date:20060306

I use PICO (nano for CentOS). Works great.



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[Asterisk-Users] Re: 20 seconds til voice transmission starts

2006-03-06 Thread Cornelius Suermann

I solved this issue by replacing the router (Netgear RP 614) with a 
newer model (Netgear DG 834 B). It seems the old router had occasionally 
problems to forward the UDP-ports to Asterisk. However, I'm glad 
everything works now! Thank you for your help!


Regards, Lius


I'm experiencing a strange problem with my Asterisk. I hope you can help:

Asterisk is running at my company behind NAT. Ports 5060 and 1-2 
are being forwarded to it. I have put the router's external IP-address 
into externip in sip.conf. At home I'm using an AVM FritzBox Fon WLAN 
7050 which is registered with the Asterisk at my company.


When I try to call Asterisk (or a phone connected to the attached 
legacy-pbx) from home, it's ringing normally and I can hear my opposite. 
But it takes about 20 seconds until my opposite hears me! When I call 
the same number again staight after, everything is working fine from the 
beginning. Also, calls from the company to my home are working perfectly.


I'm greateful for any tips!



One way to identify the issue is to run ethereal to see what's happening
with the udp ports. If that doesn't provide a clue, then run asterisk
with additional levels of debug/verboseness.



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RE: [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel

2006-03-06 Thread Bart van Daal

Hi Sina,

a detailed list of the steps you took could help. 
Did you follow the suggestions in README.udev, also
a 'make linux26' did some magic for me.

kr,
Bart 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dovid Bender
Sent: maandag 6 maart 2006 13:41
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6
kernel

plase email a detailed list of what you did. step by step.

dovid

--- Sina Bahram [EMAIL PROTECTED] wrote:

 Yes, I did
 
 Take care,
 Sina
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Dovid 
 Bender
 Sent: Sunday, March 05, 2006 7:16 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Problem compiling ztdummy on centos 4, 
 2.6 kernel
 
 did you uncommnet # from before ztdummy ?
 
 --- Sina Bahram [EMAIL PROTECTED] wrote:
 
  Hi all,
  
  I hope everyone is doing well. I just joined the
 list, and I've really
  enjoyed all I have read about asterisk so far.
  Unfortunately, I'm having a
  bit of trouble implementing this thing :).
  
  By the way ... I did my best to search the forums,
 and also to use
  google extensively, and while I have found pages
 with people with the
  same problem, ... The fix suggested on those
 sites, didn't work for
  me.
  
  Here's what I have:
  
  Results of uname -r:
  2.6.9-22.0.2.106.unsupportedsmp
  
  Arch:
  X86_64
  
  If you need more specs on the machine or OS,
 please let me know.
  
  I downloaded and have been following the asterisk
 book, and in chapter
  three I followed all the instructions on
 downloading the sources,
  untarring them, and so forth.
  
  Zaptel compiled without a hitch, as did the rest
 of the asterisk
  packages. I modified udev, and I restarted the
 box: ... I did:
  
  /etc/init.d/zaptel start
  
  I get:
  
  Loading zaptel framework:  FATAL: Module zaptel
 not found.

  
[FAILED]
  Waiting for zap to come online...Error: missing
 /dev/zap!
  
  If I do
  
  /sbin/modprobe zaptel
  
  I get:
  FATAL: Module zaptel not found. 
  
  If I do
  
  /sbin/modprobe ztdummy
  
  I get:
  
  FATAL: Module ztdummy not found.
  FATAL: Error running install command for ztdummy
  
  Also, if i run:
  
  /etc/init.d/zaptel reload
  
  I get:
  
  Reloading ztcfg:  Notice: Configuration file is
 /etc/zaptel.conf line
  0: Unable to open master device '/dev/zap/ctl'
  1 error(s) detected

  
[FAILED]
  
  If I go back to /usr/src/zaptel-1.2.4 and I do
  
  make ztdummy
  
  I get:
  
  cc   ztdummy.o   -o ztdummy
 

/usr/lib/gcc/x86_64-redhat-linux/3.4.4/../../../../lib64/crt1.o(.text+0x21):
  In
  function `_start':
  : undefined reference to `main'
  ztdummy.o(.text+0xc): In function `ztdummy_timer':
  /usr/src/zaptel-1.2.4/ztdummy.c:154: undefined
 reference to
  `zt_receive'
 

ztdummy.o(.text+0x18):/usr/src/zaptel-1.2.4/ztdummy.c:155:
  undefined
  reference t
  o `zt_transmit'
 

ztdummy.o(.text+0x1f):/usr/src/zaptel-1.2.4/ztdummy.c:156:
  undefined
  reference t
  o `jiffies'
  ztdummy.o(.text+0x4d): In function `init_module':
  include/linux/slab.h:93: undefined reference to
 `malloc_sizes'
  ztdummy.o(.text+0x52):include/linux/slab.h:93:
  undefined reference to
  `kmem_cach
  e_alloc'
  ztdummy.o(.text+0x6a): In function `init_module':
  /usr/src/zaptel-1.2.4/ztdummy.c:232: undefined
 reference to `printk'
 

ztdummy.o(.text+0x197):/usr/src/zaptel-1.2.4/ztdummy.c:192:
  undefined
  reference
  to `zt_register'
 

ztdummy.o(.text+0x1a9):/usr/src/zaptel-1.2.4/ztdummy.c:239:
  undefined
  reference
  to `printk'
 

ztdummy.o(.text+0x1b5):/usr/src/zaptel-1.2.4/ztdummy.c:240:
  undefined
  reference
  to `kfree'
 

ztdummy.o(.text+0x1e2):/usr/src/zaptel-1.2.4/ztdummy.c:261:
  undefined
  reference
  to `jiffies'
  ztdummy.o(.text+0x23d): In function `init_module':
  include/linux/timer.h:87: undefined reference to
 `__mod_timer'
  ztdummy.o(.text+0x255): In function `init_module':
  /usr/src/zaptel-1.2.4/ztdummy.c:286: undefined
 reference to `printk'
  ztdummy.o(.text+0x27c): In function
  `cleanup_module':
  /usr/src/zaptel-1.2.4/ztdummy.c:298: undefined
 reference to
  `del_timer'
 

ztdummy.o(.text+0x288):/usr/src/zaptel-1.2.4/ztdummy.c:303:
  undefined
  reference
  to `zt_unregister'
 

ztdummy.o(.text+0x294):/usr/src/zaptel-1.2.4/ztdummy.c:304:
  undefined
  reference
  to `kfree'
  ztdummy.o(.text+0x39): In function
 `ztdummy_timer':
  include/linux/timer.h:87: undefined reference to
 `__mod_timer'
  ztdummy.o(.text+0x2b0): In function
  `cleanup_module':
  /usr/src/zaptel-1.2.4/ztdummy.c:310: undefined
 reference to `printk'
  ztdummy.o(__param+0x10): undefined reference to
 `param_set_int'
  ztdummy.o(__param+0x18): undefined reference to
 `param_get_int'
  collect2: ld returned 1 exit status

Re: [Asterisk-Users] Info about mp3 which are installed with Asterisk

asterisk tends to not work well with mp3's that have
ID3 tags

--- Zach A [EMAIL PROTECTED] wrote:

 Hi,
 
 The 3 MP3 files which are installed with asterisk,
 what is their bit
 rate, are they mono and do they have ID3 tags?
 
 Zach A
 
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Re: [Asterisk-Users] login/logout agents in a specific queue

2006-03-06 Thread Adrian Carter


Johann wrote:
In Asterisk the Agent / Queue setup is kinda different than most 
people may expect.  You can use a Queue without using Agents and 
Agents can be used without Queues.  Agents however extend normal 
channels with the ability to login/logout/pause that is not available 
on Zap/SIP/IAX/etc.


Im just curious, How would one use 'agents' without a queue. Is this
what you are essentially doing using Local/XXX@ dial strings??


I assume that you are using Agent/foo on both queues.  Then you will 
need to dynamically add and remove that agent from the queues using 
AddQueueMember and RemoveQueueMember.  Anything stored in queues.conf 
will be used when Asterisk is restarted/reloaded, however you can 
add/remove later as needed.  Just keep in mind if you have the agent 
default to both queues, they remove themselves from one, then you 
reload Asterisk putting them back in both.


Reloading asterisk also undoes pause I've found...


--johann

nik600 wrote:

hi

if i have an agents that figure as a member in more than one queue,
how can i login / logout him in a specific queue an not in all queues?
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--
Adrian Carter
Technical Manager
Leading Edge Internet

Web   http://www.lei.net.au http://support.lei.net.au
Direct+61 2 6163 6162  Support 1 300 662 415
E-mail[EMAIL PROTECTED]
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Re: [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel

2006-03-06 Thread Cory Andrews

+++I am out of the office until Tuesday, March 7th attending training, I 
will be returning calls and emails at that time+++


+++Thank You+++

Cory Andrews
++
VOIPSupply.com
A Division of b2 Technologies
454 Sonwil Drive
Buffalo, NY 14225

direct - 716.250.3402
mobile - 716.907.4054
email - [EMAIL PROTECTED]
AIM - b2Cory

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

Sent: Monday, March 06, 2006 7:41 AM
Subject: RE: [Asterisk-Users] Problem compiling ztdummy on centos 4,2.6 
kernel




plase email a detailed list of what you did. step by
step.

dovid

--- Sina Bahram [EMAIL PROTECTED] wrote:


Yes, I did

Take care,
Sina

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Behalf Of Dovid Bender
Sent: Sunday, March 05, 2006 7:16 AM
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] Problem compiling
ztdummy on centos 4, 2.6
kernel

did you uncommnet # from before ztdummy ?

--- Sina Bahram [EMAIL PROTECTED] wrote:

 Hi all,

 I hope everyone is doing well. I just joined the
list, and I've really
 enjoyed all I have read about asterisk so far.
 Unfortunately, I'm having a
 bit of trouble implementing this thing :).

 By the way ... I did my best to search the forums,
and also to use
 google extensively, and while I have found pages
with people with the
 same problem, ... The fix suggested on those
sites, didn't work for
 me.

 Here's what I have:

 Results of uname -r:
 2.6.9-22.0.2.106.unsupportedsmp

 Arch:
 X86_64

 If you need more specs on the machine or OS,
please let me know.

