date:20051010

On Mon, Oct 10, 2005 at 07:32:31AM +0200, Tzafrir Cohen wrote:
 On Sun, Oct 09, 2005 at 07:37:41PM -0400, Steve Gladden wrote:
 
  The debian package installs something else called mpg321 and creates
  an alias or symlink called mpg123 to mpg321.
 
 Get the package mpg123 from non-free

That is: see http://packages.debian.org/mpg123

Follow links from there to the download site. Also look for packages
with names that contain mpg123

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] where can be find zaptel cvs change log ?

2005-10-10 Thread oncemore

asterisk-users

where can be find zaptel cvs change log ?
thanks 

oncemore
[EMAIL PROTECTED]
  2005-10-09


___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] asterisk certification - thread hijack

2005-10-10 Thread snacktime

The original poster's statement about not even receiving any proof that
he was certified is kind of amazing. That's not a certification
by any definition I know of. I would push Digium on that
because they really don't have a leg to stand on if that is true.
If they sold it as a certification then they owe you a certificate of
some shape or form, and also something that say's what the
certification covered. 

I wouldn't be too upset about it either because it is probably an
honest mistake, but I would be firm on demanding that you get what you
paid for. 



___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] AVM Fritz! + chan_capi + mISDN + PTP

2005-10-10 Thread Lionel Riem


Hello everyone,

I have been using an AVM Fritz! card with chan_capi and mISDN for  
quite a while in PTM mode and it was working finely.


Now, I needed more DID/MSN, so I switched to PTP. But now nothing  
works anymore :(


I am using Asterisk on Debian Sarge stable and installed Asterisk  
along with chan_capi from apt-get. I installed mISDN from the CVS of  
isdn4linux.de.


It is :
- Asterisk 1.0.7 with bristuff
- chan_capi 0.3.5

When I load the whole modules lot, I get the following in dmesg:

Modular ISDN Stack core $Revision: 1.25 $
mISDNd: kernel daemon started
ISAC module $Revision: 1.16 $
mISDNd: test event done
CAPI Subsystem Rev 1.1.2.8
capi20: Rev 1.1.2.7: started up with major 68 (middleware+capifs)
ISDN L1 driver version 1.11
ISDN L2 driver version 1.20
mISDN: DSS1 Rev. 1.30
mISDN Capi 2.0 driver file version 1.14
X25 DTE modul version 1.8
AVM Fritz PCI/PnP driver Rev. 1.30
ACPI: PCI interrupt :00:14.0[A] - GSI 10 (level, low) - IRQ 10
mISDN_fcpcipnp: found adapter Fritz!Card PCI v2 at :00:14.0
fritz card cd09a000 dch cd09a094 bch1 cd09a214 bch2 cd09a3a0
AVM PCI V2: stat 0x240020e
AVM PCI V2: Class E Rev 2
AVM PnP: HDLC version 2
mISDN: AVM Fritz!PCIv2 config irq:10 base:0xEC00
spin_lock_adr=cd09a024 now(d015b867)
busy_lock_adr=cd09a024 now(d015b867)
AVM PCI/PnP: reset
AVM PCI/PnP: S0/S1 40/2
Fritz1 ISAC STAR 40
Fritz1 ISAC MODE c0
Fritz1 ISAC ADF2 ff
Fritz1 ISAC ISTA 0
Fritz1 ISAC CIR0 7
mISDN_isac_init: ISACSX
Fritz1 HDLC 1 STA 8200
Fritz1 HDLC 2 STA 8200
AVM Fritz!PCI: IRQ 10 count 4
fritz 1 cards installed



Here is my /etc/asterisk/capi.conf:

;
; CAPI config
;
;
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8

[interfaces]
mode=immediate
isdnmode=ptp
msn=*
incomingmsn=*
controller=1
softdtmf=1
context=dispatcher
accountcode=
devices=2


Here is my /etc/modprobe.d/capi conf file:

alias /dev/capi20 avmfritz
alias char-major-68-0 avmfritz

install avmfritz /sbin/modprobe capi; \
/sbin/modprobe mISDN_core; \
/sbin/modprobe mISDN_l1; \
/sbin/modprobe mISDN_l2; \
/sbin/modprobe l3udss1; \
/sbin/modprobe mISDN_capi; \
/sbin/modprobe mISDN_x25dte; \
/sbin/modprobe --ignore-install avmfritz protocol=0x22

remove avmfritz /sbin/modprobe -r --ignore-remove avmfritz; \
/sbin/modprobe -r mISDN_x25dte; \
/sbin/modprobe -r mISDN_capi; \
/sbin/modprobe -r l3udss1; \
/sbin/modprobe -r mISDN_l2; \
/sbin/modprobe -r mISDN_l1; \
/sbin/modprobe -r mISDN_core; \
/sbin/modprobe -r capi



capiinfo shows me:

asterisk:/etc/asterisk# capiinfo
Number of Controllers : 1
Controller 1:
Manufacturer: mISDN CAPI controller Fritz1
CAPI Version: 2.0
Manufacturer Version: 1.0
Serial Number: 0002
BChannels: 2
Global Options: 0x0018
   DTMF supported
   Supplementary Services supported
B1 protocols support: 0x0003
   64 kbit/s with HDLC framing
   64 kbit/s bit-transparent operation
B2 protocols support: 0x0043
   ISO 7776 (X.75 SLP)
   Transparent
   Transparent (ignoring framing errors of B1 protocol)
B3 protocols support: 0x0005
   Transparent
   ISO 8208 (X.25 DTE-DTE)

  0100
  0200
  1800
  0300
  4300
  0500
       
      

Supplementary services support: 0x0012
   Terminal Portability
   Call Forwarding



In Asterisk, when an incoming call arrives, it shows me the following:

Asterisk Ready.
*CLI capi info
Contr1: 2 B channels total, 2 B channels free.
*CLI capi debug
CAPI Debugging Enabled
*CLI
*CLI
*CLI -- INFO_IND ID=001 #0x0001 LEN=0016
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x18
  InfoElement = 89

-- INFO_IND ID=001 #0x0001 LEN=0016
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x18
  InfoElement = 89

Oct 10 09:17:16 NOTICE[5948]: chan_capi.c:1184 find_pipe: PLCI doesnt  
match last pipe (PLCI = 0x101)
Oct 10 09:17:16 NOTICE[5948]: chan_capi.c:1301 pipe_msg: INFO_IND  
ID=001 #0x0001 LEN=0016

  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x18
  InfoElement = 89
-- CONNECT_IND ID=001 #0x0002 LEN=0044
  Controller/PLCI/NCCI= 0x101
  CIPValue= 0x1
  CalledPartyNumber   = 8120
  CallingPartyNumber  = 01 830123456789
  CalledPartySubaddress   = default
  CallingPartySubaddress  = default
  BC  = 80 90 a3
  LLC = default
  HLC = default
  AdditionalInfo
   BChannelinformation= default
   Keypadfacility = default
   Useruserdata   = default
   Facilitydataarray  = default

Oct 10 09:17:16 NOTICE[5948]: chan_capi.c:1931 capi_handle_msg:  
CONNECT_IND ID=001 #0x0002 LEN=0044

  Controller/PLCI/NCCI= 0x101
  CIPValue= 0x1

[Asterisk-Users] TDM400 not working

2005-10-10 Thread Rudolf Ladyzhenskii


Hi, all

I have installed TDM400 card. I can see it is there (lspci).
But Asterisk does not find it.

phonebox2*CLI zap show status
No Zaptel interface found.

I assume that driver is not loaded, but I am sure I have installed it (I 
compiled zaptel and then re-build asterisk and did make install for both 
zaptel and asterisk).


When asterisk is started I get:
Jan  2 06:28:08 WARNING[3473]: chan_zap.c:872 zt_open: Unable to open 
'/dev/zap/channel': No such file or directory
Jan  2 06:28:08 ERROR[3473]: chan_zap.c:6572 mkintf: Unable to open channel 
2: No such file or directory

here = 0, tmp-channel = 2, channel = 2
Jan  2 06:28:08 ERROR[3473]: chan_zap.c:9927 setup_zap: Unable to register 
channel '2'
Jan  2 06:28:08 WARNING[3473]: loader.c:402 __load_resource: chan_zap.so: 
load_module failed, returning -1
Jan  2 06:28:08 WARNING[3473]: loader.c:523 load_modules: Loading module 
chan_zap.so failed!


Ok, I look in the /dev and I could not find /dev/zap at all! But, there is a 
/dev/zapchannel character device.


Any ideas what can be wrong?

And last question. Does zaptel driver reads configuration file on startup? 
If so, how do I force the driver to update if config file was changed?


Thanks,
Rudolf


___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] AVM Fritz! + chan_capi + mISDN + PTP

2005-10-10 Thread Kib Eki

I think you can't use a Fritz Card for PTP. You need an active card. We use the 
the beronet ISDN Cards with misdn.


Lionel Riem wrote:

Hello everyone,

I have been using an AVM Fritz! card with chan_capi and mISDN for  quite 
a while in PTM mode and it was working finely.


Now, I needed more DID/MSN, so I switched to PTP. But now nothing  works 
anymore :(


I am using Asterisk on Debian Sarge stable and installed Asterisk  along 
with chan_capi from apt-get. I installed mISDN from the CVS of  
isdn4linux.de.


It is :
- Asterisk 1.0.7 with bristuff
- chan_capi 0.3.5

When I load the whole modules lot, I get the following in dmesg:

Modular ISDN Stack core $Revision: 1.25 $
mISDNd: kernel daemon started
ISAC module $Revision: 1.16 $
mISDNd: test event done
CAPI Subsystem Rev 1.1.2.8
capi20: Rev 1.1.2.7: started up with major 68 (middleware+capifs)
ISDN L1 driver version 1.11
ISDN L2 driver version 1.20
mISDN: DSS1 Rev. 1.30
mISDN Capi 2.0 driver file version 1.14
X25 DTE modul version 1.8
AVM Fritz PCI/PnP driver Rev. 1.30
ACPI: PCI interrupt :00:14.0[A] - GSI 10 (level, low) - IRQ 10
mISDN_fcpcipnp: found adapter Fritz!Card PCI v2 at :00:14.0
fritz card cd09a000 dch cd09a094 bch1 cd09a214 bch2 cd09a3a0
AVM PCI V2: stat 0x240020e
AVM PCI V2: Class E Rev 2
AVM PnP: HDLC version 2
mISDN: AVM Fritz!PCIv2 config irq:10 base:0xEC00
spin_lock_adr=cd09a024 now(d015b867)
busy_lock_adr=cd09a024 now(d015b867)
AVM PCI/PnP: reset
AVM PCI/PnP: S0/S1 40/2
Fritz1 ISAC STAR 40
Fritz1 ISAC MODE c0
Fritz1 ISAC ADF2 ff
Fritz1 ISAC ISTA 0
Fritz1 ISAC CIR0 7
mISDN_isac_init: ISACSX
Fritz1 HDLC 1 STA 8200
Fritz1 HDLC 2 STA 8200
AVM Fritz!PCI: IRQ 10 count 4
fritz 1 cards installed



Here is my /etc/asterisk/capi.conf:

;
; CAPI config
;
;
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8

[interfaces]
mode=immediate
isdnmode=ptp
msn=*
incomingmsn=*
controller=1
softdtmf=1
context=dispatcher
accountcode=
devices=2


Here is my /etc/modprobe.d/capi conf file:

alias /dev/capi20 avmfritz
alias char-major-68-0 avmfritz

install avmfritz /sbin/modprobe capi; \
/sbin/modprobe mISDN_core; \
/sbin/modprobe mISDN_l1; \
/sbin/modprobe mISDN_l2; \
/sbin/modprobe l3udss1; \
/sbin/modprobe mISDN_capi; \
/sbin/modprobe mISDN_x25dte; \
/sbin/modprobe --ignore-install avmfritz protocol=0x22

remove avmfritz /sbin/modprobe -r --ignore-remove avmfritz; \
/sbin/modprobe -r mISDN_x25dte; \
/sbin/modprobe -r mISDN_capi; \
/sbin/modprobe -r l3udss1; \
/sbin/modprobe -r mISDN_l2; \
/sbin/modprobe -r mISDN_l1; \
/sbin/modprobe -r mISDN_core; \
/sbin/modprobe -r capi



capiinfo shows me:

asterisk:/etc/asterisk# capiinfo
Number of Controllers : 1
Controller 1:
Manufacturer: mISDN CAPI controller Fritz1
CAPI Version: 2.0
Manufacturer Version: 1.0
Serial Number: 0002
BChannels: 2
Global Options: 0x0018
   DTMF supported
   Supplementary Services supported
B1 protocols support: 0x0003
   64 kbit/s with HDLC framing
   64 kbit/s bit-transparent operation
B2 protocols support: 0x0043
   ISO 7776 (X.75 SLP)
   Transparent
   Transparent (ignoring framing errors of B1 protocol)
B3 protocols support: 0x0005
   Transparent
   ISO 8208 (X.25 DTE-DTE)

  0100
  0200
  1800
  0300
  4300
  0500
       
      

Supplementary services support: 0x0012
   Terminal Portability
   Call Forwarding



In Asterisk, when an incoming call arrives, it shows me the following:

Asterisk Ready.
*CLI capi info
Contr1: 2 B channels total, 2 B channels free.
*CLI capi debug
CAPI Debugging Enabled
*CLI
*CLI
*CLI -- INFO_IND ID=001 #0x0001 LEN=0016
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x18
  InfoElement = 89

-- INFO_IND ID=001 #0x0001 LEN=0016
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x18
  InfoElement = 89

Oct 10 09:17:16 NOTICE[5948]: chan_capi.c:1184 find_pipe: PLCI doesnt  
match last pipe (PLCI = 0x101)
Oct 10 09:17:16 NOTICE[5948]: chan_capi.c:1301 pipe_msg: INFO_IND  
ID=001 #0x0001 LEN=0016

  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x18
  InfoElement = 89
-- CONNECT_IND ID=001 #0x0002 LEN=0044
  Controller/PLCI/NCCI= 0x101
  CIPValue= 0x1
  CalledPartyNumber   = 8120
  CallingPartyNumber  = 01 830123456789
  CalledPartySubaddress   = default
  CallingPartySubaddress  = default
  BC  = 80 90 a3
  LLC = default
  HLC = default
  AdditionalInfo
   BChannelinformation= default
   Keypadfacility = default
   Useruserdata   = default
   Facilitydataarray  = default

Oct 10 09:17:16 NOTICE[5948]: chan_capi.c:1931

Re: [Asterisk-Users] AVM Fritz! + chan_capi + mISDN + PTP

2005-10-10 Thread Lionel Riem


Hello,

Well, now, with the help of mISDN you can, according to http:// 
www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs- 
html/x3343.html :


With the introduction of the isdn4linux new mISDN architecture and  
it's capi layer, that problem is fixed. chan_capi supports PTP on the  
AVM Fritz! card in that case and you even get rid of having a tainted  
kernel, at least for this module.


L. Riem
[EMAIL PROTECTED]


Le 10 oct. 05 à 10:14, Kib Eki a écrit :

I think you can't use a Fritz Card for PTP. You need an active  
card. We use the the beronet ISDN Cards with misdn.


Lionel Riem wrote:


Hello everyone,
I have been using an AVM Fritz! card with chan_capi and mISDN for   
quite a while in PTM mode and it was working finely.
Now, I needed more DID/MSN, so I switched to PTP. But now nothing   
works anymore :(
I am using Asterisk on Debian Sarge stable and installed Asterisk   
along with chan_capi from apt-get. I installed mISDN from the CVS  
of  isdn4linux.de.

It is :
- Asterisk 1.0.7 with bristuff
- chan_capi 0.3.5
When I load the whole modules lot, I get the following in dmesg:
Modular ISDN Stack core $Revision: 1.25 $
mISDNd: kernel daemon started
ISAC module $Revision: 1.16 $
mISDNd: test event done
CAPI Subsystem Rev 1.1.2.8
capi20: Rev 1.1.2.7: started up with major 68 (middleware+capifs)
ISDN L1 driver version 1.11
ISDN L2 driver version 1.20
mISDN: DSS1 Rev. 1.30
mISDN Capi 2.0 driver file version 1.14
X25 DTE modul version 1.8
AVM Fritz PCI/PnP driver Rev. 1.30
ACPI: PCI interrupt :00:14.0[A] - GSI 10 (level, low) - IRQ 10
mISDN_fcpcipnp: found adapter Fritz!Card PCI v2 at :00:14.0
fritz card cd09a000 dch cd09a094 bch1 cd09a214 bch2 cd09a3a0
AVM PCI V2: stat 0x240020e
AVM PCI V2: Class E Rev 2
AVM PnP: HDLC version 2
mISDN: AVM Fritz!PCIv2 config irq:10 base:0xEC00
spin_lock_adr=cd09a024 now(d015b867)
busy_lock_adr=cd09a024 now(d015b867)
AVM PCI/PnP: reset
AVM PCI/PnP: S0/S1 40/2
Fritz1 ISAC STAR 40
Fritz1 ISAC MODE c0
Fritz1 ISAC ADF2 ff
Fritz1 ISAC ISTA 0
Fritz1 ISAC CIR0 7
mISDN_isac_init: ISACSX
Fritz1 HDLC 1 STA 8200
Fritz1 HDLC 2 STA 8200
AVM Fritz!PCI: IRQ 10 count 4
fritz 1 cards installed
Here is my /etc/asterisk/capi.conf:
;
; CAPI config
;
;
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
mode=immediate
isdnmode=ptp
msn=*
incomingmsn=*
controller=1
softdtmf=1
context=dispatcher
accountcode=
devices=2
Here is my /etc/modprobe.d/capi conf file:
alias /dev/capi20 avmfritz
alias char-major-68-0 avmfritz
install avmfritz /sbin/modprobe capi; \
/sbin/modprobe mISDN_core; \
/sbin/modprobe mISDN_l1; \
/sbin/modprobe mISDN_l2; \
/sbin/modprobe l3udss1; \
/sbin/modprobe mISDN_capi; \
/sbin/modprobe mISDN_x25dte; \
/sbin/modprobe --ignore-install avmfritz protocol=0x22
remove avmfritz /sbin/modprobe -r --ignore-remove avmfritz; \
/sbin/modprobe -r mISDN_x25dte; \
/sbin/modprobe -r mISDN_capi; \
/sbin/modprobe -r l3udss1; \
/sbin/modprobe -r mISDN_l2; \
/sbin/modprobe -r mISDN_l1; \
/sbin/modprobe -r mISDN_core; \
/sbin/modprobe -r capi
capiinfo shows me:
asterisk:/etc/asterisk# capiinfo
Number of Controllers : 1
Controller 1:
Manufacturer: mISDN CAPI controller Fritz1
CAPI Version: 2.0
Manufacturer Version: 1.0
Serial Number: 0002
BChannels: 2
Global Options: 0x0018
   DTMF supported
   Supplementary Services supported
B1 protocols support: 0x0003
   64 kbit/s with HDLC framing
   64 kbit/s bit-transparent operation
B2 protocols support: 0x0043
   ISO 7776 (X.75 SLP)
   Transparent
   Transparent (ignoring framing errors of B1 protocol)
B3 protocols support: 0x0005
   Transparent
   ISO 8208 (X.25 DTE-DTE)
  0100
  0200
  1800
  0300
  4300
  0500
       
      
Supplementary services support: 0x0012
   Terminal Portability
   Call Forwarding
In Asterisk, when an incoming call arrives, it shows me the  
following:

Asterisk Ready.
*CLI capi info
Contr1: 2 B channels total, 2 B channels free.
*CLI capi debug
CAPI Debugging Enabled
*CLI
*CLI
*CLI -- INFO_IND ID=001 #0x0001 LEN=0016
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x18
  InfoElement = 89
-- INFO_IND ID=001 #0x0001 LEN=0016
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x18
  InfoElement = 89
Oct 10 09:17:16 NOTICE[5948]: chan_capi.c:1184 find_pipe: PLCI  
doesnt  match last pipe (PLCI = 0x101)
Oct 10 09:17:16 NOTICE[5948]: chan_capi.c:1301 pipe_msg: INFO_IND   
ID=001 #0x0001 LEN=0016

  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x18
  InfoElement = 89
-- CONNECT_IND ID=001 #0x0002 LEN=0044
  Controller/PLCI/NCCI= 0x101
  CIPValue= 0x1
  CalledPartyNumber   = 8120
  CallingPartyNumber  = 01 830123456789

Re: [Asterisk-Users] compiling asterisk on SuSE Linux 9.3 fails: illegal instruction

2005-10-10 Thread gehrts


Hi Tzafrir !

