you define you'l get an instance.


-Original Message-
From: Matthew Boehm
To: [EMAIL PROTECTED]
Sent: 03/09/04 15:04
Subject: [Asterisk-Users] mpg123 - multiple instances, taxing CPU

Is there any reason why there should ever be more than 1 instance of
mpg123
running on a * server?

I just did an 'uptime' and noticed all 3 of my loads where over 3.00.

'top' showed 8 mpg123 processes all processing the same 3 songs (our
background music).

I tried to kill one of them but another one spawned in its place.

Any ideas?

Thanks,
Matthew

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that any review, distribution or copying of this document is strictly prohibited. If 
you have 
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Re: [Asterisk-Users] mpg123 - multiple instances, taxing CPU

[Matthew Boehm]

This is all that is in that file.

musiconhold.conf
-
;
; Music on hold class definitions
;
[classes]
default = mp3:/var/lib/asterisk/mohmp3

There are 4 mp3 files inside that dir. Any ideas?
Matthew

- Original Message - 
From: Steve Hanselman [EMAIL PROTECTED]
To: 'Matthew Boehm ' [EMAIL PROTECTED];
[EMAIL PROTECTED]
Sent: Friday, September 03, 2004 9:06 AM
Subject: RE: [Asterisk-Users] mpg123 - multiple instances, taxing CPU



 check your musiconhold.conf, for each one you define you'l get an
instance.


 -Original Message-
 From: Matthew Boehm
 To: [EMAIL PROTECTED]
 Sent: 03/09/04 15:04
 Subject: [Asterisk-Users] mpg123 - multiple instances, taxing CPU

 Is there any reason why there should ever be more than 1 instance of
 mpg123
 running on a * server?

 I just did an 'uptime' and noticed all 3 of my loads where over 3.00.

 'top' showed 8 mpg123 processes all processing the same 3 songs (our
 background music).

 I tried to kill one of them but another one spawned in its place.

 Any ideas?

 Thanks,
 Matthew

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 received  this communication in error, please notify Brendata immediately
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RE: [Asterisk-Users] mpg123 - multiple instances, taxing CPU

[Tenorio, Leandro]

 Actually, I got almost the same issue (i´m not having such load), but I got 
defines 4 different moh and got 10 process (I check every time I restart * to kill all 
the mpg123 processes also.

LTenorio

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman
Sent: Friday, September 03, 2004 11:06 AM
To: 'Matthew Boehm '; '[EMAIL PROTECTED] '
Subject: RE: [Asterisk-Users] mpg123 - multiple instances, taxing CPU

 
check your musiconhold.conf, for each one you define you'l get an instance.


-Original Message-
From: Matthew Boehm
To: [EMAIL PROTECTED]
Sent: 03/09/04 15:04
Subject: [Asterisk-Users] mpg123 - multiple instances, taxing CPU

Is there any reason why there should ever be more than 1 instance of
mpg123
running on a * server?

I just did an 'uptime' and noticed all 3 of my loads where over 3.00.

'top' showed 8 mpg123 processes all processing the same 3 songs (our background music).

I tried to kill one of them but another one spawned in its place.

Any ideas?

Thanks,
Matthew

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Re: [Asterisk-Users] Any way to _always_ execute certain commands in a dialplan context?

[Rob Fugina]

Use the 's' extension...


On Thu, 02 Sep 2004 19:42:13 -0700, Kevin P. Fleming
[EMAIL PROTECTED] wrote:
 I've got a need to do something like the following:
 
 [foo-context]
 exten = _.,1,SetCIDNum(123)
 exten = _.,2,SetCIDName(XYZ)
 include = local
 include = tollfree
 
 But of course, this example won't work. The goal here is this: if a call
 ends up being handled by the local or tollfree contexts, I want
 those SetCID*** commands executed. Otherwise, I don't want them
 executed. I don't want to embed them into the local/tollfree contexts
 themselves, because then I'd have to figure out some way to store the
 123 and XYZ values so that they could be used by commands in those
 contexts.
 
 Essentially, what I want to do is override the CALLERIDNUM/CALLERIDNAME
 data for calls that are directed outside the PBX, and leave it alone for
 calls inside the PBX. That way internal users can see John Q. Smith
 322 (different for each extension), but outside callees see Smithco
 Widgets 602-555-1212 (which would be identical for all of the
 extensions that can make outside calls).
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RE: [Asterisk-Users] digitnetworks card issues?

[Jay Milk]

The difference is that digitnetworks specifically targets Digium as
competition.  Cisco, Sipura, etc, don't directly compete with IAXy
because they have different feature sets and were around long before
IAXy was released.  Digium was first on the market with the X100P and
digitnetworks cloned their product, thus circumvented Digium's RD cost
and undermines their ability to recover that cost.  Digitnetworks are
trying to steal a slice of the pie from Digium, and that's why
supporting them on this list is objectionable.

 -Original Message-
 From: Kannaiyan Natesan [mailto:[EMAIL PROTECTED] 
 Sent: Friday, September 03, 2004 2:41 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] digitnetworks card issues?
 
 
 Does it mean that we cannot talk about Cisco or other FXS  
 products since 
 IAXy is released??
 I hope this list for every member who uses asterisk not 
 Digium's products 
 users alone.
 
 
 
 - Original Message - 
 From: Jay Milk [EMAIL PROTECTED]
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
 [EMAIL PROTECTED]
 Sent: Friday, September 03, 2004 8:09 AM
 Subject: RE: [Asterisk-Users] digitnetworks card issues?
 
 
  Have you contacted digitnetworks for support?  This list is 
 owned and 
  maintained by Digium, who already gave you Asterisk for free.  
  Probably not the best forum to ask for support for a competitive 
  product here.
 
  -Original Message-
  From: Imran Akbar [mailto:[EMAIL PROTECTED]
  Sent: Friday, September 03, 2004 1:38 AM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: [Asterisk-Users] digitnetworks card issues?
 
 
  Hi,
  I've purchased two x100p clones, and when I try 
 accessing a  line 
  from asterisk with something like this:
 
  exten = _1NXXNXX,1,Dial(Zap/2/{$EXTEN})
  (is that only supposed to put you on channel 2 or actually 
 dial the # 
  for you?)
 
  but I first hear noise, then a dial tone, but as soon as I start 
  dialing numbers I get feedback and noise, and the call doesn't go 
  through.
 
  Any suggestions?
 
  Thanks,
  Imran
 
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[Asterisk-Users] BIG ISSUE with SIP, not sure where to go but it's killing asterisk.

[Daniel Jimenez]

I frequently get this error message, it repeats itself hundred/thousands 
of times and never stops.

chan_sip.c:7467 sipsock_read: Failed to grab lock, trying again...
During this period, I can make no SIP calls what-so-ever. The only way 
I've been able to stop it is to killall -9 asterisk. Doing a restart now 
doesn't respond.

Anyone know why?
--
Daniel Jimenez djimenez[at]pobox[dot]com
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[Asterisk-Users] SIP Question

[tonini . massimo]

Is there a way for a natted client with
a dynamic ip address to receive call from the asterisk box ?

I can call from the natted phone using
tasterisk but I can't receive call in the natted phone because *
does not know the ip address of the phone

I have enabled the registration but
when I launch the show peers I have:

281/281
(Unspecified)  (D) 255.255.255.255 0  
 Unmonitored

instead in the local network phone I
have specified the ip address.

Someone can help me ?

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Re: [Asterisk-Users] Any way to _always_ execute certain commands in a dialplan context?