 I downloaded and have been following the asterisk
book, and in chapter
 three I followed all the instructions on
downloading the sources,
 untarring them, and so forth.

 Zaptel compiled without a hitch, as did the rest
of the asterisk
 packages. I modified udev, and I restarted the
box: ... I did:

 /etc/init.d/zaptel start

 I get:

 Loading zaptel framework:  FATAL: Module zaptel
not found.


   [FAILED]
 Waiting for zap to come online...Error: missing
/dev/zap!

 If I do

 /sbin/modprobe zaptel

 I get:
 FATAL: Module zaptel not found.

 If I do

 /sbin/modprobe ztdummy

 I get:

 FATAL: Module ztdummy not found.
 FATAL: Error running install command for ztdummy

 Also, if i run:

 /etc/init.d/zaptel reload

 I get:

 Reloading ztcfg:  Notice: Configuration file is
/etc/zaptel.conf line
 0: Unable to open master device '/dev/zap/ctl'
 1 error(s) detected


   [FAILED]

 If I go back to /usr/src/zaptel-1.2.4 and I do

 make ztdummy

 I get:

 cc   ztdummy.o   -o ztdummy



/usr/lib/gcc/x86_64-redhat-linux/3.4.4/../../../../lib64/crt1.o(.text+0x21):

 In
 function `_start':
 : undefined reference to `main'
 ztdummy.o(.text+0xc): In function `ztdummy_timer':
 /usr/src/zaptel-1.2.4/ztdummy.c:154: undefined
reference to
 `zt_receive'



ztdummy.o(.text+0x18):/usr/src/zaptel-1.2.4/ztdummy.c:155:

 undefined
 reference t
 o `zt_transmit'



ztdummy.o(.text+0x1f):/usr/src/zaptel-1.2.4/ztdummy.c:156:

 undefined
 reference t
 o `jiffies'
 ztdummy.o(.text+0x4d): In function `init_module':
 include/linux/slab.h:93: undefined reference to
`malloc_sizes'
 ztdummy.o(.text+0x52):include/linux/slab.h:93:
 undefined reference to
 `kmem_cach
 e_alloc'
 ztdummy.o(.text+0x6a): In function `init_module':
 /usr/src/zaptel-1.2.4/ztdummy.c:232: undefined
reference to `printk'



ztdummy.o(.text+0x197):/usr/src/zaptel-1.2.4/ztdummy.c:192:

 undefined
 reference
 to `zt_register'



ztdummy.o(.text+0x1a9):/usr/src/zaptel-1.2.4/ztdummy.c:239:

 undefined
 reference
 to `printk'



ztdummy.o(.text+0x1b5):/usr/src/zaptel-1.2.4/ztdummy.c:240:

 undefined
 reference
 to `kfree'



ztdummy.o(.text+0x1e2):/usr/src/zaptel-1.2.4/ztdummy.c:261:

 undefined
 reference
 to `jiffies'
 ztdummy.o(.text+0x23d): In function `init_module':
 include/linux/timer.h:87: undefined reference to
`__mod_timer'
 ztdummy.o(.text+0x255): In function `init_module':
 /usr/src/zaptel-1.2.4/ztdummy.c:286: undefined
reference to `printk'
 ztdummy.o(.text+0x27c): In function
 `cleanup_module':
 /usr/src/zaptel-1.2.4/ztdummy.c:298: undefined
reference to
 `del_timer'



ztdummy.o(.text+0x288):/usr/src/zaptel-1.2.4/ztdummy.c:303:

 undefined
 reference
 to `zt_unregister'



ztdummy.o(.text+0x294):/usr/src/zaptel-1.2.4/ztdummy.c:304:

 undefined
 reference
 to `kfree'
 ztdummy.o(.text+0x39): In function
`ztdummy_timer':
 include/linux/timer.h:87: undefined reference to
`__mod_timer'
 ztdummy.o(.text+0x2b0): In function
 `cleanup_module':
 /usr/src/zaptel-1.2.4/ztdummy.c:310: undefined
reference to `printk'
 ztdummy.o(__param+0x10): undefined reference to
`param_set_int'
 ztdummy.o(__param+0x18): undefined reference to
`param_get_int'
 collect2: ld returned 1 exit status
 make: *** [ztdummy] Error 1

 Any ideas? I know I posted things in some wrong
order here,

Re: [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel

Oh No! Here we go again
Cory you should know better.

--- Cory Andrews [EMAIL PROTECTED] wrote:

 +++I am out of the office until Tuesday, March 7th
 attending training, I 
 will be returning calls and emails at that time+++
 
 +++Thank You+++
 
 Cory Andrews
 ++
 VOIPSupply.com
 A Division of b2 Technologies
 454 Sonwil Drive
 Buffalo, NY 14225
 
 direct - 716.250.3402
 mobile - 716.907.4054
 email - [EMAIL PROTECTED]
 AIM - b2Cory
 
 - Original Message - 
 From: Dovid Bender [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial
 Discussion 
 asterisk-users@lists.digium.com
 Sent: Monday, March 06, 2006 7:41 AM
 Subject: RE: [Asterisk-Users] Problem compiling
 ztdummy on centos 4,2.6 
 kernel
 
 
  plase email a detailed list of what you did. step
 by
  step.
 
  dovid
 
  --- Sina Bahram [EMAIL PROTECTED] wrote:
 
  Yes, I did
 
  Take care,
  Sina
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED]
 On
  Behalf Of Dovid Bender
  Sent: Sunday, March 05, 2006 7:16 AM
  To: Asterisk Users Mailing List - Non-Commercial
  Discussion
  Subject: Re: [Asterisk-Users] Problem compiling
  ztdummy on centos 4, 2.6
  kernel
 
  did you uncommnet # from before ztdummy ?
 
  --- Sina Bahram [EMAIL PROTECTED] wrote:
 
   Hi all,
  
   I hope everyone is doing well. I just joined
 the
  list, and I've really
   enjoyed all I have read about asterisk so far.
   Unfortunately, I'm having a
   bit of trouble implementing this thing :).
  
   By the way ... I did my best to search the
 forums,
  and also to use
   google extensively, and while I have found
 pages
  with people with the
   same problem, ... The fix suggested on those
  sites, didn't work for
   me.
  
   Here's what I have:
  
   Results of uname -r:
   2.6.9-22.0.2.106.unsupportedsmp
  
   Arch:
   X86_64
  
   If you need more specs on the machine or OS,
  please let me know.
  
   I downloaded and have been following the
 asterisk
  book, and in chapter
   three I followed all the instructions on
  downloading the sources,
   untarring them, and so forth.
  
   Zaptel compiled without a hitch, as did the
 rest
  of the asterisk
   packages. I modified udev, and I restarted the
  box: ... I did:
  
   /etc/init.d/zaptel start
  
   I get:
  
   Loading zaptel framework:  FATAL: Module zaptel
  not found.
  
 
 [FAILED]
   Waiting for zap to come online...Error: missing
  /dev/zap!
  
   If I do
  
   /sbin/modprobe zaptel
  
   I get:
   FATAL: Module zaptel not found.
  
   If I do
  
   /sbin/modprobe ztdummy
  
   I get:
  
   FATAL: Module ztdummy not found.
   FATAL: Error running install command for
 ztdummy
  
   Also, if i run:
  
   /etc/init.d/zaptel reload
  
   I get:
  
   Reloading ztcfg:  Notice: Configuration file is
  /etc/zaptel.conf line
   0: Unable to open master device '/dev/zap/ctl'
   1 error(s) detected
  
 
 [FAILED]
  
   If I go back to /usr/src/zaptel-1.2.4 and I do
  
   make ztdummy
  
   I get:
  
   cc   ztdummy.o   -o ztdummy
  
 
 

/usr/lib/gcc/x86_64-redhat-linux/3.4.4/../../../../lib64/crt1.o(.text+0x21):
   In
   function `_start':
   : undefined reference to `main'
   ztdummy.o(.text+0xc): In function
 `ztdummy_timer':
   /usr/src/zaptel-1.2.4/ztdummy.c:154: undefined
  reference to
   `zt_receive'
  
 
 

ztdummy.o(.text+0x18):/usr/src/zaptel-1.2.4/ztdummy.c:155:
   undefined
   reference t
   o `zt_transmit'
  
 
 

ztdummy.o(.text+0x1f):/usr/src/zaptel-1.2.4/ztdummy.c:156:
   undefined
   reference t
   o `jiffies'
   ztdummy.o(.text+0x4d): In function
 `init_module':
   include/linux/slab.h:93: undefined reference to
  `malloc_sizes'
   ztdummy.o(.text+0x52):include/linux/slab.h:93:
   undefined reference to
   `kmem_cach
   e_alloc'
   ztdummy.o(.text+0x6a): In function
 `init_module':
   /usr/src/zaptel-1.2.4/ztdummy.c:232: undefined
  reference to `printk'
  
 
 

ztdummy.o(.text+0x197):/usr/src/zaptel-1.2.4/ztdummy.c:192:
   undefined
   reference
   to `zt_register'
  
 
 

ztdummy.o(.text+0x1a9):/usr/src/zaptel-1.2.4/ztdummy.c:239:
   undefined
   reference
   to `printk'
  
 
 

ztdummy.o(.text+0x1b5):/usr/src/zaptel-1.2.4/ztdummy.c:240:
   undefined
   reference
   to `kfree'
  
 
 

ztdummy.o(.text+0x1e2):/usr/src/zaptel-1.2.4/ztdummy.c:261:
   undefined
   reference
   to `jiffies'
   ztdummy.o(.text+0x23d): In function
 `init_module':
   include/linux/timer.h:87: undefined reference
 to
  `__mod_timer'
   ztdummy.o(.text+0x255): In function
 `init_module':
   /usr/src/zaptel-1.2.4/ztdummy.c:286: undefined
  reference to `printk'
   ztdummy.o(.text+0x27c): In function
   `cleanup_module':
   /usr/src/zaptel-1.2.4/ztdummy.c:298: undefined
  reference to
   `del_timer'
  
 
 

ztdummy.o(.text+0x288):/usr/src/zaptel-1.2.4/ztdummy.c:303:
   undefined
   reference
   to `zt_unregister'
  
 
 

ztdummy.o(.text+0x294):/usr/src/zaptel-1.2.4/ztdummy.c:304:
   undefined
   reference
   to `kfree'

Re: [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel

Oh No! Here we go again
Cory you should know better.