Thanks for your help!! Now it works. 
It took some time to find everything and to set up everything, but now it 
works. 

So I can tell: using asterisk on book pc's with cyrix processors and VIA 
chipset compiles fine.

Now I need to check what the performance is like.

thanks again,
Hans-Henning


Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com schrieb am 09.10.05 20:27:55:
 
 On Sun, Oct 09, 2005 at 05:15:36PM +0200, [EMAIL PROTECTED] wrote:
  
  Hi all!
  
  I'm running a SuSE Linux 9.3 on a little book pc which is based on a VIA 
  CPU and Chipset:
   cat /proc/cpuinfo 
  processor   : 0
  vendor_id   : CentaurHauls
  cpu family  : 6
  model   : 7
  model name  : VIA Samuel 2
  stepping: 3
  cpu MHz : 532.776
  cache size  : 64 KB
  fdiv_bug: no
  hlt_bug : no
  f00f_bug: no
  coma_bug: no
  fpu : yes
  fpu_exception   : yes
  cpuid level : 1
  wp  : yes
  flags   : fpu de tsc msr cx8 mtrr pge mmx pni 3dnow
  bogomips: 1046.52
  
  kernel is  Linux hermes 2.6.11.4-21.9-default #1 Fri Aug 19 11:58:59 UTC 
  2005 i686 i686 i386 GNU/Linux
  
  has anyone tried to compile asterisk on a cv860 book pc? I'm always getting 
  a 'illegal instruction':
  # asterisk -vvv
  Illegal instruction
 
 What optimization flags did you use?
 
 Generally setting PROC=i586 is safe.
 
 -- 
 Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
 http://tzafrir.org.il |   | a Mutt's  
 [EMAIL PROTECTED] |   |  best
 ICQ# 16849755 |   | friend
 ___
 --Bandwidth and Colocation sponsored by Easynews.com --
 
 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re[2]: [Asterisk-Users] asterisk certification - thread hijack

2005-10-10 Thread Alessio Focardi

I took the certification in Astricon Madrid, still I have to get any kind of 
proof/certificate.

I contacted the testing company and they told me it was just a matter
of time, so probably they are working on this  probably those are
just super rapid growing problems.

Regards!


s The original poster's statement about not even receiving any
s proof thathe was certified is kind of amazing.

s I wouldn't be too upset about it either because it is probably
s anhonest mistake, but I would be firm on demanding that you get
s what youpaid for.  


-- 
Best regards,
 Alessiomailto:[EMAIL PROTECTED]

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Clicks, pops and noise


 I've got a TE410p connected via T1 to four channel banks (2 FXO and 2 FXS), 
 no PRIs.
 
 Some users are complaining that they hear clicks and pops on the FXS lines, 
 generally when they pick up the phone it's noisy. This happens only after a 
 while, e.g. after a fresh restart of everything, all is fine but after some 
 time 
 these noises start appearing. From what I've read on the list, this could be 
 down to frame slips or some problem due to synchronization.
 
 Since there's no incoming PRI to sync to, this means everything needs to be 
 internally clocked. Could it be the internal clock source on the card has 
 gone 
 wonky? Or is something else in the server screwing up the clock signal?
 
 Anyone else experienced this when connecting four channel banks to the TE410?
 
 Zaptel.conf:
 
 span=1,0,0,esf,b8zs
 fxsks=1-24
 
 span=2,0,0,esf,b8zs
 fxsks=25-48
 
 span=3,0,0,esf,b8zs
 fxsls=49-72
 
 span=4,0,0,esf,b8zs
 fxsls=73-96

Timing (or sync) has nothing to do with PRI's. All T1/E1 data links
of any type require sync.

All T1/E1 links have timing/sync included in their transmit leg and
there is no way for you to turn it off. Its part of the T1/E1 spec.

If you don't have any T1/E1 connections to the outside world, then
pick one channel bank and call it your official source of sync, and
change the above definitions to sync off that channel bank. On all
other channel banks, configure them to sync off the asterisk card.

If you do have a T1/E1 that comes from your telco/pstn network, use
it as the source of clock sync, and config all channel banks to sync
from asterisk.

Your clicks will go away.


___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Problem setting SIP incoming/outgoing

2005-10-10 Thread Zafar Kazmi

Hi

I am a newbie to * and I am having a problem which appears strange as I did
not find any mention of it anywhere in my search.

Simply speaking, I have an external SIP proxy server which I am trying to
configure for incoming and outgoing calls from my asterisk installation. So
here is my configuration in sip.conf

[general]
register = user:secret:[EMAIL PROTECTED]:8080

as long as I have just the above entry, I am able to receive incoming calls.
Now I would like to setup outgoing calls too. So I create a new section in
sip.conf

[sipserverout]
type=peer
secret=secret
username=user
fromuser=user
fromdomain=sipserver.com
host=sipserver.com
port=8080
context=default

with the above configuration I can successfully dial out using
dial(SIP/[EMAIL PROTECTED])

but now when I call my incoming number, I get a busy or invalid number
signal. If I coment out sipserverout section, I could receive incoming calls
again.

So I turned on sip debug on CLI. and it appears to me that the following is
happening. astreisk takes the incoming call and tries to match it with a
section with the same hostname. Now the reverse IP lookup on 109.147.41.48
return sipserver.com (which is correct), so it is trying to send the call to
sipserverout which is essentially back to the same server where it came from
(Notice the statement Found peer 'sipserverout' in the sip debug logs
below). This creates an endless loop and the equipment at the other end
terminates the call.

According to all the examples I have seen, my setup is the correct setup and
everyone seems to be using it. but it does not work for me. I am deperately
looking for a solution. Please help.

I am using asterisk 1.2.0 beta 1 on FC1.

Here is the sip debug dump when a call is coming.

-- SIP read from 109.147.41.48:8080:
INVITE sip:[EMAIL PROTECTED]:5050 SIP/2.0
Record-Route: sip:209.47.41.48:80;ftag=2C996308-10F9;lr=on
Via: SIP/2.0/UDP 209.47.41.48:80;branch=z9hG4bK03a4.da6a926.0
Via: SIP/2.0/UDP
209.47.41.61:5060;rport=53084;x-route-tag=tgrp:sroutetor1;branch=z9hG4bK4B
B6EA6
From: sip:[EMAIL PROTECTED];tag=2C996308-10F9
To: sip:[EMAIL PROTECTED]
Date: Thu, 06 Oct 2005 08:13:58 GMT
Call-ID: [EMAIL PROTECTED]
Supported: timer
Min-SE: 1800
Cisco-Guid: 4208765565-896995802-2793406481-2459445924
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE,
NOTIFY, INFO, UPDATE, REGISTER
CSeq: 101 INVITE
Max-Forwards: 4
Remote-Party-ID:
sip:[EMAIL PROTECTED];party=calling;screen=yes;privacy=off
Timestamp: 1128586438
Contact: sip:[EMAIL PROTECTED]:53084
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 369
hint: NAThelper
hint: SDP rewritten
hint: usrloc applied
hint: NAT...

v=0
o=CiscoSystemsSIP-GW-UserAgent 5168 3221 IN IP4 209.47.41.61
s=SIP Call
c=IN IP4 109.147.41.48
t=0 0
m=audio 53870 RTP/AVP 0 8 18 3 101
c=IN IP4 109.147.41.48
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=direction:passive
a=nortpproxy:yes

--- (26 headers 16 lines)---
Using INVITE request as basis request -
[EMAIL PROTECTED]
Sending to 109.147.41.48 : 80 (non-NAT)
Found peer 'sipserverout'
Reliably Transmitting (no NAT) to 209.47.41.48:80:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 209.47.41.48:80;branch=z9hG4bK03a4.da6a926.0
Via: SIP/2.0/UDP
209.47.41.61:5060;x-route-tag=tgrp:sroutetor1;branch=z9hG4bK4BB6EA6
From: sip:[EMAIL PROTECTED] ;tag=2C996308-10F9
To: sip:[EMAIL PROTECTED] ;tag=as1b7fff99
Call-ID: [EMAIL PROTECTED]
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: sip:[EMAIL PROTECTED]:5050
Proxy-Authenticate: Digest realm=asterisk, nonce=6d00a83d
Content-Length: 0


---
Scheduling destruction of call
'[EMAIL PROTECTED]' in 15000 ms

-- SIP read from 109.147.41.48:8080:
ACK sip:[EMAIL PROTECTED]:5050 SIP/2.0
Via: SIP/2.0/UDP 109.147.41.48:8080;branch=z9hG4bK03a4.da6a926.0
From: sip:[EMAIL PROTECTED];tag=2C996308-10F9
Call-ID: [EMAIL PROTECTED]
To: sip:[EMAIL PROTECTED];tag=as1b7fff99
CSeq: 101 ACK
User-Agent: Phone Server 1
Content-Length: 0


___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] telephony that just works



Hello list,
I am looking for a way to have multiple remote Windows users download a  
package and get connected to *. My idea would be that they run a simple  
app, it connects without any setting to an * box (maybe via IAX) and then  
people press a button to talk. It would be okay if they had to enter a  
username and password, but not more than that.


Looking for such software, I keep finding how much easier for a  
non-technical end-user is to download skype and have it running than  
downloading a softphone, creating an account, configuring the softphone  
and then dialing the required number. Having a way to use skype as a  
terminal would be nice, but I fear it's impossible by now (see  
http://www.skypejournal.com/blog/archives/2005/03/skype_strategy.php ).


So, anybody has experience of something that could be used, repackaged,  
modified or you-know-what that could be helpful in this case? And don't  
you think a IAX intercom could be somehow useful? :-)


Bye
l.


--
Assum est, versa et manduca.
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Zaptel Line Build Out

2005-10-10 Thread asterisk


I don't think you will have any problems at all.  I have always used the
lowest setting no matter how long the cable (never had a very long cable
run) and have never run into any problems.

My assumption of the CSU settings are because different CSUs have different
voltage outputs (this is just a guess)

Another educated guess is that the settings just tell asterisk how much to
amplify the signal coming in (similar to RX and TX gain) due to attenuation
loss.

Thanks,
Steve


 Yeah... sorta.

 So the CSU settings may be used when the E1 is pulled down to my premises,
and I
 have a short cable connecting directly to the CSU device. I don't know why
I'd
 need to change the LBO settings in that case, but I guess that doesn't
really
 matter to me at the moment.

 In my case, I am approximately 40-50 metres (130-165 feet) from the switch
 (according to the telco's engineers), so an LBO of 1 in my span
definition
 would theoretially be correct.




 asterisk wrote:
 
http://www.adc.com/Library/Techpub/80348_1.pdf?refer=LibraryC=Copper_ConnectivityL=DS1_E1_Twisted_Pair_Products
 
  http://www.pcmag.com/encyclopedia_term/0,2542,t=DSUCSUi=42059,00.asp
 
  any help?
 
 
 
 
 
 
 
 Maybe I need to be a little more specific.
 
 I know what signal attenuation is. What I don't know, is how LBO (and
 specifically the implementation of it as used in the zaptel
 
  hardware/software)
 
 helps the situation.
 
 My servers are co-located with my carrier, and my PRI circuits are run
 
  through
 
 several patch panels, jumpers, etc. to another room, where they
terminate
 
  on a
 
 DMS-100. I have asked the carrier for an estimated cable length, so i
can
 correctly set the LBO.
 
 In the zaptel config, what is meant by DSX-1? What is CSU?
 
 Why would I use a -7.5db, -15db or -22.5db LBO?
 
 
 
 
 asterisk wrote:
 
 
http://searchnetworking.techtarget.com/sDefinition/0,,sid7_gci211613,00.html
 
 
 - Original Message - 
 From: Rod Bacon [EMAIL PROTECTED]
 To: asterisk-users@lists.digium.com
 Sent: Sunday, October 09, 2005 6:42 PM
 Subject: [Asterisk-Users] Zaptel Line Build Out
 
 
 
 
 Can someone who is knowledgable in the traditional telco space please
 
  give
 
 me a
 
 
 layman's explanation (or point me to an appropriate url) of LBO as per
 
  the
 
 zaptel configuration file?
 
 # The line build-out (or LBO) is an integer, from the following table:
 # 0: 0 db (CSU) / 0-133 feet (DSX-1)
 # 1: 133-266 feet (DSX-1)
 # 2: 266-399 feet (DSX-1)
 # 3: 399-533 feet (DSX-1)
 # 4: 533-655 feet (DSX-1)
 # 5: -7.5db (CSU)
 # 6: -15db (CSU)
 # 7: -22.5db (CSU)
 
 
 -- 
 ==
 Rod Bacon
 Empowered Communications
 Ground Floor, 102 York St. South Melbourne
 Victoria, Australia. 3205
 Phone: +613 99401600Fax: +613 99401650
 FWD: 512237   ICQ: 5662270
 ==
 ___
 --Bandwidth and Colocation sponsored by Easynews.com --
 
 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 
 -- 
 No virus found in this incoming message.
 Checked by AVG Anti-Virus.
 Version: 7.0.344 / Virus Database: 267.11.13/124 - Release Date:
 
  10/7/2005
 
 
 
 ___
 --Bandwidth and Colocation sponsored by Easynews.com --
 
 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 
 ___
 --Bandwidth and Colocation sponsored by Easynews.com --
 
 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 
 -- 
 No virus found in this incoming message.
 Checked by AVG Anti-Virus.
 Version: 7.0.344 / Virus Database: 267.11.13/124 - Release Date:
10/7/2005
 
 
 
  ___
  --Bandwidth and Colocation sponsored by Easynews.com --
 
  Asterisk-Users mailing list
  Asterisk-Users@lists.digium.com
  http://lists.digium.com/mailman/listinfo/asterisk-users
  To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 ___
 --Bandwidth and Colocation sponsored by Easynews.com --

 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users



 -- 
 No virus found in this incoming message.
 Checked by AVG Anti-Virus.
 Version: 7.0.344 / Virus Database: 267.11.13/124 -

[Asterisk-Users] Dial plan logic documentation?

2005-10-10 Thread Andrew Furey

Hi all,

What methods (software or even on paper) would you folks use /
recommend for the purposes of documenting how a dial plan is
constructed? ie. what extensions jump to other extensions, etc? This
is as a means of getting the big picture rather than having pages
and pages of printed extensions.conf output...

If you consider priorities as line numbers, extensions as
functions/subroutines and contexts as source files, you could compare
the dialplan to a regular programming language source... I've thought
of various things like flowcharts but I don't know of any really good
flowcharting programs. Besides, the analogy breaks down in that
programming languages don't generally jump to specific line numbers in
a function (whereas using priorities other than 1 is quite common).

Any thoughts?

Andrew

--
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
  -- Bill Garrett
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Problem setting SIP incoming/outgoing


 I am a newbie to * and I am having a problem which appears strange as I did
 not find any mention of it anywhere in my search.
 
 Simply speaking, I have an external SIP proxy server which I am trying to
 configure for incoming and outgoing calls from my asterisk installation. So
 here is my configuration in sip.conf
 
 [general]
 register = user:secret:[EMAIL PROTECTED]:8080
 
 as long as I have just the above entry, I am able to receive incoming calls.
 Now I would like to setup outgoing calls too. So I create a new section in
 sip.conf
 
 [sipserverout]
 type=peer
 secret=secret
 username=user
 fromuser=user
 fromdomain=sipserver.com
 host=sipserver.com
 port=8080
 context=default
 
 with the above configuration I can successfully dial out using
 dial(SIP/[EMAIL PROTECTED])
 
 but now when I call my incoming number, I get a busy or invalid number
 signal. If I coment out sipserverout section, I could receive incoming calls
 again.
 
