[Asterisk-Users] H323 config question

[Damian Minkov]

Hi,
We are using the latest CVS HEAD of asterisk and the
h323 channel By Jeremy McNamara (h323).
We want to connect to a voip provider which want from us to
authenticate with username and password.
How can we achieve* *this. If this cannot be done with this channel is it
possible with the others (oh323, ooh323)?
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Re: [Asterisk-Users] Alternative voiceprompts (new subject)

[Olle E. Johansson]

Andreas Sikkema wrote:
There should be other voices worth while...

Give other people the chance
The market is growing...

Be open :)

Asterisk as a product is in no way closed to Allison's voice prompts.
If that was the case, it would be a serious roadblock for international
use.
There are plenty of Asterisk voice prompts in various languages
(and various qualities) out there.

 I'd _love_ a different voice for the default 
 distribution. To my (European) ears Allison 
 is practically incomprehensible.
 

I am sure that if you pay an artist to record all voice prompts and
donate them to the project - and keep them up to date by adding new ones
as needed - they will happily be included in the asterisk-sounds
distribution.

Allisons voice prompts included in the Asterisk distribution is more to
be seen as a sample of a full prompt set and something that is extremely
useful for US companies setting up an in-house PBX. If you set up a full
scale service provider, you will propably want your own voice prompts in
your own language.


/O
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[Asterisk-Users] Asterisk + Voicetronix Card

[Antonio Rabena]

HI,

I'm using asterisk + voicetronix openswitch12 (using fxo). I just 
noticed when I call a pstn number (mobile number), asterisk will 
answer the call first before it actually dials the destination 
number.  Is this normal?




-- Executing Wait(SIP/192.168.1.130-081671b0, 1) in new stack
-- Executing Dial(SIP/192.168.1.130-081671b0, 
vpb/1-12/911) in new stack

  ==  1-12 requested, got: [vpb/1-12]
  == vpb/1-12: Calling 911 on vpb/1-12
  == vpb/1-12: Dial parms for vpb/1-12 1/2000ms/4000ms/4000ms/12ms
  == vpb/1-12: Dial parms for vpb/1-12 tone 7-0
  == vpb/1-12: Dial parms for vpb/1-12 tone 0-1
  == vpb/1-12: Dial parms for vpb/1-12 tone 4-2
  == vpb/1-12: Dial parms for vpb/1-12 tone 7-3
  == vpb/1-12: Dial parms for vpb/1-12 tone 3-4
-- vpb/1-12: VPB Calling 911 [t=12] on vpb/1-12 returned 0
vpb/1-12: chanreads: starting thread
-- Called 1-12/911
-- vpb/1-12 is ringing
  == vpb/1-12: Dialend
-- vpb/1-12 answered SIP/192.168.1.130-081671b0
  == vpb/1-12:Now listening for DTMF
  == vpb/1-12: Starting record mode (codec=0)[AST_FORMAT_SLINEAR:VPB_LINEAR]
vpb/1-12: vpb_write: Starting play mode 
(codec=0)[AST_FORMAT_SLINEAR:VPB_LINEAR]

  == vpb/1-12: Hangup requested
  == vpb/1-12: Ending record mode (1/yes)
  == vpb/1-12: Ending play mode on vpb/1-12


Regards,

Antonio

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Re: [Asterisk-Users] Polycom Buddy Feature

[harry gaillac]

 However, it doesn't work consistently.  Sometimes it
 does, and sometimes 
 it doesn't.  There's a thread on the asterisk-dev
 list titled 
 chan_exosip2 where I am discussing my problems
 with Olle.


Yes i posted the chan_exosip2 thread !

Harry






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Re: [Asterisk-Users] Fritz card usb v2.1 - Capi installation problem

[Elmar Haneke]

As you can see bellow, the node /dev/capi20 exits and permissions
seems to be good (read and write for user and group). Do you have
other ideas to help me to resolve this problem?


Is user asterisk member of group dialout?

Elmar
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[Asterisk-Users] E1 PRI slips on TE410P

[Nir Simionovich - CTO]

Hi All,

  I've recently encountered a very funny problem, which wasn't happening in
the past. I will describe this in detail:

During the past 4 weeks, our production Asterisk box had been experiencing
PRI (E1 lines) slips over and over at random intervals. When digging into
the available information and debug logs, I've noticed HDLC hang-ups,
followed by a complete reset of the line.

  Now, according to the lists and wiki, this is most probably caused by an
IRQ issue between the TE410P and the onboard IDE controller. So, I looked
into /proc/interrupts to find the following:

[EMAIL PROTECTED] root]# cat /proc/interrupts
   CPU0   CPU1
  0:  196927300456IO-APIC-edge  timer
  1:  2  0IO-APIC-edge  keyboard
  2:  0  0  XT-PIC  cascade
  8:  1  0IO-APIC-edge  rtc
 14:6152062 22IO-APIC-edge  ide0
 15:137  1IO-APIC-edge  ide1
 16:  0  0   IO-APIC-level  usb-uhci
 18:  0  0   IO-APIC-level  usb-uhci
 19:  0  0   IO-APIC-level  usb-uhci
 20:   21986877 21   IO-APIC-level  eth0
 54: 1965865547   4505   IO-APIC-level  wct4xxp
NMI:  0  0
LOC:  196918032  196918689
ERR:  0
MIS:  0

  As you can surely see, the wct4xxp driver and the ide0 and ide1 are
totally on different interrupts. The IDE drives are also set to do UDMA2,
as described in many other places.

  The board is a Dual XEON 2.8Ghz Intel board, with 1GB RAM and a single 
IDE 80GB Hard drive. 

  Any input would be highly appreciated.

Regards,
  Nir S

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Re: [Asterisk-Users] Can someone explain the 's' extension

[Matt Riddell]

Umair Bari wrote:
 No, i really dont think so,

we were talking about _. which I think you will find matches o, s,h,i,t :D and
a couple of others.

-- 
Cheers,

Matt Riddell
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Re: [Asterisk-Users] NAT setup

[Matt Riddell]

John Biundo wrote:
 I can't forward 1-2 with my router.  So I used rtp.conf to
 narrow the band of ports down to something like 14000-14030 and
 forwarded those ports  That seems to work fine.
 
 Am I asking for trouble down the line with this approach?

Depends on how many calls and how many rtp streams they have.  I.E. voice+video

-- 
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RE: [Asterisk-Users] ISDN card required

[Lee Archer]

For us it boils down to the card with the less 
hassle. Anyone used this sirrix quad card?

Regards

Lee


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Rob 
LithSent: 14 November 2005 18:24To: Asterisk Users Mailing 
List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] ISDN 
card required
Junghanns is a very good card, Sirrix (www.sirrix.de) also do a good card with its own 
channel driver - saves hassels with BRIstuff needed with Jungahnns. In the end 
its down to personal preference. Sirrix comes in quad version, Junghans in quad 
and octo. RegardsRob
On 11/14/05, Klaus 
Darilion [EMAIL PROTECTED] 
wrote: 
Kristof 
  Hardy wrote: Lee Archer wrote: Can anyone point me 
  in the direction of a quality, works with  Asterisk, BRI 
  card.I need minimum 2 port/4 channel. Ack. 
  Like Mark pointed out, I also used Junghanns.net cards, works fine.Hi 
  Kristof!(sorry for the empty email)Do you use it with asterisk 
  1.2 (CVS)? AFAIK the bristuff package for1.2 is quiet 
  out-of-date.btw: have you ever used chan_misdn from beronet with 
  quadBRI cards? Anyexperiences? 
  regardsklaus___--Bandwidth 
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Re: [Asterisk-Users] E1 PRI slips on TE410P

[George]

Hi Nir,

The wct4xxxp is in a different IRQ than ide0 and ide1 but the issue 
is that you have enabled the APIC (which extends the IRQ table). In 
reality  the card might share IRQ with another device. My advice is 
to disable APIC and then check again what IRQ the card gets. If you 
see that the card shares an IRQ with another device try if its 
possible to disable this device , for example USB.


George

At 10:42 AM 2005-11-15, Nir Simionovich - CTO wrote:

Hi All,

  I've recently encountered a very funny problem, which wasn't happening in
the past. I will describe this in detail:

During the past 4 weeks, our production Asterisk box had been experiencing
PRI (E1 lines) slips over and over at random intervals. When digging into
the available information and debug logs, I've noticed HDLC hang-ups,
followed by a complete reset of the line.

  Now, according to the lists and wiki, this is most probably caused by an
IRQ issue between the TE410P and the onboard IDE controller. So, I looked
into /proc/interrupts to find the following:

[EMAIL PROTECTED] root]# cat /proc/interrupts
   CPU0   CPU1
  0:  196927300456IO-APIC-edge  timer
  1:  2  0IO-APIC-edge  keyboard
  2:  0  0  XT-PIC  cascade
  8:  1  0IO-APIC-edge  rtc
 14:6152062 22IO-APIC-edge  ide0
 15:137  1IO-APIC-edge  ide1
 16:  0  0   IO-APIC-level  usb-uhci
 18:  0  0   IO-APIC-level  usb-uhci
 19:  0  0   IO-APIC-level  usb-uhci
 20:   21986877 21   IO-APIC-level  eth0
 54: 1965865547   4505   IO-APIC-level  wct4xxp
NMI:  0  0
LOC:  196918032  196918689
ERR:  0
MIS:  0

  As you can surely see, the wct4xxp driver and the ide0 and ide1 are
totally on different interrupts. The IDE drives are also set to do UDMA2,
as described in many other places.

  The board is a Dual XEON 2.8Ghz Intel board, with 1GB RAM and a single
IDE 80GB Hard drive.

  Any input would be highly appreciated.

Regards,
  Nir S

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RE: [Asterisk-Users] Fritz card usb v2.1 - Capi installation problem

[Armin Schindler]

In that case the kernel module 'capi.o' may not be loaded, or the isdn card 
driver just does not work.

Armin

On Tue, 15 Nov 2005, amaury BOSSE wrote:
 Hello,
 As you can see bellow, the node /dev/capi20 exits and permissions seems to be 
 good (read and write for user and group).
 Do you have other ideas to help me to resolve this problem?
 Thanks for your answer.
 Amaury
 
 # ls -l /dev/capi*
 crw-rw-rw-  1 root dialout 68,  0 2005-11-10 10:18 /dev/capi20
 crw-rw  1 root dialout 68,  1 2005-11-10 10:18 /dev/capi20.00
 crw-rw  1 root dialout 68,  2 2005-11-10 10:18 /dev/capi20.01
 crw-rw  1 root dialout 68,  3 2005-11-10 10:18 /dev/capi20.02
 crw-rw  1 root dialout 68,  4 2005-11-10 10:18 /dev/capi20.03
 crw-rw  1 root dialout 68,  5 2005-11-10 10:18 /dev/capi20.04
 crw-rw  1 root dialout 68,  6 2005-11-10 10:18 /dev/capi20.05
 crw-rw  1 root dialout 68,  7 2005-11-10 10:18 /dev/capi20.06
 crw-rw  1 root dialout 68,  8 2005-11-10 10:18 /dev/capi20.07
 crw-rw  1 root dialout 68,  9 2005-11-10 10:18 /dev/capi20.08
 crw-rw  1 root dialout 68, 10 2005-11-10 10:18 /dev/capi20.09
 crw-rw  1 root dialout 68, 11 2005-11-10 10:18 /dev/capi20.10
 crw-rw  1 root dialout 68, 12 2005-11-10 10:18 /dev/capi20.11
 crw-rw  1 root dialout 68, 13 2005-11-10 10:18 /dev/capi20.12
 crw-rw  1 root dialout 68, 14 2005-11-10 10:18 /dev/capi20.13
 crw-rw  1 root dialout 68, 15 2005-11-10 10:18 /dev/capi20.14
 crw-rw  1 root dialout 68, 16 2005-11-10 10:18 /dev/capi20.15
 crw-rw  1 root dialout 68, 17 2005-11-10 10:18 /dev/capi20.16
 crw-rw  1 root dialout 68, 18 2005-11-10 10:18 /dev/capi20.17
 crw-rw  1 root dialout 68, 19 2005-11-10 10:18 /dev/capi20.18
 crw-rw  1 root dialout 68, 20 2005-11-10 10:18 /dev/capi20.19
 
 /dev/capi:
 total 0
 
 
 
 -Message d'origine-
 De : Armin Schindler [mailto:[EMAIL PROTECTED] 
 Envoyé : lundi 14 novembre 2005 18:37
 À : Amaury BOSSE
 Cc : asterisk-users@lists.digium.com
 Objet : Re: [Asterisk-Users] Fritz card usb v2.1 - Capi installation problem
 
 If 'capiinfo' does not work, chan_capi will fail too.
 
 Do you have the node /dev/capi20 with correct permissions?
 
 Armin
 
 On Mon, 14 Nov 2005, Amaury BOSSE wrote:
  Hi all,
  
   
  
  I am trying to install an AVM Fritz card USB v2.1 on my Asterisk Box.
  
  I am using Debian Sarge with 2.6.8 kernel.
  
  I have compiled capi last drivers (fcusb2-suse93-3.11-07.tar.gz) and have
  copied fcusb.ko to /lib/modules/2.6.8/extra/.
  
   
  
  All modules seems loaded (capi, capifs, kernelcapi, fcusb,.)
  
