[Asterisk-Users] Skype - Bandwidth

2005-03-14 Thread César Davi Ávila do Nascimento 



Hi All,

Does anyone know the amount of memory used by skype?
regards
César
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Re: [Asterisk-Users] Skype - Bandwidth

2005-03-14 Thread César Davi Ávila do Nascimento 
Ok... take it easy... But do you know skype, don't you?

regards

- Original Message - 
From: Steven Critchfield [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, March 14, 2005 3:56 PM
Subject: Re: [Asterisk-Users] Skype - Bandwidth


On Mon, 2005-03-14 at 14:30 -0300, César Davi Ávila do Nascimento wrote:
 Hi All,

 Does anyone know the amount of memory used by skype?

Did you think about the best venue to ask this question. We are not a
skype support forum.

And BTW, TURN OFF HTML EMAIL. Exercise some thought about those who will
read your message before assuming we all need to see blue.
-- 
Steven Critchfield [EMAIL PROTECTED]

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Re: [Asterisk-Users] Skype - Bandwidth

2005-03-14 Thread César Davi Ávila do Nascimento 
Talk about skype is forbidden, but to be impolite is allowed...
Great list!

Regards

- Original Message - 
From: Steven Critchfield [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, March 14, 2005 5:16 PM
Subject: Re: [Asterisk-Users] Skype - Bandwidth


On Mon, 2005-03-14 at 16:36 -0300, César Davi Ávila do Nascimento wrote:
 Ok... take it easy... But do you know skype, don't you?

I know of it, I don't use it as I have no use for it and it isn't open
source.

Next lesson is to learn how to trim the bottoms of your messages. You
might want to learn in-line quoting as well and configure your mail app
for proper quoting as well.

 - Original Message - 
 From: Steven Critchfield [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Monday, March 14, 2005 3:56 PM
 Subject: Re: [Asterisk-Users] Skype - Bandwidth


 On Mon, 2005-03-14 at 14:30 -0300, César Davi Ávila do Nascimento wrote:
  Hi All,
 
  Does anyone know the amount of memory used by skype?

 Did you think about the best venue to ask this question. We are not a
 skype support forum.

 And BTW, TURN OFF HTML EMAIL. Exercise some thought about those who will
 read your message before assuming we all need to see blue.
-- 
Steven Critchfield [EMAIL PROTECTED]

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Re: [Asterisk-Users] Skype - Bandwidth

2005-03-14 Thread César Davi Ávila do Nascimento 
Yes... Great List!

- Original Message - 
From: Andrew Kohlsmith [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, March 14, 2005 6:30 PM
Subject: Re: [Asterisk-Users] Skype - Bandwidth


 On March 14, 2005 04:20 pm, Eric Wieling wrote:
  Skype does not interface with Asterisk in any way whatsoever.  You
  could just as well have asked if someone knows what RNA sequence 42 in
  the turnip genome is for.  About as many people on this list would be
  familiar with that as would Skype.
 
 Don't be silly.  RNA doesn't exist in turnips, RNA only exists in animal 
 cells.  :-)
 
 -A.
  
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[Asterisk-Users] IAX Client

2005-02-01 Thread César Davi Ávila do Nascimento 



Hi All,

I'd like to develop an IAX - client.
Does somebody know where can I get the source code 
for an IAX client?

Regards

César
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[Asterisk-Users] IAX Client

2005-02-01 Thread César Davi Ávila do Nascimento 
Hi All,

I'd like to develop an IAX - client.
Does anybody know where can I get the source code for an IAX client?

Regards

César

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Re: [Asterisk-Users] IAX Client

2005-02-01 Thread César Davi Ávila do Nascimento 
Thanks Guys

César

- Original Message - 
From: Denis Galvão - iSolve [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Cc: César Davi Ávila do Nascimento [EMAIL PROTECTED]
Sent: Tuesday, February 01, 2005 1:58 PM
Subject: Re: [Asterisk-Users] IAX Client



Hi Cesar.