 So I turned on sip debug on CLI. and it appears to me that the following is
 happening. astreisk takes the incoming call and tries to match it with a
 section with the same hostname. Now the reverse IP lookup on 109.147.41.48
 return sipserver.com (which is correct), so it is trying to send the call to
 sipserverout which is essentially back to the same server where it came from
 (Notice the statement Found peer 'sipserverout' in the sip debug logs
 below). This creates an endless loop and the equipment at the other end
 terminates the call.
 
 According to all the examples I have seen, my setup is the correct setup and
 everyone seems to be using it. but it does not work for me. I am deperately
 looking for a solution. Please help.
 
 I am using asterisk 1.2.0 beta 1 on FC1.

In very general terms, you probably want something like this in your sip.conf:
 [sipserver]
 type=friend
 secret=secret
 username=user
 fromuser=user
 fromdomain=sipserver.com
 host=sipserver.com
 port=8080
 insecure=very
 canreinvite=no
 dtmfmode=inband
 context=from-sipserver
 disallow=all  
 allow=ulaw

For sip stuff, notice the use of type=friend and canreinvite=no. The use
of the register statement (in this case) implies use of type=friend (for
both incoming and outgoing calls).

Then in extensions.conf, use something like this:
 exten = _1NX,3,Dial(SIP/sipserver/${EXTEN})
where SIP/sipserver is referring to the context [sipserver] in sip.conf.

Did the folks at sipserver.com tell you to use port=8080?  If not, 
remove that statement as the default for sip is port=5060.

There are other ways to accomplish the same thing, so consider the above
as only way to do it.


___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] telephony that just works

2005-10-10 Thread Anders Svensson

What you are looking for is a pc2phone dialer. This can be preconfigured
with all settings and when it connects to your * it ask for username and
password or just a pin. There are many of these out on the net. Most is
however locked to a provider but you will also find many that you can buy
with your settings in them.

Anders

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of lenz
Sent: den 10 oktober 2005 13:28
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] telephony that just works


Hello list,
I am looking for a way to have multiple remote Windows users download a  
package and get connected to *. My idea would be that they run a simple  
app, it connects without any setting to an * box (maybe via IAX) and then  
people press a button to talk. It would be okay if they had to enter a  
username and password, but not more than that.

Looking for such software, I keep finding how much easier for a  
non-technical end-user is to download skype and have it running than  
downloading a softphone, creating an account, configuring the softphone  
and then dialing the required number. Having a way to use skype as a  
terminal would be nice, but I fear it's impossible by now (see  
http://www.skypejournal.com/blog/archives/2005/03/skype_strategy.php ).

So, anybody has experience of something that could be used, repackaged,  
modified or you-know-what that could be helpful in this case? And don't  
you think a IAX intercom could be somehow useful? :-)

Bye
l.


-- 
Assum est, versa et manduca.
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] My contribution to the issue of code- reversal

2005-10-10 Thread Federico Alves

Four years ago, I faced a real dilemma in my business: the Visual Voice PRI
dll had a bug that considered unanswered any call after ringing for 20
seconds. This bug was in fact killing my business, because for international
calling, the setup of the call was already close to 20 seconds on many
cases. Furthermore, the vendor, Artisoft, had cowardly sold the software to
Dialogic, and Intel-Dialogic had killed the product. There was no support. I
had to bite the bullet and buy a reverse-assembler (IDA-Pro), from a Belgium
company. I had to lock myself down in the lab for a week, until I understood
the location of exactly the right byte that was wrong, and replaced it at a
binary level for a 40 hex. Bingo. I made a living out of selling my pre-paid
platform for another two years, until I adapted Asterisk to replace Dialogic
and now I am paying my bills thanks to Asterisk. If I had not solved, my
existing clients would have looked elsewhere for a solution, and I had
failed to sell more switches. If Visual Voice had been open-source, I would
not had faced the terrible pressure to understand every single step of
assembler code required. So we need to reverse code and it surely is a
legitimate operation. Open source is far more convenient, but how do we
charge for the product? The business model is not there: the more popular
the product is, the more remote the possibility of the creator making any
money from it. Take Digium. The more experts on Asterisk pop-up, the less
demand is for Digium services. In fact, having tried Asterisk support from
Digium and others, I think the best Asterisk people --like Jeremy, Shido and
swk286-- are somewhere else. So the question is: how do we make sure that
the creator of the product makes even one dollar from every copy put in use
of his creation? The answer is: there is no answer. There is where Microsoft
wins. Additionally, Microsoft support services do know their products, and
if they fail to behave, they fix it. Digium made me once spend $150 and they
could not make res_odbc work, etc. I stopped using Digium support because
there is no way to know how many hours or dollars is going to take to fix
anything, while with others I pay for the result, not for the time. The
success is guaranteed. Regarding open-source-closed source, the future holds
a mixed-model in the store, and we are yet to discover it.

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] [Fwd: Libpri/chan_zap problems?]

2005-10-10 Thread Igor Briski


What am I doing wrong here? Why is this happening?

libpri is version 1.0.7-1 (debian package)
asterisk is version 1.0.7.dfsg.1-2 (debian package)
zaptel is version 1.0.9.2


   -- Executing Dial(SIP/739-5935, Zap/g1/0916000739) in new stack
   -- Called g1/0916000739
   -- Channel 0/1, span 1 got hangup
Oct 10 13:14:45 WARNING[7544]: app_dial.c:412 wait_for_answer: Unable to 
forward voice

   -- Hungup 'Zap/1-1'
 == No one is available to answer at this time
   -- Executing Congestion(SIP/739-5935, ) in new stack
 == Spawn extension (siplocalclients, 0916000739, 2) exited non-zero 
on 'SIP/739-5935'
Oct 10 13:14:46 WARNING[7544]: chan_zap.c:7445 pri_fixup_principle: Call 
specified, but not found?
Oct 10 13:14:46 NOTICE[7544]: chan_zap.c:8768 pri_dchannel: hangup, did 
not find cref 32777, tei 0
Oct 10 13:14:46 WARNING[7544]: chan_zap.c:8769 pri_dchannel: Hangup on 
bad channel 0/1 on span 1
Oct 10 13:14:50 WARNING[7544]: chan_zap.c:7445 pri_fixup_principle: Call 
specified, but not found?
Oct 10 13:14:50 NOTICE[7544]: chan_zap.c:8768 pri_dchannel: hangup, did 
not find cref 32777, tei 0
Oct 10 13:14:50 WARNING[7544]: chan_zap.c:8769 pri_dchannel: Hangup on 
bad channel 0/1 on span 1


Tnx.

--
Igor Briski



___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] oh323 problem

2005-10-10 Thread asterisk

I am trying to setup an h32h channel in Asterisk

Firstly I tried to use chan_h323, but I was not able to compile the
required pwlib ad open323 version under my system (Suse Linux 9.2)
Next I tried to use oh323. I succed in compile and install the

pwlib-Mimas_patch2-src-tar.gz

then

openh323-Mimas_patch2-src-tar.gz

and finally

asterisk-oh323-0.6.7

everything was OK, and the module

chan_oh323.so

was created in the /usr/lib/asterisk/modules directory

But if I try to start asterisk, I see in the full log

Oct 10 14:29:20 VERBOSE[12461]:   == Registered translator 'lpc10tolin'
from format lpc10 to slin, cost 3
Oct 10 14:29:20 VERBOSE[12461]:   == Registered translator 'lintolpc10'
from format slin to lpc10, cost 5
Oct 10 14:29:20 VERBOSE[12461]:  [app_setcidname.so]Oct 10 14:29:20
VERBOSE[12461]:  [app_setcidname.so] = (Set CallerID Name)
Oct 10 14:29:20 VERBOSE[12461]:   == Registered application 'SetCIDName'
Oct 10 14:29:20 VERBOSE[12461]:  [chan_oh323.so]Oct 10 14:29:20
WARNING[12461]: /usr/lib/asterisk/modules/chan_oh323.so: undefined symbol:
_ZNK8PChannel7IsClassEPKc
Oct 10 14:29:20 WARNING[12461]: Loading module chan_oh323.so failed!

Does anybody know which is the problem ?


thanks in advance,

Andrea

Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla.

Visitate il sito http://www.frameweb.it

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] WiFi Phones

2005-10-10 Thread Pedro

The UTStarcom F1000 with the latest firmware (3.10st) has improved
sound volume over the default firmware shipped with the units.
Also, TFTP configuration works well so you don't have to configure the
units with the keypad. You will need to get the configuration
compiler from your vendor and be aware that the default encryption key
should be set to NULL rather than F1000 as stated in the docs when
compiling your config. At first I was not sure how I would like a
WiFi phone because I figured it would sound bad, but I have been very
impressed with the quality of the F1000. We have now added it to
our VoIP product offerings.

- Pedro
http://www.traci.netOn 10/8/05, Cory Andrews [EMAIL PROTECTED] wrote:
The F3000 is not anticipated to be available for distribution until lateDecember/January, FYI.
Cory AndrewsSenior Partner+++VOIPSupply.com454 Sonwil DriveBuffalo, NY 14225+++voice - 716.630.1555 X22email - [EMAIL PROTECTED]
fax - 716.630.1548Denis Galvão - iSolve wrote: Wait for the next UTStarCom version... Called F3000, Im not sure, but something like that. It will have better battery performance and will have
802.11g support, and many other improvements. It will be available soon. Denis. On 07 de out de 2005, at 00:54, Andy Hamilton wrote: Anyone have good words to say about any of the WiFi handsetscurrently
available? The UTStarCom F1000 (an 802.11b device) works pretty well. It's about half the $$$ of a Cisco 7920 (which are also pretty nice), but it seems like most of the config is done from the keypad. There is a TFTP
option, but it seems that isn't quite perfect. You could check the manual (I programmed the unit without that, except to find that the default password is 88). The unit, I'm guessing, was designed somewhere in Asia, and the
language translation shows it a little bit. Sound quality seems pretty good for the few calls I've passed through it. I only have one AP in my house, so I can't comment on roaming. The headset for my cell phone
is stereo, and I think the phone would be most happy with a standard 3 conductor plug, but I imagine a headset on a phone is a headset on a phone. The keypad is a touch small, and sometimes I hit the wrong key (and my
fingers aren't terribly fat). I also seemed to have a problem transferring calls (using the built in transfer function -- # should still work). Despite many vendors' pages saying that it does
802.1x authentication, it sure looks like WEP is the only available security option. Overall: I would recommend purchasing one, for testing at the very least.
They are well priced and of good quality. Battery life seems to be pretty good, too. -A ___ --Bandwidth and Colocation sponsored by
Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list
Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___
--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] customize the pager email

2005-10-10 Thread Andy Goss

I am running CVS-HEAD-04/12/05-21:44:31 and I am curious if it is
possible to customize the email message sent to the pager email address.


Thanks,
Andy
--
H. Andy Goss
Network Engineer
Network Advocates Inc.
Main: 502.412.1050
DID: 502.992.5933
Mobile: 502.387.8216
[EMAIL PROTECTED]
 
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] call to a particular 800 number never showsanswered on Zap channel

2005-10-10 Thread Andy Goss

I am still looking to solve this problem, does anyone have any ideas?

Thanks,
Andy

-Original Message-
From: Andy Goss 
Sent: Friday, October 07, 2005 5:37 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] call to a particular 800 number never 
showsanswered on Zap channel

Thanks for the reply.  Forgive me for being naïve, however have jumped in to 
this asterisk project at work due to some circumstances beyond my control and I 
don't know a lot about carriers and how this all works.  I am figuring it out, 
but it's a lot of trial by fire.  

As far as I know, we only use 1 carrier for our system.  We have a PRI from 
NuVox and we use 7 channels for our asterisk server.  So, I have a few 
questions:

Is asterisk or the carrier causing the disconnect?

Is IBM (the 800 number I am dialing) not passing the answer supervision or is 
that a function of the carrier?

Is there a way to make asterisk not drop the call or to force the answer on 
this number?  Seems like a hard-PBX would have to be able to handle this type 
of situation.

Thanks,
Andy

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Garth Summey
Sent: Friday, October 07, 2005 5:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] call to a particular 800 number never 
showsanswered on Zap channel

This one drove me crazy for a while too.  I found out that some 
companies don't exactly play fair and don't pass answer supervision on a 
call until you are actually speaking with a live person.  The person I 
spoke to about this wasn't sure if that was even legal, but he said it 
happens quite a bit.  I was lucky in that I use multiple carriers 
(voipjet and broadvoice), voipjet disconnected the call after 60 
seconds, but broadvoice did not, so when I find one of those 800 numbers 
I route it through broadvoice.

Hope that helps,

G

Andy Goss wrote:
 Whenever we call IBM, the call counter on the phone never starts and in
 the CLI the zap channel never gets the answered signal from the PRI.
 See below.

 -- Executing Dial(SIP/5933-645d, Zap/g1/18004267378) in new
 stack
 -- Requested transfer capability: 0x00 - SPEECH
 -- Called g1/18004267378

 At this point, I am in IBM's menu system.  However the call never
 indicates that it is answered either on the phone or in the CLI.  After
 60 seconds, the call disconnects.  

 -- Hungup 'Zap/1-1'
   == Spawn extension (main, 18004267378, 1) exited non-zero on
 'SIP/5933-7bff'
 -- Executing Hangup(SIP/5933-7bff, ) in new stack
   == Spawn extension (main, h, 1) exited non-zero on 'SIP/5933-7bff'

 Does anyone have any ideas?

 Thanks,
 Andy

 --
 H. Andy Goss
 Network Engineer
 Network Advocates Inc.
 Main: 502.412.1050
 DID: 502.992.5933
 Mobile: 502.387.8216
 [EMAIL PROTECTED]
 ___
 --Bandwidth and Colocation sponsored by Easynews.com --

 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] customize the pager email

2005-10-10 Thread Tom Rymes


Andy,

I may be wrong, but I think that you need to edit the code and  
recompile to change the message. I wanted to add a numeric line as  
the first line of our system's voicemail pager message so it would  
work with numeric pagers as well as text pagers. AFAIK, editing the  
code and recompiling worked.


Tom

On Oct 10, 2005, at 8:56 AM, Andy Goss wrote:


I am running CVS-HEAD-04/12/05-21:44:31 and I am curious if it is
possible to customize the email message sent to the pager email  
address.



Thanks,
Andy
--
H. Andy Goss
Network Engineer
Network Advocates Inc.
Main: 502.412.1050
DID: 502.992.5933
Mobile: 502.387.8216
[EMAIL PROTECTED]

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users



___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Bandwidth usage for codecs

2005-10-10 Thread Kanishka Somaratne


hi
how much bandwidth is used for the following codecs

723 r 5.3
723 r 6.3
723 r 8

what i know so far is the 

723 r 5.3 uses 5.3 k up and 5.3k down 
723 r 6.3 uses 6.3 k up and 6.3k down 
729 r 8 uses 8 k up and 8k down 


is this correct or is it like the following

723 r 5.3 uses 11 k up and 11k down 
723 r 6.3 uses 13 k up and 13k down 
729 r 8 uses 16 k up and 16k down 


if u guy know, please let me know.

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] oh323 problem

2005-10-10 Thread Hauke Zuehl


[EMAIL PROTECTED] wrote:

WARNING[12461]: /usr/lib/asterisk/modules/chan_oh323.so: undefined symbol:
_ZNK8PChannel7IsClassEPKc
Oct 10 14:29:20 WARNING[12461]: Loading module chan_oh323.so failed!

Does anybody know which is the problem ?


It seems Asterisk source and binary version do not fit.



Andrea



HTH,
Hauke
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] [Fwd: Libpri/chan_zap problems?]

2005-10-10 Thread Goran Skular

What am I doing wrong here? Why is this happening?

libpri is version 1.0.7-1 (debian package) asterisk is version 
1.0.7.dfsg.1-2 (debian package) zaptel is version 1.0.9.2


-- Executing Dial(SIP/739-5935, Zap/g1/0916000739) in new stack
-- Called g1/0916000739
-- Channel 0/1, span 1 got hangup
Oct 10 13:14:45 WARNING[7544]: app_dial.c:412 wait_for_answer: 
Unable to forward voice

Does the same thing happens even when you're not calling cellular number VIP
(I assume you are in Croatia, calling VIPnet) i.e. some fixed line number ?

And what connection do you use, BRI (bristuff, capi), PRI, some FXO,...

You can reach me at 01/4573573. I'll be glad to hear you if my assumptions
(on Croatia thing) were right...

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Clicks, pops and noise

2005-10-10 Thread Flynn

On 10/10/2005, Rich Adamson  [EMAIL PROTECTED] wrote:

snip

If you don't have any T1/E1 connections to the outside world, then
pick one channel bank and call it your official source of sync, and
change the above definitions to sync off that channel bank. On all
other channel banks, configure them to sync off the asterisk card.


snip

Your clicks will go away.


Rich,

Thanks for your input. I tried as you suggested, and now the first
channel bank is set as master. The three other channel banks are set as
slaves. My zaptel.conf now looks like:

span=1,1,0,esf,b8zs
fxsks=1-24

span=2,0,0,esf,b8zs
fxsks=25-48

span=3,0,0,esf,b8zs
fxsls=49-72

span=4,0,0,esf,b8zs
fxsls=73-96

so essentially saying that the first FXO channel bank's T1 is the
primary sync source. unloaded zap modules, reloaded them and restarted
asterisk. clicks and noise still appear. but it was after I did an
AutoT1 (forgot to mention we're using Rhinos) did the click and pops go
away.

However, some channels on one of the channel banks are still problematic.
I'm checking with Rhino to see if it's a channel bank problem, since
the noise always appears on the same channel no matter how many times I
reboot, unload/load etc.

Thanks again for your advice!

regards,
Flynn
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] does tellular cell phones support answer switching

2005-10-10 Thread jonny hashem

i have tellular cell phone plugged to fxo modules ,the
problem that i am facing is when i dial a number on
the fxo modules the call is been answered before it
picked up on the other side , i thought that the
analoge lines do not support the answer switching
feature but not tellular cell phones because i think
it digital.




__ 
Start your day with Yahoo! - Make it your home page! 
http://www.yahoo.com/r/hs
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Clicks, pops and noise


 If you don't have any T1/E1 connections to the outside world, then
 pick one channel bank and call it your official source of sync, and
 change the above definitions to sync off that channel bank. On all
 other channel banks, configure them to sync off the asterisk card.
 
 
 snip
 
 Your clicks will go away.
 