  Capiinfo fails and returns : capi not installed - No such device or
  address (6)
  
   
  
  I have tried to install chan_capi-cm-0.6.1.tar.gz but Asterisk no longer
  starts.
  
  /var/log/asterisk/messages returns :
  
  Nov 14 16:40:51 WARNING[4005]: CAPI not installed, CAPI disabled!
  
  Nov 14 16:40:51 WARNING[4005]: chan_capi.so: load_module failed, returning
  -1
  
  Nov 14 16:40:51 WARNING[4005]: Loading module chan_capi.so failed!
  
   
  
  I have tried to find out a solution from the web but without results.
  
  Does someone know where the problem is from?
  
   
  
  Thanks for your help
  
  Amaury
  
  
 
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[Asterisk-Users] SIP = H.323 Terminator

[Abdul Lateef]

Hi all,

I have H.323 Terminator and i want to terminate our
all SIP clients to this terminator, Is it possible to
add H.323 Terminator in Asterisk?

Please give me a little hint os i can start to
configure.





Yours,
Abdul Lateef
Computer Programmer
HATIF COM
Mob: +974 - 5405022
Tel: +974 - 4883068
ICQ: 276994704
YM!: abdul_zu
Fax: +974 - 4883063
Doha Qatar
http://www.hatif.com




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RE: [Asterisk-Users] ISDN card required

[Armin Schindler]

Less hassle? Then I would recommend the Eicon Diva Server Cards.
Active cards with full support for any ISDN line protocol, Modem, Fax and
support with Asterisk via a generic channel driver (chan_capi, tested with 
other cards too).
How less hassle do you need?

Armin

On Tue, 15 Nov 2005, Lee Archer wrote:
 For us it boils down to the card with the less hassle.  Anyone used this
 sirrix quad card?
  
 Regards
  
 Lee
 
 
 
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Rob Lith
 Sent: 14 November 2005 18:24
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] ISDN card required
 
 
 Junghanns is a very good card, Sirrix (www.sirrix.de) also do a good
 card with its own channel driver - saves hassels with BRIstuff needed
 with Jungahnns. In the end its down to personal preference. Sirrix comes
 in quad version, Junghans in quad and octo. 
 
 Regards
 Rob
 
 
 On 11/14/05, Klaus Darilion [EMAIL PROTECTED] wrote: 
 
   Kristof Hardy wrote:
Lee Archer wrote:
   
Can anyone point me in the direction of a quality, works with
 
Asterisk, BRI card.  I need minimum 2 port/4 channel.
   
   
Ack. Like Mark pointed out, I also used Junghanns.net cards,
 works fine.
   
   Hi Kristof!
   
   (sorry for the empty email)
   
   Do you use it with asterisk 1.2 (CVS)? AFAIK the bristuff
 package for
   1.2 is quiet out-of-date.
   
   btw: have you ever used chan_misdn from beronet with quadBRI
 cards? Any
   experiences? 
   
   regards
   klaus
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[Asterisk-Users] ERROR utils.c:509 tvfix:

[yusuf]

These errors just started showing on asterisk cli.  Me setup is a
 pri/e1 card, connected to a philips PBX.
google gave no answers.
Any ideas???


2005-11-15 11:25:58 ERROR[23932]: utils.c:509 tvfix: warning negative
timestamp -267121.-532000
2005-11-15 11:25:58 ERROR[23932]: utils.c:509 tvfix: warning negative
timestamp -267121.-522000
2005-11-15 11:25:58 ERROR[23932]: utils.c:509 tvfix: warning negative
timestamp -267121.-522000
2005-11-15 11:25:58 ERROR[23932]: utils.c:509 tvfix: warning negative
timestamp -267121.-512000
2005-11-15 11:25:58 ERROR[23932]: utils.c:509 tvfix: warning negative
timestamp -267121.-502000
2005-11-15 11:25:58 ERROR[23932]: utils.c:509 tvfix: warning negative
timestamp -267121.-502000
2005-11-15 11:25:58 ERROR[23932]: utils.c:509 tvfix: warning negative
timestamp -267121.-492000
2005-11-15 11:25:58 ERROR[23932]: utils.c:509 tvfix: warning negative
timestamp -267121.-482000
2005-11-15 11:25:58 ERROR[23932]: utils.c:509 tvfix: warning negative
timestamp -267121.-482000
2005-11-15 11:25:58 ERROR[23932]: utils.c:509 tvfix: warning negative
timestamp -267121.-472000
2005-11-15 11:25:58 ERROR[23932]: utils.c:509 tvfix: warning negative
timestamp -267121.-462000
2005-11-15 11:25:58 ERROR[23932]: utils.c:509 tvfix: warning negative
timestamp -267121.-462000
2005-11-15 11:25:58 ERROR[23932]: utils.c:509 tvfix: warning negative
timestamp -267121.-452000
2005-11-15 11:25:58 ERROR[23932]: utils.c:509 tvfix: warning negative
timestamp -267121.-442000
2005-11-15 11:25:58 ERROR[23932]: utils.c:509 tvfix: warning negative
timestamp -267121.-442000
2005-11-15 11:25:58 ERROR[23932]: utils.c:509 tvfix: warning negative
timestamp -267121.-432000
2005-11-15 11:25:58 ERROR[23932]: utils.c:509 tvfix: warning negative
timestamp -267121.-422000
2005-11-15 11:25:58 ERROR[23932]: utils.c:509 tvfix: warning negative
timestamp -267121.-422000


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RE: [Asterisk-Users] Fritz card usb v2.1 - Capi installation problem

[Amaury BOSSÉ]

The user Asterisk is a member of group dialout so he must have access to
/dev/capi20.
The kernel module capi.o seems to be loaded as you can see on lsmod output:

#lsmod
Module  Size  Used by
ipt_MASQUERADE  3968  3 
ipt_state   2304  6 
iptable_nat22828  2 ipt_MASQUERADE
ip_conntrack   32908  3 ipt_MASQUERADE,ipt_state,iptable_nat
iptable_filter  3072  1 
ip_tables  16896  4
ipt_MASQUERADE,ipt_state,iptable_nat,iptable_filter
af_packet  20872  8 
thermal12944  0 
fan 4236  0 
button  6680  0 
processor  13220  1 thermal
ac  5132  0 
ipv6  229764  18 
fcusb 607384  0 
capi   17728  0 
kernelcapi 46624  2 fcusb,capi
capifs  6024  2 capi
evdev   9088  0 
floppy 54992  0 
8139cp 19072  0 
pci_hotplug30640  0 
via_agp 8832  1 
agpgart31784  1 via_agp
uhci_hcd   29328  0 
usbcore   104164  4 fcusb,uhci_hcd
8139too23936  0 
mii 4864  2 8139cp,8139too
sr_mod 15780  0 
psmouse17800  0 
ide_cd 38176  0 
cdrom  35740  2 sr_mod,ide_cd
genrtc  9332  0 
ext3  109672  4 
jbd54552  1 ext3
ide_generic 1664  0 
via82cxxx  12956  1 
unix   26036  47

and dmesg shows that capi and fcusb are started :

capifs: Rev 1.1.2.3
CAPI Subsystem Rev 1.1.2.8
capi20: Rev 1.1.2.7: started up with major 68 (middleware+capifs)
fcusb: module license 'Proprietary' taints kernel.
fcusb: AVM FRITZ!Card USB driver, revision 0.6.4
fcusb: (fcusb built on Nov 10 2005 at 16:06:01)
fcusb: -- 32 bit CAPI driver --
fcusb: Loading...
usbcore: registered new driver fcusb
fcusb: Loaded.

Do you have another method to test if capi.o module is loaded or if it comes
from the isdn card driver.
Thanks.


-Message d'origine-
De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 
Envoyé : mardi 15 novembre 2005 10:14
À : amaury BOSSE
Cc : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : RE: [Asterisk-Users] Fritz card usb v2.1 - Capi installation problem

In that case the kernel module 'capi.o' may not be loaded, or the isdn card 
driver just does not work.

Armin

On Tue, 15 Nov 2005, amaury BOSSE wrote:
 Hello,
 As you can see bellow, the node /dev/capi20 exits and permissions seems to
be good (read and write for user and group).
 Do you have other ideas to help me to resolve this problem?
 Thanks for your answer.
 Amaury
 
 # ls -l /dev/capi*
 crw-rw-rw-  1 root dialout 68,  0 2005-11-10 10:18 /dev/capi20
 crw-rw  1 root dialout 68,  1 2005-11-10 10:18 /dev/capi20.00
 crw-rw  1 root dialout 68,  2 2005-11-10 10:18 /dev/capi20.01
 crw-rw  1 root dialout 68,  3 2005-11-10 10:18 /dev/capi20.02
 crw-rw  1 root dialout 68,  4 2005-11-10 10:18 /dev/capi20.03
 crw-rw  1 root dialout 68,  5 2005-11-10 10:18 /dev/capi20.04
 crw-rw  1 root dialout 68,  6 2005-11-10 10:18 /dev/capi20.05
 crw-rw  1 root dialout 68,  7 2005-11-10 10:18 /dev/capi20.06
 crw-rw  1 root dialout 68,  8 2005-11-10 10:18 /dev/capi20.07
 crw-rw  1 root dialout 68,  9 2005-11-10 10:18 /dev/capi20.08
 crw-rw  1 root dialout 68, 10 2005-11-10 10:18 /dev/capi20.09
 crw-rw  1 root dialout 68, 11 2005-11-10 10:18 /dev/capi20.10
 crw-rw  1 root dialout 68, 12 2005-11-10 10:18 /dev/capi20.11
 crw-rw  1 root dialout 68, 13 2005-11-10 10:18 /dev/capi20.12
 crw-rw  1 root dialout 68, 14 2005-11-10 10:18 /dev/capi20.13
 crw-rw  1 root dialout 68, 15 2005-11-10 10:18 /dev/capi20.14
 crw-rw  1 root dialout 68, 16 2005-11-10 10:18 /dev/capi20.15
 crw-rw  1 root dialout 68, 17 2005-11-10 10:18 /dev/capi20.16
 crw-rw  1 root dialout 68, 18 2005-11-10 10:18 /dev/capi20.17
 crw-rw  1 root dialout 68, 19 2005-11-10 10:18 /dev/capi20.18
 crw-rw  1 root dialout 68, 20 2005-11-10 10:18 /dev/capi20.19
 
 /dev/capi:
 total 0
 
 
 
 -Message d'origine-
 De : Armin Schindler [mailto:[EMAIL PROTECTED] 
 Envoyé : lundi 14 novembre 2005 18:37
 À : Amaury BOSSE
 Cc : asterisk-users@lists.digium.com
 Objet : Re: [Asterisk-Users] Fritz card usb v2.1 - Capi installation
problem
 
 If 'capiinfo' does not work, chan_capi will fail too.
 
 Do you have the node /dev/capi20 with correct permissions?
 
 Armin
 
 On Mon, 14 Nov 2005, Amaury BOSSE wrote:
  Hi all,
  
   
  
  I am trying to install an AVM Fritz card USB v2.1 on my Asterisk Box.
  
  I am using Debian Sarge with 2.6.8 kernel.
  
  I have compiled capi last drivers (fcusb2-suse93-3.11-07.tar.gz) and
have
  copied fcusb.ko to /lib/modules/2.6.8/extra/.
  
   
  
  All modules seems loaded (capi, capifs, kernelcapi, 

Re: [Asterisk-Users] Snom clients deregistering

[Richard Watson]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Richard Watson wrote:

 [888120]
 type=friend
 username=888120
 mailbox=888120
 canreinvite=no
 nat=yes
 secret=secret
 host=dynamic
 qualify=yes
 context=sipdemo
 subscribecontext=sipdemo

Just for fun I had a play yesterday using SER as a stateless proxy
ouside the nat to see if for some reason that hung on to the
registrations. The result was even worse than before.

Current situation - I've removed the Challenge Password Dialog
configuration from the Snoms and they still lose their registration.

I'm quite stumped now - anyone got any idea what to try next?

-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.1 (GNU/Linux)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org

iD8DBQFDea8LP05lUVhVYk0RAkpuAKCAgdcEPxgzqQc9S9jYvHRpQAhWCACcCpuh
crOxBqrTfSwp5dtCm9jJGxs=
=jK4e
-END PGP SIGNATURE-
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RE: [Asterisk-Users] ISDN card required

[Lee Archer]

I could do without playing with multiple versions of drivers trying to
find one that works.  And I could do with not spending days trying to
make the card work with the ISDN lines.  Bascially I could do with a
card which has linux 2.6 drivers, works with Asterix and is documented.

Regards

Lee 

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 
Sent: 15 November 2005 09:26
To: Lee Archer
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] ISDN card required

Less hassle? Then I would recommend the Eicon Diva Server Cards.
Active cards with full support for any ISDN line protocol, Modem, Fax
and support with Asterisk via a generic channel driver (chan_capi,
tested with other cards too).
How less hassle do you need?

Armin

On Tue, 15 Nov 2005, Lee Archer wrote:
 For us it boils down to the card with the less hassle.  Anyone used 
 this sirrix quad card?
  
 Regards
  
 Lee
 
 
 
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Rob Lith
 Sent: 14 November 2005 18:24
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] ISDN card required
 
 
 Junghanns is a very good card, Sirrix (www.sirrix.de) also do a good 
 card with its own channel driver - saves hassels with BRIstuff needed 
 with Jungahnns. In the end its down to personal preference. Sirrix 
 comes in quad version, Junghans in quad and octo.
 