[Kevin P. Fleming]

Rob Fugina wrote:
Use the 's' extension...
Uhh, no. That doesn't work at all.
The s extension is only used if the channel coming into this context 
doesn't have any target extension to look for. If it does, the s 
extension is never used. If you have a context for SIP phones, and one 
of them calls 1234, then:

[foo-sip]
exten = s,1,Hangup
exten = 1234,1,Dial(Zap/1/89434594)
will _not_ hangup, it will dial out on Zap/1.
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Re: [Asterisk-Users] SIP Question

[Matthew Boehm]

This means either that:
- you do not have  nat=yes in the sip.conf for that device,
- or you don't have a STUN server ip in the device settings
- or the device has not properly logged in to * (various reasons).

Turn on sip debugging and see if you see any error messages like 404 Not
Authorized and the like.

Matthew

- Original Message - 
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, September 03, 2004 9:55 AM
Subject: [Asterisk-Users] SIP Question


 Is there a way for a natted client with a dynamic ip address to receive
 call from the asterisk box ?

 I can call from the natted phone using tasterisk  but I can't receive call
 in the natted phone because * does not know the ip address of the phone

 I have enabled the registration but when I launch the show peers I have:

 281/281  (Unspecified)   (D)  255.255.255.255  0 Unmonitored

 instead in the local network phone I have specified the ip address.

 Someone can help me ?

 Thank you.






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[Asterisk-Users] Dlink Video Phone Asterisk

[Ken Wiesner]

Hello,



Just wondering if anyone has tried connecting the Dlink
Video Phone (DVC-1000) to Asterisk. It would be cool if you could use Asterisk
as an MCU.



~Ken








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Re: [Asterisk-Users] Any way to _always_ execute certain commands in a dialplan context?

[Rob Fugina]

Ah, well...  Never tried it with SIP phones.  I thought I had used
that before for inbound calls on a Zap channel, and with local Zap
extensions, too...

On Fri, 03 Sep 2004 08:11:09 -0700, Kevin P. Fleming
[EMAIL PROTECTED] wrote:
 Rob Fugina wrote:
  Use the 's' extension...
 
 
 Uhh, no. That doesn't work at all.
 
 The s extension is only used if the channel coming into this context
 doesn't have any target extension to look for. If it does, the s
 extension is never used. If you have a context for SIP phones, and one
 of them calls 1234, then:
 
 [foo-sip]
 exten = s,1,Hangup
 exten = 1234,1,Dial(Zap/1/89434594)
 
 will _not_ hangup, it will dial out on Zap/1.
 
 
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RE: [Asterisk-Users] Any way to _always_ execute certain commands in a dialplan context?

[Kris Boutilier]

You need to a method other than 'include =', which effectively concatenates
the target of the include with the current context. Consider this approach
instead:

[foo-context]
; This needs to match the criteria for tollfree, say a 91800 prefix
exten = _91800.,1,SetCIDNum(123)
exten = _91800.,2,SetCIDName(XYZ)
exten = _91800.,3,Goto(tollfree,${EXTEN},1)

; This needs to match the criteria for local, say a 9 prefix
exten = _9.,1,SetCIDNum(123)
exten = _9.,2,SetCIDName(XYZ)
exten = _9.,3,Goto(local,${EXTEN},1)


It could also be implemented as:

[foo-context]
; This needs to match the criteria for tollfree, say a 91800 prefix
exten = _91800.,1,Macro(setOutgoingCLID)
exten = _91800.,2,Goto(tollfree,${EXTEN},1)

; This needs to match the criteria for local, say a 9 prefix
exten = _9.,1,Macro(setOutgoingCLID)
exten = _9.,2,Goto(local,${EXTEN},1)

[macro-setOutgoingCLID]
exten = s,1,SetCIDNum(123)
exten = s,2,SetCIDName(XYZ)


You'll need to implement 't' and 'i' handlers in [foo-context] and,
possibly, seperate 'h' handlers in [local] and [tollfree].

Hope that helps.

Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District

-Original Message-
From: Kevin P. Fleming [mailto:[EMAIL PROTECTED]
Sent: September 2, 2004 7:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Any way to _always_ execute certain commands
in a dialplan context?


I've got a need to do something like the following:

[foo-context]
exten = _.,1,SetCIDNum(123)
exten = _.,2,SetCIDName(XYZ)
include = local
include = tollfree

But of course, this example won't work. The goal here is this: if a call 
ends up being handled by the local or tollfree contexts, I want 
those SetCID*** commands executed. Otherwise, I don't want them 
executed. I don't want to embed them into the local/tollfree contexts 
themselves, because then I'd have to figure out some way to store the 
123 and XYZ values so that they could be used by commands in those 
contexts.

Essentially, what I want to do is override the CALLERIDNUM/CALLERIDNAME 
data for calls that are directed outside the PBX, and leave it alone for 
calls inside the PBX. That way internal users can see John Q. Smith 
322 (different for each extension), but outside callees see Smithco 
Widgets 602-555-1212 (which would be identical for all of the 
extensions that can make outside calls).
{clip}
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RE: [Asterisk-Users] Any way to _always_ execute certain commands in a dialplan context?

[Kris Boutilier]

If 'immediate=yes' then the target exten in the context for the zap line
will always be 's', where you would implement digit collection or whatever.
If 'immediate=no' then the simple switch code will collect the digits and
dive in to the context with something to match against, thereby ignoring
's'.

-Original Message-
From: Rob Fugina [mailto:[EMAIL PROTECTED]
Sent: September 3, 2004 8:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Any way to _always_ execute certain
commands in a dialplan context?


Ah, well...  Never tried it with SIP phones.  I thought I had used
that before for inbound calls on a Zap channel, and with local Zap
extensions, too...

On Fri, 03 Sep 2004 08:11:09 -0700, Kevin P. Fleming
[EMAIL PROTECTED] wrote:
 Rob Fugina wrote:
  Use the 's' extension...
 
 
 Uhh, no. That doesn't work at all.
 
{clip}
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Re: [Asterisk-Users] BIG ISSUE with SIP, not sure where to go but it's killing asterisk.

[Daniel Jimenez]

To top this off, I also get PRI errors
Sep  3 10:56:52 NOTICE[196620]: chan_zap.c:7027 pri_dchannel: PRI got 
event: 6 on Primary D-channel of span 1
Sep  3 10:56:52 NOTICE[196620]: chan_zap.c:7027 pri_dchannel: PRI got 
event: 6 on Primary D-channel of span 1
Sep  3 10:56:52 NOTICE[196620]: chan_zap.c:7027 pri_dchannel: PRI got 
event: 8 on Primary D-channel of span 1
Sep  3 10:56:52 NOTICE[196620]: chan_zap.c:7027 pri_dchannel: PRI got 
event: 6 on Primary D-channel of span 1
Sep  3 10:56:52 NOTICE[196620]: chan_zap.c:7027 pri_dchannel: PRI got 
event: 6 on Primary D-channel of span 1
Sep  3 10:56:52 NOTICE[196620]: chan_zap.c:7027 pri_dchannel: PRI got 
event: 6 on Primary D-channel of span 1
Sep  3 10:56:52 NOTICE[196620]: chan_zap.c:7027 pri_dchannel: PRI got 
event: 8 on Primary D-channel of span 1
Sep  3 10:56:52 NOTICE[196620]: chan_zap.c:7027 pri_dchannel: PRI got 
event: 6 on Primary D-channel of span 1
Sep  3 10:56:52 NOTICE[196620]: chan_zap.c:7027 pri_dchannel: PRI got 
event: 6 on Primary D-channel of span 1
Sep  3 10:56:52 NOTICE[196620]: chan_zap.c:7027 pri_dchannel: PRI got 
event: 6 on Primary D-channel of span 1
Sep  3 10:56:52 NOTICE[196620]: chan_zap.c:7027 pri_dchannel: PRI got 
event: 8 on Primary D-channel of span 1
Sep  3 10:56:52 NOTICE[196620]: chan_zap.c:7027 pri_dchannel: PRI got 
event: 6 on Primary D-channel of span 1
Sep  3 10:56:52 NOTICE[196620]: chan_zap.c:7027 pri_dchannel: PRI got 
event: 6 on Primary D-channel of span 1
Sep  3 10:56:52 NOTICE[196620]: chan_zap.c:7027 pri_dchannel: PRI got 
event: 6 on Primary D-channel of span 1
Sep  3 10:56:52 NOTICE[196620]: chan_zap.c:7027 pri_dchannel: PRI got 
event: 8 on Primary D-channel of span 1

Daniel Jimenez wrote:
I frequently get this error message, it repeats itself hundred/thousands 
of times and never stops.

chan_sip.c:7467 sipsock_read: Failed to grab lock, trying again...
During this period, I can make no SIP calls what-so-ever. The only way 
I've been able to stop it is to killall -9 asterisk. Doing a restart now 
doesn't respond.