--- Cory Andrews [EMAIL PROTECTED] wrote:

 +++I am out of the office until Tuesday, March 7th
 attending training, I 
 will be returning calls and emails at that time+++
 
 +++Thank You+++
 
 Cory Andrews
 ++
 VOIPSupply.com
 A Division of b2 Technologies
 454 Sonwil Drive
 Buffalo, NY 14225
 
 direct - 716.250.3402
 mobile - 716.907.4054
 email - [EMAIL PROTECTED]
 AIM - b2Cory
 
 - Original Message - 
 From: Dovid Bender [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial
 Discussion 
 asterisk-users@lists.digium.com
 Sent: Monday, March 06, 2006 7:41 AM
 Subject: RE: [Asterisk-Users] Problem compiling
 ztdummy on centos 4,2.6 
 kernel
 
 
  plase email a detailed list of what you did. step
 by
  step.
 
  dovid
 
  --- Sina Bahram [EMAIL PROTECTED] wrote:
 
  Yes, I did
 
  Take care,
  Sina
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED]
 On
  Behalf Of Dovid Bender
  Sent: Sunday, March 05, 2006 7:16 AM
  To: Asterisk Users Mailing List - Non-Commercial
  Discussion
  Subject: Re: [Asterisk-Users] Problem compiling
  ztdummy on centos 4, 2.6
  kernel
 
  did you uncommnet # from before ztdummy ?
 
  --- Sina Bahram [EMAIL PROTECTED] wrote:
 
   Hi all,
  
   I hope everyone is doing well. I just joined
 the
  list, and I've really
   enjoyed all I have read about asterisk so far.
   Unfortunately, I'm having a
   bit of trouble implementing this thing :).
  
   By the way ... I did my best to search the
 forums,
  and also to use
   google extensively, and while I have found
 pages
  with people with the
   same problem, ... The fix suggested on those
  sites, didn't work for
   me.
  
   Here's what I have:
  
   Results of uname -r:
   2.6.9-22.0.2.106.unsupportedsmp
  
   Arch:
   X86_64
  
   If you need more specs on the machine or OS,
  please let me know.
  
   I downloaded and have been following the
 asterisk
  book, and in chapter
   three I followed all the instructions on
  downloading the sources,
   untarring them, and so forth.
  
   Zaptel compiled without a hitch, as did the
 rest
  of the asterisk
   packages. I modified udev, and I restarted the
  box: ... I did:
  
   /etc/init.d/zaptel start
  
   I get:
  
   Loading zaptel framework:  FATAL: Module zaptel
  not found.
  
 
 [FAILED]
   Waiting for zap to come online...Error: missing
  /dev/zap!
  
   If I do
  
   /sbin/modprobe zaptel
  
   I get:
   FATAL: Module zaptel not found.
  
   If I do
  
   /sbin/modprobe ztdummy
  
   I get:
  
   FATAL: Module ztdummy not found.
   FATAL: Error running install command for
 ztdummy
  
   Also, if i run:
  
   /etc/init.d/zaptel reload
  
   I get:
  
   Reloading ztcfg:  Notice: Configuration file is
  /etc/zaptel.conf line
   0: Unable to open master device '/dev/zap/ctl'
   1 error(s) detected
  
 
 [FAILED]
  
   If I go back to /usr/src/zaptel-1.2.4 and I do
  
   make ztdummy
  
   I get:
  
   cc   ztdummy.o   -o ztdummy
  
 
 

/usr/lib/gcc/x86_64-redhat-linux/3.4.4/../../../../lib64/crt1.o(.text+0x21):
   In
   function `_start':
   : undefined reference to `main'
   ztdummy.o(.text+0xc): In function
 `ztdummy_timer':
   /usr/src/zaptel-1.2.4/ztdummy.c:154: undefined
  reference to
   `zt_receive'
  
 
 

ztdummy.o(.text+0x18):/usr/src/zaptel-1.2.4/ztdummy.c:155:
   undefined
   reference t
   o `zt_transmit'
  
 
 

ztdummy.o(.text+0x1f):/usr/src/zaptel-1.2.4/ztdummy.c:156:
   undefined
   reference t
   o `jiffies'
   ztdummy.o(.text+0x4d): In function
 `init_module':
   include/linux/slab.h:93: undefined reference to
  `malloc_sizes'
   ztdummy.o(.text+0x52):include/linux/slab.h:93:
   undefined reference to
   `kmem_cach
   e_alloc'
   ztdummy.o(.text+0x6a): In function
 `init_module':
   /usr/src/zaptel-1.2.4/ztdummy.c:232: undefined
  reference to `printk'
  
 
 

ztdummy.o(.text+0x197):/usr/src/zaptel-1.2.4/ztdummy.c:192:
   undefined
   reference
   to `zt_register'
  
 
 

ztdummy.o(.text+0x1a9):/usr/src/zaptel-1.2.4/ztdummy.c:239:
   undefined
   reference
   to `printk'
  
 
 

ztdummy.o(.text+0x1b5):/usr/src/zaptel-1.2.4/ztdummy.c:240:
   undefined
   reference
   to `kfree'
  
 
 

ztdummy.o(.text+0x1e2):/usr/src/zaptel-1.2.4/ztdummy.c:261:
   undefined
   reference
   to `jiffies'
   ztdummy.o(.text+0x23d): In function
 `init_module':
   include/linux/timer.h:87: undefined reference
 to
  `__mod_timer'
   ztdummy.o(.text+0x255): In function
 `init_module':
   /usr/src/zaptel-1.2.4/ztdummy.c:286: undefined
  reference to `printk'
   ztdummy.o(.text+0x27c): In function
   `cleanup_module':
   /usr/src/zaptel-1.2.4/ztdummy.c:298: undefined
  reference to
   `del_timer'
  
 
 

ztdummy.o(.text+0x288):/usr/src/zaptel-1.2.4/ztdummy.c:303:
   undefined
   reference
   to `zt_unregister'
  
 
 

ztdummy.o(.text+0x294):/usr/src/zaptel-1.2.4/ztdummy.c:304:
   undefined
   reference
   to `kfree'

[Asterisk-Users] Outbound Proxy Support

2006-03-06 Thread hgaillac-sip

Hi all,

May I have to patch asterisk-1.2.x with this patch
http://bugs.digium.com/bug_view_page.php?bug_id=0002859
to configure an outbound sip proxy in sip.conf ?

Regards
Harry











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Re: [Asterisk-Users] login/logout agents in a specific queue

On 00:07, Tue 07 Mar 06, Adrian Carter wrote:
 Im just curious, How would one use 'agents' without a queue. Is this
 what you are essentially doing using Local/XXX@ dial strings??

kindda.
Maybe an example makes it a bit more clear.
Say you have 10 desks, all with a phone on them.
Users dont have their own desk, but take a free one whenever
they come in the office.
Users sits down, types a number, gives his pernonal number
and his password, and from that moment on he/she is logged
in as agent personal number
In the dialplan you can now reach this user with: Dial(Agent/personal_number)

When the user switches desk, they simply login on the new
phone.

Does that make it a bit more clear ?

-- 
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D

Why is it drug addicts and computer afficionados are both called users?

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Re: [Asterisk-Users] login/logout agents in a specific queue

2006-03-06 Thread Adrian Carter





Yeah, "Hot Desking"  but ok.. if you'll indulge me further, why
would the likes of AMP use astdb to implement that combined with some
clunky macros? 
Im after that exact solution, but have various issues on occasion with
the AMP implementation of 'user login/logoff'. I'd love for you to
share an example extensions.conf snippet with how to "login" the agent
to nowhere... 

This also means.. in essence.. rather than DND'ing or logging out, an
'agent' can just pause themselves for a period from all calls - another
benefit I seek. 

So any actual dialplan code you could share I'd love :)

Michiel van Baak wrote:

  On 00:07, Tue 07 Mar 06, Adrian Carter wrote:
  
  
Im just curious, How would one use 'agents' without a queue. Is this
what you are essentially doing using Local/XXX@ dial strings??

  
  
kindda.
Maybe an example makes it a bit more clear.
Say you have 10 desks, all with a phone on them.
Users dont have their own desk, but take a free one whenever
they come in the office.
Users sits down, types a number, gives his pernonal number
and his password, and from that moment on he/she is logged
in as agent personal number
In the dialplan you can now reach this user with: Dial(Agent/personal_number)

When the user switches desk, they simply login on the new
phone.

Does that make it a bit more clear ?

  


-- 
Adrian Carter
Technical Manager
Leading Edge Internet

Web	  http://www.lei.net.au http://support.lei.net.au
Direct+61 2 6163 6162  Support 1 300 662 415
E-mail[EMAIL PROTECTED]


-- 
Adrian Carter
Technical Manager
Leading Edge Internet

Web	  http://www.lei.net.au http://support.lei.net.au
Direct+61 2 6163 6162  Support 1 300 662 415
E-mail[EMAIL PROTECTED]



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Re: [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel

2006-03-06 Thread Rich Adamson

I just sent an email to one of his coworkers to disable that stuff.

Rich


  From: Dovid Bender [EMAIL PROTECTED]
  Subject: Re: [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 
kernel
  Date: Mon, 6 Mar 2006 05:18:13 -0800 (PST) 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com


 Oh No! Here we go again
 Cory you should know better.
 
 --- Cory Andrews [EMAIL PROTECTED] wrote:
 
  +++I am out of the office until Tuesday, March 7th
  attending training, I 
  will be returning calls and emails at that time+++
  
  +++Thank You+++
  
  Cory Andrews
  ++
  VOIPSupply.com
  A Division of b2 Technologies
  454 Sonwil Drive
  Buffalo, NY 14225
  
  direct - 716.250.3402
  mobile - 716.907.4054
  email - [EMAIL PROTECTED]
  AIM - b2Cory
  
  - Original Message - 
  From: Dovid Bender [EMAIL PROTECTED]
  To: Asterisk Users Mailing List - Non-Commercial
  Discussion 
  asterisk-users@lists.digium.com
  Sent: Monday, March 06, 2006 7:41 AM
  Subject: RE: [Asterisk-Users] Problem compiling
  ztdummy on centos 4,2.6 
  kernel
  
  
   plase email a detailed list of what you did. step
  by
   step.
  
   dovid
  
   --- Sina Bahram [EMAIL PROTECTED] wrote:
  
   Yes, I did
  
   Take care,
   Sina
  
   -Original Message-
   From: [EMAIL PROTECTED]
   [mailto:[EMAIL PROTECTED]
  On
   Behalf Of Dovid Bender
   Sent: Sunday, March 05, 2006 7:16 AM
   To: Asterisk Users Mailing List - Non-Commercial
   Discussion
   Subject: Re: [Asterisk-Users] Problem compiling
   ztdummy on centos 4, 2.6
   kernel
  
   did you uncommnet # from before ztdummy ?
  