 Regards
 Rob
 
 
 On 11/14/05, Klaus Darilion [EMAIL PROTECTED] wrote: 
 
   Kristof Hardy wrote:
Lee Archer wrote:
   
Can anyone point me in the direction of a quality, works with
 
Asterisk, BRI card.  I need minimum 2 port/4 channel.
   
   
Ack. Like Mark pointed out, I also used Junghanns.net cards,
works 
 fine.
   
   Hi Kristof!
   
   (sorry for the empty email)
   
   Do you use it with asterisk 1.2 (CVS)? AFAIK the bristuff
package for
   1.2 is quiet out-of-date.
   
   btw: have you ever used chan_misdn from beronet with quadBRI
cards? 
 Any
   experiences? 
   
   regards
   klaus
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[Asterisk-Users] remove asterisk?

[Matteo Piazza]

Is there a command to remove completely asterisk?
I want clean the server before the installation of 1.2 version.
Matteo
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RE: [Asterisk-Users] Fritz card usb v2.1 - Capi installation problem

[Armin Schindler]

The capi subsystem seems to be loaded and available. But as you can see in 
the dmesg, the fcusb driver does not register with capi (there should be 
additional messages like Controller 1 added or similar).

I don't know anything about the fcusb stuff, but this seems to be the 
problem.

Armin

On Tue, 15 Nov 2005, Amaury BOSSÉ wrote:
 The user Asterisk is a member of group dialout so he must have access to
 /dev/capi20.
 The kernel module capi.o seems to be loaded as you can see on lsmod output:
 
 #lsmod
 Module  Size  Used by
 ipt_MASQUERADE  3968  3 
 ipt_state   2304  6 
 iptable_nat22828  2 ipt_MASQUERADE
 ip_conntrack   32908  3 ipt_MASQUERADE,ipt_state,iptable_nat
 iptable_filter  3072  1 
 ip_tables  16896  4
 ipt_MASQUERADE,ipt_state,iptable_nat,iptable_filter
 af_packet  20872  8 
 thermal12944  0 
 fan 4236  0 
 button  6680  0 
 processor  13220  1 thermal
 ac  5132  0 
 ipv6  229764  18 
 fcusb 607384  0 
 capi   17728  0 
 kernelcapi 46624  2 fcusb,capi
 capifs  6024  2 capi
 evdev   9088  0 
 floppy 54992  0 
 8139cp 19072  0 
 pci_hotplug30640  0 
 via_agp 8832  1 
 agpgart31784  1 via_agp
 uhci_hcd   29328  0 
 usbcore   104164  4 fcusb,uhci_hcd
 8139too23936  0 
 mii 4864  2 8139cp,8139too
 sr_mod 15780  0 
 psmouse17800  0 
 ide_cd 38176  0 
 cdrom  35740  2 sr_mod,ide_cd
 genrtc  9332  0 
 ext3  109672  4 
 jbd54552  1 ext3
 ide_generic 1664  0 
 via82cxxx  12956  1 
 unix   26036  47
 
 and dmesg shows that capi and fcusb are started :
 
 capifs: Rev 1.1.2.3
 CAPI Subsystem Rev 1.1.2.8
 capi20: Rev 1.1.2.7: started up with major 68 (middleware+capifs)
 fcusb: module license 'Proprietary' taints kernel.
 fcusb: AVM FRITZ!Card USB driver, revision 0.6.4
 fcusb: (fcusb built on Nov 10 2005 at 16:06:01)
 fcusb: -- 32 bit CAPI driver --
 fcusb: Loading...
 usbcore: registered new driver fcusb
 fcusb: Loaded.
 
 Do you have another method to test if capi.o module is loaded or if it comes
 from the isdn card driver.
 Thanks.
 
 
 -Message d'origine-
 De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 
 Envoyé : mardi 15 novembre 2005 10:14
 À : amaury BOSSE
 Cc : Asterisk Users Mailing List - Non-Commercial Discussion
 Objet : RE: [Asterisk-Users] Fritz card usb v2.1 - Capi installation problem
 
 In that case the kernel module 'capi.o' may not be loaded, or the isdn card 
 driver just does not work.
 
 Armin
 
 On Tue, 15 Nov 2005, amaury BOSSE wrote:
  Hello,
  As you can see bellow, the node /dev/capi20 exits and permissions seems to
 be good (read and write for user and group).
  Do you have other ideas to help me to resolve this problem?
  Thanks for your answer.
  Amaury
  
  # ls -l /dev/capi*
  crw-rw-rw-  1 root dialout 68,  0 2005-11-10 10:18 /dev/capi20
  crw-rw  1 root dialout 68,  1 2005-11-10 10:18 /dev/capi20.00
  crw-rw  1 root dialout 68,  2 2005-11-10 10:18 /dev/capi20.01
  crw-rw  1 root dialout 68,  3 2005-11-10 10:18 /dev/capi20.02
  crw-rw  1 root dialout 68,  4 2005-11-10 10:18 /dev/capi20.03
  crw-rw  1 root dialout 68,  5 2005-11-10 10:18 /dev/capi20.04
  crw-rw  1 root dialout 68,  6 2005-11-10 10:18 /dev/capi20.05
  crw-rw  1 root dialout 68,  7 2005-11-10 10:18 /dev/capi20.06
  crw-rw  1 root dialout 68,  8 2005-11-10 10:18 /dev/capi20.07
  crw-rw  1 root dialout 68,  9 2005-11-10 10:18 /dev/capi20.08
  crw-rw  1 root dialout 68, 10 2005-11-10 10:18 /dev/capi20.09
  crw-rw  1 root dialout 68, 11 2005-11-10 10:18 /dev/capi20.10
  crw-rw  1 root dialout 68, 12 2005-11-10 10:18 /dev/capi20.11
  crw-rw  1 root dialout 68, 13 2005-11-10 10:18 /dev/capi20.12
  crw-rw  1 root dialout 68, 14 2005-11-10 10:18 /dev/capi20.13
  crw-rw  1 root dialout 68, 15 2005-11-10 10:18 /dev/capi20.14
  crw-rw  1 root dialout 68, 16 2005-11-10 10:18 /dev/capi20.15
  crw-rw  1 root dialout 68, 17 2005-11-10 10:18 /dev/capi20.16
  crw-rw  1 root dialout 68, 18 2005-11-10 10:18 /dev/capi20.17
  crw-rw  1 root dialout 68, 19 2005-11-10 10:18 /dev/capi20.18
  crw-rw  1 root dialout 68, 20 2005-11-10 10:18 /dev/capi20.19
  
  /dev/capi:
  total 0
  
  
  
  -Message d'origine-
  De : Armin Schindler [mailto:[EMAIL PROTECTED] 
  Envoyé : lundi 14 novembre 2005 18:37
  À : Amaury BOSSE
  Cc : asterisk-users@lists.digium.com
  Objet : Re: [Asterisk-Users] Fritz card usb v2.1 - Capi installation
 problem
  
  If 'capiinfo' does not work, chan_capi will fail 

RE: [Asterisk-Users] ISDN card required

[David Waugh]

If you want to use a precompiled driver, then you need to download the one that 
matches your kernel version exactly. It is not a case of having to try 
different drivers to see which one works.

If you are in doubt, then download the source driver from the website. As long 
as you have the kernel build packages installed for your system, then it is 
just a case of running the ./Build command.

This will then build the driver for your specific kernel. Kernel 2.6.X is 
supported!!

Regarding spending days making the ISDN Line work with the card, it is first 
important to know about your settings of the line. Switch Type, Layer 2 connect 
mode, whether it is point-to-point or point-to-multipoint. 

If you have a QSIG PBX, then there are also some additional parameters that you 
need to know.

Unfortunately, no card is clever enough to work out what sort of ISDN line you 
have connected. This information needs to come from the ISDN Line provider 
itself. I'm sure this is true for all ISDN cards.

As the eicon diva server card has a CAPI 2.0 compliant driver, then using the 
chan_capi from Sourceforge works.

Documentation on the card is provided by the Diva Server for Linux reference 
guide and assistance is also provided via the telephone support line.

In short, the Diva Server range of cards, works with Asterisk, works with 
Kernel 2.6.X as is well documented.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Lee Archer
Sent: 15 November 2005 09:52
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] ISDN card required


I could do without playing with multiple versions of drivers trying to
find one that works.  And I could do with not spending days trying to
make the card work with the ISDN lines.  Bascially I could do with a
card which has linux 2.6 drivers, works with Asterix and is documented.

Regards

Lee 

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 
Sent: 15 November 2005 09:26
To: Lee Archer
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] ISDN card required

Less hassle? Then I would recommend the Eicon Diva Server Cards.
Active cards with full support for any ISDN line protocol, Modem, Fax
and support with Asterisk via a generic channel driver (chan_capi,
tested with other cards too).
How less hassle do you need?

Armin

On Tue, 15 Nov 2005, Lee Archer wrote:
 For us it boils down to the card with the less hassle.  Anyone used 
 this sirrix quad card?
  
 Regards
  
 Lee
 
 
 
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Rob Lith
 Sent: 14 November 2005 18:24
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] ISDN card required
 
 
 Junghanns is a very good card, Sirrix (www.sirrix.de) also do a good 
 card with its own channel driver - saves hassels with BRIstuff needed 
 with Jungahnns. In the end its down to personal preference. Sirrix 
 comes in quad version, Junghans in quad and octo.
 
 Regards
 Rob
 
 
 On 11/14/05, Klaus Darilion [EMAIL PROTECTED] wrote: 
 
   Kristof Hardy wrote:
Lee Archer wrote:
   
Can anyone point me in the direction of a quality, works with
 
Asterisk, BRI card.  I need minimum 2 port/4 channel.
   
   
Ack. Like Mark pointed out, I also used Junghanns.net cards,
works 
 fine.
   
   Hi Kristof!
   
   (sorry for the empty email)
   
   Do you use it with asterisk 1.2 (CVS)? AFAIK the bristuff
package for
   1.2 is quiet out-of-date.
   
   btw: have you ever used chan_misdn from beronet with quadBRI
cards? 
 Any
   experiences? 
   
   regards
   klaus
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RE: [Asterisk-Users] ISDN card required

[Armin Schindler]

On Tue, 15 Nov 2005, Lee Archer wrote:
 I could do without playing with multiple versions of drivers trying to
 find one that works.  And I could do with not spending days trying to
 make the card work with the ISDN lines.

Yes, you could do that, but it is not fun ;-)

 Bascially I could do with a
 card which has linux 2.6 drivers, works with Asterix and is documented.

Well, Diva Server cards from Eicon do have our latest driver in kernel 2.6.
Only if you need support for newer cards (e.g. new PRI or 4PRI), then you 
can use an RPM from Eicon with new driver.
Also, I use 4BRI with Asterisk at home and in our company. I even use it as 
an embedded project (diskless boot and runs completely in RAM, a designated
small server for Asterisk/ISDN jobs only, without Debian,SuSe,etc installed 
and created with ELinOS - www.elinos.com)... So I guess I could say it works.

For 'documented', I cannot offer a book, but samples, README, Wiki and of 
course this mailinglist have everything you need.

Armin
 
 Regards
 
 Lee 
 
 -Original Message-
 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 
 Sent: 15 November 2005 09:26
 To: Lee Archer
 Cc: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] ISDN card required
 
 Less hassle? Then I would recommend the Eicon Diva Server Cards.
 Active cards with full support for any ISDN line protocol, Modem, Fax
 and support with Asterisk via a generic channel driver (chan_capi,
 tested with other cards too).
 How less hassle do you need?
 
 Armin
 
 On Tue, 15 Nov 2005, Lee Archer wrote:
  For us it boils down to the card with the less hassle.  Anyone used 
  this sirrix quad card?
   
  Regards
   
  Lee
  
  
  
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of Rob Lith
  Sent: 14 November 2005 18:24
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [Asterisk-Users] ISDN card required
  
  
  Junghanns is a very good card, Sirrix (www.sirrix.de) also do a good 
  card with its own channel driver - saves hassels with BRIstuff needed 
  with Jungahnns. In the end its down to personal preference. Sirrix 
  comes in quad version, Junghans in quad and octo.
  
  Regards
  Rob
  
  
  On 11/14/05, Klaus Darilion [EMAIL PROTECTED] wrote: 
  
  Kristof Hardy wrote:
   Lee Archer wrote:
  
   Can anyone point me in the direction of a quality, works with
  
   Asterisk, BRI card.  I need minimum 2 port/4 channel.
  
  
   Ack. Like Mark pointed out, I also used Junghanns.net cards,
 works 
  fine.
  
  Hi Kristof!
  
  (sorry for the empty email)
  
  Do you use it with asterisk 1.2 (CVS)? AFAIK the bristuff
 package for
  1.2 is quiet out-of-date.
  
  btw: have you ever used chan_misdn from beronet with quadBRI
 cards? 
  Any
  experiences? 
  
  regards
  klaus
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[Asterisk-Users] A2Billing problems. still.