Anyone know why?
--
Daniel Jimenez djimenez[at]pobox[dot]com
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[Asterisk-Users] AgentCallbackLogin by other means

[Corey S. McFadden]

Hi,

We’re looking at options for logging agents into the system
programmatically via Perl/PHP and I was wondering if anyone else is doing
this and if so, how.  We're using AgentCallbackLogin now but would like to
set up a web interface instead.  I've been looking at Asterisk::Manager
and didn't see anything relevant and wanted to ask the group before we
dove into the Asterisk source.

Any input would be immensely appreciated...

-Corey



--
Corey S. McFadden ([EMAIL PROTECTED])
McFadden Associates - Technology Consultants
phone 215-825-2121 x510  - web.csma.biz




*
This message has been scanned for viruses and
dangerous content, and is believed to be clean.

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RE: [Asterisk-Users] Dell PowerEdge 750 rackmount

[Scott Stingel]

Hi Angel-

Had trouble getting Dell's in Portugal, however customer can get HP Proliant
DL320's.  I had one shipped to me here, and ran it through some load tests.
Seems fine.

Thanks for responding!
Scott 

Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California  London England
www.evtmedia.com 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Angel Gomez
Sent: Thursday, September 02, 2004 10:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Dell PowerEdge 750 rackmount

Hi Scott.

I have used servers from advansor, one with a 2 Xeon cpus, 2 nics, hw
raid and a te405p card, and another with 1 P4 cpu and 1 t100p, both working
veri well.

The only bad thing is that advansor site has an Altigen add ;-p

Scott Stingel wrote:

Hi-

I have an upcoming order for a bunch of asterisk boxes, and I'm 
considering using an assembled package for the server, instead of 
building them from components as I usually do.

Does anyone have experience with the Dell PowerEdge 750 server, or any 
other 1U rackmount server for use with asterisk?

Thanks in advance
Scott Stingel
 
 
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California  London England
www.evtmedia.com


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[Asterisk-Users] Dropping incompatible voice frame

[Carlos Gabriel Drach]

Hi: i have a problem.



Mi extensions.conf:



exten = _N.,1,Setvar(VOICEMAILREQ=${EXTEN})

exten = _N.,2,SetAccount(${customer})

exten = _N.,3,SetCDRUserField(${VOICEMAILREQ:1})

exten = _N.,4,ResponseTimeout(5)

exten = _N.,5,Background(ifyou)

exten = _N.,6,Background(silence/1)

exten = _N.,7,Background(ifyou)

exten = _N.,8,Background(silence/5)

exten = _N.,9,Background(ifyou)

exten = _N.,10,Background(silence/5)

exten = _N.,11,Background(adio)

exten = _N.,12,Wait,1

exten = _N.,13,Hangup





but in step 5:



 -- Executing BackGround(Local/[EMAIL PROTECTED],1,
ifyou) in new stack

Sep 3 11:59:22 WARNING[14350]: format_wav.c:123
check_header: Does not say fmt

Sep 3 11:59:22 WARNING[14350]: file.c:406
ast_filehelper: Unable to open fd on /opt/asterisk/var/lib/sounds/ifyou.wav

Sep 3 11:59:22 WARNING[14350]: file.c:761
ast_streamfile: Unable to open ifyou (format SLINR): No such file or directory

Sep 3 11:59:22 WARNING[14350]: pbx.c:4484
pbx_builtin_background: ast_streamfile failed on Local/[EMAIL PROTECTED],1
fro ifyou

 -- Executing BackGround(Local/[EMAIL PROTECTED],1,
silence/1) in new stack

 -- Playing 'silence/1' (language 'en')

 == Spawn extension (callout, x, 2)
exited non-zero on 'Local/[EMAIL PROTECTED],2'

Sep 3 11:59:22 NOTICE[14350]: channel.c:1287
ast_read: Dropping incompatible voice frame on IAX2/voiptalk/1 of format SLINR
since our native format has changed to GSM



Then step 7 is ok.



Any help?



Thanks






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RE: [Asterisk-Users] digitnetworks card issues?

[Rich Adamson]

Not that it makes any significant difference, but the x100p was an
off-the-shelf card that digium integrated into * and spent the time
writing the drivers, etc. The TDM card is a digium copyright design.


 The difference is that digitnetworks specifically targets Digium as
 competition.  Cisco, Sipura, etc, don't directly compete with IAXy
 because they have different feature sets and were around long before
 IAXy was released.  Digium was first on the market with the X100P and
 digitnetworks cloned their product, thus circumvented Digium's RD cost
 and undermines their ability to recover that cost.  Digitnetworks are
 trying to steal a slice of the pie from Digium, and that's why
 supporting them on this list is objectionable.
 
  -Original Message-
  From: Kannaiyan Natesan [mailto:[EMAIL PROTECTED] 
  Sent: Friday, September 03, 2004 2:41 AM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [Asterisk-Users] digitnetworks card issues?
  
  
  Does it mean that we cannot talk about Cisco or other FXS  
  products since 
  IAXy is released??
  I hope this list for every member who uses asterisk not 
  Digium's products 
  users alone.
  
  
  
  - Original Message - 
  From: Jay Milk [EMAIL PROTECTED]
  To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
  [EMAIL PROTECTED]
  Sent: Friday, September 03, 2004 8:09 AM
  Subject: RE: [Asterisk-Users] digitnetworks card issues?
  
  
   Have you contacted digitnetworks for support?  This list is 
  owned and 
   maintained by Digium, who already gave you Asterisk for free.  
   Probably not the best forum to ask for support for a competitive 
   product here.
  
   -Original Message-
   From: Imran Akbar [mailto:[EMAIL PROTECTED]
   Sent: Friday, September 03, 2004 1:38 AM
   To: Asterisk Users Mailing List - Non-Commercial Discussion
   Subject: [Asterisk-Users] digitnetworks card issues?
  
  
   Hi,
   I've purchased two x100p clones, and when I try 
  accessing a  line 
   from asterisk with something like this:
  
   exten = _1NXXNXX,1,Dial(Zap/2/{$EXTEN})
   (is that only supposed to put you on channel 2 or actually 
  dial the # 
   for you?)
  
   but I first hear noise, then a dial tone, but as soon as I start 
   dialing numbers I get feedback and noise, and the call doesn't go 
   through.
  
   Any suggestions?
  
   Thanks,
   Imran
  
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[Asterisk-Users] Re: Sorry, Newbie here

[Jason Kawakami]

- Original Message - 
 Subject: [Asterisk-Users] Sorry, Newbie here
 To: [EMAIL PROTECTED]
 Message-ID: [EMAIL PROTECTED]
 Content-Type: text/plain; charset=iso-8859-1

 I never heard of Asterisk before today, but from what i'm looking at
on the website and hearing, it sounds pretty incredibly.  If I understand
correctly with a 1,500.00 Wildcard TE410p T1 card, a good BSD or Linux
Server, and a couple IP phones or Netmeeting on a few workstations, and of
course, Asterisk which is free; I call have a small call center.