   --- Sina Bahram [EMAIL PROTECTED] wrote:
  
Hi all,
   
I hope everyone is doing well. I just joined
  the
   list, and I've really
enjoyed all I have read about asterisk so far.
Unfortunately, I'm having a
bit of trouble implementing this thing :).
   
By the way ... I did my best to search the
  forums,
   and also to use
google extensively, and while I have found
  pages
   with people with the
same problem, ... The fix suggested on those
   sites, didn't work for
me.
   
Here's what I have:
   
Results of uname -r:
2.6.9-22.0.2.106.unsupportedsmp
   
Arch:
X86_64
   
If you need more specs on the machine or OS,
   please let me know.
   
I downloaded and have been following the
  asterisk
   book, and in chapter
three I followed all the instructions on
   downloading the sources,
untarring them, and so forth.
   
Zaptel compiled without a hitch, as did the
  rest
   of the asterisk
packages. I modified udev, and I restarted the
   box: ... I did:
   
/etc/init.d/zaptel start
   
I get:
   
Loading zaptel framework:  FATAL: Module zaptel
   not found.
   
  
  [FAILED]
Waiting for zap to come online...Error: missing
   /dev/zap!
   
If I do
   
/sbin/modprobe zaptel
   
I get:
FATAL: Module zaptel not found.
   
If I do
   
/sbin/modprobe ztdummy
   
I get:
   
FATAL: Module ztdummy not found.
FATAL: Error running install command for
  ztdummy
   
Also, if i run:
   
/etc/init.d/zaptel reload
   
I get:
   
Reloading ztcfg:  Notice: Configuration file is
   /etc/zaptel.conf line
0: Unable to open master device '/dev/zap/ctl'
1 error(s) detected
   
  
  [FAILED]
   
If I go back to /usr/src/zaptel-1.2.4 and I do
   
make ztdummy
   
I get:
   
cc   ztdummy.o   -o ztdummy
   
  
  
 
 /usr/lib/gcc/x86_64-redhat-linux/3.4.4/../../../../lib64/crt1.o(.text+0x21):
In
function `_start':
: undefined reference to `main'
ztdummy.o(.text+0xc): In function
  `ztdummy_timer':
/usr/src/zaptel-1.2.4/ztdummy.c:154: undefined
   reference to
`zt_receive'
   
  
  
 
 ztdummy.o(.text+0x18):/usr/src/zaptel-1.2.4/ztdummy.c:155:
undefined
reference t
o `zt_transmit'
   
  
  
 
 ztdummy.o(.text+0x1f):/usr/src/zaptel-1.2.4/ztdummy.c:156:
undefined
reference t
o `jiffies'
ztdummy.o(.text+0x4d): In function
  `init_module':
include/linux/slab.h:93: undefined reference to
   `malloc_sizes'
ztdummy.o(.text+0x52):include/linux/slab.h:93:
undefined reference to
`kmem_cach
e_alloc'
ztdummy.o(.text+0x6a): In function
  `init_module':
/usr/src/zaptel-1.2.4/ztdummy.c:232: undefined
   reference to `printk'
   
  
  
 
 ztdummy.o(.text+0x197):/usr/src/zaptel-1.2.4/ztdummy.c:192:
undefined
reference
to `zt_register'
   
  
  
 
 ztdummy.o(.text+0x1a9):/usr/src/zaptel-1.2.4/ztdummy.c:239:
undefined
reference
to `printk'
   
  
  
 
 ztdummy.o(.text+0x1b5):/usr/src/zaptel-1.2.4/ztdummy.c:240:
undefined
reference
to `kfree'
   
  
  
 
 ztdummy.o(.text+0x1e2):/usr/src/zaptel-1.2.4/ztdummy.c:261:
undefined
reference
to `jiffies'
ztdummy.o(.text+0x23d):

Re: [Asterisk-Users] low call volume

2006-03-06 Thread Mike Clark


billy wrote:


i have AAH connected to pstn via digium TDM01B
 
had been testing it on telewest line (UK cable company) with very 
little issues.
now moved to a BT line and had several that i anticipated from 
infomation on this list.

the one that has caught me out is low volume from the caller via pstn.
 
using sipura spa-941's and have to push the volume up to hear.

is there a setting that can correct this


If you are talking about the volume of the caller from the PSTN to your 
Asterisk systems, then you can try bumping up the rxgain value in 
zapata.conf. However, this can potententially cause other problems, like 
echo.


Mike Clark
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[Asterisk-Users] Unable to make hints function properly

2006-03-06 Thread Per Møller

Ive been trying for quite some time now to make hints work correctly, so
that I may use the BLF (busy lamp field) features of the Snom and
Grandstream models that support it.

My problem is NOT a subscription problem.

I have a running Asterisk system, everything is as it should be, hints are
in the dialplan as they should be etc...

My problem is:

If I configure my phones in sip.conf as type=friend, then hints stop working
correctly. Using the 'show hints' in the console shows me, that the only 2
states a phone can be in is 'Idle' or 'Unavailable' (when I pull the power
on the phones). The 'Ringing' or 'InUse' state will never happen no matter
what my phones are doing.

If I configure my phones in sip.conf as type=peer, then hints work a little
better, meaning that phones can now be 'Idle', 'Unavailable' and 'InUse'
state. But I still do not have the 'Ringing' state. When a phone is ringing
the state is 'InUse'.

This works fine in 1.0.7, so I'm wondering if it is broken in 1.2.x? I tried
it with 1.2.1 and 1.2.4.

Is this really broken in 1.2.x? - Should I report a bug?


// Per


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RE: [Asterisk-Users] Unable to make hints function properly

2006-03-06 Thread Mimmus

Sorry for my ignorance but what are 'HINTS'?

Thanks
Mimmus

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[Asterisk-Users] RE: Extension 's' in Realtime

2006-03-06 Thread Kaleb L. Kunzler

varchar(5)

I have realtime working fine on my box, Instead of type tinyint for the
priority column, I use type varchar(5), this allows me to not only use
,t,s,and i, but also hint.


 

-Original Message-
--

Message: 21
Date: Mon,  6 Mar 2006 05:28:00 -0600
From: [EMAIL PROTECTED]
Subject: [Asterisk-Users] 
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain;   charset=ISO-8859-1

Hi All,

I was able to insert some extensions in Mysql DB and use them successfully.
In
Mysql extensions table the priority column is of type tinyint and when I
give
's' value for it, it is not accepting that value as it takes only tinyints.
Please tell how can I make that column accept values like t,s,i and make it
work with asterisk in realtime without any problem? If I change the type of
that column to something else then I think I will get errors as asterisk
querying Mysql might go wrong. Please tell me how can I get this to work?

Thanks,
Manoj.



--

Message: 22
Date: Mon, 6 Mar 2006 11:26:46 + (GMT)
From: arun arora [EMAIL PROTECTED]
Subject: [Asterisk-Users] problems in changing Festival's Default
Voice inAsterisk
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=iso-8859-1

Hi all,
  I m in a trouble using festival voices in asterisk. I am not able to
change the default male voice of festival. Although i downloaded the us1
female voice and it iw working good in festival's CLI but it is not coming
when i am usinf Festival in asterisk.
  I changed the default-voice-priority list directive and set us1_mbrole as
first entry and also changed voice in festival CLI. But they didn't helped
me anyways.
  so please anyone can tell me if there is any setting i am missing??
  How to use festival female voice in asterisk.
   
  Thanks
  aRUnaR
   


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Message: 23
Date: Mon, 6 Mar 2006 10:12:02 -0300
From: Pablo Allietti [EMAIL PROTECTED]
Subject: [Asterisk-Users] hangup on silence?
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=iso-8859-1

is possible to define a parameter to, hangup the line on silent? or ping
dead or something? 

because all line have busy after the pc hangup :(
-- 


--

Message: 24
Date: Mon, 6 Mar 2006 13:31:32 +0100
From: Giordano Grandis [EMAIL PROTECTED]
Subject: [Asterisk-Users] Capturing DTMF during a call
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii

Hi all,
I have a simple and maybe also stupid question: if i'm in coversation on
a Zap channel and the remote party send me a DTMF, could I capture it?
 
Thanks all
 

Giordano 

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Message: 25
Date: Mon, 6 Mar 2006 04:41:13 -0800 (PST)
From: Dovid Bender [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Problem compiling ztdummy  on centos 4,
2.6 kernel
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=iso-8859-1

plase email a detailed list of what you did. step by
step.

dovid

--- Sina Bahram [EMAIL PROTECTED] wrote:

 Yes, I did
 
 Take care,
 Sina 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On
 Behalf Of Dovid Bender
 Sent: Sunday, March 05, 2006 7:16 AM
 To: Asterisk Users Mailing List - Non-Commercial
 Discussion
 Subject: Re: [Asterisk-Users] Problem compiling
 ztdummy on centos 4, 2.6
 kernel
 
 did you uncommnet # from before ztdummy ?
 
 --- Sina Bahram [EMAIL PROTECTED] wrote:
 
  Hi all,
  
  I hope everyone is doing well. I just joined the
 list, and I've really 
  enjoyed all I have read about asterisk so far.
  Unfortunately, I'm having a
  bit of trouble implementing this thing :).
  
  By the way ... I did my best to search the forums,
 and also to use 
  google extensively, and while I have found pages
 with people with the 
  same problem, ... The fix suggested on those
 sites, didn't work for 
  me.
  
  Here's what I have:
  
  Results of uname -r:
  2.6.9-22.0.2.106.unsupportedsmp
  
  Arch:
  X86_64
  
  If you need more specs on the machine or OS,
 please let me know.
  
  I downloaded and have been

Re: [Asterisk-Users] Two asterisks on one machine

2006-03-06 Thread vivek

Hi friend,
  I am running asterisk in production and it is being used by many people using 
h323. I cannot afford to change all their configurations. Also, the newer 
asterisk dosenot support inband for h323 properly. Thats why I want two 
asterisks one for backward compatibility and one for sip which I want to 
implement.