[John Fraser]

does anybody know what i am doing wrong? help please

gzip: stdin: unexpected end of file
tar: Read 8572 bytes from Open_A2Billing_version_Raccoon.tar.gz
tar: Unexpected EOF in archive
tar: Unexpected EOF in archive
tar: Error is not recoverable: exiting now
asterisk:/usr/src/a2billing#


thanks
john fraser
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RE: [Asterisk-Users] E1 PRI slips on TE410P

[Nir Simionovich - CTO]

Well,

  I've disabled APIC via lilo nolapic argument. However, upon reboot,
/proc/interrupts still shows:

   CPU0   CPU1
  0:  27139  1IO-APIC-edge  timer
  1:  2  0IO-APIC-edge  keyboard
  2:  0  0  XT-PIC  cascade
  8:  1  0IO-APIC-edge  rtc
 14:  45257  2IO-APIC-edge  ide0
 15:137  0IO-APIC-edge  ide1
 16:  0  0   IO-APIC-level  usb-uhci
 18:  0  0   IO-APIC-level  usb-uhci
 19:  0  0   IO-APIC-level  usb-uhci
 20:   2575  0   IO-APIC-level  eth0
 54: 250262  0   IO-APIC-level  wct4xxp
NMI:  0  0
LOC:  27057  27039
ERR:  0
MIS:  0

  Any idea how to disable it completely? 

Nir S 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of George
Sent: Tuesday, November 15, 2005 11:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion; 'Asterisk Users
Mailing List - Non-Commercial Discussion'
Subject: Re: [Asterisk-Users] E1 PRI slips on TE410P

Hi Nir,

The wct4xxxp is in a different IRQ than ide0 and ide1 but the issue is that
you have enabled the APIC (which extends the IRQ table). In reality  the
card might share IRQ with another device. My advice is to disable APIC and
then check again what IRQ the card gets. If you see that the card shares an
IRQ with another device try if its possible to disable this device , for
example USB.

George

At 10:42 AM 2005-11-15, Nir Simionovich - CTO wrote:
Hi All,

   I've recently encountered a very funny problem, which wasn't 
happening in the past. I will describe this in detail:

During the past 4 weeks, our production Asterisk box had been 
experiencing PRI (E1 lines) slips over and over at random intervals. 
When digging into the available information and debug logs, I've 
noticed HDLC hang-ups, followed by a complete reset of the line.

   Now, according to the lists and wiki, this is most probably caused 
by an IRQ issue between the TE410P and the onboard IDE controller. So, 
I looked into /proc/interrupts to find the following:

[EMAIL PROTECTED] root]# cat /proc/interrupts
CPU0   CPU1
   0:  196927300456IO-APIC-edge  timer
   1:  2  0IO-APIC-edge  keyboard
   2:  0  0  XT-PIC  cascade
   8:  1  0IO-APIC-edge  rtc
  14:6152062 22IO-APIC-edge  ide0
  15:137  1IO-APIC-edge  ide1
  16:  0  0   IO-APIC-level  usb-uhci
  18:  0  0   IO-APIC-level  usb-uhci
  19:  0  0   IO-APIC-level  usb-uhci
  20:   21986877 21   IO-APIC-level  eth0
  54: 1965865547   4505   IO-APIC-level  wct4xxp
NMI:  0  0
LOC:  196918032  196918689
ERR:  0
MIS:  0

   As you can surely see, the wct4xxp driver and the ide0 and ide1 are 
totally on different interrupts. The IDE drives are also set to do 
UDMA2, as described in many other places.

   The board is a Dual XEON 2.8Ghz Intel board, with 1GB RAM and a 
single IDE 80GB Hard drive.

   Any input would be highly appreciated.

Regards,
   Nir S

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Re: [Asterisk-Users] A2Billing problems. still.

[Vahan Yerkanian]

John Fraser wrote:

does anybody know what i am doing wrong? help please

gzip: stdin: unexpected end of file
tar: Read 8572 bytes from Open_A2Billing_version_Raccoon.tar.gz
tar: Unexpected EOF in archive
tar: Unexpected EOF in archive
tar: Error is not recoverable: exiting now
asterisk:/usr/src/a2billing#


Sounds like a truncated .tar.gz. Make sure the download finishes 
successfuly. I had no problems downloading the tarball from the website.
begin:vcard
fn:Vahan Yerkanian
n:Yerkanian;Vahan
org:ARMINCO Global Telecommunications;Head, Research  Development dept.
adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia
email;internet:[EMAIL PROTECTED]
x-mozilla-html:FALSE
url:http://www.arminco.com/
version:2.1
end:vcard

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Re: [Asterisk-Users] Alternative voiceprompts (new subject)

[Avi Miller]

Olle E. Johansson wrote:

be seen as a sample of a full prompt set and something that is extremely


This actually leads to a question I've had for a while: Is there a list 
somewhere of all the prompts (by filename) and what is said? I've 
searched the Wiki but haven't found anything. Having a list would be a 
lot easier than transcribing each file. :)


Thanks,
Avi

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[Asterisk-Users] Polycom Softkeys Voicemail Button

[Brian]

I have the Polycom 30x/50x/60x IP Phones running a combination of 1.5.x 
and 1.6.x SIP Software (I'm finishing testing the 1.6.x before I switch 
all phones over to that platform).


I would like to alter some of the softkey options on the phone, such as 
removing Buddies/MyStat buttons, and possibly replacing them with other 
options -- as well as altering the transfer/conference softkey options 
for situations where a call is in progress.  I have read through the 
Admin Guide, but still remain unsure on how to actually make these 
changes -- and what configuration options to use.  Has anyone does this 
before, if so, do you have the configuration sections that need to be 
modified?


Also, I have noticed that when multiple registrations are placed onto a 
Polycom Phone, the phone then presents you with the Message Center 
when you press the Messages button.  Is it possible to configure the 
phone to always place you in the VMB of the first line registration, as 
the phone does if you only have a single registration in progress?


Thanks.

-Brian
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Re: [Asterisk-Users] A2Billing problems. still.

[John Fraser]

Thank you for the reply.  I downloaded again and ran the command
tar -xzvf Open_A2Billing_version_Raccoon.tar.gz

but still getting the same error.  may I ask the url you used for the 
download please?
thank you
john fraser


asterisk:/usr/src/a2billing# wget 
http://www.areski.net/a2billing/Open_A2Billing_version_Raccoon.tar.gz
--11:49:03--  
http://www.areski.net/a2billing/Open_A2Billing_version_Raccoon.tar.gz
   = `Open_A2Billing_version_Raccoon.tar.gz.2'
Resolving www.areski.net... 213.186.33.19
Connecting to www.areski.net[213.186.33.19]:80... connected.
HTTP request sent, awaiting response... 200 OK
Length: 2,559,178 [application/x-tar]

100%
[=
=] 2,559,178 50.67K/sETA 00:00

11:49:47 (58.41 KB/s) - `Open_A2Billing_version_Raccoon.tar.gz.2' saved 
[2559178/2559178]

asterisk:/usr/src/a2billing#







On Tue, 15 Nov 2005 14:35:03 +0400, Vahan Yerkanian wrote
 John Fraser wrote:
  does anybody know what i am doing wrong? help please
  
  gzip: stdin: unexpected end of file
  tar: Read 8572 bytes from Open_A2Billing_version_Raccoon.tar.gz
  tar: Unexpected EOF in archive
  tar: Unexpected EOF in archive
  tar: Error is not recoverable: exiting now
  asterisk:/usr/src/a2billing#
 
 Sounds like a truncated .tar.gz. Make sure the download finishes 
 successfuly. I had no problems downloading the tarball from the website.

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[Asterisk-Users] Cisco 7905 sccp Hold and Message buttons

[Francesco Angi]

Hi all.
Two simple questions about Cisco 7905 on Asterisk using chan_sccp.
1) using both sccp firmware 5.0 and 6.1 I cannot put a call in hold,
because there's no Hold Button at all! Is there a way to configure
buttons? Perhaps through XML?

2) How can I configure Message button to dial my voicemail number (and
not default 8500)? I tried to enter the number into the messagesURL
element in SEPmac_address.cnf.xml, I also put it into sccp.conf, but
pushing Message always dials 8500.

Thank you,
_fangi_
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Re: [Asterisk-Users] Alternative voiceprompts (new subject)

[Tor Setane]

On Tue, 2005-11-15 at 21:42 +1100, Avi Miller wrote:
 Olle E. Johansson wrote:
  be seen as a sample of a full prompt set and something that is extremely
 
 This actually leads to a question I've had for a while: Is there a list 
 somewhere of all the prompts (by filename) and what is said? I've 
 searched the Wiki but haven't found anything. Having a list would be a 
 lot easier than transcribing each file. :)
 
 Thanks,
 Avi
 

I think sounds.txt i the source directory of asterisk, and also
sounds-extra.txt in asterisk-sounds-x directory will tell you what the
soundfiles says.

Regards,
Tor
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Re: [Asterisk-Users] A2Billing problems. no longer

[John Fraser]

figured it out. thanks anyway.

the first dload had stopped at 12 percent.  second dload was named
Open_A2Billing_version_Raccoon.tar.gz.1
ran the command
tar -xzvf Open_A2Billing_version_Raccoon.tar.gz.1
everything good now at least up to this point.

hope this helps the other non linux people out there

john fraser


On Tue, 15 Nov 2005 02:52:46 -0800, John Fraser wrote
 Thank you for the reply.  I downloaded again and ran the command
 tar -xzvf Open_A2Billing_version_Raccoon.tar.gz
 
 but still getting the same error.  may I ask the url you used for 
 the download please? thank you john fraser
 
 asterisk:/usr/src/a2billing# wget 
 http://www.areski.net/a2billing/Open_A2Billing_version_Raccoon.tar.gz
 --11:49:03--  
 http://www.areski.net/a2billing/Open_A2Billing_version_Raccoon.tar.gz
= `Open_A2Billing_version_Raccoon.tar.gz.2'
 Resolving www.areski.net... 213.186.33.19
 Connecting to www.areski.net[213.186.33.19]:80... connected.
 HTTP request sent, awaiting response... 200 OK
 Length: 2,559,178 [application/x-tar]
 
 100%
 
[=
 =] 2,559,178 50.67K/sETA 00:00
 
 11:49:47 (58.41 KB/s) - `Open_A2Billing_version_Raccoon.tar.gz.2' 
 saved [2559178/2559178]
 
 asterisk:/usr/src/a2billing#
 
 On Tue, 15 Nov 2005 14:35:03 +0400, Vahan Yerkanian wrote
  John Fraser wrote:
   does anybody know what i am doing wrong? help please
   
   gzip: stdin: unexpected end of file
   tar: Read 8572 bytes from Open_A2Billing_version_Raccoon.tar.gz
   tar: Unexpected EOF in archive
   tar: Unexpected EOF in archive
   tar: Error is not recoverable: exiting now
   asterisk:/usr/src/a2billing#
  
  Sounds like a truncated .tar.gz. Make sure the download finishes 
  successfuly. I had no problems downloading the tarball from the website.
 
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Re: [Asterisk-Users] remove asterisk?

[Martin Vit]

make uninstall?

Matteo Piazza wrote:

Is there a command to remove completely asterisk?
I want clean the server before the installation of 1.2 version.
Matteo
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RE: [Asterisk-Users] E1 PRI slips on TE410P

[Mark Ackroyd]

   Any idea how to disable it completely?

You can disable it in the BIOS as well.

Mark


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Re: [Asterisk-Users] Cisco 7905 sccp Hold and Message buttons

[Sergio Chersovani]

Francesco Angi ha scritto:


Two simple questions about Cisco 7905 on Asterisk using chan_sccp.
1) using both sccp firmware 5.0 and 6.1 I cannot put a call in hold,
because there's no Hold Button at all! Is there a way to configure
 

The 7905 has an hard button for the hold stuff, the button is the one on 
the top of the button 1



element in SEPmac_address.cnf.xml, I also put it into sccp.conf, but
pushing Message always dials 8500.
 


vmnum = 123456

in the line section

Sergio
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[Asterisk-Users] errors with chan_zap.c when installing asterisk-1.2.0-rc2

[Steven Langley]

Title: errors with chan_zap.c when installing asterisk-1.2.0-rc2






Hi

I am trying to install Asterisk-1.2.0-rc2 on Ubuntu Linux and am getting a lot of errors with chan_zap.c. Most of the errors look like this:

Chan_zap.c:10927: error: dereferencing pointer to incomplete type

I have already successfully installed the same version of Zaptel. Any ideas what the problem could be?

Thanks

Steven


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[Asterisk-Users] usb cellphone

[Alejandro Vargas]

Many cellphones has usb port, and I noticed that there are software
for windows to use it as e soft-phone. It is useless in a workstation,
but I think it could be greet if asterisk can use this feature of the
phones to create a channel for dialing out.

 All I've found is related to bluetooth. Is there a way to do this
(using a cellphone conected through USB port) already included?
--
Alejandro Vargas
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Re: [Asterisk-Users] remove asterisk?

[Lenz]

rm -rf for the following dirs (taken from  
http://www.oinko.net/astrecipes/index.php?n=93 ):


/etc/asterisk
/var/log/asterisk
/var/lib/asterisk
/var/spool/asterisk
/usr/lib/asterisk

any other?
l.