 
 Rich,
 
 Thanks for your input. I tried as you suggested, and now the first
 channel bank is set as master. The three other channel banks are set as
 slaves. My zaptel.conf now looks like:
 
 span=1,1,0,esf,b8zs
 fxsks=1-24
 
 span=2,0,0,esf,b8zs
 fxsks=25-48
 
 span=3,0,0,esf,b8zs
 fxsls=49-72
 
 span=4,0,0,esf,b8zs
 fxsls=73-96
 
 so essentially saying that the first FXO channel bank's T1 is the
 primary sync source. unloaded zap modules, reloaded them and restarted
 asterisk. clicks and noise still appear. but it was after I did an
 AutoT1 (forgot to mention we're using Rhinos) did the click and pops go
 away.
 
 However, some channels on one of the channel banks are still problematic.
 I'm checking with Rhino to see if it's a channel bank problem, since
 the noise always appears on the same channel no matter how many times I
 reboot, unload/load etc.
 

One other item to check is to ensure the digium T1 card is on its own
dedicated interrupt. Use 'cat /proc/interrupts' from the system command
line.



___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] sip register incoming call contexts?

2005-10-10 Thread Steve Gladden

Sorry this is a bit of a newbie question, I've been at this for a few
months and still have not quite figured this one out.


I've been able to setup one itsp (incoming calls) (sip account) with a
register line like this:

register = nnn:[EMAIL PROTECTED]

-or-

register = nnn:[EMAIL PROTECTED]/nnn
to come directly into an extension in the dialplan


It seems that this only works with the default context in the dialplan.


I have another sip account from another provider that I would like
all of it's incoming calls to come into the s, extension of
a new context but I have been unable to figure out
how to bring calls from a register line into an alternate context.

It seems that register lines are limited to only being used in the
general section of sip.conf and you are limited to one context=
statement there.

Is there a way to register a second account and have it's calls come into
another context in the dialplan?

register lines only seem to work in [general] and it seems like you
are limited to only one inbound context here.

I would like the two inbound call accounts to be 'isolated' from each other
and not have to come in on the same incoming context in the dialplan.

I'd also like to be able to have them have their own contexts with thier
own s, (start) extension available.


Thanks!

Steve












___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] telephony that just works

2005-10-10 Thread Ivan Stepaniuk

On Mon, 2005-10-10 at 13:28 +0200, lenz wrote:
 I am looking for a way to have multiple remote Windows users download a  
 package and get connected to *. My idea would be that they run a simple  
 app, it connects without any setting to an * box (maybe via IAX) and then  
 people press a button to talk. It would be okay if they had to enter a  
 username and password, but not more than that.

i've tried IaxComm 
http://iaxclient.sourceforge.net/iaxcomm/

it works, it's iax, and it's open source so you can re-package -
re-compile it with you own default settings (or even hide those settings
you don't want final users to see)  

-- 
Ivan Stepaniuk [EMAIL PROTECTED]

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Re: www.openpbx.org

2005-10-10 Thread Mike M

On Sat, Oct 08, 2005 at 06:50:54PM -0300, Doug Meredith wrote:
 Dinesh Nair [EMAIL PROTECTED] wrote:
 
 too much divergence and we have two pieces of software competing for each 
 other.
 
 My guess is that if they succeed, they will diverge significantly.

We will have two pieces of software that work with each other at
well-defined interfaces.  The development of internal workings may diverge.

-- 
Mike
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Distorted VM with iax2 with ilbc and jitterbuffer - bug?


  Two asterisk boxes 150 miles apart, both cvs-head as of this morning
  (and since Sept 27th), connected via iax2 with low-utilized ds3 internet,
  C7960 calls exten on remote system (also C7960), and call goes to VM.
  No other calls in either system (eg, no load).
  
  Both boxes have iax config'ed as:
   trunk=yes
   allow=ilbc
   jitterbuffer=yes
  Recorded VM messages are very distorted.
  
  Changing only jitterbuffer=no (and * restart), recorded VM messages are
  very clean. With jitterbuffer=yes and trunk=no, messages are very clean.
  
  Both boxes config'ed as:
   trunk=yes
   allow=gsm
   jitterbuffer=yes
  Recorded VM messages are very clean.
  
  Conclusion: looks like the combination of trunk=yes and jitterbuffer=yes
  with ilbc is causing the distorted VM messages. Normal answered calls
  have no distortion.
  
  Is this an unacceptable iax config or does this represent a bug?
  (Problem can be recreated at will and is very consistent.)
  
 
  
  Try using trunktimestamps as well..
 
 That didn't help at all; exactly same distorted vm audio.
 
 An example from the iax.conf looks like this:
 [npi-out]
 type=peer
 username=coz-in   
 secret=mysecret
 auth=plaintext  
 host=1.2.3.4
 trunk=yes
 trunktimestamps=yes
 jitterbuffer=yes
 disallow=all
 allow=ilbc
 
 Note: using type=user and type=peer on both cvs-head systems. Normal
 calls sound fine, but recorded vm messages are distorted.
 
Bug #5420 opened for this issue.


___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] sip register incoming call contexts?


 Sorry this is a bit of a newbie question, I've been at this for a few
 months and still have not quite figured this one out.
 
 
 I've been able to setup one itsp (incoming calls) (sip account) with a
 register line like this:
 
 register = nnn:[EMAIL PROTECTED]
 
 -or-
 
 register = nnn:[EMAIL PROTECTED]/nnn
 to come directly into an extension in the dialplan
 
 
 It seems that this only works with the default context in the dialplan.
 
 
 I have another sip account from another provider that I would like
 all of it's incoming calls to come into the s, extension of
 a new context but I have been unable to figure out
 how to bring calls from a register line into an alternate context.
 
 It seems that register lines are limited to only being used in the
 general section of sip.conf and you are limited to one context=
 statement there.
 
 Is there a way to register a second account and have it's calls come into
 another context in the dialplan?
 
 register lines only seem to work in [general] and it seems like you
 are limited to only one inbound context here.
 
 I would like the two inbound call accounts to be 'isolated' from each other
 and not have to come in on the same incoming context in the dialplan.
 
 I'd also like to be able to have them have their own contexts with thier
 own s, (start) extension available.

Try using something like:
 deny=0.0.0.0/0.0.0.0  
 permit=147.135.8.129/255.255.255.0 
 permit=147.135.0.129/255.255.255.0
 permit=147.135.4.128/255.255.255.0

in each sip.conf itsp definition to limit which contexts will match.
Obviously, replace the above permit's IP addresses with the correct
ones for your provider.


___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] TDM02B card difficulties

2005-10-10 Thread Min Qiu

Thank you for your respond, please see more detail inline...

Min

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Tzafrir Cohen
 Sent: Friday, October 07, 2005 4:57 PM
 To: asterisk-users@lists.digium.com
 Subject: Re: [Asterisk-Users] TDM02B card difficulties

 On Fri, Oct 07, 2005 at 03:41:43PM -0400, Min Qiu wrote:

  Hi all,

  I just installed an TDM02B.  My system is a dell pc with
  linux 2.6.12-1.1456_FC4
  asterisk-1.2.0-beta1
  zaptel-1.2.0-beta1
  libpri-1.2.0-beta1

  in /etc/zaptel.conf I have (all others are default):
  fxsks=3-4   --- I saw light in the ports
  channels=1-2--- change it to 3-4 has 
 same result

 cat /proc/zaptel/1 to see the channel numbers.

[EMAIL PROTECTED] mqiu]# cat /proc/zaptel/1
Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1

   1 WCTDM/0/0
   2 WCTDM/0/1
   3 WCTDM/0/2
   4 WCTDM/0/3

 Anyway, the last line is incorrect. It should be used in 
 zapata.conf and
 not in zaptel.conf .

The zaptel.conf has channels=... as an example.  Took the line
out I have:

 [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': Found
Oct 10 10:28:18 WARNING[28754]: chan_zap.c:890 zt_open: Unable to specify 
channel 1: No such device or address
Oct 10 10:28:18 ERROR[28754]: chan_zap.c:6650 mkintf: Unable to open channel 1: 
No such device or address
here = 0, tmp-channel = 1, channel = 1
Oct 10 10:28:18 ERROR[28754]: chan_zap.c:10030 setup_zap: Unable to register 
channel '1'
Oct 10 10:28:18 WARNING[28754]: loader.c:403 __load_resource: chan_zap.so: 
load_module failed, returning -1
Oct 10 10:28:18 WARNING[28754]: loader.c:543 load_modules: Loading module 
chan_zap.so failed!
Ouch ... error while writing audio data: : Broken pipe

  but...
  [EMAIL PROTECTED] asterisk]# /etc/rc.d/init.d/zaptel restart
  Unloading zaptel hardware drivers: wctdm.
  Removing zaptel module:[  OK  ]
  Loading zaptel framework:  [  OK  ]
  Waiting for zap to come online...OK
  Loading zaptel hardware modules:Running ztcfg:  Notice: 
 Configuration file is /etc/zaptel.conf
  line 207: Cannot get number of tones chanel 1
  line 207: Cannot init tones chanel 1
  /etc/init.d/functions: line 408: 13058 Segmentation fault  $*
 [FAILED]

  Wires are checked.  Can anyone point me to next step?

  Thanks a lot,

  Min

  ___
  --Bandwidth and Colocation sponsored by Easynews.com --

  Asterisk-Users mailing list
  Asterisk-Users@lists.digium.com
  http://lists.digium.com/mailman/listinfo/asterisk-users
  To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users

 -- 
 Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
 http://tzafrir.org.il |   | a Mutt's  
 [EMAIL PROTECTED] |   |  best
 ICQ# 16849755 |   | friend
 ___
 --Bandwidth and Colocation sponsored by Easynews.com --

 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Billing/SPA-841/CDR Log

2005-10-10 Thread Waldo Rubinstein


Hi list,

I have a couple of questions related to asterisk billing and the  
generation of cdr logs. I've searched the wiki but have not found my  
answers, hopefully you guys can help.


1) When are asterisk CDR logs _normally_ generated? When the call  
arrives, when the call hangs up, or both? I have looked at the  
records created and it seems to only generate it at the time the call  
is hung up in order to write the call duration, etc, but I just want  
to be sure.


2) If in fact it is the case that the CDR entry is written after the  
call terminates, what would be the best place for me to intercept the  
handler so that I can write my own CDR log? Because I only need to  
log certain calls, I was thinking of doing something with FastAGI so  
that only when certain calls terminate, I would write my custom CDR.


3) Related to the same theme (billing), I have a client who uses the  
SPA-841. This phone, by default has two lines. When I only configure  
Line 1, the phone still allows me to make/receive calls on Line 2.  
For the purposes of billing, I could understand allowing my client  
for two simultaneous conversations if s/he uses the call waiting  
feature of Line 1. But by default and without configuring Line 2 on  
the phone, the customer is able to, potentially, establish 4  
simultaneous calls and I'm only billing for one account. Is there a  
way to restrict the SPA-841 from Asterisk so that I don't depend on  
Line 2 being disabled on the SPA-841 (which the client could always  
change)?


4) Because this (item 3) has already happened to me, is there any  
free tool out there that will allow me to parse the CDR logs in order  
to determine the maximum number of simultaneous calls that a  
particular SIP peer has made within a specific timeframe? That way, I  
could potentially bill the client for 2 accounts instead of 1.


Thank you again,
Waldo 
___

--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] AVM Fritz! + chan_capi + mISDN + PTP

2005-10-10 Thread Craig Guy


Hi,

Yes, you can use the Fritz! in PTP mode, though only if you are using the 
mISDN drivers.  The mISDN driver should be called like this:


   modprobe avmfritz protocol=34

Craig

- Original Message - 
From: Lionel Riem [EMAIL PROTECTED]

To: asterisk-users@lists.digium.com
Sent: Monday, October 10, 2005 4:04 PM
Subject: [Asterisk-Users] AVM Fritz! + chan_capi + mISDN + PTP



Hello everyone,

I have been using an AVM Fritz! card with chan_capi and mISDN for  quite a 
while in PTM mode and it was working finely.


Now, I needed more DID/MSN, so I switched to PTP. But now nothing  works 
anymore :(


I am using Asterisk on Debian Sarge stable and installed Asterisk  along 
with chan_capi from apt-get. I installed mISDN from the CVS of 
isdn4linux.de.


It is :
- Asterisk 1.0.7 with bristuff
- chan_capi 0.3.5

When I load the whole modules lot, I get the following in dmesg:

Modular ISDN Stack core $Revision: 1.25 $
mISDNd: kernel daemon started
ISAC module $Revision: 1.16 $
mISDNd: test event done
CAPI Subsystem Rev 1.1.2.8
capi20: Rev 1.1.2.7: started up with major 68 (middleware+capifs)
ISDN L1 driver version 1.11
ISDN L2 driver version 1.20
mISDN: DSS1 Rev. 1.30
mISDN Capi 2.0 driver file version 1.14
X25 DTE modul version 1.8
AVM Fritz PCI/PnP driver Rev. 1.30
ACPI: PCI interrupt :00:14.0[A] - GSI 10 (level, low) - IRQ 10
mISDN_fcpcipnp: found adapter Fritz!Card PCI v2 at :00:14.0
fritz card cd09a000 dch cd09a094 bch1 cd09a214 bch2 cd09a3a0
AVM PCI V2: stat 0x240020e
AVM PCI V2: Class E Rev 2
AVM PnP: HDLC version 2
mISDN: AVM Fritz!PCIv2 config irq:10 base:0xEC00
spin_lock_adr=cd09a024 now(d015b867)
busy_lock_adr=cd09a024 now(d015b867)
AVM PCI/PnP: reset
AVM PCI/PnP: S0/S1 40/2
Fritz1 ISAC STAR 40
Fritz1 ISAC MODE c0
Fritz1 ISAC ADF2 ff
Fritz1 ISAC ISTA 0
Fritz1 ISAC CIR0 7
mISDN_isac_init: ISACSX
Fritz1 HDLC 1 STA 8200
Fritz1 HDLC 2 STA 8200
AVM Fritz!PCI: IRQ 10 count 4
fritz 1 cards installed



Here is my /etc/asterisk/capi.conf:

;
; CAPI config
;
;
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8

[interfaces]
mode=immediate
isdnmode=ptp
msn=*
incomingmsn=*
controller=1
softdtmf=1
context=dispatcher
accountcode=
devices=2


Here is my /etc/modprobe.d/capi conf file:

alias /dev/capi20 avmfritz
alias char-major-68-0 avmfritz

install avmfritz /sbin/modprobe capi; \
/sbin/modprobe mISDN_core; \
/sbin/modprobe mISDN_l1; \
/sbin/modprobe mISDN_l2; \
/sbin/modprobe l3udss1; \
/sbin/modprobe mISDN_capi; \
/sbin/modprobe mISDN_x25dte; \
/sbin/modprobe --ignore-install avmfritz protocol=0x22

remove avmfritz /sbin/modprobe -r --ignore-remove avmfritz; \
/sbin/modprobe -r mISDN_x25dte; \
/sbin/modprobe -r mISDN_capi; \
/sbin/modprobe -r l3udss1; \
/sbin/modprobe -r mISDN_l2; \
/sbin/modprobe -r mISDN_l1; \
/sbin/modprobe -r mISDN_core; \
/sbin/modprobe -r capi



capiinfo shows me:

asterisk:/etc/asterisk# capiinfo
Number of Controllers : 1
Controller 1:
Manufacturer: mISDN CAPI controller Fritz1
CAPI Version: 2.0
Manufacturer Version: 1.0
Serial Number: 0002
BChannels: 2
Global Options: 0x0018
   DTMF supported
   Supplementary Services supported
B1 protocols support: 0x0003
   64 kbit/s with HDLC framing
   64 kbit/s bit-transparent operation
B2 protocols support: 0x0043
   ISO 7776 (X.75 SLP)
   Transparent
   Transparent (ignoring framing errors of B1 protocol)
B3 protocols support: 0x0005
   Transparent
   ISO 8208 (X.25 DTE-DTE)

  0100
  0200
  1800
  0300
  4300
  0500
       
      

Supplementary services support: 0x0012
   Terminal Portability
   Call Forwarding



In Asterisk, when an incoming call arrives, it shows me the following:

Asterisk Ready.
*CLI capi info
Contr1: 2 B channels total, 2 B channels free.
*CLI capi debug
CAPI Debugging Enabled
*CLI
*CLI
*CLI -- INFO_IND ID=001 #0x0001 LEN=0016
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x18
  InfoElement = 89

-- INFO_IND ID=001 #0x0001 LEN=0016
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x18
  InfoElement = 89

Oct 10 09:17:16 NOTICE[5948]: chan_capi.c:1184 find_pipe: PLCI doesnt 
match last pipe (PLCI = 0x101)
Oct 10 09:17:16 NOTICE[5948]: chan_capi.c:1301 pipe_msg: INFO_IND  ID=001 
#0x0001 LEN=0016

  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x18
  InfoElement = 89
-- CONNECT_IND ID=001 #0x0002 LEN=0044
  Controller/PLCI/NCCI= 0x101
  CIPValue= 0x1
  CalledPartyNumber   = 8120
  CallingPartyNumber  = 01 830123456789
  CalledPartySubaddress   = default
  CallingPartySubaddress  = default
  BC  = 80 90 a3
  LLC = default
  HLC =

[Asterisk-Users] DTMF Question (misunderstood '*' button)

2005-10-10 Thread Giovanni Barbis

Hi all!

I'm experimenting a strange problem in my Asterisk PBX:
I've got an Asterisk pbx in the office: I dial an external number; the dialled 
number answers me correctly, but as soon as I press the '*' button (i.e. to 
navigate through the menus or to enter a password) my Asterisk box put me on 
hold.

(CLI transcription follows:
-- Executing ChanIsAvail(SIP/222-23da, Zap/g1Zap/g2) in new stack
-- Executing Cut(SIP/222-23da, theChannel=AVAILCHAN||1) in new stack
-- Executing NoOp(SIP/222-23da, Zap/1) in new stack
-- Executing Dial(SIP/222-23da, Zap/1/34844503450||tTH) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called 1/34844503450
-- Zap/1-1 is ringing
-- Zap/1-1 answered SIP/222-23da
[Here I press the '*' button]
-- Started music on hold, class 'default', on Zap/1-1
-- Unable to find extension '' in context 'internal'
-- Playing 'pbx-invalid' (language 'en')
-- Stopped music on hold on Zap/1-1
-- Hungup 'Zap/1-1'
  == Spawn extension (internal, 034844503450, 4) exited non-zero on 
'SIP/222-23da')


I think that Asterisk understands my postselection '*' DTMF tone like a 
command, not simply a tone to forward to the remote destination.