 This can't be?  I was looking at tens of thousands for a Cisco
solution.  Any comments or insight is welcome.


after working the telecom industry for the past 10 years i can tell you to
believe it.  your statement is absolutely true

dont kid yourself though, * has some gotchas especially in call center
functionality, and * requires learning from scratch how open source software
developers interpreted what hardware engineers have done for the past 30
years.  if you have experience in implementing open source solutions and
some telephony background you can build just about anything you want to do
with a telephone and a computer with *.  usually there is a trade off in
cost (read capital expenditure) and installation and maint of these
solutions.

i would suggest to you contacting a consultant (check the listings on
voip-info.org) and contact someone near you about your requirements.  or do
what we all did and download the software from CVS and dive in.

welcome to the brave new world

Jason Kawakami
www.optellabs.com

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Re: [Asterisk-Users] Re: Sorry, Newbie here

[Chris Shaw]

I think one of the greatest things about * is that not only do you get the
most flexible PBX I've ever worked with, but it also can act as a IP gateway
for much less than traditional hardware IP gateways (a. la.
Cisco/Mediatrix/etc...). You can use it to extend an existing PBX and save
thousands per month by terminating your PSTN calls via IP...

 -Chris

- Original Message -
From: Jason Kawakami [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, September 03, 2004 9:24 AM
Subject: [Asterisk-Users] Re: Sorry, Newbie here



 - Original Message -
  Subject: [Asterisk-Users] Sorry, Newbie here
  To: [EMAIL PROTECTED]
  Message-ID: [EMAIL PROTECTED]
  Content-Type: text/plain; charset=iso-8859-1
 
  I never heard of Asterisk before today, but from what i'm looking at
 on the website and hearing, it sounds pretty incredibly.  If I understand
 correctly with a 1,500.00 Wildcard TE410p T1 card, a good BSD or Linux
 Server, and a couple IP phones or Netmeeting on a few workstations, and of
 course, Asterisk which is free; I call have a small call center.
 
  This can't be?  I was looking at tens of thousands for a Cisco
 solution.  Any comments or insight is welcome.
 

 after working the telecom industry for the past 10 years i can tell you to
 believe it.  your statement is absolutely true

 dont kid yourself though, * has some gotchas especially in call center
 functionality, and * requires learning from scratch how open source
software
 developers interpreted what hardware engineers have done for the past 30
 years.  if you have experience in implementing open source solutions and
 some telephony background you can build just about anything you want to do
 with a telephone and a computer with *.  usually there is a trade off in
 cost (read capital expenditure) and installation and maint of these
 solutions.

 i would suggest to you contacting a consultant (check the listings on
 voip-info.org) and contact someone near you about your requirements.  or
do
 what we all did and download the software from CVS and dive in.

 welcome to the brave new world

 Jason Kawakami
 www.optellabs.com

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RE: [Asterisk-Users] digitnetworks card issues?

[Kevin Walsh]

Kannaiyan Natesan [EMAIL PROTECTED] wrote:
 If you could learn from the previous mails around here, as far i have seen
 the issues were discussed based on the use of asterisk with and without
 devices, not just supporting digium alone. You can see mails from
 broadvoice, voicepulse, iconnecthere. Do they support Digium? never mind
 about it. The issue here is why it is not working with asterisk, how that
 can be resolved and how the users around here solved those problems.
 
Well said.  My followup wouldn't have been quite so polite.  It's lucky
I read the whole thread before responding.

-- 
   _/   _/  _/_/_/_/  _/_/  _/_/_/  _/_/
  _/_/_/   _/_/  _/_/_/_/_/  _/   K e v i n   W a l s h
 _/ _/_/  _/ _/ _/_/  _/_/[EMAIL PROTECTED]
_/   _/  _/_/_/_/  _/_/_/_/  _/_/

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RE: [Asterisk-Users] Group Dial

[Tomica Crnek]

Title: Message



TRUNKBP=Zap/g2

This is E1 trunk to Ericsson BusinessPhone 
PBX.
The channel is not answered in that moment. First ring goes 
to all phones, and after that only first phone continues ringing and only this 
one can be answered.

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Robinson 
  Tim-W10277Sent: Wednesday, September 01, 2004 6:01 PMTo: 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: 
  [Asterisk-Users] Group Dial
  
  What 
  is your definition of TRUNKBP ?
  
  It 
  is probably because that channel is being answered 
  first
  
  Rgds
  Tim
  

-Original Message-From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Tomica 
CrnekSent: 01 September 2004 15:19To: 
[EMAIL PROTECTED]Subject: [Asterisk-Users] Group 
Dial
Hi 
everyone,

I want to have 
a group and dial multiple phones/lines simultaneously. If I use this Dial 
command:

exten = 
222,2,Dial(${TRUNKBP}/246SIP/258${TRUNKBP}/243,20,tTr)

... all phones 
ring just once, after that only the first one continues ringing and only 
that one can answer. Can anyone tell me why?

thanks!
Tomica

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RE: [Asterisk-Users] Group Dial

[Tomica Crnek]

The new one, it was upgraded few days ago

CVS-HEAD-08/29/04-13:17:08 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Kevin Walsh
 Sent: Wednesday, September 01, 2004 5:08 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Group Dial
 
 Tomica Crnek [EMAIL PROTECTED] wrote:
  (Article auto-converted from unnecessary HTML to nice plain text.)
  
  I want to have a group and dial multiple phones/lines 
 simultaneously. 
  If I use this Dial command:
  
  exten = 222,2,Dial(${TRUNKBP}/246SIP/258${TRUNKBP}/243,20,tTr)
  
  ... all phones ring just once, after that only the first 
 one continues 
  ringing and only that one can answer. Can anyone tell me why?
  
 I haven't noticed that on my setups; all phones ring as 
 expected.  Are you using the latest CVS version or some old 
 version.  Perhaps an upgrade will help.
 
 By the way, I don't use the [tTr] flags either, but I don't 
 think that makes a difference in this case.
 
 -- 
_/   _/  _/_/_/_/  _/_/  _/_/_/  _/_/
   _/_/_/   _/_/  _/_/_/_/_/  _/   K e v i n   
 W a l s h
  _/ _/_/  _/ _/ _/_/  _/_/[EMAIL PROTECTED]
 _/   _/  _/_/_/_/  _/_/_/_/  _/_/
 
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Re: [Asterisk-Users] BIG ISSUE with SIP, not sure where to go but it's killing asterisk.

[Chad Scott]

Do these two events coincide?  If so, I'd suspect memory problems.

If they don't coincide, I'd still suspect memory, but I'd also look at
IRQ sharing issues.

On Fri, 2004-09-03 at 09:16, Daniel Jimenez wrote:
 To top this off, I also get PRI errors
 
 
 Sep  3 10:56:52 NOTICE[196620]: chan_zap.c:7027 pri_dchannel: PRI got 
 event: 6 on Primary D-channel of span 1
 Sep  3 10:56:52 NOTICE[196620]: chan_zap.c:7027 pri_dchannel: PRI got 
 event: 6 on Primary D-channel of span 1
 Sep  3 10:56:52 NOTICE[196620]: chan_zap.c:7027 pri_dchannel: PRI got 
 event: 8 on Primary D-channel of span 1
 Sep  3 10:56:52 NOTICE[196620]: chan_zap.c:7027 pri_dchannel: PRI got 
 event: 6 on Primary D-channel of span 1
 Sep  3 10:56:52 NOTICE[196620]: chan_zap.c:7027 pri_dchannel: PRI got 
 event: 6 on Primary D-channel of span 1
 Sep  3 10:56:52 NOTICE[196620]: chan_zap.c:7027 pri_dchannel: PRI got 
 event: 6 on Primary D-channel of span 1
 Sep  3 10:56:52 NOTICE[196620]: chan_zap.c:7027 pri_dchannel: PRI got 
 event: 8 on Primary D-channel of span 1
 Sep  3 10:56:52 NOTICE[196620]: chan_zap.c:7027 pri_dchannel: PRI got 
 event: 6 on Primary D-channel of span 1
 Sep  3 10:56:52 NOTICE[196620]: chan_zap.c:7027 pri_dchannel: PRI got 
 event: 6 on Primary D-channel of span 1
 Sep  3 10:56:52 NOTICE[196620]: chan_zap.c:7027 pri_dchannel: PRI got 
 event: 6 on Primary D-channel of span 1
 Sep  3 10:56:52 NOTICE[196620]: chan_zap.c:7027 pri_dchannel: PRI got 
 event: 8 on Primary D-channel of span 1
 Sep  3 10:56:52 NOTICE[196620]: chan_zap.c:7027 pri_dchannel: PRI got 
 event: 6 on Primary D-channel of span 1
 Sep  3 10:56:52 NOTICE[196620]: chan_zap.c:7027 pri_dchannel: PRI got 
 event: 6 on Primary D-channel of span 1
 Sep  3 10:56:52 NOTICE[196620]: chan_zap.c:7027 pri_dchannel: PRI got 
 event: 6 on Primary D-channel of span 1
 Sep  3 10:56:52 NOTICE[196620]: chan_zap.c:7027 pri_dchannel: PRI got 
 event: 8 on Primary D-channel of span 1
 