On Tue, 15 Nov 2005 11:02:08 +0100, Matteo Piazza  
[EMAIL PROTECTED] wrote:



Is there a command to remove completely asterisk?
I want clean the server before the installation of 1.2 version.
Matteo




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[Asterisk-Users] Configuring Asterisk Queues using real time(MySQL)

[Bharat M. Sarvan]

Hi all,

 I am configuring
Asterisk Queues using real time( that is using the MySQL DB). I am thru with
having a table for queues as given on the site http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue
. And I have configured the extconfig.conf file as 



Queue = mysql,localpbx,queue_table



Now I was wondering do I need to specify some parameter in
the queues.conf file for things to work out right.



Please let me know am going the right way, to achieve queues
using the real time? And let me know the right way to achieve it









Regards,

Bharat 








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Re: [Asterisk-Users] remove asterisk?

[Matteo Piazza]

Are you sure that with make unistall all asterisk's three is cancelled?


Martin Vit wrote:

make uninstall?

Matteo Piazza wrote:


Is there a command to remove completely asterisk?
I want clean the server before the installation of 1.2 version.
Matteo
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--
===
 Matteo Piazza, Junior Researcher
 CREATE-NET
 Via Solteri, 38 - 38100 Trento - Italy
 email: [EMAIL PROTECTED]
 Tel: +39-0461-408400ext:308
 www.create-net.it
===
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[Asterisk-Users] app_conference compiling for asterisk

[Dominik Simon]

Hi all,

today I download the app_conference from iaxclient-dvs. I edit the  
Makefile to my paths:


INSTALL_PREFIX := /usr
INSTALL_MODULES_DIR := $(INSTALL_PREFIX)/lib/asterisk/modules
ASTERISK_INCLUDE_DIR := $(INSTALL_PREFIX)/src/asterisk-1.2.0-rc2/ 
include/asterisk


and then try make, but I only get the following errors:

[EMAIL PROTECTED] app_conference]# make
gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g  -I/ 
usr/src/asterisk-1.2.0-rc2/include/asterisk  -D_REENTRANT - 
D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop- 
arrays -fsingle-precision-constant -march=pentium3 -msse -mfpmath=sse, 
387  -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2   -c -o  
app_conference.o app_conference.c

In file included from /usr/include/unistd.h:26,
 from /usr/include/asterisk/channel.h:89,
 from /usr/include/asterisk/pbx.h:27,
 from app_conference.h:23,
 from app_conference.c:19:
/usr/src/asterisk-1.2.0-rc2/include/asterisk/features.h:44: Fehler:  
syntax error before »AST_LIST_ENTRY«

In file included from /usr/include/bits/types.h:31,
 from /usr/include/unistd.h:186,
 from /usr/include/asterisk/channel.h:89,
 from /usr/include/asterisk/pbx.h:27,
 from app_conference.h:23,
 from app_conference.c:19:
/usr/lib/gcc/i386-redhat-linux/4.0.1/include/stddef.h:214: Fehler:  
syntax error before »typedef«

In file included from /usr/include/asterisk/channel.h:89,
 from /usr/include/asterisk/pbx.h:27,
 from app_conference.h:23,
 from app_conference.c:19:
/usr/include/unistd.h:256: Fehler: syntax error before »__THROW«
/usr/include/unistd.h:287: Fehler: syntax error before »__THROW«
/usr/include/unistd.h:313: Fehler: syntax error before »__wur«
/usr/include/unistd.h:319: Fehler: syntax error before »__wur«
/usr/include/unistd.h:370: Fehler: syntax error before »__THROW«
/usr/include/unistd.h:379: Fehler: syntax error before »__THROW«
/usr/include/unistd.h:420: Fehler: syntax error before »__THROW«
/usr/include/unistd.h:435: Fehler: syntax error before »__THROW«
/usr/include/unistd.h:449: Fehler: syntax error before »__THROW«
/usr/include/unistd.h:468: Fehler: syntax error before »__THROW«
/usr/include/unistd.h:471: Fehler: syntax error before »__THROW«
/usr/include/unistd.h:483: Fehler: syntax error before »__THROW«
/usr/include/unistd.h:495: Fehler: syntax error before »__THROW«
/usr/include/unistd.h:500: Fehler: syntax error before »__THROW«
/usr/include/unistd.h:505: Fehler: syntax error before »__THROW«
/usr/include/unistd.h:510: Fehler: syntax error before »__THROW«
/usr/include/unistd.h:516: Fehler: syntax error before »__THROW«
In file included from /usr/include/asterisk/channel.h:89,
 from /usr/include/asterisk/pbx.h:27,
 from app_conference.h:23,
 from app_conference.c:19:
/usr/include/unistd.h:536: Fehler: syntax error before »__THROW«
/usr/include/unistd.h:539: Fehler: syntax error before »__THROW«
/usr/include/unistd.h:542: Fehler: syntax error before »__THROW«
/usr/include/unistd.h:551: Fehler: syntax error before »__THROW«
/usr/include/unistd.h:554: Fehler: syntax error before »__THROW«
/usr/include/unistd.h:559: Fehler: syntax error before »__THROW«
/usr/include/unistd.h:569: Fehler: syntax error before »__THROW«
/usr/include/unistd.h:578: Fehler: syntax error before »__THROW«
/usr/include/unistd.h:612: Fehler: syntax error before »__THROW«
/usr/include/unistd.h:620: Fehler: syntax error before »__THROW«
/usr/include/unistd.h:623: Fehler: syntax error before »__THROW«



Can anybody help? I tried different options, but I dont find the  
mistake


Best regards and many thanks
Dominik Simon
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[Asterisk-Users] Re: remove asterisk?

[Tony Mountifield]

In article [EMAIL PROTECTED],
Lenz [EMAIL PROTECTED] wrote:
 
 rm -rf for the following dirs (taken from  
 http://www.oinko.net/astrecipes/index.php?n=93 ):
 
 /etc/asterisk
 /var/log/asterisk
 /var/lib/asterisk
 /var/spool/asterisk
 /usr/lib/asterisk
 
 any other?

/usr/sbin/asterisk
/usr/sbin/safe_asterisk
/etc/rc.d/init.d/asterisk
/etc/rc.d/rc*.d/*asterisk

There might be zaptel stuff too.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[Asterisk-Users] Monthly tips for the community?

[Matt Riddell]

Is it just me or have the monthly tips from Olle stopped.  I just opened my
mail client and the last few posts were about 80% HTML.

Please Olle if you already posted it this month, can you step it up to once
every couple of weeks!

:)

hehe

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RE: [Asterisk-Users] voicemial maxmsg

[Bharat M. Sarvan]

Hi Joseph,




The parameter that you have specified works pretty well. I checked it putting
it in the [general] context of the voicemail.conf. It worked pretty nicely on voicemail
configured on real time as well. Where exactly are you facing the problem? Let
me know.









Regards,

Bharat 











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joseph Rothstein
Sent: Tuesday, November 15, 2005
3:13 PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
voicemial maxmsg





Has anyone tested the maxmsg parameter in the voicemail.conf
file? I am trying to restrict the number of messages for each mailbox, but I
cant seem to get this parameter to have any effect. I also could not
find a single reference to this parameter on the wiki.



If anyone has gotten this to work, or know of another way to
restrict the number of allowable messages I would sincerely appreciate the
help.



Regards to all,

Joe 












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Re: [Asterisk-Users] open asterisk?

[don vanfossen]

No...I don't...because if you want to defend someone
you do it somewhere 
other then this type of list...

- Original Message - 
From: Paul [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
Discussion 
asterisk-users@lists.digium.com
Sent: Monday, November 14, 2005 11:05 PM
Subject: Re: [Asterisk-Users] open asterisk?


 Oh, you don't think it's okay for a few of us to say
things in favor of
 someone when she is slandered?

 If you think it's crap why is your mailbox full of
it? Just delete it
 and stop whining.

 don vanfossen wrote:

Heh...who really cares about this topic?
I have a mailbox full of crap about who is gonna
make
recordings where...



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RE: [Asterisk-Users] E1 PRI slips on TE410P

[Nir Simionovich - CTO]

Hey Mark,

  Looks like I'll be heading your way soon in that case ;-)

Nir S 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Ackroyd
Sent: Tuesday, November 15, 2005 1:11 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] E1 PRI slips on TE410P


   Any idea how to disable it completely?

You can disable it in the BIOS as well.

Mark


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Re: [Asterisk-Users] Asterisk hobby box

[Roman]

On Tuesday 15 November 2005 09:30, Dmitry Ivanov wrote:
 On Tuesday 15 November 2005 06:26, [EMAIL PROTECTED] wrote:
  Hi. I'm setting up an Asterisk hobby box for me to play around with.
  Is it possible to use a regular 56k modem and a regular home phone
  for it?

 Yes, but forget G.711.

Well, actually, no.
Some modems (with Intel and Motorola chipsets) will work as FXO with zaptel 
driver but most of them won't. FXS ports are pretty expensive -- it's cheaper 
to by an ATA (f.e. Grandstream) and use it with regular phone.
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Re: [Asterisk-Users] SIP = H.323 Terminator

[Martin Vit]

i would recomend this channel for h323:
http://www.inaccessnetworks.com/projects/asterisk-oh323

Abdul Lateef wrote:

Hi all,

I have H.323 Terminator and i want to terminate our
all SIP clients to this terminator, Is it possible to
add H.323 Terminator in Asterisk?

Please give me a little hint os i can start to
configure.





Yours,
Abdul Lateef
Computer Programmer
HATIF COM
Mob: +974 - 5405022
Tel: +974 - 4883068
ICQ: 276994704
YM!: abdul_zu
Fax: +974 - 4883063
Doha Qatar
http://www.hatif.com




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[Asterisk-Users] Asterisk and Agents

[mbodbg]

Hallo all,

I've a question regarding the agent concept of asterisk. If I login an agent
(using AgentLogin), this agent is directly ready to receive calls.

From the most other ACD-systems I know that an agent first logs into the
system and then has to set himself ready to receive calls. So the most
common agent states are login, ready, not ready, wrapup and logoff.

How is ready/notready/wrapup implemented in asterisk?

Thanks and Regards

Markus





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Re: [Asterisk-Users] voicemial maxmsg

[Leif Neland]

 Original Message 
From: Joseph Rothstein [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, November 15, 2005 10:42 AM
Subject: [Asterisk-Users] voicemial maxmsg


Has anyone tested the maxmsg parameter in the voicemail.conf file? I
am trying to restrict the number of messages for each mailbox, but I
can't seem to get this parameter to have any effect. I also could
not find a single reference to this parameter on the wiki.



If anyone has gotten this to work, or know of another way to
restrict the number of allowable messages I would sincerely
appreciate the help.



Try putting a silly value like -1, then asterisk should complain:

Invalid number of messages per folder maxmsg=%s. Using default value %i\n, 
value, MAXMSG


If it doesn't complain asterisk isn't reading your value

The default and max is:
#define MAXMSG 100
#define MAXMSGLIMIT 

Leif

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[Asterisk-Users] bug in asterisk 1.2.0.rc2

[Thomas Hoellriegel]

Hi, i found 2 bugs in asterisk 1.2.0rc1.
I using debian stable.
I start asterisk with:
/usr/sbin/asterisk -U thomas
or an different user,
Asterisk is starting. 
Autodialing are Ignored. 
(/var/spool/asterisk/outgoing).
Asterisk ignore to dial a Number / Extension, automaticlly. 

When i start asterisk 
as: su thomas
/usr/sbin/asterisk
All autodialings  are going on.

bug 2 in meetme

i press * and hear the prompt, the prompt is playing and i press 8 to 
exit,
i can.t hear any poeple in the conference. 
all poeple can hear me.


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[Asterisk-Users] Multiple Outbound SIP Trunks

[Pikoro]

Here is my situation:

I have an office with around 10 users.  Inbound calls will come in via 4 
PSTN lines.  Outbound calls will be routed across  a maximum of 10 SIP 
trunks.


How can I set up a group of outbound trunks which will rotate use 
dependant on how many outbound calls need to be made.


There will be no discrimination or routes based on outbound calling, 
like a certain trunk for international calls, another for local calls, 
etc... Only a group of 10 SIP trunks to be rotated for all outbound calls.


For example:

Customer Support person 1 makes an outbound call on trunk1 (selected 
randomly by asterisk).  Tech support person 1 needs to make an outbound 
call but for some reason is getting routed to trunk1 instead of to the 
next available open SIP trunk.


Can anyone offer any suggestions, links, websites, or conf files that I 
could refer to in order to make all of this work.


Thanks in advance.

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Re: [Asterisk-Users] Multiple Outbound SIP Trunks

[steve]

On Tue, 15 Nov 2005, Pikoro wrote:

 There will be no discrimination or routes based on outbound calling, 
 like a certain trunk for international calls, another for local calls, 
 etc... Only a group of 10 SIP trunks to be rotated for all outbound calls.


Can you explain what you mean by a SIP trunk?

SIP just has addresses - sometimes slightly hidden away in sip.conf behind 
a SIP peer.  So if you Dial(SIP/remotehost/number), a SIP invite is sent 
to the host IP address defined in the SIP peer in sip.conf.  If you 
Dial(SIP/[EMAIL PROTECTED]) then the invite is sent to the host hostname.  
Normally it makes no difference to either side how many other calls may 
already by in progress between the two sides.

Steve
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Re: [Asterisk-Users] Multiple Outbound SIP Trunks

[asterisk]

 On Tue, 15 Nov 2005, Pikoro wrote:

  There will be no discrimination or routes based on outbound calling,
  like a certain trunk for international calls, another for local calls,
  etc... Only a group of 10 SIP trunks to be rotated for all outbound
calls.