With warm regards.

Vivek J. Joshi.

[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.

--New opinions often appear first as jokes and fancies, then as blasphemies and 
treason, then as questions open to discussion, and finally as established 
truths.



Joseph Tanner wrote:
 You could run a virtual machine.  I'd try xen, uml, and vmware in that
 order (vmware would be the easiest/quickest to setup, but is more of a
 resource-hog than xen or uml).  Assign a separate ip to the virtual
 server, setup asterisk, and you're all set.
 
 BTW, just curious but why can't you run one asterisk install with both
 h323 and sip?  It'd simplify things and use less resources than
 running a virtual server, assuming it works for you.
 
 Another idea, if one's solely for h323 and the other's solely for sip
 (neither will be running both), then you could compile asterisk twice,
 using different directories for each install.  I don't think this
 would work if both needed to use the same ports.  I'm guessing you
 want to bridge the h323 asterisk to the sip asterisk?  If not, but you
 do want to use sip on both, perhaps you can use port 5060 on one and
 5061 for the other.  Couldn't bridge them, but both could talk to the
 outside world (that is, maybe they could, I haven't tried this and do
 not know what's involved).  Running one in a virtual server is
 probably going to be the easiest way to get two asterisk processes to
 coexist on the same physical server.
 
 Joseph Tanner
 
 On 3/6/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
  Hello friends,
 Can I run two asterisks running simultaneously on the same machine? I 
  want one to run v1.0.2 for h323 ( which is an old and running production 
  system ) and one for sip implementation. I wonder how it can be done since 
  they will want access to the same ports and ip addresses.
 Does anyone know to do this or has done this before?
 Please share your experiences please.
 
 
 
 
 
  With warm regards.
 
  Vivek J. Joshi.
 
  [EMAIL PROTECTED]
  Trikon electronics Pvt. Ltd.
 
  --New opinions often appear first as jokes and fancies, then as blasphemies 
  and treason, then as questions open to discussion, and finally as 
  established truths.
 
 
 
 
 
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 http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 


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[Asterisk-Users] grandstream handytone 286 sometimes dials out wrong number

2006-03-06 Thread Giorgio Incantalupo


Hi,
I have an asterisk 1.2.1 (on a debian sarge) box with a TDM400P card. I 
connected the TDM400P to a grandstream 286 to use a VoIP provider.
It seems all right except for a little problem: one call every 30 is 
made to a wrong number.

Is there anybody who had the same problem and solved it?

TIA

Giorgio Incantalupo
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[Asterisk-Users] Re: Problem with libpri?

2006-03-06 Thread McQuiggan, Mark xt46480

Title: Re: Problem with libpri?





In addition, I have created a possibly larger dump of the issue, as below. Can someone help me determine what the problem is? Is there more information that I can provide? I am running libpri 1.2.2, zaptel 1.2.4 and asterisk 1.2.5:

gdb dump:


Program received signal SIGSEGV, Segmentation fault.
[Switching to Thread -1211937872 (LWP 16798)]
0x00207138 in pri_disconnect_timeout (data="" at q931.c:2619
2619 if (pri-debug  PRI_DEBUG_Q931_STATE)
(gdb) bt
#0 0x00207138 in pri_disconnect_timeout (data="" at q931.c:2619
#1 0x002013db in __pri_schedule_run (pri=0x8a04010, tv=0xb7c33e3c) at prisched.c:98
#2 0x00201446 in pri_schedule_run (pri=0x8a04010) at prisched.c:110
#3 0x001d1282 in pri_dchannel (vpri=0x1e0c40) at chan_zap.c:8190
#4 0x004eb341 in start_thread () from /lib/tls/libpthread.so.0
#5 0x004406fe in clone () from /lib/tls/libc.so.6
(gdb) list
8190 e = pri_schedule_run(pri-dchans[which]);
8191 if (e)
8192 break;
8193 }
8194 } else if (res  -1) {
8195 for (which=0;whichNUM_DCHANS;which++) {
8196 if (!pri-dchans[which])
8197 break;
8198 if (fds[which].revents  POLLPRI) {
8199 /* Check for an event */



console dump (phone numbers have been removed)


Mar 6 08:58:19 VERBOSE[16799] logger.c: -- Channel 3/21, span 4 got hangup request
Mar 6 08:58:19 VERBOSE[16916] logger.c: -- Hungup 'Zap/3-1'
Mar 6 08:58:19 VERBOSE[16916] logger.c: == Spawn extension (macro-dialextNoCallid, s, 3) exited non-zero on 'Zap/93-1' in macro 'dialextNoCallid'

Mar 6 08:58:19 VERBOSE[16916] logger.c: == Spawn extension (macro-dialextNoCallid, s, 3) exited non-zero on 'Zap/93-1'

Mar 6 08:58:19 VERBOSE[16916] logger.c: -- Hungup 'Zap/93-1'
Mar 6 08:58:27 VERBOSE[17053] logger.c: -- Executing Macro(SIP/46583-82b2, dialOutToronto|Zap/g1/9416xxx|ADP BROKER SVC|416xxx) in new stack

Mar 6 08:58:27 VERBOSE[17053] logger.c: -- Executing Answer(SIP/46583-82b2, ) in new stack
Mar 6 08:58:27 VERBOSE[17053] logger.c: -- Executing SetCallerID(SIP/46583-82b2, ADP BROKER SVC 416xxx) in new stack

Mar 6 08:58:27 VERBOSE[17053] logger.c: -- Executing Dial(SIP/46583-82b2, Zap/g1/9416xxx) in new stack
Mar 6 08:58:27 VERBOSE[17053] logger.c: -- Requested transfer capability: 0x00 - SPEECH
Mar 6 08:58:27 VERBOSE[17053] logger.c: -- Called g1/9416xxx
Mar 6 08:58:27 VERBOSE[17053] logger.c: -- Zap/2-1 is proceeding passing it to SIP/46583-82b2
Mar 6 08:58:27 VERBOSE[17053] logger.c: -- Zap/2-1 is ringing
Mar 6 08:58:41 VERBOSE[17053] logger.c: -- Zap/2-1 answered SIP/46583-82b2



zaptel.conf


defaultzone=us


span=1,1,0,d4,b8zs
bchan=1-12
dchan=24


span=2,3,0,d4,b8zs
bchan=25-36
dchan=48


# span=3,0,0,esf,b8zs
# bchan=49-71
# dchan=72


span=4,2,0,esf,b8zs
bchan=73-95
dchan=96



zapata.conf
[trunkgroups]
trunkgroup = 1,24,48
trunkgroup = 2,96


spanmap = 1,1,0
spanmap = 2,1,1
spanmap = 4,2,3



[channels]
rxgain=8.0
txgain=-4.5
echocancel=yes
echotraining=yes
echocancelwhenbridged=yes


group = 1
context = trunk
usecallerid=yes
callerid = asreceived
switchtype = national
nsf = none
overlapdial = no
signalling = pri_net
channel = 1-12,25-36



rxgain=-5


group = 2
context = trunk
usecallerid=yes
callerid = asreceived
switchtype = national
overlapdial = no
signalling = pri_net
relaxdtmf = yes
channel = 73-95




--


Message: 4


Date: Sun, 5 Mar 2006 15:20:15 -0500 


From: McQuiggan, Mark xt46480 [EMAIL PROTECTED]


Subject: [Asterisk-Users] Problem with libpri?


To: asterisk-users@lists.digium.com


Message-ID:


[EMAIL PROTECTED]


Content-Type: text/plain; charset=windows-1252


While testing a problem with spontaeously and occasionally rebooting


asterisk, I came upon this problem:


Program received signal SIGSEGV, Segmentation fault.


[Switching to Thread -1210770512 (LWP 11346)]


0x002e3fe1 in pri_release_timeout (data="" at q931.c:2589


2589 q931.c: No such file or directory.


in q931.c


q931.c is in libpri, function pri_release_timeout, and line 2589 reads: 


if (pri-debug  PRI_DEBUG_Q931_STATE)


pri_message(pri, Timed out looking for release


complete\n);


PRI Debug was not on in the asterisk console. 


Any ideas? My asterisk restarts about twice a day, and drops any current


calls in the process.


Regards, 


Mark McQuiggan



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SV: [Asterisk-Users] Unable to make hints function properly

2006-03-06 Thread Per Møller

Try 'show hints' in the console...

Or read http://www.voip-info.org/wiki-Asterisk+standard+extensions

It's Asterisk way of knowing the state of a phone so that phones may
subscribe to this information and make small led light up if a phone is
busy, and flash if it's ringing.

// Per


 Sorry for my ignorance but what are 'HINTS'?

 Thanks
 Mimmus


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[Asterisk-Users] spa3000 asterisk fxo gateway

2006-03-06 Thread Roberto Pereyra

Hi

Somebody knows a tutorial or help me for use a SPA3000 like fxo Asterisk interface ?

I would like to send and receive calls from/to my asterisk extensions from PSTN by spa3000 fxo.

Thanks in advance.

roberto-- Ing. Roberto PereyraContenidosOnlineServidores BSD, Solaris y LinuxSoporte técnico ISPsJabber ID: [EMAIL PROTECTED]
For reliable and professional DNS, use DNS Made Easy!http://www.dnsmadeeasy.com/u/14989
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Re: [Asterisk-Users] spa3000 asterisk fxo gateway

2006-03-06 Thread Rich Adamson


 Somebody knows a tutorial or help me for use a SPA3000 like fxo Asterisk 
 interface ?
 
 I would like to send and receive  calls from/to my asterisk extensions from 
 PSTN by spa3000 fxo.

Go to www.voxilla.com and look for a setup wizard. Also, lots of other
good references/user-experiences at that site.


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RE: [Asterisk-Users] Polycom 501 power over ethernet

No, some IP 501's have the inline cable and some have the power jack.

-Original Message-
From: Paul Hales [mailto:[EMAIL PROTECTED]
Sent: Sunday, March 05, 2006 8:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Polycom 501 power over ethernet

The IP300/301 has the power jack, the IP500/501 the inline cable.

PaulH

On Sun, 2006-03-05 at 20:56 -0700, Douglas Garstang wrote:
 Not true. Some do and some don't. Some have a place to plug a separate DC 
 adapter, and some have the inline power, where the adapter plugs into the 
 ethernet cable. Not sure which ones are newer, and which are older.