 Daniel Jimenez wrote:
  I frequently get this error message, it repeats itself hundred/thousands 
  of times and never stops.
  
  chan_sip.c:7467 sipsock_read: Failed to grab lock, trying again...
  
  During this period, I can make no SIP calls what-so-ever. The only way 
  I've been able to stop it is to killall -9 asterisk. Doing a restart now 
  doesn't respond.
  
  Anyone know why?

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[Asterisk-Users] Call Parking with Queues

[Ronan Eckelberry]

Quick questionI have queues setup, when an agent parks a customer
and the park times out, it goes back to the queue.  Is there any way to
get it to go back to the extension of the agent that parked them without
using the ParkAndAnnounce cmd?

Thanks,

-Ronan

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RE: [Asterisk-Users] digitnetworks card issues?

[Kevin Walsh]

Jay Milk [EMAIL PROTECTED] lazily top-posted:
 The difference is that digitnetworks specifically targets Digium as
 competition.

Competition is a good thing, in my view.

I didn't find out about the non-Digium X100P cards until after I'd
bought mine (for use at home).  If I'd known then I probably would have
avoided the massive markup and bought one of the clones.  These days,
I'd recommend a Sipura SPA-3000.

Perhaps you'd like to boycott Sipura products, as they represent direct
competition to the likes of the X100P and TDM cards.


 Digium was first on the market with the X100P and
 digitnetworks cloned their product, thus circumvented Digium's RD cost
 and undermines their ability to recover that cost.  Digitnetworks are
 trying to steal a slice of the pie from Digium, and that's why
 supporting them on this list is objectionable.

You've not seen the http://www.zapatatelephony.org/ website then, I
take it.  All of the X100P cards are clones, and I remember when you
could download the X100P artwork from that website.  The X100P artwork
and specs seem to have been removed now, for whatever reason.

The T400P (and E400P) are clones of the Zapata Tormenta II, and anyone
can download the artwork to build and sell their own version.  If the
owners of the Zapata Telephony project didn't want people to use their
designs then they would not have released them under the GPL and
published them on a public website.

As they say on their website:

¡Viva la revolución de las computadoras telefónicas! ¡Viva Zapata!

Live with it.

--
   _/   _/  _/_/_/_/  _/_/  _/_/_/  _/_/
  _/_/_/   _/_/  _/_/_/_/_/  _/   K e v i n   W a l s h
 _/ _/_/  _/ _/ _/_/  _/_/[EMAIL PROTECTED]
_/   _/  _/_/_/_/  _/_/_/_/  _/_/

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Re: [Asterisk-Users] digitnetworks card issues?

[Chris Shaw]

 The T400P (and E400P) are clones of the Zapata Tormenta II, and anyone
 can download the artwork to build and sell their own version.  If the
 owners of the Zapata Telephony project didn't want people to use their
 designs then they would not have released them under the GPL and
 published them on a public website.



Last time I looked on there I think they even published the gerber files so
you could feed them into a CAM

-Chris

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Re: [Asterisk-Users] digitnetworks card issues?

[Lee Howard]

On Friday, September 03, 2004 8:45 AM William Suffill wrote:
Digitnetworks is profiting off the cards so they should support them.
I think that it wasn't so much an issue of Digitnetworks vs. Digium 
supporting them, but rather Asterisk supporting them.

If it wasn't for Digium there wouldn't be Asterisk anyway.
So what you're trying to indicate is that Asterisk should only support 
Digium, and I think that many, many people here would disagree with you 
on that.

So doesn't
that make it better to support the primary company for software that
many of you use every day at home and work?
No, it doesn't.  Competition among hardware vendors is good for 
Asterisk.

Digium GPL'ed Asterisk as a business decision, and they have profited 
from that decision.  Good for them.  But, Digium's bottom-line 
shouldn't be Asterisk's primary concern.

Lee.
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Re: [Asterisk-Users] FXO Disconnect supervision problem

[Glen Johnson]

On  September 01, 2004 12:06 PM, Scott Laird wrote:
This brings up an interesting point--disconnect supervision *mostly* works
for me with a X100P in the US.  The exception is when calls go to
voicemail; I frequently end up with ~90 seconds of dialtone instead of a
message or a clean disconnect.  This has remained constant for 6 months,
up through RC1.
Thanks for mentioning this Scott, it made me try some different tests.
We are using Asterisk as an H323-PSTN gateway.  So the FXS interfaces are 
never used, only FXO.   And it doesn't seem to matter which direction, PSTN 
 H323 or vice versa, Asterisk never catches the PSTN disconnect.

I just tried dialing from an internal line (FXS) out to a pstn number and 
then hung up the far-end.  Asterisk caught it.   So it appears DS is working 
when bridging Zaptel to Zaptel but not Zaptel to (some) applications and 
channel drivers.   With SIP, DS appears to work when the SIP-phone calls out 
and the (pstn) far-end disconnects, but not the other way around.

According to the asterisk-console, when a pstn callers connects:  after they 
hang up, asterisk will always timeout and then hang up.  It never catches 
the hang up when it actually happens.   And also, zap show channel x 
reports the channel is offhook even though it isn't (and will still 
answers calls).

At Digium-support's request, I updated to CVS-HEAD-08/31/04-07:58:19.   But 
the problem persists.

Anyone else having (or had or fixed) this problem?
Cheers
Glen 

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[Asterisk-Users] Slow Robotic or like underwater voice

[Celedonio Albarran]

Hello All:



We have latest cvs version running on FC2 with one digium
card for PSTN.



When we call the asterisk server the demo greeting answer
but we hear a unintelligible voice with a robotic or like underwater voice. Any
ideas on this issue will be appreciated.



Thanks



Cele






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RE: [Asterisk-Users] which distro for asterisk?

[Paul Mahler]

The Mepis Debian distro is pre-configured for *, www.mepis.org  They spent a
lot of time making Mepis work with * out of the box. 

Everyone has their own very strong opinions on which distro is better. I'm
not about to get into that. All I can say is Mepis is probably your fastest
easiest way to get * running. You can get Linux installed and * running VERY
quickly if you start with Mepis. 

Hope this helps,

Paul


Paul Mahler 
[EMAIL PROTECTED]   
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
 94107-1901

 Asterisk Services and Training

 

 

 

 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Tzafrir Cohen
 Sent: Tuesday, August 31, 2004 6:07 AM
 To: Asterisk Users List
 Subject: [Asterisk-Users] which distro for asterisk?
 
 Hi
 
 I want to play a bit with Asterisk. I currentlly install a 
 new system for that and I would like to get your 
 recommendations regarding the linux distro to use there.
 