 Can you explain what you mean by a SIP trunk?


I took it to mean different accounts or providers.

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[Asterisk-Users] Multiple emails

[don vanfossen]

Anyoen else getting multiple copies of each email now?





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Re: [Asterisk-Users] Multiple emails

[don vanfossen]

Seems to have stopped now...wow was getting like 10
copies of every email to 
the group hehe.

- Original Message - 
From: don vanfossen [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
Discussion 
asterisk-users@lists.digium.com
Sent: Tuesday, November 15, 2005 8:34 AM
Subject: [Asterisk-Users] Multiple emails


 Anyoen else getting multiple copies of each email
now?





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[Asterisk-Users] Queue Callback

[Tyler]

Hello,

Does anyone have any information on configuring app_icd (or know of any
way to do it with the dialplan) that would allow a user holding in a
queue to hang up, and have the system call them back when their place in
line comes up next?  

I can (obviously) allow them to '0' out to voicemail or something, but I
can only find vague references to app_icd and 'OrderlyQ' for doing what
I want to do...

Anyone?

Bueller? ;-)

Thanks

tf.



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Re: [Asterisk-Users] Polycom Softkeys Voicemail Button

[Matthew T. O'Connor]

Brian wrote:
I would like to alter some of the softkey options on the phone, such 
as removing Buddies/MyStat buttons, and possibly replacing them with 
other options -- as well as altering the transfer/conference softkey 
options for situations where a call is in progress.  I have read 
through the Admin Guide, but still remain unsure on how to actually 
make these changes -- and what configuration options to use.  Has 
anyone does this before, if so, do you have the configuration sections 
that need to be modified?


As far as I can tell, nobody has figured this out.  There are however 
many people on this list who would be VERY interested in getting these 
questions answered.


Also, I have noticed that when multiple registrations are placed onto 
a Polycom Phone, the phone then presents you with the Message Center 
when you press the Messages button.  Is it possible to configure the 
phone to always place you in the VMB of the first line registration, 
as the phone does if you only have a single registration in progress?


I don't know, I only use one single registration.
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Re: [Asterisk-Users] Multiple emails

[[EMAIL PROTECTED]]

Some email servers do that from time to time. They seem to believ that 
send towards an address fail and try again. It was probably you own 
server that did not respond properly for a short time.


Jan
don vanfossen wrote:


Seems to have stopped now...wow was getting like 10
copies of every email to 
the group hehe.


- Original Message - 
From: don vanfossen [EMAIL PROTECTED]

To: Asterisk Users Mailing List - Non-Commercial
Discussion 
asterisk-users@lists.digium.com

Sent: Tuesday, November 15, 2005 8:34 AM
Subject: [Asterisk-Users] Multiple emails


 


Anyoen else getting multiple copies of each email
   


now?
 





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Re: [Asterisk-Users] Multiple emails

[Andrew Kohlsmith]

On Tuesday 15 November 2005 08:34, don vanfossen wrote:
 Anyoen else getting multiple copies of each email now?

What?

-A.
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Re: [Asterisk-Users] Multiple emails

[Andrew Kohlsmith]

On Tuesday 15 November 2005 08:34, don vanfossen wrote:
 Anyoen else getting multiple copies of each email now?

What? 

-A.
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Re: [Asterisk-Users] Multiple emails

[Andrew Kohlsmith]

On Tuesday 15 November 2005 08:34, don vanfossen wrote:
 Anyoen else getting multiple copies of each email now?

What?  :-)

(yes there must be something funny in the coffee this morning)

-A.
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Re: [Asterisk-Users] SIP = H.323 Terminator

[Reli Loin]

hello,

I using asterisk-oh323

i have a problem for send dtmf

see log error:

 reason 24 (Call ended with Q.931  cause [28 - Invalid number format])

thanks for you help


2005/11/15, Martin Vit [EMAIL PROTECTED]:
 i would recomend this channel for h323:
 http://www.inaccessnetworks.com/projects/asterisk-oh323

 Abdul Lateef wrote:
  Hi all,
 
  I have H.323 Terminator and i want to terminate our
  all SIP clients to this terminator, Is it possible to
  add H.323 Terminator in Asterisk?
 
  Please give me a little hint os i can start to
  configure.
 
 
 
 
  
  Yours,
  Abdul Lateef
  Computer Programmer
  HATIF COM
  Mob: +974 - 5405022
  Tel: +974 - 4883068
  ICQ: 276994704
  YM!: abdul_zu
  Fax: +974 - 4883063
  Doha Qatar
  http://www.hatif.com
 
 
 
 
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Re: [Asterisk-Users] Multiple Outbound SIP Trunks

[Leif Neland]

 Original Message 
From: [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com Sent: Tuesday, November 15, 2005 2:27
PM Subject: Re: [Asterisk-Users] Multiple Outbound SIP Trunks


On Tue, 15 Nov 2005, Pikoro wrote:


There will be no discrimination or routes based on outbound calling,
like a certain trunk for international calls, another for local
calls, etc... Only a group of 10 SIP trunks to be rotated for all
outbound calls. 



Can you explain what you mean by a SIP trunk?

SIP just has addresses - sometimes slightly hidden away in sip.conf
behind a SIP peer.  So if you Dial(SIP/remotehost/number), a SIP
invite is sent to the host IP address defined in the SIP peer in
sip.conf.  If you Dial(SIP/[EMAIL PROTECTED]) then the invite is sent
to the host hostname. Normally it makes no difference to either
side how many other calls may already by in progress between the two
sides. 


Some providers allow only one outgoing call at a time.

Leif

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[Asterisk-Users] RE: open asterisk?

[Aidan Van Dyk]

[EMAIL PROTECTED] wrote:

 As for the people who would suspect digium is strong-arming anyone,
 hell, if it weren't for them you wouldn't have asterisk would you?  And
 therefore probably no openpbx either, and we all would be spending
 thousands to do what asterisk can do for free.

And if it weren't for the community of indentured slaves and testers, where
would Digium be, with no users, contributors, or bug-reporters?  Never mind
the no millions of dollars of hardware sales.  It *should* be a symbiotic
relationship.



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Re: [Asterisk-Users] Mixmonitor

[Brian Roy]

On 11/14/05, BJ Weschke [EMAIL PROTECTED] wrote:
There is a known issue right now where using mixmonitor withchan_local is going to cause an unintentional disconnect. Are you
using Local/ with this setup?


BJ,

Thanks for the response. No, I've got nothing going though chan/local at all. It's a real straigh-forward zap to sip bridge. Nothing fancy. I'm going to try and route my calls over to another box via iax today and see if that makes any difference. The mixmonitor will be looking at sip to iax then.


Let me know if you think I should file a bug on this.

-Brian

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Re: [Asterisk-Users] RE: open asterisk?

[Andrew Kohlsmith]

On Tuesday 15 November 2005 08:51, Aidan Van Dyk wrote:
 And if it weren't for the community of indentured slaves and testers, where
 would Digium be, with no users, contributors, or bug-reporters?  Never mind
 the no millions of dollars of hardware sales.  It *should* be a symbiotic
 relationship.

I'm sorry, indentured slaves?  Were you chained to your cubicle wall and 
forced to use Asterisk?

-A.
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[Asterisk-Users] Cisco 7960 Multiple Line Appearance

[Matt Hoskins]

Hello:

I'm currently using asterisk 1.0.7 with cisco 7960 SIP 7.5 phones.  I'd 
like to use this phone for a receptionist so that she can take calls for 
 4 other people.  Is this possible?


I tried setting buttons 2 - 5 to register with the extensions of the 
people she'd be taking calls from, but then the other people couldn't 
receive calls.  I'm assuming this is because only one phone can register 
with the same extension.


I also tried ringing multiple channels with the extension, ie:

1234 = 1,Dial(SIP/1234SIP/4321)

This led to the intended result (both phones rang), but it did not show 
which extension was being called.  (IE: on the phone display, it should 
that I was getting a call from 5551212, but did not say who 5551212 was 
calling)


Is there any way to do this with SIP and the 7960?  I've seen the 7914 
but then I'd have to use SCCP and I'm not sure if it is stable enough 
for production use.


Thank you for your time,

Matt Hoskins

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[Asterisk-Users] Possible bug in agent monitoring

[Julian Lyndon-Smith]

using CVS-HEAD (CVS-D2005.10.28.07.54.07-11/13/05-08:33:54)

We monitor (record) all inbound calls to our queues, using 
recordagentcalls=yes and recordformat=gsm in the agents.conf file.


If a call comes in to a queue, and is answered by an agent (let's say 
6001) then I have a recording for agent-6001-xxx-yyy.gsm.
if the agent wants to transfer the call to another agent (an attended 
xfer), then the recording is terminated at the exact time the inbound 
call is transferred to the second agent.


Anyone seen this ?

Julian
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[Asterisk-Users] canreinvite=yes

[Trond Andersen]

Hi,

Just one question.  The documentation I have seen says that the RTP
audio stream is routed directly(if allowed ...), but never anything
about video streams? Is this just because documents are pre 1.2 or is it
true that audio can go directly, but video must pass through Asterisk?

Anyone?

Does anyone have experience with H263 on the 1.2.rc1 version? I think
there is a bug, and will trace and submit it to Bugzilla..??


Trond

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RE: [Asterisk-Users] Polycom Softkeys Voicemail Button

[Sean Cook]

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Matthew T. O'Connor
 Sent: Tuesday, November 15, 2005 8:41 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Polycom Softkeys  Voicemail Button
 
 Brian wrote:
  I would like to alter some of the softkey options on the phone, such
  as removing Buddies/MyStat buttons, and possibly replacing them with
  other options -- as well as altering the transfer/conference softkey
  options for situations where a call is in progress.  I have read
  through the Admin Guide, but still remain unsure on how to actually
  make these changes -- and what configuration options to use.  Has
  anyone does this before, if so, do you have the configuration sections
  that need to be modified?
 
 As far as I can tell, nobody has figured this out.  There are however
 many people on this list who would be VERY interested in getting these
 questions answered.

Not so... 
They do require ftp or tftp configuration:

To remove the Buddies/MyStat soft buttons set both enabled flags here to 0
feature feature.1.name=presence feature.1.enabled=0
feature.2.name=messaging feature.2.enabled=0


To Change the Messages Button for 1 touch dial:

msg msg.bypassInstantMessage=1
  mwi msg.mwi.1.subscribe= msg.mwi.1.callBackMode=contact
msg.mwi.1.callBack=4999 msg.mwi.2.subscrib
e= msg.mwi.2.callBackMode=disabled msg.mwi.2.callBack=
msg.mwi.3.subscribe= msg.mwi.3.callBackMode=di
sabled msg.mwi.3.callBack= msg.mwi.4.subscribe=
msg.mwi.4.callBackMode=disabled msg.mwi.4.callBack= m
sg.mwi.5.subscribe= msg.mwi.5.callBackMode=disabled
msg.mwi.5.callBack= msg.mwi.6.subscribe= msg.mwi.6
.callBackMode=disabled msg.mwi.6.callBack=/
   /msg



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Re: [Asterisk-Users] Multiple Outbound SIP Trunks

[Pikoro]

By "trunk" I mean each trunk is a different account on the same SIP
provider. Yes, they only allow one call per account. We are an
internet provider so I can obtain as many trunks(accounts) as I need.

Cheers


asterisk wrote:

  
  
  

On Tue, 15 Nov 2005, Pikoro wrote:



  There will be no discrimination or routes based on outbound calling,
like a certain trunk for international calls, another for local calls,
etc... Only a group of 10 SIP trunks to be rotated for all outbound
  

  
  calls.
  
  

Can you explain what you mean by a "SIP trunk"?


  
  
I took it to mean different accounts or providers.

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Re: [Asterisk-Users] Cisco 7960 Multiple Line Appearance

[Sergio Chersovani]

Matt Hoskins ha scritto:

I'm currently using asterisk 1.0.7 with cisco 7960 SIP 7.5 phones.  
I'd like to use this phone for a receptionist so that she can take 
calls for  4 other people.  Is this possible?


The SIP firmware does not support it.
You have to use SCCP to do that

Is there any way to do this with SIP and the 7960?  I've seen the 7914 
but then I'd have to use SCCP and I'm not sure if it is stable enough 
for production use.


Well give it a chance :-)

http://chan-sccp.berlios.de

Sergio
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[Asterisk-Users] speech to text for only digits

[Jerry Geis]

Is there a speech to text app in asterisk that only looks/listens
for digits to be spoken?

Not a full any word speech to text - just the digits. 0 - 9?

Thanks,

Jerry
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[Asterisk-Users] Message waiting notification

[Sixto Diaz]

Hi i want to notify a user that he has an 
unreadvoicemail waiting to be read.
How can i do this for sip users?

THanks 
Sixto
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Re: [Asterisk-Users] Alternative voiceprompts (new subject)

[Olle E. Johansson]

Avi Miller wrote:
 Olle E. Johansson wrote:
 
 be seen as a sample of a full prompt set and something that is extremely
 
 
 This actually leads to a question I've had for a while: Is there a list
 somewhere of all the prompts (by filename) and what is said? I've
 searched the Wiki but haven't found anything. Having a list would be a
 lot easier than transcribing each file. :)
 
http://www.asterisk.org/doxygen/SoundFiles.html

It's included in the source file distribution!