How can I solve the problem?

Tnx in advance



Giovanni
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Multitenant Call Center Setup

2005-10-10 Thread Waldo Rubinstein


Hi list (again),

I have another question which I have not been able to resolve from  
neither the wiki nor Google.


I've been able to configure a multi-tenant setup of asterisk for 2  
small call centers with no problem, by simply playing with contexts  
(which I guess is how everyone else is doing it).


The problem I have is that I've only been able to configure one  
global agents.conf file. This restricts my setup in a way that I  
cannot have two agents 1001, for example if my clients wanted to. Is  
there a way to overcome this?


Thanks,
Waldo
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] compiling asterisk on SuSE Linux 9.3 fails: illegal instruction

On Mon, Oct 10, 2005 at 12:28:43PM +0200, [EMAIL PROTECTED] wrote:
 
 Hi Tzafrir !
 
 Thanks for your help!! Now it works. 

Now, how would we detect that to avoid needless manual editing of the
CPU? 

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] telephony that just works

On Mon, Oct 10, 2005 at 01:28:17PM +0200, lenz wrote:
 
 Hello list,
 I am looking for a way to have multiple remote Windows users download a  
 package and get connected to *. My idea would be that they run a simple  
 app, it connects without any setting to an * box (maybe via IAX) and then  
 people press a button to talk. It would be okay if they had to enter a  
 username and password, but not more than that.

Using IAX to register to a server and call through it would just work.

This still would not handle instant-messaging.

 
 Looking for such software, I keep finding how much easier for a  
 non-technical end-user is to download skype and have it running than  
 downloading a softphone, creating an account, configuring the softphone  
 and then dialing the required number. Having a way to use skype as a  
 terminal would be nice, but I fear it's impossible by now (see  
 http://www.skypejournal.com/blog/archives/2005/03/skype_strategy.php ).
 
 So, anybody has experience of something that could be used, repackaged,  
 modified or you-know-what that could be helpful in this case? And don't  
 you think a IAX intercom could be somehow useful? :-)

iaxcomm is a start. Not the best in terms of usability, but a start. You
could hard-wire the server and make the accounts setup dialog a bit
firendlier.

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] TDM400 not working

2005-10-10 Thread Min Qiu

I failed to make TDM400 working too myself.  But I believed
I passed the the driver stuff... by installing zaptel-1.2.0-beta1.
Inside the package, there is a script zaptel.init that should
take care of loading/unloading the driver.

Min

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Rudolf Ladyzhenskii
 Sent: Monday, October 10, 2005 5:26 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] TDM400 not working
 
 
 Hi, all
 
 I have installed TDM400 card. I can see it is there (lspci).
 But Asterisk does not find it.
 
 phonebox2*CLI zap show status
 No Zaptel interface found.
 
 I assume that driver is not loaded, but I am sure I have 
 installed it (I 
 compiled zaptel and then re-build asterisk and did make 
 install for both 
 zaptel and asterisk).
 
 When asterisk is started I get:
 Jan  2 06:28:08 WARNING[3473]: chan_zap.c:872 zt_open: Unable to open 
 '/dev/zap/channel': No such file or directory
 Jan  2 06:28:08 ERROR[3473]: chan_zap.c:6572 mkintf: Unable 
 to open channel 
 2: No such file or directory
 here = 0, tmp-channel = 2, channel = 2
 Jan  2 06:28:08 ERROR[3473]: chan_zap.c:9927 setup_zap: 
 Unable to register 
 channel '2'
 Jan  2 06:28:08 WARNING[3473]: loader.c:402 __load_resource: 
 chan_zap.so: 
 load_module failed, returning -1
 Jan  2 06:28:08 WARNING[3473]: loader.c:523 load_modules: 
 Loading module 
 chan_zap.so failed!
 
 Ok, I look in the /dev and I could not find /dev/zap at all! 
 But, there is a 
 /dev/zapchannel character device.
 
 Any ideas what can be wrong?
 
 And last question. Does zaptel driver reads configuration 
 file on startup? 
 If so, how do I force the driver to update if config file was changed?
 
 Thanks,
 Rudolf
 
 
 ___
 --Bandwidth and Colocation sponsored by Easynews.com --
 
 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
 
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Zaptel Line Build Out

2005-10-10 Thread Andrew Kohlsmith

On Sunday 09 October 2005 18:42, Rod Bacon wrote:
 Can someone who is knowledgable in the traditional telco space please give
 me a layman's explanation (or point me to an appropriate url) of LBO as per
 the zaptel configuration file?

Unless something has changed in the last two years, zaptel totally ignores the 
LBO setting you provide. 

-A.
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Clicks, pops and noise

2005-10-10 Thread El Flynn


Rich Adamson wrote:

snip


One other item to check is to ensure the digium T1 card is on its own
dedicated interrupt. Use 'cat /proc/interrupts' from the system command
line.



It is on one interrupt, first thing I checked when the problem cropped up. One 
thing I did notice was interrupt latency when doing a 'lspci -v'.. should that 
number be 0? If so, does anyone know how to set that at boot time?


Flynn

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] AVM Fritz! + chan_capi + mISDN + PTP

2005-10-10 Thread Lionel Riem


... which is equivalent to my protocol=0x22 ;)

Nevertheless. I think it was a problem with chan_capi being too old  
and not supporting protocol=0x22 layermask=0xf (it would not work  
without layermask=0xf).


I am currently trying to get it working with chan_misdn. Will let you  
know how it goes. It was a pain in the arse to find some document  
about how to get it running, so I hope other people may use my  
findings somehow.


L. Riem
[EMAIL PROTECTED]


Le 10 oct. 05 à 16:34, Craig Guy a écrit :


Hi,

Yes, you can use the Fritz! in PTP mode, though only if you are  
using the mISDN drivers.  The mISDN driver should be called like this:


   modprobe avmfritz protocol=34

Craig

- Original Message - From: Lionel Riem [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, October 10, 2005 4:04 PM
Subject: [Asterisk-Users] AVM Fritz! + chan_capi + mISDN + PTP




Hello everyone,

I have been using an AVM Fritz! card with chan_capi and mISDN for   
quite a while in PTM mode and it was working finely.


Now, I needed more DID/MSN, so I switched to PTP. But now nothing   
works anymore :(


I am using Asterisk on Debian Sarge stable and installed Asterisk   
along with chan_capi from apt-get. I installed mISDN from the CVS  
of isdn4linux.de.


It is :
- Asterisk 1.0.7 with bristuff
- chan_capi 0.3.5

When I load the whole modules lot, I get the following in dmesg:

Modular ISDN Stack core $Revision: 1.25 $
mISDNd: kernel daemon started
ISAC module $Revision: 1.16 $
mISDNd: test event done
CAPI Subsystem Rev 1.1.2.8
capi20: Rev 1.1.2.7: started up with major 68 (middleware+capifs)
ISDN L1 driver version 1.11
ISDN L2 driver version 1.20
mISDN: DSS1 Rev. 1.30
mISDN Capi 2.0 driver file version 1.14
X25 DTE modul version 1.8
AVM Fritz PCI/PnP driver Rev. 1.30
ACPI: PCI interrupt :00:14.0[A] - GSI 10 (level, low) - IRQ 10
mISDN_fcpcipnp: found adapter Fritz!Card PCI v2 at :00:14.0
fritz card cd09a000 dch cd09a094 bch1 cd09a214 bch2 cd09a3a0
AVM PCI V2: stat 0x240020e
AVM PCI V2: Class E Rev 2
AVM PnP: HDLC version 2
mISDN: AVM Fritz!PCIv2 config irq:10 base:0xEC00
spin_lock_adr=cd09a024 now(d015b867)
busy_lock_adr=cd09a024 now(d015b867)
AVM PCI/PnP: reset
AVM PCI/PnP: S0/S1 40/2
Fritz1 ISAC STAR 40
Fritz1 ISAC MODE c0
Fritz1 ISAC ADF2 ff
Fritz1 ISAC ISTA 0
Fritz1 ISAC CIR0 7
mISDN_isac_init: ISACSX
Fritz1 HDLC 1 STA 8200
Fritz1 HDLC 2 STA 8200
AVM Fritz!PCI: IRQ 10 count 4
fritz 1 cards installed



Here is my /etc/asterisk/capi.conf:

;
; CAPI config
;
;
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8

[interfaces]
mode=immediate
isdnmode=ptp
msn=*
incomingmsn=*
controller=1
softdtmf=1
context=dispatcher
accountcode=
devices=2


Here is my /etc/modprobe.d/capi conf file:

alias /dev/capi20 avmfritz
alias char-major-68-0 avmfritz

install avmfritz /sbin/modprobe capi; \
/sbin/modprobe mISDN_core; \
/sbin/modprobe mISDN_l1; \
/sbin/modprobe mISDN_l2; \
/sbin/modprobe l3udss1; \
/sbin/modprobe mISDN_capi; \
/sbin/modprobe mISDN_x25dte; \
/sbin/modprobe --ignore-install avmfritz protocol=0x22

remove avmfritz /sbin/modprobe -r --ignore-remove avmfritz; \
/sbin/modprobe -r mISDN_x25dte; \
/sbin/modprobe -r mISDN_capi; \
/sbin/modprobe -r l3udss1; \
/sbin/modprobe -r mISDN_l2; \
/sbin/modprobe -r mISDN_l1; \
/sbin/modprobe -r mISDN_core; \
/sbin/modprobe -r capi



capiinfo shows me:

asterisk:/etc/asterisk# capiinfo
Number of Controllers : 1
Controller 1:
Manufacturer: mISDN CAPI controller Fritz1
CAPI Version: 2.0
Manufacturer Version: 1.0
Serial Number: 0002
BChannels: 2
Global Options: 0x0018
   DTMF supported
   Supplementary Services supported
B1 protocols support: 0x0003
   64 kbit/s with HDLC framing
   64 kbit/s bit-transparent operation
B2 protocols support: 0x0043
   ISO 7776 (X.75 SLP)
   Transparent
   Transparent (ignoring framing errors of B1 protocol)
B3 protocols support: 0x0005
   Transparent
   ISO 8208 (X.25 DTE-DTE)

  0100
  0200
  1800
  0300
  4300
  0500
       
      

Supplementary services support: 0x0012
   Terminal Portability
   Call Forwarding



In Asterisk, when an incoming call arrives, it shows me the  
following:


Asterisk Ready.
*CLI capi info
Contr1: 2 B channels total, 2 B channels free.
*CLI capi debug
CAPI Debugging Enabled
*CLI
*CLI
*CLI -- INFO_IND ID=001 #0x0001 LEN=0016
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x18
  InfoElement = 89

-- INFO_IND ID=001 #0x0001 LEN=0016
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x18
  InfoElement = 89

Oct 10 09:17:16 NOTICE[5948]: chan_capi.c:1184 find_pipe: PLCI  
doesnt match last pipe (PLCI = 0x101)
Oct 10 09:17:16 NOTICE[5948]: chan_capi.c:1301 pipe_msg: INFO_IND   
ID=001 #0x0001 LEN=0016

  Controller/PLCI/NCCI=

RE: [Asterisk-Users] sip register incoming call contexts?

2005-10-10 Thread Dennis Walker

Ok :)

--
From:   Rich Adamson[SMTP:[EMAIL PROTECTED]
Reply To:   Asterisk Users Mailing List - Non-Commercial Discussion
Sent:   Monday, October 10, 2005 11:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:Re: [Asterisk-Users] sip register incoming call contexts?

 Sorry this is a bit of a newbie question, I've been at this for a few
 months and still have not quite figured this one out.

 I've been able to setup one itsp (incoming calls) (sip account) with a
 register line like this:

 register = nnn:[EMAIL PROTECTED]

 -or-

 register = nnn:[EMAIL PROTECTED]/nnn
 to come directly into an extension in the dialplan

 It seems that this only works with the default context in the dialplan.

 I have another sip account from another provider that I would like
 all of it's incoming calls to come into the s, extension of
 a new context but I have been unable to figure out
 how to bring calls from a register line into an alternate context.

 It seems that register lines are limited to only being used in the
 general section of sip.conf and you are limited to one context=
 statement there.

 Is there a way to register a second account and have it's calls come into
 another context in the dialplan?

 register lines only seem to work in [general] and it seems like you
 are limited to only one inbound context here.

 I would like the two inbound call accounts to be 'isolated' from each other
 and not have to come in on the same incoming context in the dialplan.

 I'd also like to be able to have them have their own contexts with thier
 own s, (start) extension available.

Try using something like:
 deny=0.0.0.0/0.0.0.0  
 permit=147.135.8.129/255.255.255.0 
 permit=147.135.0.129/255.255.255.0
 permit=147.135.4.128/255.255.255.0

in each sip.conf itsp definition to limit which contexts will match.
Obviously, replace the above permit's IP addresses with the correct
ones for your provider.

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Re: www.openpbx.org

2005-10-10 Thread Paul


Mike M wrote:


On Sat, Oct 08, 2005 at 06:50:54PM -0300, Doug Meredith wrote:
 


Dinesh Nair [EMAIL PROTECTED] wrote:

   

too much divergence and we have two pieces of software competing for each 
other.
 


My guess is that if they succeed, they will diverge significantly.
   



We will have two pieces of software that work with each other at
well-defined interfaces.  The development of internal workings may diverge.

 

Well-defined interfaces is what I like to see even if there are never 
any forks. Things like XMLRPC or SOAP interfaces that are blessed by 
asterisk will motivate some of us to contribute more to the community. I 
find it easier to settle down and produce a complete application(even 
with some comments in the source) when I know it works against a 
well-defined interface - one that will persist for several releases.


Disclaimer - don't construe my mention of XMLRPC/SOAP as an endorsement 
or preference. That would start a whole new thread.


___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Open Source Content Management System - Joomla

2005-10-10 Thread Kanuri, Seshu $Company IT$

There was some discussion in the past about which one is the best
Content Management System that can be used in conjunction with Asterisk.
Mambo was supposed to be the best out there under GPL. The guys who
developed Mambo have a new product now - Joomla. I am using this and it
appears to be better than Mambo in many respects. Read the gist about
Joomla below.

-
If you've read anything at all about Content Management Systems (CMS),
you'll probably know at least three things: CMS are the most exciting
way to do business, CMS can be really, I mean really, complicated and
lastly Portals are absolutely, outrageously, often unaffordably
expensive. 

Joomla! is set to change all that ... Joomla! is different from the
normal models for portal software. For a start, it's not complicated.
Joomla! has been developed for the masses. It's licensed under the
GNU/GPL license, easy to install and administer and reliable. Joomla!
doesn't even require the user or administrator of the system to know
HTML to operate it once it's up and running.

http://www.joomla.org/

--

Seshu Kanuri


NOTICE: If received in error, please destroy and notify sender.  Sender does 
not waive confidentiality or privilege, and use is prohibited.
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Clicks, pops and noise


 snip
  
  One other item to check is to ensure the digium T1 card is on its own
  dedicated interrupt. Use 'cat /proc/interrupts' from the system command
  line.
  
 
 It is on one interrupt, first thing I checked when the problem cropped up. 
 One 
 thing I did notice was interrupt latency when doing a 'lspci -v'.. should 
 that 
 number be 0? If so, does anyone know how to set that at boot time?

I played around a fair amount with the latency thing and could not 
identify any noticable differences. I doubt that making changes there
will have any impact.


___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Open Source Content Management System - Joomla

2005-10-10 Thread Dean Collins

I'd also like to through the into the discussion my recommendation for
www.xoops.org much better solution than Mambo as far shorter learning
curve.

Also larger development team behind Xoops, particularly as Mambo is now
split messily into two camps and the whole issue about who owns what.

For a demo of a Xoops site check out mine www.AussieNYmeetup.net

This is a site that probably took me about 20-25 hours to put together
(could probably build it in about half that time now that I know what
I'm doing).

I'd also like to suggest for your research you check out
www.opensourcecms.com it's an amzing site that has a whole heap of open
source software able to be custom defined so you can get a feel for the
'backend' of the various CMS platforms and after 30 minutes it
automatically resets itself back to normal.

That's how I found out about www.xoops.org

Cheers,
Dean
(btw what this has to do with asterisk I have no idea, just responding
to the initial question)





 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Kanuri, Seshu (Company IT)
 Sent: Monday, 10 October 2005 11:11 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] Open Source Content Management System -
Joomla
 
 There was some discussion in the past about which one is the best
 Content Management System that can be used in conjunction with
Asterisk.
 Mambo was supposed to be the best out there under GPL. The guys who
 developed Mambo have a new product now - Joomla. I am using this and
it
 appears to be better than Mambo in many respects. Read the gist about
 Joomla below.
 
 -
 If you've read anything at all about Content Management Systems (CMS),
 you'll probably know at least three things: CMS are the most exciting
 way to do business, CMS can be really, I mean really, complicated and
 lastly Portals are absolutely, outrageously, often unaffordably
 expensive.
 
 Joomla! is set to change all that ... Joomla! is different from the
 normal models for portal software. For a start, it's not complicated.
 Joomla! has been developed for the masses. It's licensed under the
 GNU/GPL license, easy to install and administer and reliable. Joomla!
 doesn't even require the user or administrator of the system to know
 HTML to operate it once it's up and running.
 
 http://www.joomla.org/
 
 --
 
 Seshu Kanuri
 
 
 NOTICE: If received in error, please destroy and notify sender.
Sender
 does not waive confidentiality or privilege, and use is prohibited.
 ___
 --Bandwidth and Colocation sponsored by Easynews.com --
 
 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Multitenant Call Center Setup

2005-10-10 Thread BJ Weschke

There isn't a way to do it in agents.conf. 

That being said though, there are folks that have forgone agents.conf and have used the AddQueueMember and RemoveQueueMember commands via both the dial plan and manager interfaces to work their own agents approach that certainly could be designed to support a multi-tenant environment. 