 This is NOT intended to become a general distro flame war. My 
 favorite distro is  and no argument that you flame 
 will convince me here (probably because I've heard it before).
 
 However I would like to minimize the OS maintinance task. I 
 really wouldn't like to start worrying about upgrading sshd 
 due to some stupid secuirty hole, and to worry what will it 
 break on my system. I expect my distro to do that for me. 
 
 I'd also like to have solid astrisk packages that won't break 
 unnecessarily when the sshd package is updated next time. 
 Hopefully also some sort of integration of zaptel in the 
 distro's kernel package.
 
 I saw numerous complaints about unofficial RPM packages of asterisk.
 Besides them, the following free distros include asterisk packages:
 
 1. Debian: http://packages.debian.org/asterisk . 
 2. Gentoo: Current package seems to be version 0.9.0 from 
 10-May-2004 3. The DAG repository for RH/Fedora:
http://dag.wieers.com/packages/asterisk/
 
 I have some experince with Debian, Mandrake and 
 RedHat/Fedora. I'm unfamiliar with Gentoo and I have no 
 good/bad experince with DAG packages with respect to quality 
 and stability.
 
 Any recommendations, relevant experince and other learned opinions?
 
 thx
 
 -- 
 Tzafrir Cohen   +---+
 http://www.technion.ac.il/~tzafrir/ |vim is a mutt's best friend|
 mailto:[EMAIL PROTECTED]   +---+
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[Asterisk-Users] Sending multi-line sms text

[Asterisk]

I can send sms messages just fine via a calling file, however, I cannot send
messages that have more than one line. How do I encode the message to 

This is line 1
This is line 2
This is line 3

* complains about 2 syntax errors (I presume because the calling file has
three lines for the message), and sends line 1.

Anyone had anything similar ?

Julian.

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Re: [Asterisk-Users] Any way to _always_ execute certain commands in a dialplan context?

[Kevin P. Fleming]

Kris Boutilier wrote:
[foo-context]
; This needs to match the criteria for tollfree, say a 91800 prefix
exten = _91800.,1,SetCIDNum(123)
exten = _91800.,2,SetCIDName(XYZ)
exten = _91800.,3,Goto(tollfree,${EXTEN},1)
This is the direction I started going; however, I need to implement this 
for multiple clients, and I'm not keen on duplicating the pattern 
matching in separate contexts for each client. That's why I was trying 
to find a solution that would let me use an included context, but 
still provide commands to be executed if that included context found a 
match. I may work on coding this up, as I think it could be very useful.

It could also be implemented as:
[foo-context]
; This needs to match the criteria for tollfree, say a 91800 prefix
exten = _91800.,1,Macro(setOutgoingCLID)
exten = _91800.,2,Goto(tollfree,${EXTEN},1)
[macro-setOutgoingCLID]
exten = s,1,SetCIDNum(123)
exten = s,2,SetCIDName(XYZ)
I also considered this, and if there was any way to have something like:
exten = _91800.,1,Macro(setCID#${ACCOUNTCODE})
then it would work well. I have not yet tried this, but based on the way 
the dialplan is imported into Asterisk (and displayed via show 
dialplan), I don't think it's possible for the decision of which macro 
to call to be made at run-time (as opposed to config file parsing time).

Another option would be to use Goto(Local/${ACCOUNTCODE}-CLID), to make 
the subroutine call at runtime, but I'm leery of doing that for a 
couple of reasons: it could seriously mess up my CDR, and I don't know 
(without testing) if SetCIDNum/SetCIDName changes made in the called 
context will propagate back (since going to Local sort-of creates a 
new channel).
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Re: [Asterisk-Users] digitnetworks card issues?

[Chris Shaw]

Lol... This never clicked before... It's called Zapata Tormenta (Shoe
Storm)... Like a bunch of women at a shoe sale I guess...

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[Asterisk-Users] X100P blows up after a while (really loud noise)

[Marconi Rivello]

Two days ago, I was talking on the phone from the FXO, to a SIP phone.
After some time (like 1h30m), all of a sudden, there's a huge noise,
like a buzz... Really loud. So I hungup, and called my asterisk box
again... All I could hear was that sound. Someone called me from the
internet, and as Asterisk dialed the FXO, all she heard was that noise
too.

So, I logged in my Asterisk server, restarted the Asterisk (just the
software). Didn't work. So I stopped it, unloaded the wcfxo module,
loaded it up again and it was just fine. I could call the FXO and use
it just fine. Weird.

Last night, I talked for about 2 hours straight. No problem. But, this
morning, when someone called the FXO all that could be heard was that
loud noise.

I could make a stop-asterisk; reload modules; start-asterisk script,
and a cron entry or something to do it periodically, even check to see
if there's any call on progress before restarting, but that's just a
very ugly solution... If I could check the wcfxo status and get some
info that tells me if it's in buzzer-mode-on, I could come up with a
more elegant solution.

I don't know if it helps: the FXO card, at the first day, was sharing
IRQ with the soundcard. But there wasn't any software using the
soundcard. Yesterday, I unloaded all the sound modules, and checked
/proc/interrupts. No IRQ sharing... But the problem occurred again
later... In this cheap MoBo there's no option to mess around with IRQs
in the BIOS. Today, I'm gonna disable onboard sound, to see if it
helps at all, but I think that without modules loaded, it would have
the same effect. I'll try this just to make sure...

Did someone have this problem too? Any ideas, thoughts, suggestions...?

Thanks,
Marconi Rivello.
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[Asterisk-Users] New to *

[Bill Andersen]

I just ran across the * site.  Looks great.  I do not need
a PBX at this time, but DO need to replace an old voice mail
system.  I'll do my homework and figure out the specifics,
but before I dive into it all and spend a bunch of time only
to find out I didn't understand, is it reasonable to think I
could configure * to simply act as a voicemail system off an
existing PBX?  It looks possible to me.

Who knows, I might learn enough about how it all works to
actually end up replacing my PBX.  But for now, with proper
configuration, could it act as a voice mail system?

TIA
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RE: [Asterisk-Users] which distro for asterisk?

[Mike Chapman]

Are the test versions configured for * out of the box?

Mike C.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Mahler
Sent: Friday, September 03, 2004 1:27 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] which distro for asterisk?

The Mepis Debian distro is pre-configured for *, www.mepis.org  They spent a
lot of time making Mepis work with * out of the box. 

Everyone has their own very strong opinions on which distro is better. I'm
not about to get into that. All I can say is Mepis is probably your fastest
easiest way to get * running. You can get Linux installed and * running VERY
quickly if you start with Mepis. 

Hope this helps,

Paul


Paul Mahler 
[EMAIL PROTECTED]   
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
 94107-1901

 Asterisk Services and Training

 

 

 

 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Tzafrir Cohen
 Sent: Tuesday, August 31, 2004 6:07 AM
 To: Asterisk Users List
 Subject: [Asterisk-Users] which distro for asterisk?
 
 Hi
 
 I want to play a bit with Asterisk. I currentlly install a 
 new system for that and I would like to get your 
 recommendations regarding the linux distro to use there.
 
 This is NOT intended to become a general distro flame war. My 
 favorite distro is  and no argument that you flame 
 will convince me here (probably because I've heard it before).
 
 However I would like to minimize the OS maintinance task. I 
 really wouldn't like to start worrying about upgrading sshd 
 due to some stupid secuirty hole, and to worry what will it 
 break on my system. I expect my distro to do that for me. 
 
 I'd also like to have solid astrisk packages that won't break 
 unnecessarily when the sshd package is updated next time. 
 Hopefully also some sort of integration of zaptel in the 
 distro's kernel package.
 