Reading the docs is a good thing. ;-)

/O
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Re: [Asterisk-Users] Monthly tips for the community?

[Olle E. Johansson]

Matt Riddell wrote:
 Is it just me or have the monthly tips from Olle stopped.  I just opened my
 mail client and the last few posts were about 80% HTML.
 
 Please Olle if you already posted it this month, can you step it up to once
 every couple of weeks!
 
Well, the monthly tip of this month is:

**
***   
*** PLEASE TEST THE 1.2 RELEASE CANDIDATE 
***   
**

Thank you for reminding me Matt, I'll start planning for next month.

/Olle
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[Asterisk-Users] Re: Multiple Outbound SIP Trunks

[Steven]

My question is the same/similar.

I want to test a VOIP provider.
I only want one LD call to that provider at a time so that I can check with 
the users on the quality, etc.

I want the first LD call to go to the VOIP provider, if one session to that 
provider is in use, I want to use ZAP for any additional LD calls.
Preferably I want to be able to change it from 1 session in use to 2, then 3 
etc. until I reach a level of quality vs. savings.
If I switch over completely, then the day that we make 15 simultaneous LD 
calls will ruin our quality.

Zaptel seems to have this functionality built in, where in a group of 5 
trunks, asterisk will use the next unused trunk.
But SIP and IAX do not seem to get tagged as in use as far as I can see.

-- 
-- 
Steven

May you have the peace and freedom that come from abandoning all hope of 
having a better past.
----  ---  - - -   -- -   -   --  - - - --- - --   - 
 - --- - - -- -  -- --   -   --
Pikoro [EMAIL PROTECTED] wrote in message 
news:[EMAIL PROTECTED]
 Here is my situation:

 I have an office with around 10 users.  Inbound calls will come in via 4 
 PSTN lines.  Outbound calls will be routed across  a maximum of 10 SIP 
 trunks.

 How can I set up a group of outbound trunks which will rotate use 
 dependant on how many outbound calls need to be made.

 There will be no discrimination or routes based on outbound calling, like 
 a certain trunk for international calls, another for local calls, etc... 
 Only a group of 10 SIP trunks to be rotated for all outbound calls.

 For example:

 Customer Support person 1 makes an outbound call on trunk1 (selected 
 randomly by asterisk).  Tech support person 1 needs to make an outbound 
 call but for some reason is getting routed to trunk1 instead of to the 
 next available open SIP trunk.

 Can anyone offer any suggestions, links, websites, or conf files that I 
 could refer to in order to make all of this work.

 Thanks in advance.

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Re: [Asterisk-Users] Intel Desktop MotherBoards Unsuitable for DigiumBoards

[Kevin Hanson]

Piotr A. Sygula wrote:


Having an issue with shared interrupts with a setup with 3 TDM400P cards,
and dealing with Digium support, I'd like to share with the list the fact
that Digium claims the following:

The following output from lspci -vb (shows IRQ from PCI-bus perspective,
rather than the APIC perspective) shows one of your Digium cards sharing
with another device on the system.

:01:02.0 Communication controller: Tiger Jet Network Inc. Tiger3XX
Modem/ISDN interface
   Subsystem: Unknown device b119:0003
   Flags: bus master, medium devsel, latency 64, IRQ 5
   I/O ports at de00
   Memory at feafe000 (32-bit, non-prefetchable)
   Capabilities: [40] Power Management version 2

:00:1f.3 SMBus: Intel Corp. 82801EB/ER (ICH5/ICH5R) SMBus Controller
(rev 02)
   Subsystem: Dell: Unknown device 019d
   Flags: medium devsel, IRQ 5
   I/O ports at efe0



I.e. although APIC is splitting up IRQ's rather nicely, the tech support guy
is saying that it doesn't matter what the APIC layer says.  Would someone
out there break the tie?  I'd like an educated opinion/statement on
whether APIC support solves the IRQ sharing issue, or simply masks it.

What this boils down to for me is an elemental issue; either the TDM400P
cards are just flat out crap, and Digium is using any excuse in the book to
keep saps like me hoping that the problem can be fixed by getting another
motherboard, or, APIC shmapic, an IRQ sharing issue is an IRQ sharing issue.

Anyone care to comment???

 

If what you say is true, then I'm hosed.  I've got six things sharing 
IRQ 255 according to lspci -vb:


Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) USB UHCI 
Controller #1
Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) USB UHCI 
Controller #2
Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) USB UHCI 
Controller #3

Intel Corporation 82801DB/DBM (ICH4/ICH4-M) USB2 EHCI Controller
Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) SMBus Controller
Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface

Is this what is causing all my echo?

I'll try disabling USB in bios and see what happens.

Cheers,
Kevin
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RE: [Asterisk-Users] Message waiting notification

[Jonathan k. Creasy]

mailbox= in the sip.conf











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sixto Diaz
Sent: Tuesday, November 15, 2005
9:33 AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Message
waiting notification







Hi i want to notify a user that he has an
unreadvoicemail waiting to be read.





How can i do this for sip users?











THanks 





Sixto








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Re: [Asterisk-Users] Intel Desktop MotherBoards Unsuitable for DigiumBoards

[Kevin Hanson]

Kevin Hanson wrote:

If what you say is true, then I'm hosed.  I've got six things sharing 
IRQ 255 according to lspci -vb:


Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) USB UHCI 
Controller #1
Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) USB UHCI 
Controller #2
Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) USB UHCI 
Controller #3

Intel Corporation 82801DB/DBM (ICH4/ICH4-M) USB2 EHCI Controller
Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) SMBus Controller
Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface

Is this what is causing all my echo?

I'll try disabling USB in bios and see what happens.

Cheers,
Kevin 


Well, my bios doesn't let me disable usb.  Drat.

Cheers,
Kevin
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Re: [Asterisk-Users] Asterisk and Agents

[Tom Rymes]

On Nov 15, 2005, at 7:38 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:


Hallo all,

I've a question regarding the agent concept of asterisk. If I login  
an agent

(using AgentLogin), this agent is directly ready to receive calls.

From the most other ACD-systems I know that an agent first logs  
into the

system and then has to set himself ready to receive calls. So the most
common agent states are login, ready, not ready, wrapup and logoff.

How is ready/notready/wrapup implemented in asterisk?


Well, Asterisk 1.2 implements agent pause, so paused/unpaused would  
correspond to not ready/ready.


Wrapup is handled as a fixed time in the the queue setup for  
Asterisk. You define wrapuptime=xx as a number of seconds. Asterisk  
then waits that long before considering the agent to be available.


Tom


Tom Rymes
Cascade Link Systems
www.cascadelinksystems.com
(603) 375-1414

Intelligent technology solutions for small businesses.


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Re: [Asterisk-Users] Intel Desktop MotherBoards Unsuitable for DigiumBoards

[Tom Rymes]

On Nov 15, 2005, at 10:02 AM, Kevin Hanson wrote:


Kevin Hanson wrote:

If what you say is true, then I'm hosed.  I've got six things  
sharing IRQ 255 according to lspci -vb:


Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) USB UHCI  
Controller #1
Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) USB UHCI  
Controller #2
Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) USB UHCI  
Controller #3

Intel Corporation 82801DB/DBM (ICH4/ICH4-M) USB2 EHCI Controller
Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) SMBus  
Controller

Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface

Is this what is causing all my echo?

I'll try disabling USB in bios and see what happens.

Cheers,
Kevin


Well, my bios doesn't let me disable usb.  Drat.

Cheers,
Kevin


Kevin,

Have you tried swapping the PCI slot to see if that helps? Does your  
BIOS allow you to reassign IRQ numbers?


Tom


Tom Rymes
Cascade Link Systems
www.cascadelinksystems.com
(603) 375-1414

Intelligent technology solutions for small businesses.


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Re: [Asterisk-Users] Message waiting notification

[Sixto Diaz]

i want to ring the phone user or change the tone 
is this posible with mailbox= ?



  - Original Message - 
  From: 
  Jonathan 
  k. Creasy 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Tuesday, November 15, 2005 11:58 
  AM
  Subject: RE: [Asterisk-Users] Message 
  waiting notification
  
  
  mailbox= in the 
  sip.conf
  
  
  
  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Sixto DiazSent: Tuesday, November 15, 2005 9:33 
  AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Message waiting 
  notification
  
  
  Hi i want to notify a user that he has an 
  unreadvoicemail waiting to be read.
  
  How can i do this for sip 
  users?
  
  
  
  THanks 

  
  Sixto
  
  

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Re: [Asterisk-Users] Asterisk hobby box

[Tom Rymes]

On Nov 15, 2005, at 2:30 AM, Dmitry Ivanov wrote:


On Tuesday 15 November 2005 06:26, [EMAIL PROTECTED] wrote:

Hi. I'm setting up an Asterisk hobby box for me to play around with.
Is it possible to use a regular 56k modem and a regular home phone
for it?


Yes, but forget G.711.

BTW, some SIP-phones have built-in modem :)


Unless I'm mistaken, this is not true. Some modems will work, but  
they are extremely rare. Your average USR/Rockwell/etc. modem will  
not work. Search on voip-info.org for X100P clone and read up on  
which will work. A regular phone line will work just fine, though,  
assuming that you get a way to interface it with your system:


Digium X100P
Digium TDM400P w/FXO Port
Digium TDM2400P w/FXO Port
ATA with FXO Port (Like Sipura SPA-3000)

Tom


Tom Rymes
Cascade Link Systems
www.cascadelinksystems.com
(603) 375-1414

Intelligent technology solutions for small businesses.


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Re: [Asterisk-Users] Cisco 7960 Multiple Line Appearance

[Matt Hoskins]

Alright, I'm inspired.  I'll give it a shot.  Should I use the asterisk 
hint system or is line appearance done in the sccp config file seperately?


Do you have a configuration example?

Thanks!

Matt

Sergio Chersovani wrote:

Matt Hoskins ha scritto:

I'm currently using asterisk 1.0.7 with cisco 7960 SIP 7.5 phones.  
I'd like to use this phone for a receptionist so that she can take 
calls for  4 other people.  Is this possible?



The SIP firmware does not support it.
You have to use SCCP to do that

Is there any way to do this with SIP and the 7960?  I've seen the 7914 
but then I'd have to use SCCP and I'm not sure if it is stable enough 
for production use.



Well give it a chance :-)

http://chan-sccp.berlios.de

Sergio
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begin:vcard
fn:Matt Hoskins
n:Hoskins;Matt
email;internet:[EMAIL PROTECTED]
tel;work:816-273-0336
tel;cell:816-261-2260
note:To reach our IS Helpline dial: 816-273-0350
x-mozilla-html:FALSE
version:2.1
end:vcard

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Re: [Asterisk-Users] Snom clients deregistering

[Richard Watson]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Richard Watson wrote:
 Richard Watson wrote:
 
 
[888120]
type=friend
username=888120
mailbox=888120
canreinvite=no
nat=yes
secret=secret
host=dynamic
qualify=yes
context=sipdemo
subscribecontext=sipdemo
 
 
 Just for fun I had a play yesterday using SER as a stateless proxy
 ouside the nat to see if for some reason that hung on to the
 registrations. The result was even worse than before.

OK, well I think I'm getting some progress.

So far they've been logged in for about 2 hours without losing
registration and the lights are working fine.

For the record I had to do all of the following:

in asterisk:
qualify=yes to send keepalives to the nat

in snom config:
challenge password dialog = no
support broken registrar = yes
suggest session time of 1min

So far if I apply all of these I get a more stable solution than I have
for a while.

-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.1 (GNU/Linux)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org

iD8DBQFDefx2P05lUVhVYk0RAlz8AJ9LtdXzV97NgAx7uzlBnuSTcDF8MwCeJg1M
1B3G+yUbuHpbZ85Aw8aNc4E=
=uIPU
-END PGP SIGNATURE-
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Re: [Asterisk-Users] NAT setup

[Martinez Felix]

problably you need a stun server...vovida.org works fineOn 11/15/05, Matt Riddell [EMAIL PROTECTED]
 wrote:John Biundo wrote: I can't forward 1-2 with my router.So I used 
rtp.conf to narrow the band of ports down to something like 14000-14030 and forwarded those portsThat seems to work fine. Am I asking for trouble down the line with this approach?
Depends on how many calls and how many rtp streams they have.I.E. voice+video--Cheers,Matt Riddell___
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[Asterisk-Users] Restore Asterisk log files after deleting...

[Chuck Bunn]

Hi,

I did something stupid - I deleted the Asterisk log files instead of 
copying empty information into the file (I know dumb) to clear out the 
data. I recreated the files with the same rights as before (644) using 
the asterisk user and group but asterisk does not appear to recognize 
them. Is there a way to get asterisk to use /var/log/asterisk/full and 
/var/log/asterisk/messages again... I restarted the server and Asterisk 
but nothing appears in the logs even though several SIP devices have 
logged in etc. (I know this as I was running the Asterisk console).


Thanks
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[Asterisk-Users] help and guidance needed from gurus

[Amir Aziz]

Hello,

First let me apologize if I am asking the wrong questions from this forum. I am totally new to VoIP but I am very much interested in the technology. I want to do the following.

1. Setup asterisk server so that it can take incoming calls initially from analog lines.
2. Setup asterisk so that it give voice responses.
3. Route the calls to local voip telephone or software.
4. Route the calls using voip internationally.

my questions are pretty basic.