   -Original Message- 
   From: Michael Welter [mailto:[EMAIL PROTECTED] 
   Sent: Sun 3/5/2006 6:50 PM 
   To: Asterisk Users Mailing List - Non-Commercial Discussion 
   Cc: 
   Subject: Re: [Asterisk-Users] Polycom 501 power over ethernet

   The IP501 does not have a power jack.  You'll need one of the Polycom
   cables.

   William M Conlon wrote:
My recollection of the marketing fluff was that we would just use our
legacy network (cables) and the devices at both ends would figure out
whether they were sourcing, sinking, or neither.  In the case of the
501, it's the special Polycom cable, either with or without provision
for an AC power adapter, that powers the phone.  That's what I meant 
 by
saying the '501' itself is not compliant with 802.3af -- it needs a
separate thingamajig [tech jargon :)]to be powered.

Anyway I had hoped that I could just plug a CAT-5 patch cable from my
RJ45 wall outlet into the phone.

On Mar 5, 2006, at 5:17 PM, Michael Welter wrote:

As I understand 802.3af, the phones go through a negotiation with the
unit supplying the power.  I don't think it's a matter of -48VDC on a
particular pair.  I remember a schematic from years ago--it had each
of the receive pair and the transmit pair going into a transformer
winding,  and that winding had a center tap for PoE.  This is not
something that *I* am going to screw with.

The IP501 telephone set is the same for both PoE and local power. 
With the PoE cable, the 802.3af electronics (the negotiator) is a
plastic thing in the cable.  For the local power, there is a plastic
thingie toward the wall end of the cable, and you plug the wall wart
into the plastic thingie.  Notice the advanced technical jargon 
 here

With local power, there is still only one cable one the desk--the
power plugs into the cable towards the wall.  Except for a power
interruption, this has all the advantages of PoE.

William M Conlon wrote:
I saw that Polycom offered a cable (not stocked anywhere), at $40 a
pop for 802.3af connections.  That's what made me think the phone
itself is NOT 802.3af compliant.
Presumably, for $40, there's more than a fuse in that special cable.
On Mar 5, 2006, at 4:31 PM, Paul Hales wrote:
For Polycom IP500/501's and IP300/301's you need a special polycom 
 POE
cable.

When you buy Polycom phones you can usually specify POE or 
 powerpack.

PaulH

On Sun, 2006-03-05 at 16:23 -0800, William M Conlon wrote:
When I bought two Polycom 501 SIP phones, I naively thought they 
 were
Power-over-Ethernet (IEEE 802.3af) because they were powered over
ethernet.  Silly me.

Polycom must have some odd voltage or funny way of injecting the
power, because the POE switch I bought for them (Netgear [EMAIL 
 PROTECTED])
won't power them, though if I use the Polycom-supplied AC adapter 
 and
ethernet power injector cable, they work with the switch in either
its powered or unpowered ports.

Anyhow, I hadn't seen any mention of how people power these 
 phones,
as I had planned on centralizing phone power on a UPS to supply my
Asterisk server and POE switch.  Now the question is:

Can the Polycom AC-powered injector be used with a standard 
 ethernet
patch cable:

switch :: Polycom injector cable :: RJ45 coupler :: patch 
 cable ::
Polycom 501

which would allow me to power the Polycom AC adapters by my UPS.  
 Or
do I need to provide a UPS at each phone and run the ethernet like

switch :: patch cable :: RJ45 coupler :: Polycom injector 
 cable ::
Polycom 501

thanks.
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[Asterisk-Users] Background() App From AGI

I have the following python AGI script.
I know it's been abstracted, but it's still pretty easy to see what's happening.

self.agi.channelAnswer()
self.agi.wait(1)
self.agi.execCmd(background,enter-conf-call-number,)
self.agi.execCmd(Read,confNum|||,)
confNum = self.agi.getVar(confNum)

I enter DTMF digits, and read the result with Read() while the sound file is 
still playing. I always lose the first digit. The docs aren't clear but it 
appears that Background() is designed to grab the first DTMF digit it sees. I 
don't want Background() to chomp my first DTMF digit! I want to read them all 
with Read(). How can I play a sound file, while still waiting for DTMF input 
and get all the DTMF digits entered?

Thanks,
Doug.
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[Asterisk-Users] Call Transfer - Both legs must reside on Asterisk box to transfer at this time

I have a SIP user, 2944093 that dialled 3254102. I'm trying to transfer the 
call from 3254102 to 3254104. When I try and transfer the call, I get the 
following on the Asterisk console.

Mar  3 15:14:18 NOTICE[23124]: chan_sip.c:6731 get_refer_info: Supervised 
transfer requested, but unable to find callid '[EMAIL PROTECTED]'.  Both legs 
must reside on Asterisk box to transfer at this time.

Below is what my SIP debug console output shows me. IP 216.188.128.11 is the 
phone that the transferer is on (3254102). It sends a REFER message to 
Asterisk. Asterisk turns around and says 'Not found' eventhough the destination 
user, 3254104, is in it's database. I wonder if this is because the REFER has 
Asterisks's IP address and not the IP address of the phone? How could it have 
gotten that way? 

Thanks,
Doug.

--- (10 headers 0 lines)---
-- SIP/3254104-a911 is ringing

-- SIP read from 216.188.128.11:5060: 
REFER sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 216.188.128.11;branch=z9hG4bKb3056f7489B0729B
From: sip:[EMAIL PROTECTED];tag=AD42A97D-626BB596
To: Douglas Garstang sip:[EMAIL PROTECTED];tag=as6202b08e
CSeq: 2 REFER
Call-ID: [EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.3.0067
Refer-To: sip:[EMAIL 
PROTECTED];user=phone?Replaces=77a7b64e-f546fcbc-f206df35%40172.31.16.67%3Bto-tag%3Das4744b9fa%3Bfrom-tag%3D200C85AA-7A3B0AE3
Referred-By: sip:[EMAIL PROTECTED]
Max-Forwards: 70
Content-Length: 0


--- (12 headers 0 lines)---
Transfer to 3254104 in From_OneEighty
Transfer from 3254102 in From_OneEighty
Mar  3 14:32:49 NOTICE[16519]: chan_sip.c:6731 get_refer_info: Supervised 
transfer requested, but unable to find callid '[EMAIL PROTECTED]'.  Both legs 
must reside on Asterisk box to transfer at this time.
Reliably Transmitting (no NAT) to 216.188.128.11:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 
216.188.128.11;branch=z9hG4bKb3056f7489B0729B;received=216.188.128.11
From: sip:[EMAIL PROTECTED];tag=AD42A97D-626BB596
To: Douglas Garstang sip:[EMAIL PROTECTED];tag=as6202b08e
Call-ID: [EMAIL PROTECTED]
CSeq: 2 REFER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:[EMAIL PROTECTED]
Accept: application/sdp
Content-Length: 0

Here's the database entry for the destination number:
/SIP/Registry/3254104 : 
216.188.128.12:5060:3600:3254104:sip:[EMAIL PROTECTED]

As you can see, that isn't what the REFER has. It has 216.188.140.203, which is 
Asterisks IP address. I don't know if that's the issue or not. Asterisk _IS_ in 
the RTP path.

Doug.


-Original Message-
From: David Thomas [mailto:[EMAIL PROTECTED]
Sent: Friday, March 03, 2006 2:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Hardware Requirements for 1M minutes


Sorry, I saw that right after I posted.

It is per month. And almost all during business hours.

regards,
David

On 3/3/06, Martin Joseph [EMAIL PROTECTED] wrote:

 On Mar 3, 2006, at 9:49 AM, David Thomas wrote:

  I'm doing an install for a client with the following requirements.
 
  - 1 Million minutes of outbound calling

 Per what?

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RE: [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel

2006-03-06 Thread Sina Bahram

Hi there,

I did do both of those things, yes, but it's not necessary to do the udev
permisions and rules modifications is it, since the makefile appears to do
that for you. At least it did it for me; however, just to make sure I did it
manually as well.

I'll send the output of the script command as I go through the process, to
the list.

Take care,
Sina

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bart van Daal
Sent: Monday, March 06, 2006 7:47 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6
kernel

Hi Sina,

a detailed list of the steps you took could help. 
Did you follow the suggestions in README.udev, also a 'make linux26' did
some magic for me.

kr,
Bart 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dovid Bender
Sent: maandag 6 maart 2006 13:41
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6
kernel

plase email a detailed list of what you did. step by step.

dovid

--- Sina Bahram [EMAIL PROTECTED] wrote:

 Yes, I did
 
 Take care,
 Sina
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Dovid 
 Bender
 Sent: Sunday, March 05, 2006 7:16 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Problem compiling ztdummy on centos 4,
 2.6 kernel
 
 did you uncommnet # from before ztdummy ?
 
 --- Sina Bahram [EMAIL PROTECTED] wrote:
 
  Hi all,
  
  I hope everyone is doing well. I just joined the
 list, and I've really
  enjoyed all I have read about asterisk so far.
  Unfortunately, I'm having a
  bit of trouble implementing this thing :).
  
  By the way ... I did my best to search the forums,
 and also to use
  google extensively, and while I have found pages
 with people with the
  same problem, ... The fix suggested on those
 sites, didn't work for
  me.
  
  Here's what I have:
  
  Results of uname -r:
  2.6.9-22.0.2.106.unsupportedsmp
  
  Arch:
  X86_64
  
  If you need more specs on the machine or OS,
 please let me know.
  
  I downloaded and have been following the asterisk
 book, and in chapter
  three I followed all the instructions on
 downloading the sources,
  untarring them, and so forth.
  
  Zaptel compiled without a hitch, as did the rest
 of the asterisk
  packages. I modified udev, and I restarted the
 box: ... I did:
  
  /etc/init.d/zaptel start
  
  I get:
  
  Loading zaptel framework:  FATAL: Module zaptel
 not found.

  
[FAILED]
  Waiting for zap to come online...Error: missing
 /dev/zap!
  
  If I do
  
  /sbin/modprobe zaptel
  
  I get:
  FATAL: Module zaptel not found. 
  