 I saw numerous complaints about unofficial RPM packages of asterisk.
 Besides them, the following free distros include asterisk packages:
 
 1. Debian: http://packages.debian.org/asterisk . 
 2. Gentoo: Current package seems to be version 0.9.0 from 
 10-May-2004 3. The DAG repository for RH/Fedora:
http://dag.wieers.com/packages/asterisk/
 
 I have some experince with Debian, Mandrake and 
 RedHat/Fedora. I'm unfamiliar with Gentoo and I have no 
 good/bad experince with DAG packages with respect to quality 
 and stability.
 
 Any recommendations, relevant experince and other learned opinions?
 
 thx
 
 -- 
 Tzafrir Cohen   +---+
 http://www.technion.ac.il/~tzafrir/ |vim is a mutt's best friend|
 mailto:[EMAIL PROTECTED]   +---+
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Re: [Asterisk-Users] digitnetworks card issues?

[Scott Laird]

On Sep 3, 2004, at 10:12 AM, Kevin Walsh wrote:
Competition is a good thing, in my view.
I didn't find out about the non-Digium X100P cards until after I'd
bought mine (for use at home).  If I'd known then I probably would have
avoided the massive markup and bought one of the clones.  These days,
I'd recommend a Sipura SPA-3000.
I think the specific point of the pro-Digium anti-clone argument is 
this:

There's nothing inherently special about the X100P.  It's really just a 
$10 winmodem.  The *only* reason that anyone cares about it is because 
Digium spent the money to develop an Asterisk driver for it.  They 
recoup their costs for developing the X100 driver (and Asterisk itself) 
by selling the card at an impressive markup.  From a simple economic 
standpoint, this isn't really a rational move on their part--there's no 
simple reason for people to pay $100 to them when they could pay $15 to 
newegg or someone on ebay.  However, the very fact that they're willing 
to go out on a limb like this is rather endearing to a lot of us.  
They've spent years building Asterisk and giving it away for free.  In 
exchange, we've paid the markup on their PCI cards as a sort of proxy 
for paying for Asterisk itself, and we encourage others to do the same. 
 It's our way of keeping Asterisk economically viable while waiting for 
the VoIP market to mature.  Digium gave us Asterisk, along with 
relatively inexpensive hardware, and in exchange, we've given them a 
bit more cash then we strictly had to.

Scott
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Re: [Asterisk-Users] X100P blows up after a while (really loud noise)

[Tor Roberts]

Marconi,
I don't know if this is will help you, but I had problems with some 
TDM400p cards. They worked fine, but after about 10 minutes in use there 
was a very loud static, humming noise. The cards where brand new, rev. 
G. I spoke with Digium about the problem, and they suggested that I 
update to the latest Asterisk, as there was a driver change in the last 
month (I was running a version from July). So I updated Asterisk, 
rebooted, and now my cards work great.  Hope that helps!

-Tor
Marconi Rivello wrote:
Two days ago, I was talking on the phone from the FXO, to a SIP phone.
After some time (like 1h30m), all of a sudden, there's a huge noise,
like a buzz... Really loud. So I hungup, and called my asterisk box
again... All I could hear was that sound. Someone called me from the
internet, and as Asterisk dialed the FXO, all she heard was that noise
too.
So, I logged in my Asterisk server, restarted the Asterisk (just the
software). Didn't work. So I stopped it, unloaded the wcfxo module,
loaded it up again and it was just fine. I could call the FXO and use
it just fine. Weird.
Last night, I talked for about 2 hours straight. No problem. But, this
morning, when someone called the FXO all that could be heard was that
loud noise.
I could make a stop-asterisk; reload modules; start-asterisk script,
and a cron entry or something to do it periodically, even check to see
if there's any call on progress before restarting, but that's just a
very ugly solution... If I could check the wcfxo status and get some
info that tells me if it's in buzzer-mode-on, I could come up with a
more elegant solution.
I don't know if it helps: the FXO card, at the first day, was sharing
IRQ with the soundcard. But there wasn't any software using the
soundcard. Yesterday, I unloaded all the sound modules, and checked
/proc/interrupts. No IRQ sharing... But the problem occurred again
later... In this cheap MoBo there's no option to mess around with IRQs
in the BIOS. Today, I'm gonna disable onboard sound, to see if it
helps at all, but I think that without modules loaded, it would have
the same effect. I'll try this just to make sure...
Did someone have this problem too? Any ideas, thoughts, suggestions...?
Thanks,
Marconi Rivello.
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Re: [Asterisk-Users] New to *

[Greg Hill]

On Fri, 3 Sep 2004, Bill Andersen wrote:

 I just ran across the * site.  Looks great.  I do not need a PBX at this
 time, but DO need to replace an old voice mail system.  I'll do my
 homework and figure out the specifics, but before I dive into it all and
 spend a bunch of time only to find out I didn't understand, is it
 reasonable to think I could configure * to simply act as a voicemail
 system off an existing PBX?  It looks possible to me.

 Who knows, I might learn enough about how it all works to actually end
 up replacing my PBX.  But for now, with proper configuration, could it
 act as a voice mail system?

yes, that's entirely reasonable. Probably the trickiest bit will be the
actual connection between your PBX and your Asterisk box. This connection
could be made via an x100p card connected to an analog station port on
your PBX (or multiple connections of the same style). From there you'd
have to work out how to transfer a call in the PBX out to Asterisk via the
analog port extension, and how to signal to Asterisk what mailbox is
wanted (or simply make the transfer and use an IVR in Asterisk so that the
caller can choose a mailbox him/herself).

In any case, it's a relatively inexpensive experiment.

Greg

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[Asterisk-Users] Help setting 2 Offices in US and India

[Ofer Dagan]

I am new to Asterisk  and VoIP. I have been given the task of setting up a telephone 
network in US and India. When customers call the US location, the calls should route 
to India (using VoIP) and handle there. The Indian location should be able to call Us 
numbers using the Voip to save money. The solution should be flexible enough to 
support initial of 5 simultaneous calls with the option to expand to 20+ within a year.

1) Can anyone direct me what is the minimum hardware needed. (or most inexpensive 
solution)

2) If we use dedicated T1 in both location, will the voice quality be good enough? 

3) Can we use Vonage or a company like that for the voip to save on T1 cost?

Thanks for the help,
Mike




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Re: [Asterisk-Users] X100P blows up after a while (really loud noise)

[Marconi Rivello]

Tor,

Unfortunately (?), my Asterisk, Zapata, and Zaptel versions are already 1.0-RC2.

I apreciate your help, though. :)

Best regards,
Marconi.

On Fri, 03 Sep 2004 11:44:29 -0700, Tor Roberts [EMAIL PROTECTED] wrote:
 Marconi,
 I don't know if this is will help you, but I had problems with some
 TDM400p cards. They worked fine, but after about 10 minutes in use there
 was a very loud static, humming noise. The cards where brand new, rev.
 G. I spoke with Digium about the problem, and they suggested that I
 update to the latest Asterisk, as there was a driver change in the last
 month (I was running a version from July). So I updated Asterisk,
 rebooted, and now my cards work great.  Hope that helps!
 
 -Tor
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Re: [Asterisk-Users] Lower cost router suitable for VOIP ?

[Marconi Rivello]

Hi,

I believe what you're looking for is QoS. I didn't mess around with it
yet... But I know you can setup a cheap linux router with it, so your
VoIP traffic will get more priority.

Here's an idea: setup one linux box as a router, with 1 ethernet for
inside voip, another one for the rest, and the last one to the outside
world. I'm sure you'll find the necessary tools for linux QoS.

Maybe you could have only one inside ethernet connection, and the QoS
thing will let the voip traffic pass with higher priority, but I don't
really know about that. The 2 inside ethernet setup sounds easier to
configure...

Hope it helps...
Marconi.