1. What hardware do I need for the server to accept incoming and outgoing analog calls.
2. What books, guides or companies or individuals can help me setup.
3. I need scalable hardware.
4. What do I need in terms of technology and hardware to route calls internationally.

I would appreciate if anyone can set me up on right path I want to learn this and maybe later down the road start my own business. I thank you all for reading the message and I appreciate anyone willing to guide me to learn. Thank you.

Best Regards,
Amir Aziz

		 Yahoo! FareChase - Search multiple travel sites in one click.

 

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[Asterisk-Users] FW: Asterisk 1.0/1.2 on cobalt Raq2-4

[bram kortleven]

Title: Asterisk 1.0/1.2 on cobalt Raq2-4






Anyone ever tried to install on a Cobalt Raq device? 
This is a 1U 19" rack computer, the Raq2 using a Mips processor, the Raq4 using 
a K6-2/3 processor.As I do have a few of these as spares, I was 
wondering if I could use them as my pbx system, because of their low 
power-system and dence system box.I simply need the pbx to serve 2 
phones in my appartment, a SIP- connection for 4 external internet devices (my 
brother, living in the USA, my parents, living a few miles from here, and my 
nefew living in France and his mother, living here in Belgium too)Has 
anyone done this setup on a Raq2? Or do I need to use the extra power of the 
Raq4 (faster cpu and mem, bigger faster disk, ...)Anyone having a 
pkg-installer for the raq devices, as they are used for updates 
etc...?ThanksBram 


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RE: [Asterisk-Users] Message waiting notification

[Jonathan k. Creasy]

On my phones (polycoms) its
an option in the configuration to change the tone, etc. 



-Jonathan











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sixto Diaz
Sent: Tuesday, November 15, 2005
10:13 AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
Message waiting notification







i want to ring the phone user or change the
tone is this posible with mailbox= ?



















- Original Message - 





From: Jonathan k.
Creasy 





To: Asterisk Users Mailing List -
Non-Commercial Discussion 





Sent: Tuesday, November
15, 2005 11:58 AM





Subject: RE:
[Asterisk-Users] Message waiting notification









mailbox= in the sip.conf











From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sixto Diaz
Sent: Tuesday, November 15, 2005
9:33 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Message
waiting notification







Hi i want to notify a user that he has an
unreadvoicemail waiting to be read.





How can i do this for sip users?











THanks 





Sixto









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[Asterisk-Users] * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW *

[Olle E. Johansson]

Welcome to the Asterisk users community!


Asterisk is the leading Open Source Telephony platform,
with support both for classical telephony and IP telephony.
Asterisk.org is a fast moving project. New code is added every
day.

These are exciting times for Asterisk, with a release candidate
for 1.2 out and a release hopefully coming soon. Check the new
features on http://www.astricon.net/asterisk1-2/

Our community is also growing fast and we're having a lot
of interaction, on the IRC and on the mailing lists.

It's great to have you participating in this Open Source project
- building an Open Source PBX. Here are a few things to know and
remember while working with the project.

Again, welcome to the Asterisk.org Open Source PBX Project!

Meet you on the IRC channel, the bug tracker or
on the mailing list!

/oej

PS. There's also a new mailing list on lists.digium.com called
asterisk-i18n for discussion on Asterisk internationalization.
As soon as 1.2 is out of the door, let's meet there and discuss what
we can do to improve how Asterisk works in different languages
and character sets. Subscribe today if you want to participate!

** Asterisk version information

At this moment we have two current versions of Asterisk, the
developer version and the release version. The release version
is distributed as .tar.gz archives on several servers. The
current released version of Asterisk is 1.0.9. The release version
is fixed, we are adding no new functions and only changes it
when bugs are fixed.

The development version is to be used by people that can test
new functions and live with bugs and unexpected shortcomings.
The development version is branded 1.1 and will be the basis
for the next release version, version 1.2. This version is to be
released any day now, and development will continue on the
1.3 version.

** The mailing list is growing

Today, we propably have over 10,000 readers on the -users list. This
means that everything anyone write to this mailing list, is sent to
thousands of mailboxes that are already flowing over with messages.
That's why we all need to follow some simple rules on how to use
the mailing list and the other tools that are available.

** Think before sending a message, think twice

I would like to stress the fact that you have to think before you send a
message to such a big list. Do *not* send out personal replies on the list.

If you offer services to someone, do *not* CC: or reply to the list, it
will annoy more potential customers than get you new customers. If you
send out a message by mistake, you don't have to apologize to all of us,
we understand you're embarassed. We will get more annoyed by your
apology than over your first message.

And please do not send out test messages to the list.

** Try finding the answer first, then ask the list

The Asterisk Wiki at http://www.voip-info.org is an important
knowledge base for the project.

Go there to find your answer first, then search the mailing list
archives (Google or http://search.voip-forum.com) and then
go to the IRC channel. The IRC channel is populated with Asterisk gurus
around the clock (literally) and they'll help you move forward.

* IRC info: http://www.asterisk.org/index.php?menu=support#irc
* There's many links to Asterisk web pages on the documentation
  page at http://www.asterisk.org
* The Asterisk FAQ is found on the wiki
  http://www.voip-info.org/wiki-Asterisk+FAQ
* The Asterisk documentation project (which needs your help)
  is at http://www.asteriskdocs.org
  You can download their new book from the web site or buy
  it from the bookstore.
* Asterisk Daily news is at
  http://www.sineapps.com/news.php
* VoIP-search (Asterisk mailing list etc)
  http://search.voip-forum.com

Finally, if you don't find the answer elsewhere, try the list.

** Mailing lists
For developers, there is a developer's list, asterisk-dev.
Do not use this list as a secondary support line if you do
not get an answer on the -users list. It is meant for developer
discussions, not advanced support. If you need answers, there
is a better chance that you will get help on the irc channel.

For BSD users (FreeBSD, NetBSD, OpenBSD and OS/X) there's a
list called asterisk-bsd. There is also a business list
for those that want to ask for commercial services and
inform their community about new services (asterisk-biz).

You'll find all lists on http://lists.digium.com, which is the
site where you manage your subscription to this list as well.

Please, do not crosspost the same message to multiple mailing
lists. It will not help you, it will only add to the mail flow
and get people that read both lists irritated. If you are
unsure which list to use, send only to the -users list.

Make sure that you remove unnecessary text when you reply,
to make it easy to browse the mailing list quickly. And please
do not send HTML mail to a mailing list.

** Reporting bugs
If you think you have found a bug, report it. We 

Re: [Asterisk-Users] canreinvite=yes

[Kevin P. Fleming]

Trond Andersen wrote:


Just one question.  The documentation I have seen says that the RTP
audio stream is routed directly(if allowed ...), but never anything
about video streams? Is this just because documents are pre 1.2 or is it
true that audio can go directly, but video must pass through Asterisk?


All RTP streams are handled identically, regardless of their content.
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[Asterisk-Users] Problem: Can't make outgoing call

[Jean-Yves Avenard]

HelloI'm having trouble since I recently upgraded to Asterisk 1.2.x.I have a Sipura SPA3000 which is registered to my asterisk server.It can receive VoIP call perfectly, but can't make call.In the Asterisk SIP debug I see things like:"SIP/2.0 407 Proxy Authentication Required"Googling gave me some clues, and I found that by removing the "secret=" in sip.conf and leaving blank the password field on the sipura configuration page, actually allowed me to make calls just fine.Obviously, I do not want to leave in place a SIP client that connects without any password !Any idea on why I would be able to make calls if their no password needed, but as soon as I put a password then it fails ?Is this an issue with Asterisk 1.2 ?RegardsJean-Yves --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 8573 5200 | office +61 3 8573 5299 | direct +61 3 8573 5200  ___
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Re: [Asterisk-Users] Intel Desktop MotherBoards Unsuitable for DigiumBoards

[Kevin Hanson]

Tom Rymes wrote:


On Nov 15, 2005, at 10:02 AM, Kevin Hanson wrote:


Kevin Hanson wrote:

If what you say is true, then I'm hosed.  I've got six things  
sharing IRQ 255 according to lspci -vb:


Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) USB UHCI  
Controller #1
Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) USB UHCI  
Controller #2
Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) USB UHCI  
Controller #3

Intel Corporation 82801DB/DBM (ICH4/ICH4-M) USB2 EHCI Controller
Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) SMBus  
Controller

Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface

Is this what is causing all my echo?

I'll try disabling USB in bios and see what happens.

Cheers,
Kevin



Well, my bios doesn't let me disable usb.  Drat.

Cheers,
Kevin



Kevin,

Have you tried swapping the PCI slot to see if that helps? Does your  
BIOS allow you to reassign IRQ numbers?


Tom


Tom Rymes
Cascade Link Systems
www.cascadelinksystems.com
(603) 375-1414

Haven't tried swapping slots yet.  And, no, bios does not allow IRQ 
assignment.


Cheers
Kevin
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[Asterisk-Users] Play a message at the begining of a call

[asterisk]

Hello

is it possible to play a message just when the call starts that can be heard
both caller and called?

Thanks for any help!
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Re: [Asterisk-Users] Cisco 7960 Multiple Line Appearance

[Sergio Chersovani]

Matt Hoskins ha scritto:

Alright, I'm inspired.  I'll give it a shot.  Should I use the 
asterisk hint system or is line appearance done in the sccp config 
file seperately?

Do you have a configuration example?


the configuration example is in the package conf/sccp.conf or take a 
look at the site

http://chan-sccp.org/

Sergio
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Re: [Asterisk-Users] Polycom Softkeys Voicemail Button

[Kevin Hanson]

Sean Cook wrote:


To Change the Messages Button for 1 touch dial:

msg msg.bypassInstantMessage=1
 mwi msg.mwi.1.subscribe= msg.mwi.1.callBackMode=contact
msg.mwi.1.callBack=4999 msg.mwi.2.subscrib
e= msg.mwi.2.callBackMode=disabled msg.mwi.2.callBack=
msg.mwi.3.subscribe= msg.mwi.3.callBackMode=di
sabled msg.mwi.3.callBack= msg.mwi.4.subscribe=
msg.mwi.4.callBackMode=disabled msg.mwi.4.callBack= m
sg.mwi.5.subscribe= msg.mwi.5.callBackMode=disabled
msg.mwi.5.callBack= msg.mwi.6.subscribe= msg.mwi.6
.callBackMode=disabled msg.mwi.6.callBack=/
  /msg



 

For one touch messages you also need to set up.oneTouchVoiceMail=1 like 
the following in sip.cfg:


user_preferences up.headsetMode=0 up.useDirectoryNames=0 
up.oneTouchVoiceMail=1 up.welcomeSoundEnabled=1 
up.welcomeSoundOnWarmBootEnabled=1 up.localClockEnabled=1/

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Re: [Asterisk-Users] Media gateway recommendations?

[Dustin Wenz]

I've been looking a little as the Cisco AS5000 series (specifically  
the AS5350) as a SIP gateway for our PRI T1. Does anybody know how  
well these work with Asterisk?


- .Dustin Wenz

On Nov 14, 2005, at 4:09 PM, Dustin Wenz wrote:

Thanks for the info. Are you finding the Lucent gateway to play as  
nicely as people say it should with Asterisk? The data sheet claims  
that it can manage 720 concurrent calls. I think that piece of  
hardware is a little too extreme for our purposes. Even something  
that offered 1/10th the capacity would be more than enough. Does  
Lucent offer any sort of TNT Universal Gateway Mini?




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[Asterisk-Users] Re: Message waiting notification

[Doug Meredith]

Sixto Diaz [EMAIL PROTECTED] wrote:

i want to ring the phone user or change the tone is this posible with mailbox= 
 ?

These would be settings in your UA, not in Asterisk.

Doug
-- 
Doug Meredith ([EMAIL PROTECTED])
SystemGuard - Oracle remote support
877-974-8273 (87-SYSGUARD)
506-854-7997
www.systemguard.com

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RE: [Asterisk-Users] FW: Asterisk 1.0/1.2 on cobalt Raq2-4

[Neil K]

Title: Asterisk 1.0/1.2 on cobalt Raq2-4



Bram,

I'm not sure this issue will be one of performance. I think you will find 
cross-compiling Asterisk to the MIPS platform more 
challenging.

For this reason, I would recommend using the RAQ4 on the basis that at 
K6-2/3 is based on Intel Architecture and would make the process of getting your 
box operational much easier.

Hope this helps,

Neil


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of bram 
kortlevenSent: Tuesday, November 15, 2005 7:35 AMTo: 
asterisk-users@lists.digium.comSubject: [Asterisk-Users] FW: Asterisk 
1.0/1.2 on cobalt Raq2-4


Anyone ever tried to install on a Cobalt Raq device? 
This is a 1U 19" rack computer, the Raq2 using a Mips processor, the Raq4 using 
a K6-2/3 processor.As I do have a few of these as spares, I was 
wondering if I could use them as my pbx system, because of their low 
power-system and dence system box.I simply need the pbx to serve 2 
phones in my appartment, a SIP- connection for 4 external internet devices (my 
brother, living in the USA, my parents, living a few miles from here, and my 
nefew living in France and his mother, living here in Belgium too)Has 
anyone done this setup on a Raq2? Or do I need to use the extra power of the 
Raq4 (faster cpu and mem, bigger faster disk, ...)Anyone having a 
pkg-installer for the raq devices, as they are used for updates 
etc...?ThanksBram 
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