  If I do
  
  /sbin/modprobe ztdummy
  
  I get:
  
  FATAL: Module ztdummy not found.
  FATAL: Error running install command for ztdummy
  
  Also, if i run:
  
  /etc/init.d/zaptel reload
  
  I get:
  
  Reloading ztcfg:  Notice: Configuration file is
 /etc/zaptel.conf line
  0: Unable to open master device '/dev/zap/ctl'
  1 error(s) detected

  
[FAILED]
  
  If I go back to /usr/src/zaptel-1.2.4 and I do
  
  make ztdummy
  
  I get:
  
  cc   ztdummy.o   -o ztdummy
 

/usr/lib/gcc/x86_64-redhat-linux/3.4.4/../../../../lib64/crt1.o(.text+0x21):
  In
  function `_start':
  : undefined reference to `main'
  ztdummy.o(.text+0xc): In function `ztdummy_timer':
  /usr/src/zaptel-1.2.4/ztdummy.c:154: undefined
 reference to
  `zt_receive'
 

ztdummy.o(.text+0x18):/usr/src/zaptel-1.2.4/ztdummy.c:155:
  undefined
  reference t
  o `zt_transmit'
 

ztdummy.o(.text+0x1f):/usr/src/zaptel-1.2.4/ztdummy.c:156:
  undefined
  reference t
  o `jiffies'
  ztdummy.o(.text+0x4d): In function `init_module':
  include/linux/slab.h:93: undefined reference to
 `malloc_sizes'
  ztdummy.o(.text+0x52):include/linux/slab.h:93:
  undefined reference to
  `kmem_cach
  e_alloc'
  ztdummy.o(.text+0x6a): In function `init_module':
  /usr/src/zaptel-1.2.4/ztdummy.c:232: undefined
 reference to `printk'
 

ztdummy.o(.text+0x197):/usr/src/zaptel-1.2.4/ztdummy.c:192:
  undefined
  reference
  to `zt_register'
 

ztdummy.o(.text+0x1a9):/usr/src/zaptel-1.2.4/ztdummy.c:239:
  undefined
  reference
  to `printk'
 

ztdummy.o(.text+0x1b5):/usr/src/zaptel-1.2.4/ztdummy.c:240:
  undefined
  reference
  to `kfree'
 

ztdummy.o(.text+0x1e2):/usr/src/zaptel-1.2.4/ztdummy.c:261:
  undefined
  reference
  to `jiffies'
  ztdummy.o(.text+0x23d): In function `init_module':
  include/linux/timer.h:87: undefined reference to
 `__mod_timer'
  ztdummy.o(.text+0x255): In function `init_module':
  /usr/src/zaptel-1.2.4/ztdummy.c:286: undefined
 reference to `printk'
  ztdummy.o(.text+0x27c): In function
  `cleanup_module':
  /usr/src/zaptel-1.2.4/ztdummy.c:298: undefined
 reference to

[Asterisk-Users] Set(LANGUAGE()=language) - for queue

2006-03-06 Thread Tomislav Parčina

Hi group!

How to set language for queue?
I have several queue's. In every queue, agents speaks different language. I 
need to announce queue-youarenext and similar on different languages.

This is what I have in my extensions.conf and it does set language, but when 
calls enters queue, it doesn't use that language.

exten = 313,1,Answer 
exten = 313,n,Set(LANGUAGE()=de)
exten = 313,n,Playback(callcentar/qnjemacki,skip)
exten = 313,n,Queue(njemacki|t|||3600)
exten = 313,n,GotoIfTime(8:00-16:00|mon-fri|*|*?313,8)
exten = 313,n,Playback(callcentar/rvnjemacki,skip)
exten = 313,n,VoiceMail,u221
exten = 313,n,Hangup
exten = 313,n,VoiceMail,b221
exten = 313,n,Hangup


And this is how it looks on CLI.

-- Executing Goto(SIP/211-793f, callcentre|313|1) in new stack
-- Goto (callcentre,313,1)
-- Executing Answer(SIP/211-793f, ) in new stack
-- Executing Set(SIP/211-793f, LANGUAGE()=de) in new stack
-- Executing Playback(SIP/211-793f, callcentar/qnjemacki|skip) in new 
stack
-- Executing Queue(SIP/211-793f, njemacki|t|||3600) in new stack
-- outgoing agentcall, to agent '401', on 'Local/[EMAIL PROTECTED],1'
-- Called Agent/401
-- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/211|20|wWtT) in new 
stack
-- Called 211
-- SIP/211-5996 is ringing
-- Agent/401 is ringing
-- SIP/211-5996 answered Local/[EMAIL PROTECTED],2
-- Agent/401 answered SIP/211-793f
-- Playing 'callcentar/gpnjemacki' (language 'en')
  == Spawn extension (internal, 211, 1) exited non-zero on 'Local/[EMAIL 
PROTECTED],2'
-- Playing 'queue-reporthold' (language 'en')
-- Playing 'queue-less-than' (language 'en')
-- Playing 'digits/2' (language 'en')
-- Playing 'queue-minutes' (language 'en')
  


--
Tomislav Parcina
tparcina#lama.hr
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RE: [Asterisk-Users] Polycom 501 power over ethernet

2006-03-06 Thread Chad Osmond

I was just thinking, about this..

Move your Polycom Power Injecting Patch cable (Black Cable with AC
Adapter Input) into the cabling closet. You could then infuse the power
at the cabling closet and then just use a standard patch cable to patch
the phone in. 

You would be looking at a line loss of 40 Ohms per 1000 ft, or about 12
Ohms per 300ft run.

Max output of the transformer is 400mA @ 12V

The Voltage drop of a 12 Ohm load on a 400mA circuit is 0.03V... So that
should be more the acceptable.

I just don't know what would happen if a user plugged a phone into the
line..

Chad
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of William M
Conlon
Sent: March 5, 2006 8:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom 501 power over ethernet

My recollection of the marketing fluff was that we would just use our
legacy network (cables) and the devices at both ends would figure out
whether they were sourcing, sinking, or neither.  In the case of the
501, it's the special Polycom cable, either with or without provision
for an AC power adapter, that powers the phone.  That's what I meant by
saying the '501' itself is not compliant with 802.3af -- it needs a
separate thingamajig [tech jargon :)]to be powered.

Anyway I had hoped that I could just plug a CAT-5 patch cable from my
RJ45 wall outlet into the phone.

On Mar 5, 2006, at 5:17 PM, Michael Welter wrote:

 As I understand 802.3af, the phones go through a negotiation with the 
 unit supplying the power.  I don't think it's a matter of -48VDC on a 
 particular pair.  I remember a schematic from years ago--it had each 
 of the receive pair and the transmit pair going into a transformer 
 winding,  and that winding had a center tap for PoE.  This is not 
 something that *I* am going to screw with.

 The IP501 telephone set is the same for both PoE and local power.   
 With the PoE cable, the 802.3af electronics (the negotiator) is a 
 plastic thing in the cable.  For the local power, there is a plastic 
 thingie toward the wall end of the cable, and you plug the wall wart 
 into the plastic thingie.  Notice the advanced technical jargon here

 With local power, there is still only one cable one the desk--the 
 power plugs into the cable towards the wall.  Except for a power 
 interruption, this has all the advantages of PoE.



 William M Conlon wrote:
 I saw that Polycom offered a cable (not stocked anywhere), at $40  
 a pop for 802.3af connections.  That's what made me think the  
 phone itself is NOT 802.3af compliant.
 Presumably, for $40, there's more than a fuse in that special cable.
 On Mar 5, 2006, at 4:31 PM, Paul Hales wrote:
 For Polycom IP500/501's and IP300/301's you need a special  
 polycom POE
 cable.

 When you buy Polycom phones you can usually specify POE or  
 powerpack.

 PaulH

 On Sun, 2006-03-05 at 16:23 -0800, William M Conlon wrote:
 When I bought two Polycom 501 SIP phones, I naively thought they  
 were
 Power-over-Ethernet (IEEE 802.3af) because they were powered over
 ethernet.  Silly me.

 Polycom must have some odd voltage or funny way of injecting the
 power, because the POE switch I bought for them (Netgear [EMAIL PROTECTED])
 won't power them, though if I use the Polycom-supplied AC  
 adapter and
 ethernet power injector cable, they work with the switch in either
 its powered or unpowered ports.

 Anyhow, I hadn't seen any mention of how people power these phones,
 as I had planned on centralizing phone power on a UPS to supply my
 Asterisk server and POE switch.  Now the question is:

 Can the Polycom AC-powered injector be used with a standard  
 ethernet
 patch cable:

 switch :: Polycom injector cable :: RJ45 coupler :: patch  
 cable ::
 Polycom 501

 which would allow me to power the Polycom AC adapters by my  
 UPS.  Or
 do I need to provide a UPS at each phone and run the ethernet like

 switch :: patch cable :: RJ45 coupler :: Polycom injector  
 cable ::
 Polycom 501

 thanks.
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 Bill
 William M. Conlon, P.E., Ph.D.
 To the Point
 345 California Avenue Suite 2
 Palo Alto, CA 94306
vox:  650.327.2175 (direct)
fax:  650.329.8335
 mobile:  650.906.9929
 e-mail:  mailto:[EMAIL PROTECTED]
web:  http://www.tothept.com
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RE: [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel

2006-03-06 Thread Sina Bahram

Here is the compilation process of zaptel

I did edit the makefile and uncommented the #ztdummy, although, after I did
that, I get the make error of ztdummy being defined more than once.