- Original Message -
From: James H. Thompson [EMAIL PROTECTED]
Date: Fri, 3 Sep 2004 09:22:00 -1000
Subject: Re: [Asterisk-Users] Lower cost router suitable for VOIP ?
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]

 
This wiki page has some information on routers that support VOIP: 
  
http://www.voip-info.org/wiki-VOIP+Routers 
  

Jim 
  
James H. Thompson
[EMAIL PROTECTED]
 
 
- Original Message - 
From: Robert Rozman 
To: Asterisk Users Mailing List - Non-Commercial Discussion 
Sent: Thursday, September 02, 2004 11:30 PM 
Subject: [Asterisk-Users] Lower cost router suitable for VOIP ? 

Hi,

we're testing Asterisk 1  RC 2 behind ordinary router and NAT. Since we're
sharing network with web server it seems like voip packets are not coming
through fast enough (Digium demo dies after few seconds...). It's the same
if I make direct calls (passing Asterisk) so we conclude it's network
problem - it also work normally outside our router...

I wonder what solutions can we use to give voice packets higher priority.
I'm avare of VOIP routers, but they are pricey. Can some of common routers
help, or maybe implementing router on another simple Linux box?

Any advice, pointers to more info ?
How to trace network and debug Asterisk in convenient way ?

Thanks in advance,

Robert Rozman

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[Asterisk-Users] MySQL Friends

[imail]

Is it a good idea to use this option? Or its not stable and going to be
replaced soon anyways?
I'm looking for a stable solution to provision users from a db.  Anything
working well w/ *?

TIA
-jon

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[Asterisk-Users] Using AVM Fritz!PCI as zap interface

[Roland Zagler]

Hello!

Is there a way to use AVM Fritz!PCI as a ZAP interface and have it
configured for ZAP channels?

Thanx in advance!

Roland Zagler
mailto:[EMAIL PROTECTED]
@fog smart partners
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Re: [Asterisk-Users] Problem with HasNewVoicemail()

[Umar Sear]

Try to specify the the context, it seems to be using default which may
or may not be right. 


exten = s,1,HasNewVoicemail([EMAIL PROTECTED]|NEWMSGCOUNT)
Umar

On Thu, 2004-09-02 at 12:51, Nick Barnes wrote:
 Hi all,
 
 Maybe I'm being thick here, but I've had a look through the mailing list and
 the Wiki, and I can't seem to see details of anybody else with this
 problem
 
 
 With the following line:
 
   exten = s,1,HasNewVoicemail(201)
 
 I am getting the following error:
 
 -- Executing HasNewVoicemail(SIP/201-2f1e, 201) in new stack
   Sep  2 12:41:09 NOTICE[819221]: app_hasnewvoicemail.c:104
 hasvoicemail_exec: Voice mailbox 201 at
 /var/spool/asterisk/voicemail/default/201/(null) does not exist
   Sep  2 12:41:09 WARNING[819221]: ast_expr.y:474 ast_yyerror:
 ast_yyerror(): syntax error: parse error; Input:
 0 +
 ^
 ^
 
 And if I add the optional variable name to put the new count into:
 
   exten = s,1,HasNewVoicemail(201,NEWMSGCOUNT)
 
 The error message is an even more puzzling:
 
   -- Executing HasNewVoicemail(SIP/201-3277,
 [EMAIL PROTECTED]|NEWMSGCOUNT) in new stack
   Sep  2 12:45:33 NOTICE[851989]: app_hasnewvoicemail.c:104
 hasvoicemail_exec: Voice mailbox 201 at
 /var/spool/asterisk/voicemail/default|NEWMSGCOUNT/201/(null) does not exist
   Sep  2 12:45:33 WARNING[851989]: ast_expr.y:474 ast_yyerror:
 ast_yyerror(): syntax error: parse error; Input:
 0 +
 ^
 ^
 
 
 Which seems to be taking the variable name as part of the mailbox path.
 
 I have tried various combinations of ',' and '|', changing the mailbox to
 '[EMAIL PROTECTED]' and also surrounding parts with '', but the errors are all
 the same. The path '/var/spool/asterisk/voicemail/default/201/' definitely
 exists.
 
 The Asterisk version is - CVS-HEAD-08/13/04-12:00:00-BRI-stuffed-0.1.0-RC4a
 
 Has anybody else seen this error or knows what stupid mistake/assumption
 I've made?
 
 Nick Barnes
 Senior IT Consultant. 
 
 
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[Asterisk-Users] Rejecting Calls in Cisco 7960 --

[Kannaiyan Natesan]

Can Anybody help how to reject an incoming call using 7960?
-Kannaiyan
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Re: [Asterisk-Users] Lower cost router suitable for VOIP ?

[Chris Shaw]

I'd be more than happy to send you some info off-list on how to do this in
Linux... It's much cheaper and more flexible than a low-end hardware
solution...

-Chris

- Original Message -
From: Robert Rozman [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Friday, September 03, 2004 2:30 AM
Subject: [Asterisk-Users] Lower cost router suitable for VOIP ?


 Hi,

 we're testing Asterisk 1  RC 2 behind ordinary router and NAT. Since we're
 sharing network with web server it seems like voip packets are not coming
 through fast enough (Digium demo dies after few seconds...). It's the same
 if I make direct calls (passing Asterisk) so we conclude it's network
 problem - it also work normally outside our router...

 I wonder what solutions can we use to give voice packets higher priority.
 I'm avare of VOIP routers, but they are pricey. Can some of common routers
 help, or maybe implementing router on another simple Linux box?

 Any advice, pointers to more info ?
 How to trace network and debug Asterisk in convenient way ?

 Thanks in advance,

 Robert Rozman

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Re: [Asterisk-Users] X100P blows up after a while (really loud noise)

[Ryan Courtnage]

Marconi,
Marconi Rivello wrote:
Two days ago, I was talking on the phone from the FXO, to a SIP phone.
After some time (like 1h30m), all of a sudden, there's a huge noise,
like a buzz... Really loud. 
You are not alone.  This problem has also been experienced by many with 
tdm400p cards.

There is a thread you can read here:
http://lists.digium.com/pipermail/asterisk-users/2004-July/053630.html
I don't think anyone knows exactly why this happens.  I've made several 
calls to Digium without resolution.

Currently, all my installs are stable.  After standardizing on 
Intel-only hardware, fed with filtered power.  This suggests that the 
MoBo (other other HW) may be to blame.

Good luck
Ryan
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Re: [Asterisk-Users] Using AVM Fritz!PCI as zap interface

[Tim Robinson]

Hi -
no, you can't use the Fritz card as a Zap interface.  Use a card that 
has the HFC chipset.  e.g. Billion, Asustek, etc.  They are around EUR15 
if you shop around.  This works using the bri-stuff drivers from 
www.junghanns.net

Rgds
Tim
Roland Zagler wrote:
Hello!
Is there a way to use AVM Fritz!PCI as a ZAP interface and have it
configured for ZAP channels?
Thanx in advance!
Roland Zagler
mailto:[EMAIL PROTECTED]
@fog smart partners
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RE: [Asterisk-Users] Lower cost router suitable for VOIP ?

[Colin Anderson]

 Any advice, pointers to more info ?

MeshBox'll work:

http://www.locustworld.com/modules.php?op=modloadname=Newsfile=articlesid
=52mode=threadorder=0thold=0

SIP prioritization is supposed to happen regardless if the clients are wired
or wireless. 

The distro is free:

http://www.locustworld.com/modules.php?op=modloadname=Downloadsfile=index;
req=getitlid=5

I've played with it, and it's nice. 
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Re: [Asterisk-Users] Lower cost router suitable for VOIP ?

[Marconi Rivello]

Chris,

I believe it would be nice to send the info also to the list. So
others would be able to benefit as well. You've got at least 2 people
interested :)

Marconi.

On Fri, 3 Sep 2004 13:41:30 -0700, Chris Shaw [EMAIL PROTECTED] wrote:
 I'd be more than happy to send you some info off-list on how to do this in
 Linux... It's much cheaper and more flexible than a low-end hardware
 solution...
 
 -Chris
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