On 10/10/05, Waldo Rubinstein [EMAIL PROTECTED] wrote:
Hi list (again),I have another question which I have not been able to resolve fromneither the wiki nor Google.
I've been able to configure a multi-tenant setup of asterisk for 2small call centers with no problem, by simply playing with contexts(which I guess is how everyone else is doing it).The problem I have is that I've only been able to configure one
global agents.conf file. This restricts my setup in a way that Icannot have two agents 1001, for example if my clients wanted to. Isthere a way to overcome this?Thanks,Waldo___
--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Clicks, pops and noise

2005-10-10 Thread El Flynn


Rich Adamson wrote:


It is on one interrupt, first thing I checked when the problem cropped up. One 
thing I did notice was interrupt latency when doing a 'lspci -v'.. should that 
number be 0? If so, does anyone know how to set that at boot time?



I played around a fair amount with the latency thing and could not 
identify any noticable differences. I doubt that making changes there

will have any impact.



Rich,

Thanks for the info! That'll save me some time since I don't have to bark up the 
wrong tree :)


On another note, I was told to double-check the memory on the server, _just_ in 
case that might be the source of all my problems. We're running the Memtest86 
app overnight, maybe something will turn up tomorrow.


Cheers,
Flynn


___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Help please

2005-10-10 Thread Carlos Trujillo

Hello, i`m carlos, i`m just begining to use Asterisk at Home, so i have learned to configure a several extensions, but now i have a FXO target and i wanna to connect to PSTN, but i dunno how to do. i`d like to receive support from you...thanks

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Incoming Calls causing Protocol Error (6)

2005-10-10 Thread Douglas Lane

Hi Everyone,

Got a setup as follows:

Telco  Siemens HiCom 300E  Asterisk1 IAX2 Trunk
Asterisk2  Siemens HiPath 4xxx

The solution works except for one problem. Incoming calls from the telco get
redirected to the Asterisk1 box with the correct extention, only if there is
a callerid set on the call, the Asterisk1 box drops the call (it doesn't
even get to asterisk) with a Unable to handle pre-handled call and
Protocol Error (6). If you disable your callerid on your phone and phone
again via the telco, it gets passed through. Asterisk1 reports Accepting
overlap call from '' to '5804'

Currently using ECMA.1 on the Siemens HiCom 300E, and Asterisk1 is setup
using euroisdn. I am using Asterisk 1.2.0-Beta1. Asterisk1 is running as
pri_cpe as well as secondary sync source.

Any ideas on how to fix this problem?

Would it be better to change the switchtype to Q.SIG on Asterisk and on the
Siemens HiCom 300E ?

Or am I missing a configuration line?

Thanks in advance.
Doug.


___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Bandwidth usage for codecs

2005-10-10 Thread Joey Kelly

On Monday October 10 2005 08:18, Kanishka Somaratne spake:
 hi
 how much bandwidth is used for the following codecs

http://www.voip-info.org/wiki/index.php?page=Bandwidth+consumption


-- 
Joey Kelly
 Minister of the Gospel | Linux Consultant 
http://joeykelly.net

I may have invented it, but Bill made it famous.
 --- David Bradley, the IBM employee that invented CTRL-ALT-DEL


pgpIckap6836R.pgp
Description: PGP signature
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Fredericksburg ZPUG Meeting will have an Asterisk Flavor this Month

2005-10-10 Thread Hadar Pedhazur

I've been asked to forward this announcement to the list. It's a little 
short notice as the meeting is this Wednesday night. I'm one of the 
presenters as well :-)

From: Gary Poster [EMAIL PROTECTED]
Date: October 10, 2005 11:51:10 AM EDT
To: zope-announce@zope.org, python-announce-list@python.org, 
[EMAIL PROTECTED]

Subject: Fifth Fredericksburg, VA ZPUG Meeting

Please join us October 12, 7:30-9:00 PM, for the fifth
meeting of the Fredericksburg, VA Zope and Python User Group
(ZPUG). Learn about Python configuration of Asterisk, an
open source VOIP! Free food!

Rob Page, Zope Corporation CEO and President, will present a
technical session on Asterisk [1] installation,
configuration and operation. A brief discussion of
connections to the public telephone network and internet
telephony providers will be presented.

Hadar Pedhazur, Zope Corporation Chairman of the Board, will
present a technical session on call handling and processing
using Python extensions to Asterisk.

We will also serve delicious fruit, cheese, and soft drinks.

We've had a nice group for all the meetings. Please come and
bring friends!

We also are now members of the O'Reilly and Apress user
group programs, which gives us nice book discounts (prices
better than Amazon's, for instance) and the possibility of
free review copies.  Ask me about details at the meeting if
you are interested.

General ZPUG information
When: second Wednesday of every month, 7:30-9:00.

Where: Zope Corporation offices. 513 Prince Edward Street;
Fredericksburg, VA 22408 (tinyurl for map is
http://tinyurl.com/ duoab).

Parking: Zope Corporation parking lot; entrance on Prince
Edward Street.

Topics: As desired (and offered) by participants, within the
constraints of having to do with Python.

Contact: Gary Poster ([EMAIL PROTECTED])

[1] From www.asterisk.org: Asterisk is a complete PBX in
software.  It runs on Linux, BSD and MacOSX and provides all
of the features you would expect from a PBX and
more. Asterisk does voice over IP in many protocols, and can
interoperate with almost all standards- based telephony
equipment using relatively inexpensive hardware.

Asterisk provides Voicemail services with Directory, Call
Conferencing, Interactive Voice Response and Call
Queuing. It has support for three-way calling, caller ID
services, ADSI, SIP and H. 323 (as both client and
gateway). Check the Features section for a more complete
list.

Asterisk needs no additional hardware for Voice over IP. For
interconnection with digital and analog telephony equipment,
Asterisk supports a number of hardware devices, most notably
all of the hardware manufactured by Asterisk's sponsors,
Digium�¹. Digium has single and quad span T1 and E1
interfaces for interconnection to PRI lines and channel
banks as well as a single port FXO card and a one to
four-port modular FXS and FXO card.
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Outgoing quality

2005-10-10 Thread Fabrizio Mazzoni

I'm having slight problems with outgoing audio quality on Zap channels.
People hear an interrupted voice.

Can anyone help..?

Regards,

Fabrizio Mazzoni
Macron SPA
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Incoming SIP getting in, but not ringing.

2005-10-10 Thread Paul Goodyear

Hi all.

Just as a quote note, can I thank everyone on this list. I find my
self finding pretty much every answer I am looking for on here. And a
big thanks to all thoughs helping us out. Mass Respect :)

Ok, I'm using a SIP provider (SipGate UK) to do my international
dialing etc, working great from extension 8 on phones. However some
more friends/contacts have started using SipGate also, and I want to
be able to do some SipGate to SipGate calls. As I said I can dial out
on SipGate no issues, but I cannot get my [EMAIL PROTECTED] box to receive
SipGate calls.

I have attached a text file with the sip debug option for a full
log. requests are coming in from SipGates server etc but my asterisk
box is not transfering the calls to the phones.

I have the register string in my sip.conf as so:

register=6698221:(MYSECRET)@sipgate.co.uk/6698221

Port on my IPCOP box as follows:

UDP/5060
UDP/1:2
UDP/8000:8012
UDP-TCP/3478

Thanks for your time.

Paul.
Sip read: 
INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Record-Route: sip:[EMAIL PROTECTED];ftag=as6a04ebdf;lr=on
Max-Forwards:  9
Record-Route: sip:[EMAIL PROTECTED];ftag=as6a04ebdf;lr=on
Via: SIP/2.0/UDP 217.10.79.219;branch=z9hG4bKeafc.0df1fb45.0
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKeafc.dd5c7802.0
Via: SIP/2.0/UDP 217.10.79.218:5060;branch=z9hG4bK66be0c1c
From: 07976xx sip:[EMAIL PROTECTED];tag=as6a04ebdf
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: sipgate asterisk
Date: Mon, 10 Oct 2005 15:53:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 448

v=0
o=root 5903 5903 IN IP4 217.10.79.218
s=session
c=IN IP4 217.10.79.55
t=0 0
m=audio 44214 RTP/AVP 8 0 3 97 18 2 4 5 110 7 10
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:110 speex/8000
a=rtpmap:7 LPC/8000
a=rtpmap:10 L16/8000
a=silenceSupp:off - - - -
a=direction:active
a=nortpproxy:yes

17 headers, 20 lines
Using latest request as basis request
Sending to 217.10.79.219 : 5060 (non-NAT)
Found peer 'SipGate'
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 217.10.79.219;branch=z9hG4bKeafc.0df1fb45.0
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKeafc.dd5c7802.0
Via: SIP/2.0/UDP 217.10.79.218:5060;branch=z9hG4bK66be0c1c
From: 07976xx sip:[EMAIL PROTECTED];tag=as6a04ebdf
To: sip:[EMAIL PROTECTED];tag=as60d08779
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Proxy-Authenticate: Digest realm=asterisk, nonce=557d3579
Content-Length: 0


 to 217.10.79.219:5060
Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms
asterisk1*CLI 

Sip read: 
ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 217.10.79.219;branch=z9hG4bKeafc.0df1fb45.0
From: 07976xx sip:[EMAIL PROTECTED];tag=as6a04ebdf
Call-ID: [EMAIL PROTECTED]
To: sip:[EMAIL PROTECTED];tag=as60d08779
CSeq: 102 ACK
User-Agent: sipgate ser
Content-Length: 0


8 headers, 0 lines
asterisk1*CLI 

Sip read: 
INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Record-Route: sip:[EMAIL PROTECTED];ftag=as6a04ebdf;lr=on
Max-Forwards:  9
Record-Route: sip:[EMAIL PROTECTED];ftag=as6a04ebdf;lr=on
Via: SIP/2.0/UDP 217.10.79.219;branch=z9hG4bKfafc.4aae3986.0
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKfafc.a85e7d75.0
Via: SIP/2.0/UDP 217.10.79.218:5060;branch=z9hG4bK49e9a0ad
From: 07976xx sip:[EMAIL PROTECTED];tag=as6a04ebdf
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 103 INVITE
User-Agent: sipgate asterisk
Date: Mon, 10 Oct 2005 15:53:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 448

v=0
o=root 5903 5904 IN IP4 217.10.79.218
s=session
c=IN IP4 217.10.79.55
t=0 0
m=audio 44214 RTP/AVP 8 0 3 97 18 2 4 5 110 7 10
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:110 speex/8000
a=rtpmap:7 LPC/8000
a=rtpmap:10 L16/8000
a=silenceSupp:off - - - -
a=direction:active
a=nortpproxy:yes

17 headers, 20 lines
Using latest request as basis request
Sending to 217.10.79.219 : 5060 (non-NAT)
Found peer 'SipGate'
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 217.10.79.219;branch=z9hG4bKfafc.4aae3986.0
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKfafc.a85e7d75.0
Via: SIP/2.0/UDP 217.10.79.218:5060;branch=z9hG4bK49e9a0ad
From: 07976xx sip:[EMAIL PROTECTED];tag=as6a04ebdf
To: sip:[EMAIL PROTECTED];tag=as60d08779
Call-ID: [EMAIL PROTECTED]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Proxy-Authenticate: Digest realm=asterisk, nonce=70112d01

Re: [Asterisk-Users] adding new indication tones

2005-10-10 Thread El Flynn


oner asterisk wrote:

Hi all,
 I would like to add indication tones ,
 What I did is
enter data to zonedata.c and indications.conf
than compile zaptel. and restart asterisk.
 But it's not working what else I should do ?
 Regards,
 Öner



did you check that the new tones are loaded in zaptel.conf?

flynn

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Multitenant Call Center Setup

2005-10-10 Thread Waldo Rubinstein

BJ,Thanks for the prompt response. Both my clients work by using the AgentCallBackLogin so that * can send queued calls to them regardless of which SIP phone they're sitting on (sorry I didn't include this in my original email)You mean to say that if I use AddQueueMember, I could do the same and still be able to have two agents 1001?Thanks,WaldoOn Oct 10, 2005, at 11:38 AM, BJ Weschke wrote: There isn't a way to do it in agents.conf.     That being said though, there are folks that have forgone agents.conf and have used the AddQueueMember and RemoveQueueMember commands via both the dial plan and manager interfaces to work their own agents approach that certainly could be designed to support a multi-tenant environment.   On 10/10/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: Hi list (again),I have another question which I have not been able to resolve fromneither the wiki nor Google. I've been able to configure a multi-tenant setup of asterisk for 2small call centers with no problem, by simply playing with contexts(which I guess is how everyone else is doing it).The problem I have is that I've only been able to configure one global agents.conf file. This restricts my setup in a way that Icannot have two agents 1001, for example if my clients wanted to. Isthere a way to overcome this?Thanks,Waldo___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] sip register incoming call contexts?

2005-10-10 Thread Steve Gladden

Thank you for your reply and your help.
I am still confused here and apologize.
To some degree I still do not know what I am doing.

We use 2 ITSP's and one of them we have multiple SIP accounts
on so I will not be able to do this by IP address.

For incoming calls we use a register line in the [general]
section of sip.conf like: register = nnn:[EMAIL PROTECTED]

We do not have an 'itsp section' for incoming calls.

incoming calls come into the context defined in the general section
of sip.conf.

This is how we learned how to do it from the documentation

my understanding is that anything else that involves sections like

[itsp-provider out]
yada=
yada=
yada=
-or-
[itsp-provider-in]
yada=
yada=
yada=

Works for permanent non-registered types of connections.

I've experimented with trying to put register lines within anything else
other than [general] in sip.conf and it does not work
and causes a busy signal for an incoming caller.

My further under(possibly-mis)undertanding is that with our type
of itsp (sip) it requires us to register for incoming calls,
and there may be no other way to accept incoming calls from our
ITSP,

It also seems that register lines only work in the [general] section
of sip.conf which only allows me to define one single incoming context
 is this correct?


So the matching by IP address is interesting but confusing and may not
apply to what I am trying to do.

I will not be able to match by ip with seeveral incoing sip (phone numbers)
that I would like to come into their own context but come from the same IP
address.

Thanks!!

Steve






 Ok :)

 --
 From: Rich Adamson[SMTP:[EMAIL PROTECTED]
 Reply To: Asterisk Users Mailing List - Non-Commercial Discussion
 Sent: Monday, October 10, 2005 11:25 AM
 To:   Asterisk Users Mailing List - Non-Commercial Discussion
 Subject:  Re: [Asterisk-Users] sip register incoming call contexts?


 Sorry this is a bit of a newbie question, I've been at this for a few
 months and still have not quite figured this one out.


 I've been able to setup one itsp (incoming calls) (sip account) with a
 register line like this:

 register = nnn:[EMAIL PROTECTED]

 -or-

 register = nnn:[EMAIL PROTECTED]/nnn
 to come directly into an extension in the dialplan


 It seems that this only works with the default context in the dialplan.


 I have another sip account from another provider that I would like
 all of it's incoming calls to come into the s, extension of
 a new context but I have been unable to figure out
 how to bring calls from a register line into an alternate context.

 It seems that register lines are limited to only being used in the
 general section of sip.conf and you are limited to one context=
 statement there.

 Is there a way to register a second account and have it's calls come
 into
 another context in the dialplan?

 register lines only seem to work in [general] and it seems like you
 are limited to only one inbound context here.

 I would like the two inbound call accounts to be 'isolated' from each
 other
 and not have to come in on the same incoming context in the dialplan.

 I'd also like to be able to have them have their own contexts with thier
 own s, (start) extension available.

 Try using something like:
  deny=0.0.0.0/0.0.0.0
  permit=147.135.8.129/255.255.255.0
  permit=147.135.0.129/255.255.255.0
  permit=147.135.4.128/255.255.255.0

 in each sip.conf itsp definition to limit which contexts will match.
 Obviously, replace the above permit's IP addresses with the correct
 ones for your provider.


 ___
 --Bandwidth and Colocation sponsored by Easynews.com --

 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

 ___
 --Bandwidth and Colocation sponsored by Easynews.com --

 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] telephony that just works

2005-10-10 Thread Michael Van Donselaar

On Mon, 10 Oct 2005 11:01:12 -0300, Ivan Stepaniuk [EMAIL PROTECTED]
wrote:

On Mon, 2005-10-10 at 13:28 +0200, lenz wrote:
 I am looking for a way to have multiple remote Windows users download a  
 package and get connected to *. My idea would be that they run a simple  
 app, it connects without any setting to an * box (maybe via IAX) and then  
 people press a button to talk. It would be okay if they had to enter a  
 username and password, but not more than that.

i've tried IaxComm 
http://iaxclient.sourceforge.net/iaxcomm/

it works, it's iax, and it's open source so you can re-package -
re-compile it with you own default settings (or even hide those settings
you don't want final users to see)  

Rather than recompile with presets, you'd probably want to change the reg keys
used in the installer.  When I was first developing iaxComm for family and
friends, I distributed the executable with a .reg file with their
username/password, the asterisk server , and a few speed dials preset.

I finally wrote the installer script for an ITSP that has a really neat
approach:

The user provides username and password on the web page.  The server modifies
the username and password in the nsi script, and rebuilds a new installer for
each user.


-- 
Ivan Stepaniuk [EMAIL PROTECTED]

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] CLIR in chan_mISDN

2005-10-10 Thread Andreas Mavrides


Hi all

I have configured an ISDN bri card to work in TE mode using chan_mISDN in 
asterisk. I can both place and receive calls through my ISDN line with no 
problems. I am trying to restrict sending my caller id (CLIR) but I don't 
seem to find how to do it. Does anyone know how to restrict sedning the 
caller id?


Many thanks in advance

Andreas Mavrides 



___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Hang up call

2005-10-10 Thread Rene Nelson

For some reason, every morning at 8:30 I get a call on my main
extension. When this call is picked up, it promptly
disconnects. Is there some sort of Wake up call or something
that may inadvertently be set in * that could be causing this? It
has been happening for quite some time, and I always just brushed it
off, but it's consistency and regularity has caused me to wonder.

Thanks for any/all help.

Neru

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] sip register incoming call contexts?