Try it out:
http://iaxclient.sourceforge.net

-- 
D e n i s   G a l v ã o
iSolve - Solve Is Our Business
Av. Candido de Abreu, 526 1206B
CEP: 80530-000 - Curitiba - PR
+55 41 252-2977
http://www.isolve.com.br



Em Ter 01 Fev 2005 15:22, César Davi Ávila do Nascimento escreveu:
 Hi All,

 I'd like to develop an IAX - client.
 Does somebody know where can I get the source code for an IAX client?

 Regards

 César


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Re: [Asterisk-Users] IAX Client

2005-02-01 Thread César Davi Ávila do Nascimento 
Other... VxWorks

Regards

César

- Original Message - 
From: [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, February 01, 2005 3:39 PM
Subject: Re: [Asterisk-Users] IAX Client


  I'd like to develop an IAX - client.
  Does somebody know where can I get the source code for an IAX client?
 Please be more precise : is it a Windows/Linux/MacOSX/other that you want
?

 Anyway, have a look here :
http://iaxclient.sourceforge.net/iaxcomm/index.html
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[Asterisk-Users] SIP x NAT

2005-01-31 Thread César Davi Ávila do Nascimento 



Hi All,

I have a question for you:

- "SIP doesn't work behind NAT very 
well"

Do you agree with this sentence?

regards

César
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[Asterisk-Users] SIP x NAT

2005-01-31 Thread César Davi Ávila do Nascimento 
Hi All,

I have a question for you:

- SIP doesn't work behind NAT very well

Do you agree with this sentence?

regards

César

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Re: [Asterisk-Users] SIP x NAT

2005-01-31 Thread César Davi Ávila do Nascimento 
Thanks a lot!

Regards

César

- Original Message - 
From: Rich Adamson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, January 31, 2005 9:18 AM
Subject: Re: [Asterisk-Users] SIP x NAT


  I have a question for you:
 
  - SIP doesn't work behind NAT very well
 
  Do you agree with this sentence?

 Depends. Asterisk behind a nat box tends to be an implementation
 problem limited by the knowledge of the person doing the implementation
 and somewhat by the functionality implemented within the nat box.

 Sip phones behind a nat box (with asterisk on a registered IP address)
 tends to be rather easy, and how well it works depends a lot on how
 well the sip phone vendor implemented nat support.

 Both asterisk and sip phones behind different nat boxes tends to be
 the most difficult to implement and requires the greatest amount of
 knowledge/experience to implement. Again, depends a lot on the
 functionality provided in the nat boxes.

 The issue with sip is that session startup and control occurs across
 udp port 5060, and the two endpoints (* and phone) negotiate another
 set of udp ports for the rtp (voice) session. The choice of which rtp
 ports to use was left up to each sip phone vendor, so the udp port
 number in use could be anything from about 8000 (xlite) to something
 greater then 32,000.

 Some firewall/nat boxes will actually watch the sip rtp negotiation
 process by inspecting the contents of the sip packets, and open up the
 wanted ports. However, most cheap nat boxes don't do that, and leave
 it up to you to statically define/map the ports required.


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Re: [Asterisk-Users] Cisco 79XX phones not talking to asterisk

2005-01-13 Thread César Davi Ávila do Nascimento 
Hi,

I'm using a cisco Phone and it works fine. See below my sip.conf file:

;
;Cisco IP Phone
;
[45]
type=friend
username=xx
secret=xx
nat=yes
dtmfmode=rfc2833
canreinvite=no  --- this is
important!
qualify=1000
defaultip=xx.xx.xx.xx
host=dynamic
disallow=all
allow=ulaw
allow=alaw
allow=g723.1
allow=g726
allow=g729
allow=gsm
allow=ilbc
allow=lpc10
allow=speex

Regards

César

- Original Message - 
From: Jerry Geis [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, January 13, 2005 1:54 PM
Subject: [Asterisk-Users] Cisco 79XX phones not talking to asterisk


 Hi all,

 I have setup my Cisco 79XX phone. Did the tftp, put the config files in
the
 right location with the right names. Booted my phone, it does the tftp
 things,
 the screen shows my extensions everything seems fine. However, when I
 come offhook and try to dial 11 which is just a playback of demo-congrats
 in the dialplan the phone says

 Calling Out (INV)

 below is my sip.conf file - I presume it is not correcly registering
 with asterisk.
 The phone boots DHCP gets an address, loads the SIP software and sets
there
 for me to dial. However, I get the INV when I dial.