2005-10-10 Thread Olle E. Johansson

Steve Gladden wrote:
 Sorry this is a bit of a newbie question, I've been at this for a few
 months and still have not quite figured this one out.
 
 
 I've been able to setup one itsp (incoming calls) (sip account) with a
 register line like this:
 
 register = nnn:[EMAIL PROTECTED]
 
 -or-
 
 register = nnn:[EMAIL PROTECTED]/nnn
 to come directly into an extension in the dialplan
 
 
 It seems that this only works with the default context in the dialplan.
 
 
 I have another sip account from another provider that I would like
 all of it's incoming calls to come into the s, extension of
 a new context but I have been unable to figure out
 how to bring calls from a register line into an alternate context.

Create a peer with a host= setting that matches the IP of the service
provider's proxy. Set context for this peer. There are several examples
out there, one is http://edvina.net/broadvoice/

/Olle
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] 2 line SIP ATAs with Asterisk using RealTime

2005-10-10 Thread Dave Wise

I am running CVS Head i686 running Linux on 2005-06-30 22:55:14.  I have 
SIP Buddies installed using MySQL.


If I try to set up a ATA that has 2 two phone lines (resulting in 2 
lines on 1 IP address), my second line can never authenticate to dial out.
I ran ethereal and found that Asterisk is looking at the IP the request 
came from and then, apparently looking up the IP  address in the SIP 
table and responding to the first match of username to the IP address 
(this also happens if I plug in one phone to test it and use a 
designated IP address and then remove that phone and test with a 
different phone but with the same IP address, it uses the data from the 
lowest row number that the IP field matches).


Is there any work around to this.  I know that the SIP port is different 
for line 1 and line 2.  Like I mentioned above, ethereal shows that 
Asterisk is changing the responses to a different user (or that is what 
I interpreted it to be doing).


I also tried changing insecure to try to ignore the port number with no 
success.

I tried the following values in insecure:
port
port, invite
invite
yes

I looked on the WIKI and could not find a solution either.  I would 
appreciate any help.



___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] CDR problem with DST Channel

2005-10-10 Thread pbx

I have 3 different SIP extensions in my DIAL string.

i.e. I have HOME_PHONES_TO_RING=SIP/2000SIP/2001SIP/2002

so in my Extensions file i have Dial(${HOME_PHONES_TO_RING},30,tTr)

So... when the home phone line rings, all three phones ring.

Anyways.. the problem is.. in the CDR log, sometimes the log entry shows
2000, sometimes 2001, sometimes 2002

Only extension 2000 answered the call, yet, 2001 is listed as the
answering channel, or 2002 is listed as the answering channel? The LASTAPP
column all show Hangup, and disposition shows ANSWERED.

Is there a way to to force a flush to the CDR to make it reflect the
correct phone that answered?

Thanks

./Ben

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] PSTN CALLER ID FRANCE TELECOM

2005-10-10 Thread guillaume


Hi all

I try to get the caller id of a incoming call through a X100P generic card.
I have tried many configuration on the zapata.conf, but i never succeed 
to have a correct CALLERIDNUM.


What is the cid signaling provided by FranceTelecom (v23 ?)
Is there some specific stuff to do ?

Could you help me please ?


Guillaume



The provider is france telecom,
The card is a X100P
asterisk -V = 1.0.9
The error message is :
   Oct 10 20:17:02 WARNING[702]: chan_zap.c:5476 ss_thread: Calleerror 
on channel 'Zap/1-1'




my zapata.conf is ~

context=from-ft
language=fr
signalling=fxs_ks
busydetect=yes
busycount=1
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
usecallerid=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=6.0
txgain=4.0
immediate=no
callerid=asreceived
musiconhold=default
channel = 1








___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Hang up call

2005-10-10 Thread Andy Hamilton

 For some reason, every morning at 8:30 I get a call on my main extension.
 When this call is picked up, it promptly disconnects.  Is there some sort of
 Wake up call or something that may inadvertently be set in * that could be
 causing this?  It has been happening for quite some time, and I always just
 brushed it off, but it's consistency and regularity has caused me to wonder.

Asterisk can make calls based on files that appear in
/var/spool/asterisk/outgoing

You may want to check the contents of this directory around 08:28 or
08:29. If there is a file in there, that's probably why you're getting
the call (Asterisk should delete the file after the call is has been
answered/timed out). So check the directory around 08:30; if the file
is gone then but reappears the next morning, then it sounds like
something is causing that file to be generated.

If there is never a file in there, then the calls are coming from elsewhere ;)
What does callerid show? How about Asterisk console?

-Andy
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Realtime regseconds update

2005-10-10 Thread Miguel Cavazos

Hi guys, im using realtime and I want to show registered users or  
online users on a webpage and offline users. Im taking regseconds  
field to make this happend


If regseconds value is 0 then user appers offline, it regseconds is  
something else then its online, but sometimes this works and  
sometimes it does not. Im using the following options


rtcachefriends=yes
rtnoupdate=yes
rtautoclear=yes

anyone has any idea? im using 1.2.0beta1, im not sure if its updating  
this field, i have on also set in my sip.conf file


defaultexpirey=300
maxexpirey=300

Also my atas, are set with this value, so it should expire in 300  
seconds but sometimes this doesnt occure.


Miguel
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Realtime regseconds update

2005-10-10 Thread Trevor Peirce


Miguel Cavazos wrote:

Hi guys, im using realtime and I want to show registered users or  
online users on a webpage and offline users. Im taking regseconds  
field to make this happend


If regseconds value is 0 then user appers offline, it regseconds is  
something else then its online, but sometimes this works and  
sometimes it does not. Im using the following options



regseconds is when the registration expires, in unix time
You need to check to see if regseconds is in the past or in the 
future... past = expired, future = registered

If it's 0, the user has never been online.
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Help, please help -- IAX2 softphone to server on LAN

2005-10-10 Thread Wolfgang Borgon

I've already sunk several hours into this without any
real progress, so I'd really appreciate any help  My
task is simple -- establish a connection between a
softphone on XP ProSP2 to a Asterisk server on Linux
FC4 over a LAN through a Netgear router. The server
will then go out to a PSTN termination service.

Thus far, the PSTN termination connection works fine
-- I've opened up 4569 with iptables, and forwarded
4569 to the server IP.  I am not, however, having any
luck connecting the softphone to the server.

I can telnet, ftp, and http to the server, but not
IAX2. Iaxping times out, registration by Idefisk and
Firefly also times out.  

The server fails to see the client as well.  

Here's a portion of my iax.conf:

[client]
type=friend
username=client
secret=**
host=192.168.1.40
context=clientcon

and extensions.conf:

[clientcon]
exten = 2278,1,Dial(IAX2/client)


Here's the output of 'iax show peers':

Name/UsernameHost Mask
Port  Status
voxee/#   66.246.246.52   (S)  255.255.255.255
 4569  Unmonitored
client/client  192.168.1.40 (S) 
255.255.255.255  0 Unmonitored
demo/asterisk216.207.245.47  (S)  255.255.255.255 
4569  Unmonitored

Note the port listed at 0.


Debug reponse to 'dial [EMAIL PROTECTED]':

-- Executing Dial(ALSA/default, IAX2/client)
in new stack
-- Called client
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type:
IAX Subclass: NEW
   Timestamp: 00016ms  SCall: 1  DCall: 0
[192.168.1.40:0]
   VERSION : 2
   CALLED NUMBER   : s
   CODEC_PREFS : (ilbc|ulaw|alaw|gsm)
   CALLING PRESNTN : 67
   CALLING TYPEOFN : 0
   CALLING TRANSIT : 0
   LANGUAGE: en
   USERNAME: client
   FORMAT  : 2
   CAPABILITY  : 64526
   ADSICPE : 0
   DATE TIME   : 2005-10-10  00:04:14

Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type:
IAX Subclass: NEW
   Timestamp: 00016ms  SCall: 1  DCall: 0
[192.168.1.40:0]
   VERSION : 2
   CALLED NUMBER   : s
   CODEC_PREFS : (ilbc|ulaw|alaw|gsm)
   CALLING PRESNTN : 67
   CALLING TYPEOFN : 0
   CALLING TRANSIT : 0
   LANGUAGE: en
   USERNAME: client
   FORMAT  : 2
   CAPABILITY  : 64526
   ADSICPE : 0
   DATE TIME   : 2005-10-10  00:04:14

-- IAX2/client-1 is circuit-busy
Oct 10 00:04:19 NOTICE[3615]: chan_iax2.c:2754
auto_congest: Auto-congesting cal l due to slow
response
-- Hungup 'IAX2/client-1'
  == Everyone is busy/congested at this time (1:0/1/0)
  == Auto fallthrough, channel 'ALSA/default' status
is 'CONGESTION'

















__ 
Yahoo! Music Unlimited 
Access over 1 million songs. Try it free.
http://music.yahoo.com/unlimited/
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] 2 line SIP ATAs with Asterisk using RealTime

2005-10-10 Thread William Lloyd


Setup the two ports completely separately.

Each should have it's own entry in realtime with a unique username.

-bill

On 10-Oct-05, at 1:15 PM, Dave Wise wrote:

I am running CVS Head i686 running Linux on 2005-06-30 22:55:14.  I  
have SIP Buddies installed using MySQL.


If I try to set up a ATA that has 2 two phone lines (resulting in 2  
lines on 1 IP address), my second line can never authenticate to  
dial out.
I ran ethereal and found that Asterisk is looking at the IP the  
request came from and then, apparently looking up the IP  address  
in the SIP table and responding to the first match of username to  
the IP address (this also happens if I plug in one phone to test it  
and use a designated IP address and then remove that phone and test  
with a different phone but with the same IP address, it uses the  
data from the lowest row number that the IP field matches).


Is there any work around to this.  I know that the SIP port is  
different for line 1 and line 2.  Like I mentioned above, ethereal  
shows that Asterisk is changing the responses to a different user  
(or that is what I interpreted it to be doing).


I also tried changing insecure to try to ignore the port number  
with no success.

I tried the following values in insecure:
port
port, invite
invite
yes

I looked on the WIKI and could not find a solution either.  I would  
appreciate any help.



___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users



___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] *8 and group pickup not working

2005-10-10 Thread Alberto Risco

I don't know if this will help you, but we had the same problem, we also
have Polycom 500s and I changed the pickupexten to *9 (anything other
than *8), because I read somewhere that for some reason Asterisk has a
problem with this feature and *8.  It worked for us.


Alberto

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Angus
Comber
Sent: Sunday, October 09, 2005 2:48 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] *8 and group pickup not working

No that's not problem.

On my current configs I get:

Oct  9 20:43:18 NOTICE[2990]: chan_sip.c:7455 handle_request: Nothing to

pick up

every time I try *8

Why does the phone think there is nothing to pickup?

Angus




- Original Message - 
From: Alan Harrison [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Sunday, October 09, 2005 2:35 PM
Subject: Re: [Asterisk-Users] *8 and group pickup not working


 On Sun, 9 Oct 2005 21:32, Angus Comber wrote:
 Hi

 I have Polycom 600s and 500s but I find that we need to dial *8 then
send. 
 If
 we pickup then dial *8 the phone or Asterisk re-aranges it to 8*. 
 Likewise
 with *97 and *98 foes to 9*7 and 9*8.

 This might help.

 Hello

 I have a Junghanns ISDN BRI card for incoming calls and use SIP
Polycom
 IP300 phones.

 My config files look like this:

 features.conf
 pickupextn = *8

 zapata.conf
 context=frompstnisdn
 group=1
 callgroup=1
 pickupgroup=1

 I also edited sip.conf like this:
 group=1
 callgroup=1
 pickupgroup=1


 But on internal and incoming calls if I dial *8 from any phone I
cannot
 pickup.  Do I need to add anything to extensions.conf?  do something 
 else.
 I also tested with a Snom 190 and that cannot pickup using *8 either!

 Angus



 ___
 --Bandwidth and Colocation sponsored by Easynews.com --

 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

 -- 
 Regards,

 Alan Harrison
 PABX Advisory Services Pty Ltd
 PH  02 9893 7888
 Email [EMAIL PROTECTED]
 ___
 --Bandwidth and Colocation sponsored by Easynews.com --

 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
 


___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

The contents of this email message and any attachments are confidential and are 
intended solely for addressee. The information may also be legally privileged. 
This transmission is sent in trust, for the sole purpose of delivery to the 
intended recipient. If you have received this transmission in error, any use, 
reproduction or dissemination of this transmission is strictly prohibited. If 
you are not the intended recipient, please immediately notify the sender by 
reply email and delete this message and its attachments, if any.


___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Faking it: queue_log and addQueueMember


Hello list,
today I have been busy playing with addQueueMember, and it is well known  
that it does not log to the queue_log file.


The answer - bad as it may seem - is to add a fake queue_log data for each  
logon and logoff. This was covered previously by  
http://lists.digium.com/pipermail/asterisk-dev/2005-February/009615.html


Unfortunately this solution is not so simple, because:
1. when an agent logs off, they log how much time they have been on line
2. multiple logon/logoff lines are *bad* for log analysis

I have written a simple logon-logoff script that might help you in faking  
it. Of course you should likely add authentication for a real-world  
usage. To do the adding, you dial 422XX, where XX is your local extension;  
the same goes to be removed from queue.


; addqueuemember - 422
exten = _422XX,1,Answer
exten = _422XX,2,AddQueueMember(my-queue,SIP/${EXTEN:3})
exten = _422XX,3,System( echo  
${EPOCH}|${UNIQUEID}|NONE|SIP/${EXTEN:3}|AGENTLOGIN|-  /var/

log/asterisk/queue_log )
exten = _422XX,4,DBput(dynlogin/log_Agent-${EXTEN:3}=${EPOCH})
exten = _422XX,5,Hangup

; removequeuemember - 423
exten = _423XX,1,Answer
exten = _423XX,2,RemoveQueueMember(my-queue,SIP/${EXTEN:3})
exten = _423XX,3,DBget(ORGEPOCH=dynlogin/log_Agent-${EXTEN:3})
exten = _423XX,4,Set(RV=$[${EPOCH} - ${ORGEPOCH}])
exten = _423XX,5,GotoIf($[${RV} = 0]?8:6)
exten = _423XX,6,System( echo  
${EPOCH}|${UNIQUEID}|NONE|SIP/${EXTEN:3}|AGENTLOGOFF|-|${RV}

/var/log/asterisk/queue_log )

exten = _423XX,7,DBdel(dynlogin/log_Agent-${EXTEN:3})
exten = _423XX,8,Hangup

Hope this helps. With this setup, I verified that the queue_log can be  
analyzed by QueueMetrics and the dynamic agent shows up fine (albeit with  
the name of a terminal, like SIP/23, instead of the usual Agent/23 string,  
but you can modify it in QM itself).


This setup might even be used in a call center where agents are not  
actually used but queues connect straight to terminals to fake agent  
logon/logoff, in order to have such data available for reporting.


Any comment is welcome!
l.



--
Assum est, versa et manduca.
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] telephony that just works

In data Mon, 10 Oct 2005 18:57:20 +0200, Michael Van Donselaar  
[EMAIL PROTECTED] ha scritto:


Rather than recompile with presets, you'd probably want to change the  
reg keys
used in the installer.  When I was first developing iaxComm for family  
and

friends, I distributed the executable with a .reg file with their
username/password, the asterisk server , and a few speed dials preset.

I finally wrote the installer script for an ITSP that has a really neat
approach:

The user provides username and password on the web page.  The server  
modifies
the username and password in the nsi script, and rebuilds a new  
installer for

each user.



This is more or less what I wanted to do. But I think I can simply send my  
users a different .reg file every time.
DIAX also was promising - uses text files, do you just copy the directory  
and go - but I fear the licence.

Bye
l.


--
Assum est, versa et manduca.
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] *8 and group pickup not working

2005-10-10 Thread Angus Comber


When I added
group=1
callgroup=1
pickupgroup=1

under each extension then it worked.  I assume it is the pickupgroup=1 that 
did it.  I will experiment to see.


Angus

- Original Message - 
From: Alberto Risco [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com; [EMAIL PROTECTED]

Sent: Monday, October 10, 2005 7:14 PM
Subject: RE: [Asterisk-Users] *8 and group pickup not working


I don't know if this will help you, but we had the same problem, we also
have Polycom 500s and I changed the pickupexten to *9 (anything other
than *8), because I read somewhere that for some reason Asterisk has a
problem with this feature and *8.  It worked for us.


Alberto

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Angus
Comber
Sent: Sunday, October 09, 2005 2:48 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] *8 and group pickup not working

No that's not problem.

On my current configs I get:

Oct  9 20:43:18 NOTICE[2990]: chan_sip.c:7455 handle_request: Nothing to

pick up

every time I try *8

Why does the phone think there is nothing to pickup?

Angus




- Original Message - 
From: Alan Harrison [EMAIL PROTECTED]

To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, October 09, 2005 2:35 PM
Subject: Re: [Asterisk-Users] *8 and group pickup not working



On Sun, 9 Oct 2005 21:32, Angus Comber wrote:
Hi

I have Polycom 600s and 500s but I find that we need to dial *8 then

send.

If
we pickup then dial *8 the phone or Asterisk re-aranges it to 8*.
Likewise
with *97 and *98 foes to 9*7 and 9*8.

This might help.


Hello

I have a Junghanns ISDN BRI card for incoming calls and use SIP

Polycom

IP300 phones.

My config files look like this:

features.conf
pickupextn = *8

zapata.conf
context=frompstnisdn
group=1
callgroup=1
pickupgroup=1

I also edited sip.conf like this:
group=1
callgroup=1
pickupgroup=1


But on internal and incoming calls if I dial *8 from any phone I

cannot

pickup.  Do I need to add anything to extensions.conf?  do something
else.
I also tested with a Snom 190 and that cannot pickup using *8 either!

Angus



___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


--
Regards,

Alan Harrison
PABX Advisory Services Pty Ltd
PH  02 9893 7888
Email [EMAIL PROTECTED]
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users




___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

The contents of this email message and any attachments are confidential and 
are intended solely for addressee. The information may also be legally 
privileged. This transmission is sent in trust, for the sole purpose of 
delivery to the intended recipient. If you have received this transmission 
in error, any use, reproduction or dissemination of this transmission is 
strictly prohibited. If you are not the intended recipient, please 
immediately notify the sender by reply email and delete this message and its 
attachments, if any.



___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] CallerID Outbound on VOXEE

2005-10-10 Thread Jerry James









Has anyone successfully managed to get outbound CallerID to
work on outbound calls through VOXEE?

On CallerID I mean NUMBER ( I know how the PSTN works)

My callees keeps getting private call,
of course the call is blocked when callee has anonymous call block.

I have searched through the wikki and have added VOXEE
suggested line (from their web page) into extensions.conf

Support at Voxee claims they have no control of what the
ILEC does with their info they parse to them.

I have tried to explain to them the difference of private
and unknown 

It has always been my experience that private
is only used when one of the following is done:

ISDN setup or ss7 IAM message is set with presentation
restricted.

Any technical difficulty such as no numbers being sent,
traversing over a non SS7 or ISDN path results in

A UNKNOWN 

At this point I would be happy with a UNKNOWN.



Jerry










--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.344 / Virus Database: 267.11.13/126 - Release Date: 10/9/2005
 
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] telephony that just works

In data Mon, 10 Oct 2005 14:03:55 +0200, tim panton  
[EMAIL PROTECTED] ha scritto:

Yep, I'm working on such a thing.

I have a demo version running at http://www.westhawk.co.uk/software/ 
faceless/CallUs.html


You don't even need to install it, it runs in the user's browser.
( you will need IE6 and java installed -  I'll get other browsers  
supported later this week).


Email me if you want to test it out, and I'll arrange to answer the  
phone :-)


Tim.



I'd say it's excellent! worked fine with my Opera browser.
how do you plan to release this thingie? looks very very promising!
Thanks
l.



--
Assum est, versa et manduca.
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] CallerID Outbound on VOXEE

2005-10-10 Thread Cirelle Enterprises


in sip.conf
[yourextensioncontext]
callerid=1234567890 1234567890

in extensions.conf
[voxee]
exten = _1NX,1,SetCallerID(${CALLERID} |a)

HTH

Best Regards

Greg Cirino

Spam and Virus Free Email
included with every email account

Cirelle Enterprises Inc.
25 Indian Rock Rd #421
Windham NH, 03087
603-425-2221

Jerry James wrote:

Has anyone successfully managed to get outbound CallerID to work on outbound
calls through VOXEE?

On CallerID I mean NUMBER ( I know how the PSTN works)

My callee’s keeps getting “private call”, of course the call is blocked when
callee has anonymous call block.

I have searched through the wikki and have added VOXEE suggested  line (from
their web page) into extensions.conf

Support at Voxee claims they have no control of what the ILEC does with
their info they parse to them.

I have tried to explain to them the difference of “private” and “unknown”  


It has always been my experience that “private” is only used when one of the
following is done:

ISDN setup or ss7 IAM message is set with presentation restricted.

Any technical difficulty such as no numbers being sent, traversing over a
non SS7 or ISDN path results in

A “UNKNOWN” 


At this point I would be happy with a UNKNOWN.

 


Jerry

 







___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Help, please help -- IAX2 softphone to server on LAN

2005-10-10 Thread David J Carter

Wolfgang wrote:  -

I've already sunk several hours into this without any
real progress, so I'd really appreciate any help  My
task is simple -- establish a connection between a
softphone on XP ProSP2 to a Asterisk server on Linux
FC4 over a LAN through a Netgear router. The server
will then go out to a PSTN termination service.

Thus far, the PSTN termination connection works fine
-- I've opened up 4569 with iptables, and forwarded
4569 to the server IP.  I am not, however, having any
luck connecting the softphone to the server.

I can telnet, ftp, and http to the server, but not
IAX2. Iaxping times out, registration by Idefisk and
Firefly also times out.  

The server fails to see the client as well.  

Here's a portion of my iax.conf:

[client]
type=friend
username=client
secret=**
host=192.168.1.40
context=clientcon

and extensions.conf:

[clientcon]
exten = 2278,1,Dial(IAX2/client)


==
You say you have 4569 configured in iptables, what about the netgear router?

Have you port forwarded 4569 there?

Dave

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] What is this error? Is there a bug?

2005-10-10 Thread Dan Journo

Im getting this warning on the CLI.

Please im having problems getting extensions to register while using the realm instead of the IP address.
Oct 10 20:17:15 WARNING[3105]: chan_sip.c:11178 add_realm_authentication: Format for authentication entry is user[:[EMAIL PROTECTED] at line 0

Can anyone shed some light on this?

Dan
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Help, please help -- IAX2 softphone to server on LAN

2005-10-10 Thread Wolfgang Borgon

David,
Yes, I've also forwarded port 4569 to the server. 
Since the router is forwarding to the server, I cannot
forward it to the client as well -- however, as the
client isn't going out past the LAN, it shouldn't
matter... unless there's something else going on that
I don't know about.
Thanks
Wolfgang


--- David J Carter [EMAIL PROTECTED] wrote:

 Wolfgang wrote:  -
 
 I've already sunk several hours into this without
 any
 real progress, so I'd really appreciate any help  My
 task is simple -- establish a connection between a
 softphone on XP ProSP2 to a Asterisk server on Linux
 FC4 over a LAN through a Netgear router. The server
 will then go out to a PSTN termination service.
 
 Thus far, the PSTN termination connection works fine
 -- I've opened up 4569 with iptables, and forwarded
 4569 to the server IP.  I am not, however, having
 any
 luck connecting the softphone to the server.
 
 I can telnet, ftp, and http to the server, but not
 IAX2. Iaxping times out, registration by Idefisk and
 Firefly also times out.  
 
 The server fails to see the client as well.  
 
 Here's a portion of my iax.conf:
 
 [client]
 type=friend
 username=client
 secret=**
 host=192.168.1.40
 context=clientcon
 
 and extensions.conf:
 
 [clientcon]
 exten = 2278,1,Dial(IAX2/client)
 
 

==
 You say you have 4569 configured in iptables, what
 about the netgear router?
 
 Have you port forwarded 4569 there?
 
 Dave
 
 ___
 --Bandwidth and Colocation sponsored by Easynews.com
 --
 
 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com

http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
   

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





__ 
Yahoo! Mail - PC Magazine Editors' Choice 2005 
http://mail.yahoo.com
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Help, please help -- IAX2 softphone to server on LAN

2005-10-10 Thread pbx


David:

Also port 1:2 is a good idea to forward to the server as well..


 David,
 Yes, I've also forwarded port 4569 to the server.
 Since the router is forwarding to the server, I cannot
 forward it to the client as well -- however, as the
 client isn't going out past the LAN, it shouldn't
 matter... unless there's something else going on that
 I don't know about.
 Thanks
 Wolfgang


 --- David J Carter [EMAIL PROTECTED] wrote:

 Wolfgang wrote:  -

 I've already sunk several hours into this without
 any
 real progress, so I'd really appreciate any help  My
 task is simple -- establish a connection between a
 softphone on XP ProSP2 to a Asterisk server on Linux
 FC4 over a LAN through a Netgear router. The server
 will then go out to a PSTN termination service.

 Thus far, the PSTN termination connection works fine
 -- I've opened up 4569 with iptables, and forwarded
 4569 to the server IP.  I am not, however, having
 any
 luck connecting the softphone to the server.

 I can telnet, ftp, and http to the server, but not
 IAX2. Iaxping times out, registration by Idefisk and
 Firefly also times out.

 The server fails to see the client as well.

 Here's a portion of my iax.conf:

 [client]
 type=friend
 username=client
 secret=**
 host=192.168.1.40
 context=clientcon

 and extensions.conf:

 [clientcon]
 exten = 2278,1,Dial(IAX2/client)



 ==
 You say you have 4569 configured in iptables, what
 about the netgear router?

 Have you port forwarded 4569 there?

 Dave

 ___
 --Bandwidth and Colocation sponsored by Easynews.com
 --

 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com

 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:


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






 __
 Yahoo! Mail - PC Magazine Editors' Choice 2005
 http://mail.yahoo.com
 ___
 --Bandwidth and Colocation sponsored by Easynews.com --

 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users



___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Problem with Oh323 on 1.2Beta on CENTOS 3.5

2005-10-10 Thread VoIP Carib




Hello List,

I have a problem with my Oh323 install on 1.2Beta. 
I have 1.2Beta installed on CENTOS 3.5 and downloaded the following 
packages:

pwlib-Mimas_patch2-src-tar.gz
openh323-Mimas_patch2-src-tar.gz
asterisk-oh323-0.7.3.tar.gz

However when I go into the  pwlib_Mimas_patch2 and 
run a ./configure, I get an error:

[EMAIL PROTECTED] pwlib_Mimas_patch2]# 
/configurechecking for g++... g++checking for C++ compiler default 
output file name... a.outchecking whether the C++ compiler works... 
yeschecking whether we are cross compiling... nochecking for suffix of 
executables... checking for suffix of object files... ochecking whether 
we are using the GNU C++ compiler... yeschecking whether g++ accepts -g... 
yesconfigure: PTLib version is 1.8.7checking build system type... 
i686-pc-linux-gnuchecking host system type... i686-pc-linux-gnuchecking 
target system type... i686-pc-linux-gnuconfigure: OSTYPE set to 
linuxconfigure: OSRELEASE set to "2.4.21-37.EL"configure: MACHTYPE set 
to x86configure: gcc version is 3.2.3checking checking if pragma 
implementation should be used... yeschecking whether byte ordering is 
bigendian... nochecking if linker accepts -felide-constructors... 
yeschecking if linker accepts -Wreorder... nochecking if compiler uses 
RTTI by default... yeschecking for working long double with more range or 
precision than double... yeschecking if readdir_r has 2 parms... 
nochecking if readdir_r has 3 parms... yeschecking for recvmsg... 
yeschecking if using STL streams... yeschecking if atomic integer 
available... yeschecking if __exchange_and_add is in __gnu_cxx namespace... 
nochecking if Unix semaphores are available... yeschecking for 
pthread_create in -lpthread... nochecking for pthread_create in -lc_r... 
noconfigure: error: must have pthreads![EMAIL PROTECTED] 
pwlib_Mimas_patch2]# 

Has anyone sucessfully installed the latest oh323 
on 1.2 beta running on CENTOS 3.5?

Best Regards,

Phillip

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Beronet app_saynumber-beta-rc1

2005-10-10 Thread Ricardo Poppi


Hi list!

Do anybody has success histories about using the Beronet app_saynumber 
application? For those that are hearing about for the first time, it´s a 
piece of software that enables the use of another language in say_number 
commands in asterisk dialplan or AGI scripts.


Link to download: 
http://www.beronet.com/download/app_saynumber-beta-rc1.tar.gz


I´m trying to compile it in asterisk-1.2.0-beta and 1.0.7 - both kernel 
2.4.31 and 2.6.5-1.358 -  and have the same error. Do anybody has a clue 
for what is going on?


Compilation trial:

c -ggdb -fPIC -I/usr/src/asterisk-1.2.0-beta1/include 
-DAST_CONFIG_DIR=\/etc/asterisk/\   -c -o app_say_number.o 
app_say_number.c

In file included from app_say_number.c:14:
/usr/src/asterisk-1.2.0-beta1/include/asterisk/lock.h: In function 
`ast_mutex_init':
/usr/src/asterisk-1.2.0-beta1/include/asterisk/lock.h:330: error: 
`PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function)
/usr/src/asterisk-1.2.0-beta1/include/asterisk/lock.h:330: error: (Each 
undeclared identifier is reported only once
/usr/src/asterisk-1.2.0-beta1/include/asterisk/lock.h:330: error: for 
each function it appears in.)

app_say_number.c: In function `skel_exec':
app_say_number.c:99: warning: use of cast expressions as lvalues is 
deprecated

app_say_number.c:153: error: structure has no member named `callerid'
make: ** [app_say_number.o] Erro 1


Best regards,

Ricardo Poppi.
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] What is this error? Is there a bug?

2005-10-10 Thread Doug Lytle


Dan Journo wrote:

 
 
Can anyone shed some light on this?



In your sip.conf

auth=md5 was the cause for me.

Doug

--

Ben Franklin quote:

Those who give up essential liberties for temporary safety deserve neither liberty 
nor safety.


___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Help, please help -- IAX2 softphone to server on LAN

2005-10-10 Thread David J Carter

David:

Also port 1:2 is a good idea to forward to the server as well..

Only needed for SIP. 4569 is all that is required for IAX2.

 David,
 Yes, I've also forwarded port 4569 to the server.
 Since the router is forwarding to the server, I cannot
 forward it to the client as well -- however, as the
 client isn't going out past the LAN, it shouldn't
 matter... unless there's something else going on that
 I don't know about.
 Thanks
 Wolfgang

You might try: -

[2278]
type=friend
secret=**
host=dynamic
context=clientcon

 --- David J Carter [EMAIL PROTECTED] wrote:

 Wolfgang wrote:  -

 I've already sunk several hours into this without
 any
 real progress, so I'd really appreciate any help  My
 task is simple -- establish a connection between a
 softphone on XP ProSP2 to a Asterisk server on Linux
 FC4 over a LAN through a Netgear router. The server
 will then go out to a PSTN termination service.

 Thus far, the PSTN termination connection works fine
 -- I've opened up 4569 with iptables, and forwarded
 4569 to the server IP.  I am not, however, having
 any
 luck connecting the softphone to the server.

 I can telnet, ftp, and http to the server, but not
 IAX2. Iaxping times out, registration by Idefisk and
 Firefly also times out.

 The server fails to see the client as well.

 Here's a portion of my iax.conf:

 [client]
 type=friend
 username=client
 secret=**
 host=192.168.1.40
 context=clientcon

 and extensions.conf:

 [clientcon]
 exten = 2278,1,Dial(IAX2/client)



 ==
 You say you have 4569 configured in iptables, what
 about the netgear router?

 Have you port forwarded 4569 there?

 Dave

 ___
 --Bandwidth and Colocation sponsored by Easynews.com
 --

 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com

 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:


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






 __
 Yahoo! Mail - PC Magazine Editors' Choice 2005
 http://mail.yahoo.com
 ___
 --Bandwidth and Colocation sponsored by Easynews.com --

 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users



___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Outgoing quality

2005-10-10 Thread Mojo with Horan Company, LLC

Are you calling from a soft- or hardphone on a network with a high 
amount of latency?  If your (for example) SIP phone can't deliver voice 
packets to asterisk in time for asterisk to put them where they belong 
in the Zap channel, things like this might happen.  Usually the 
interruptions could be described as clicks or crackles.  In this case, 
you could reduce the network traffic by utilizing a codec with a smaller 
bandwidth usage, like g729 or gsm if your phone supports it.


Fabrizio Mazzoni wrote:

I'm having slight problems with outgoing audio quality on Zap channels.
People hear an interrupted voice.

Can anyone help..?

Regards,

Fabrizio Mazzoni
Macron SPA
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users



--
Mojo [EMAIL PROTECTED]
Office Manger, Horan  Company, LLC
(907) 747- x112
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Beronet app_saynumber-beta-rc1

On Mon, Oct 10, 2005 at 04:44:16PM -0300, Ricardo Poppi wrote:
 Hi list!
 
 Do anybody has success histories about using the Beronet app_saynumber 
 application? For those that are hearing about for the first time, it´s a 
 piece of software that enables the use of another language in say_number 
 commands in asterisk dialplan or AGI scripts.
 
 Link to download: 
 http://www.beronet.com/download/app_saynumber-beta-rc1.tar.gz

Haven't examined it, but the files in it are over a year old. Asterisk
does have this functionality built in, and not just as part of an
application. So I figure you shouldn't bother.

Any specific language you have problem with?

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Open Source Content Management System - Joomla

On Mon, Oct 10, 2005 at 11:10:59AM -0400, Kanuri, Seshu (Company IT) wrote:
 There was some discussion in the past about which one is the best
 Content Management System that can be used in conjunction with Asterisk.
 Mambo was supposed to be the best out there under GPL. 

It depends who you ask. There are quite a few of them. 

And how exactly is Asterisk relevant to a CMS? could you give a more
specific example?

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] What is this error? Is there a bug?

2005-10-10 Thread Dan Journo

I thought to add that back in after i removed it because i was having other problems

Any idea why, when i use the IP address for the realm/domain on the SipPhone, it connects ok. But when i use the domain name, it doesnt authenticate?

Thanks
Dan
On 10/10/05, Doug Lytle [EMAIL PROTECTED] wrote:
Dan Journo wrote: Can anyone shed some light on this?In your sip.conf
auth=md5 was the cause for me.Doug--Ben Franklin quote:Those who give up essential liberties for temporary safety deserve neither liberty nor safety.___
--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] TDM02B card difficulties

On Mon, Oct 10, 2005 at 10:27:19AM -0400, Min Qiu wrote:
 Thank you for your respond, please see more detail inline...
 
 Min
 
  -Original Message-
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of 
  Tzafrir Cohen
  Sent: Friday, October 07, 2005 4:57 PM
  To: asterisk-users@lists.digium.com
  Subject: Re: [Asterisk-Users] TDM02B card difficulties
  
  
  On Fri, Oct 07, 2005 at 03:41:43PM -0400, Min Qiu wrote:
   
   Hi all,
   
   I just installed an TDM02B.  My system is a dell pc with
   linux 2.6.12-1.1456_FC4
   asterisk-1.2.0-beta1
   zaptel-1.2.0-beta1
   libpri-1.2.0-beta1
   
   in /etc/zaptel.conf I have (all others are default):
   fxsks=3-4   --- I saw light in the ports
   channels=1-2--- change it to 3-4 has 
  same result
  
  cat /proc/zaptel/1 to see the channel numbers.
 
 [EMAIL PROTECTED] mqiu]# cat /proc/zaptel/1
 Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1
 
1 WCTDM/0/0
2 WCTDM/0/1
3 WCTDM/0/2
4 WCTDM/0/3

None of them has been configured by ztcfg, right?

What's the output of ztcfg -vv

 
  
  Anyway, the last line is incorrect. It should be used in 
  zapata.conf and
  not in zaptel.conf .
 
 The zaptel.conf has channels=... as an example.  Took the line
 out I have:

It shouldn't . 

Could you please post /etc/zaptel.conf and /etc/asterisk/zapata.conf ?

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] TDM400 not working