 Any ideas on why the phone is displaying invalid and what to do about
it???

 Thanks,

 jerry


 sip.conf
 
 [201]
 type=friend
 dtmfmode=rfc2833
 username=201
 secret=201
 disallow=all
 allow=ulaw
 allow=alaw
 host=dynamic
 context=smvoice-sip
 callerid=Media Assistant 201
 [202]
 type=friend
 dtmfmode=rfc2833
 username=202
 secret=202
 disallow=all
 allow=ulaw
 allow=alaw
 host=dynamic
 context=smvoice-sip
 callerid=Media Assistant 201
 [203]
 type=friend
 dtmfmode=rfc2833
 username=203
 secret=203
 disallow=all
 allow=ulaw
 allow=alaw
 host=dynamic
 context=smvoice-sip
 callerid=Media Assistant 201
 [204]
 type=friend
 dtmfmode=rfc2833
 username=204
 secret=204
 disallow=all
 allow=ulaw
 allow=alaw
 host=dynamic
 context=smvoice-sip
 callerid=Media Assistant 201
 [205]
 type=friend
 dtmfmode=rfc2833
 username=205
 secret=205
 disallow=all
 allow=ulaw
 allow=alaw
 host=dynamic
 context=smvoice-sip
 callerid=Media Assistant 201
 [206]
 type=friend
 dtmfmode=rfc2833
 username=206
 secret=206
 disallow=all
 allow=ulaw
 allow=alaw
 host=dynamic
 context=smvoice-sip
 callerid=Media Assistant 201

 extension.conf
 
 [smvoice-sip]
 exten = 11,1,Playback(demo-congrats)
 exten = 11,2,Hangup

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[Asterisk-Users] TDM400P - Segmentation fault - Help!

2005-01-07 Thread César Davi Ávila do Nascimento 



Hi all,

I'm trying to install a TDM400P card, and I need some 
help.
Please, see below...

after dmesg command:

[EMAIL PROTECTED] root]# dmesgvia82cxxx: board #1 at 
0xD800, IRQ 5Zapata Telephony Interface Registered on major 196PCI: 
Found IRQ 3 for device 00:09.0PCI: Sharing IRQ 3 with 00:10.1Freshmaker 
version: 71Freshmaker passed register testModule 0: Installed -- AUTO 
FXS/DPOModule 1: Installed -- AUTO FXS/DPOModule 2: Installed -- AUTO 
FXO (FCC mode)Module 3: Installed -- AUTO FXO (FCC mode)Found a Wildcard 
TDM: Wildcard TDM400P REV E/F (4 modules)

after [EMAIL PROTECTED] root]# asterisk -cp 
command:

[chan_phone.so] = (Linux Telephony API 
Support) == Parsing '/etc/asterisk/phone.conf': Found == 
Registered channel type 'Phone' (Standard Linux Telephony API 
Driver)[chan_zap.so] = (Zapata Telephony w/PRI) == 
Parsing '/etc/asterisk/zapata.conf': FoundJan 6 14:57:50 
WARNING[-1084944256]: chan_zap.c:665 zt_open: Unable to 
specify channel 1: No such device or addressJan 6 14:57:50 
ERROR[-1084944256]: chan_zap.c:5340 mkintf: Unable to open channel 1: No such 
device or addresshere = 0, tmp-channel = 1, channel = 1Jan 6 
14:57:50 ERROR[-1084944256]: chan_zap.c:7377 setup_zap: Unable to register 
channel '1-2'Jan 6 14:57:50 WARNING[-1084944256]: loader.c:313 
ast_load_resource: chan_zap.so: load_module failed, returning -1 == 
Unregistered channel type 'Tor' == Unregistered channel type 
'Zap'Segmentation fault[EMAIL PROTECTED] 
root]#
Please, see.conf files 
below:

zaptel.conf

fxoks=1-2
fxsks=3-4
loadzone = us
defaultzone=us

zapata.conf

[channels]
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=3.5
txgain=3.5
immediate=no
busydetect=yes
busycount=5
callprogress=no
usecallerid=yes
hidecallerid=no
;calleridcallwaiting=yes
callerid=asreceived
musiconhold=default
relaxdtmf=yes
accountcode=pstn_local
amaflags=billing
echotraining=yes
context=fxs ;Context to 
FXS ports
group=1
signalling=fxo_ks
channel=1-2
context=fxo ;Context to 
FXO ports
group=2
signalling=fxs_ks
channel=3-4

extensions.conf

[fxs]
exten = 100,1,Dial,Zap/1
exten = 100,1,Dial,Zap/2
exten = 
_9X.,1,Dial,Zap/3/${EXTEN:1}
[fxo]
exten = s,1,Dial,Zap/4


thanks

César

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[Asterisk-Users] TDM400P - Problems

2005-01-06 Thread César Davi Ávila do Nascimento 



Hi All



I've bought a TDM400P and need some help with configuration. Can you tell me 
what to do ?

I've tried to install and the message below has appeared:

[EMAIL PROTECTED] asterisk]# modprobe zaptel[EMAIL PROTECTED] asterisk]# modprobe 
wcfxo/lib/modules/2.4.22-1.2115.nptl/misc/wcfxo.o: init_module: No such 
deviceHint: insmod errors can be caused by incorrect module parameters, 
including invalid IO or IRQ parameters. You 
may find more information in syslog or the output from 
dmesg/lib/modules/2.4.22-1.2115.nptl/misc/wcfxo.o: insmod 
/lib/modules/2.4.22-1.2115.nptl/misc/wcfxo.o 
failed/lib/modules/2.4.22-1.2115.nptl/misc/wcfxo.o: insmod wcfxo 
failed

César
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[Asterisk-Users] TDM400P - Segmentation fault

2005-01-06 Thread César Davi Ávila do Nascimento 



Hi all,

I'm trying to install a TDM400P card, and I need some 
help.
Please, see below...

after dmesg command:

[EMAIL PROTECTED] root]# dmesgvia82cxxx: board #1 at 
0xD800, IRQ 5Zapata Telephony Interface Registered on major 196PCI: 
Found IRQ 3 for device 00:09.0PCI: Sharing IRQ 3 with 00:10.1Freshmaker 
version: 71Freshmaker passed register testModule 0: Installed -- AUTO 
FXS/DPOModule 1: Installed -- AUTO FXS/DPOModule 2: Installed -- AUTO 
FXO (FCC mode)Module 3: Installed -- AUTO FXO (FCC mode)Found a Wildcard 
TDM: Wildcard TDM400P REV E/F (4 modules)

after [EMAIL PROTECTED] root]# asterisk -cp 
command:

[chan_phone.so] = (Linux Telephony API 
Support) == Parsing '/etc/asterisk/phone.conf': Found == 
Registered channel type 'Phone' (Standard Linux Telephony API 
Driver)[chan_zap.so] = (Zapata Telephony w/PRI) == 
Parsing '/etc/asterisk/zapata.conf': FoundJan 6 14:57:50 
WARNING[-1084944256]: chan_zap.c:665 zt_open: Unable to 
specify channel 1: No such device or addressJan 6 14:57:50 
ERROR[-1084944256]: chan_zap.c:5340 mkintf: Unable to open channel 1: No such 
device or addresshere = 0, tmp-channel = 1, channel = 1Jan 6 
14:57:50 ERROR[-1084944256]: chan_zap.c:7377 setup_zap: Unable to register 
channel '1-2'Jan 6 14:57:50 WARNING[-1084944256]: loader.c:313 
ast_load_resource: chan_zap.so: load_module failed, returning -1 == 
Unregistered channel type 'Tor' == Unregistered channel type 
'Zap'Segmentation fault[EMAIL PROTECTED] 
root]#
Please, see.conf files 
below:

zaptel.conf

fxoks=1-2
fxsks=3-4
loadzone = us
defaultzone=us

zapata.conf

[channels]
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=3.5
txgain=3.5
immediate=no
busydetect=yes
busycount=5
callprogress=no
usecallerid=yes
hidecallerid=no
;calleridcallwaiting=yes
callerid=asreceived
musiconhold=default
relaxdtmf=yes
accountcode=pstn_local
amaflags=billing
echotraining=yes
context=fxs ;Context to 
FXS ports
group=1
signalling=fxo_ks
channel=1-2
context=fxo ;Context to 
FXO ports
group=2
signalling=fxs_ks
channel=3-4

extensions.conf

[fxs]
exten = 100,1,Dial,Zap/1
exten = 100,1,Dial,Zap/2
exten = 
_9X.,1,Dial,Zap/3/${EXTEN:1}
[fxo]
exten = s,1,Dial,Zap/4

Can someone help me?

thanks

César
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[Asterisk-Users] TDM400P - Segmentation fault

2005-01-06 Thread César Davi Ávila do Nascimento 



Hi all,

I'm trying to install a TDM400P card, and I need some 
help.
Please, see below...

after dmesg command:

[EMAIL PROTECTED] root]# dmesgvia82cxxx: board #1 at 
0xD800, IRQ 5Zapata Telephony Interface Registered on major 196PCI: 
Found IRQ 3 for device 00:09.0PCI: Sharing IRQ 3 with 00:10.1Freshmaker 
version: 71Freshmaker passed register testModule 0: Installed -- AUTO 
FXS/DPOModule 1: Installed -- AUTO FXS/DPOModule 2: Installed -- AUTO 
FXO (FCC mode)Module 3: Installed -- AUTO FXO (FCC mode)Found a Wildcard 
TDM: Wildcard TDM400P REV E/F (4 modules)

after [EMAIL PROTECTED] root]# asterisk -cp 
command:

[chan_phone.so] = (Linux Telephony API 
Support) == Parsing '/etc/asterisk/phone.conf': Found == 
Registered channel type 'Phone' (Standard Linux Telephony API 
Driver)[chan_zap.so] = (Zapata Telephony w/PRI) == 
Parsing '/etc/asterisk/zapata.conf': FoundJan 6 14:57:50 
WARNING[-1084944256]: chan_zap.c:665 zt_open: Unable to 
specify channel 1: No such device or addressJan 6 14:57:50 
ERROR[-1084944256]: chan_zap.c:5340 mkintf: Unable to open channel 1: No such 
device or addresshere = 0, tmp-channel = 1, channel = 1Jan 6 
14:57:50 ERROR[-1084944256]: chan_zap.c:7377 setup_zap: Unable to register 
channel '1-2'Jan 6 14:57:50 WARNING[-1084944256]: loader.c:313 
ast_load_resource: chan_zap.so: load_module failed, returning -1 == 
Unregistered channel type 'Tor' == Unregistered channel type 
'Zap'Segmentation fault[EMAIL PROTECTED] 
root]#
Please, see.conf files 
below:

zaptel.conf

fxoks=1-2
fxsks=3-4
loadzone = us
defaultzone=us

zapata.conf

[channels]
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=3.5
txgain=3.5
immediate=no
busydetect=yes
busycount=5
callprogress=no
usecallerid=yes
hidecallerid=no
;calleridcallwaiting=yes
callerid=asreceived
musiconhold=default
relaxdtmf=yes
accountcode=pstn_local
amaflags=billing
echotraining=yes
context=fxs ;Context to 
FXS ports
group=1
signalling=fxo_ks
channel=1-2
context=fxo ;Context to 
FXO ports
group=2
signalling=fxs_ks
channel=3-4

extensions.conf

[fxs]
exten = 100,1,Dial,Zap/1
exten = 100,1,Dial,Zap/2
exten = 
_9X.,1,Dial,Zap/3/${EXTEN:1}
[fxo]
exten = s,1,Dial,Zap/4

Can someone help me?

thanks

César



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