[EMAIL PROTECTED] src]# cd zaptel-1.2.4/
[EMAIL PROTECTED] zaptel-1.2.4]# make clean
Makefile:214: target `ztdummy.o' given more than once in the same rule.
rm -f torisatool makefw tor2fw.h radfw.h
rm -f ztcfg torisatool makefw ztmonitor ztspeed  zttest fxotune
rm -f *.o ztcfg tzdriver sethdlc sethdlc-new
rm -f zonedata.lo tonezone.lo libtonezone.so *.lo
rm -f *.ko *.mod.c .*o.cmd
rm -f xpp/*.ko xpp/*.mod.c xpp/.*o.cmd
rm -f xpp/*.o xpp/*.mod.o
rm -rf .tmp_versions
rm -f gendigits tones.h
rm -f libtonezone*
rm -f tor2ee
rm -f fxotune
rm -f core
rm -f ztcfg-shared fxstest
[EMAIL PROTECTED] zaptel-1.2.4]# make linux26
Makefile:214: target `ztdummy.o' given more than once in the same rule.
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE   -m64 -DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\   -c -o gendigits.o gendigits.c
cc -o gendigits gendigits.o -lm
./gendigits
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE   -m64 -DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\makefw.c   -o makefw
./makefw tormenta2.rbt tor2fw  tor2fw.h
Loaded 69900 bytes from file
./makefw pciradio.rbt radfw  radfw.h
Loaded 42096 bytes from file
ZAPTELVERSION=1.2.4 build_tools/make_version_h  version.h.tmp
if cmp -s version.h.tmp version.h ; then echo; else \
mv version.h.tmp version.h ; \
fi

rm -f version.h.tmp
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE   -m64 -DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\   -c -o ztcfg.o ztcfg.c
cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE   -m64 -DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o zonedata.lo
zonedata.c
cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE   -m64 -DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o tonezone.lo
tonezone.c
ar rcs libtonezone.a zonedata.lo tonezone.lo
cc -o ztcfg ztcfg.o libtonezone.a -lm
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE   -m64 -DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\   -c -o torisatool.o torisatool.c
cc -o torisatool torisatool.o
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE   -m64 -DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\   -c -o ztmonitor.o ztmonitor.c
cc -o ztmonitor ztmonitor.o
cc -o ztspeed.o -c ztspeed.c
cc -o ztspeed ztspeed.o
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE   -m64 -DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\zttest.c   -o zttest
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE   -m64 -DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\   -c -o fxotune.o fxotune.c
cc -o fxotune fxotune.o -lm
/lib/modules/2.6.9-22.0.2.106.unsupportedsmp/build
make -C /lib/modules/2.6.9-22.0.2.106.unsupportedsmp/build
SUBDIRS=/usr/src/zaptel-1.2.4 XPPMOD= modules
make[1]: Entering directory
`/usr/src/kernels/2.6.9-22.0.2.106.unsupported-x86_64'
/usr/src/zaptel-1.2.4/Makefile:214: target `ztdummy.o' given more than once
in the same rule.
  CC [M]  /usr/src/zaptel-1.2.4/zaptel.o
n/usr/src/zaptel-1.2.4/zaptel.c:188: warning: 'fcstab' defined but not used
  CC [M]  /usr/src/zaptel-1.2.4/tor2.o
  CC [M]  /usr/src/zaptel-1.2.4/torisa.o
/usr/src/zaptel-1.2.4/torisa.c:1145: warning: 'set_tor_base' defined but not
used
  CC [M]  /usr/src/zaptel-1.2.4/wcusb.o
  CC [M]  /usr/src/zaptel-1.2.4/wcfxo.o
  CC [M]  /usr/src/zaptel-1.2.4/wctdm.o
  CC [M]  /usr/src/zaptel-1.2.4/wctdm24xxp.o
  CC [M]  /usr/src/zaptel-1.2.4/ztdynamic.o
  CC [M]  /usr/src/zaptel-1.2.4/ztd-eth.o
  CC [M]  /usr/src/zaptel-1.2.4/wct1xxp.o
  CC [M]  /usr/src/zaptel-1.2.4/wct4xxp.o
;  CC [M]  /usr/src/zaptel-1.2.4/wcte11xp.o
  CC [M]  /usr/src/zaptel-1.2.4/pciradio.o
/usr/src/zaptel-1.2.4/pciradio.c:1810: warning: `MODULE_PARM_' is deprecated
(declared at include/linux/module.h:552)
  CC [M]  /usr/src/zaptel-1.2.4/ztd-loc.o
  CC [M]  /usr/src/zaptel-1.2.4/ztdummy.o
  Building modules, stage 2.
  MODPOST
Warning: could not find versions for .tmp_versions/zaptel.mod
  CC  /usr/src/zaptel-1.2.4/pciradio.mod.o
  LD [M]  /usr/src/zaptel-1.2.4/pciradio.ko
  CC  /usr/src/zaptel-1.2.4/tor2.mod.o
  LD [M]  /usr/src/zaptel-1.2.4/tor2.ko
  CC  /usr/src/zaptel-1.2.4/torisa.mod.o
  LD [M]  /usr/src/zaptel-1.2.4/torisa.ko
  CC  /usr/src/zaptel-1.2.4/wcfxo.mod.o
  LD [M]  /usr/src/zaptel-1.2.4/wcfxo.ko
  CC  /usr/src/zaptel-1.2.4/wct1xxp.mod.o
  LD [M]  /usr/src/zaptel-1.2.4/wct1xxp.ko
  CC  /usr/src/zaptel-1.2.4/wct4xxp.mod.o
  LD [M]  /usr/src/zaptel-1.2.4/wct4xxp.ko
  CC  /usr/src/zaptel-1.2.4/wctdm.mod.o
  LD [M]  /usr/src/zaptel-1.2.4/wctdm.ko
  CC  /usr/src/zaptel-1.2.4/wctdm24xxp.mod.o
  LD [M]  /usr/src/zaptel-1.2.4/wctdm24xxp.ko
  CC  /usr/src/zaptel-1.2.4/wcte11xp.mod.o
  LD [M]  /usr/src/zaptel-1.2.4/wcte11xp.ko
  CC  /usr/src/zaptel-1.2.4/wcusb.mod.o
  LD [M]  /usr/src/zaptel-1.2.4/wcusb.ko
  CC  /usr/src/zaptel-1.2.4/zaptel.mod.o
  LD [M]  /usr/src/zaptel-1.2.4/zaptel.ko
  CC

Re: [Asterisk-Users] spa3000 asterisk fxo gateway

2006-03-06 Thread john

With pen in hand, Roberto Pereyra succussfully stormed bulwarks which
others armed with sword and excommunication have been repulsed, and said
...
 Hi

 Somebody knows a tutorial or help me for use a SPA3000 like fxo Asterisk
 interface ?


Here are a couple, although you may need to make some adjustments
depending on your setup, CID, etc. I used the first one, dropping the CID
portion. Works like a charm.

http://nerdvittles.com/index.php?p=65
http://www.geekgazette.com/index.php?option=com_contenttask=viewid=28Itemid=0limit=1limitstart=3

John C.



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[Asterisk-Users] One Extension - Two Calls?

2006-03-06 Thread casasterisk

I'm trying to figure out how to allow an extension to register more than once.  
For instance, I have all of these 4 line IP phones that I use with Asterisk and 
I would like to have a persons extension (say 101) ring at all four lines so 
that if the person is on the phone they can take another call, but it appears 
as though if you try to register the same extension more than once then the 
most recent registration is the only one that works (this determined by calling 
that extension and seeing which 'line' rings).

This would also be handy for those working from home, this way their extension 
follows them whereever they are.

Any thoughts on this?  From what I have read so far it appears as though 
asterisk cannot do this and I wondered if anyone else had done something 
similar.

Thank You!

Craig Shortreed

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[Asterisk-Users] Meetme Participant Announcement

I have the following in extensions.conf:

exten = 1000,1,Meetme(|dMic|)

According to the 'show application meetme' docs:

'i' - announce user join/leave (new in Asterisk 1.2) 

Well, when users join the conference, Asterisk records their name, but does not 
broadcast it into the conference. I have Asterisk version 1.2.4. I know this 
has worked in the past. This sure as heck seems like a bug to me!

Doug.

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[Asterisk-Users] Question: When i Diall a group

2006-03-06 Thread didier


Hello,
This question is probabely recurrent, i apologize, but i haven't found a 
limpid explanation (for me)  in mail list, google, and hum source 
code):
When use the Command DIAL to ring a group, WHERE is stored the name of 
the 'winner' who pick up the call ? ($variable = ?), and, step beyond: 
WHEN (or on EVENT = ?) could we get this variable ?


many threads but nothing very very decisive (use event link, importvar, 
dialpeername (broken), bridged etc etc)

Thank's for your help to get the right process

Best regards
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[Asterisk-Users] Information to program a new driver for Asterisk

2006-03-06 Thread Álvaro Palma

I'm interested in developing a new channel driver for a thrid party 
telephony card for Asterisk. Is there any official document that 
explains how to do this? We've been looking the doc/channel.txt and 
doc/modules.txt in the source, but that's not a very complete source of 
info :)


Thanks a lot for your attention.

--
Atly.
Alvaro Palma

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Re: [Asterisk-Users] Capturing DTMF during a call

2006-03-06 Thread Kristian Kielhofner


Giordano Grandis wrote:

Hi all,
I have a simple and maybe also stupid question: if i'm in coversation on 
a Zap channel and the remote party send me a DTMF, could I capture it?
 
Thanks all
 


*Giordano *


show application Read

--
Kristian Kielhofner
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[Asterisk-Users] NEWS: SIP Firmware Available for Cisco 7970

2006-03-06 Thread Julien Goodwin

I've just recieved a copy of the new SIP firmware for the Cisco 7970,
those of you with Cisco accounts may wish to try it (shock horror I'm
sticking with SCCP).

This coincides with the release of v8 firmware for all Cisco phones (and
for those of you running Sergio's chan_sccp v8 works fine)

The firmware is now also (and for the 7970 SIP, only) distributed in
.cop files, these are actually just tarballs (.tar.gz) with a new
name. The names are mangled, but relativly easy to figure out.

Please note that I will not give this firmware out, nor point people to
places where they may pirate it.

Thanks,
Julien


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Re: [Asterisk-Users] Variable

2006-03-06 Thread C F

Not without some dialplan magic. You could have the setgroup for every
call, then use groupcount to figure out how many.

On 3/5/06, Paul Hales [EMAIL PROTECTED] wrote:

 Is there a variable to read to see how many calls are currently open?
 (related to channel status?)

 PaulH

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[Asterisk-Users] Asterisk on MacOS?

2006-03-06 Thread Christian


Hi,
I am just curious, does anyone know if I can run Asterisk on the Mac? I've 
read something that it should be possible, but cant find an eventual 
download page or what is supported. And also if the Zaptel driver is 
supported as well as Ztdummy.

Many thanks,
Christian 


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RE: [Asterisk-Users] Asterisk on MacOS?

2006-03-06 Thread Colin Anderson

http://www.voip-info.org/tiki-index.php?page=Asterisk%20MacOSX%20Support

It works but it's bitchy as hell to run because of root issues in OSX. I
run it on my Mini. Zaptel is not supported. You have to use an external
gateway of some kind. Zaptel development support is stalled, most likely
because of the Intel thing, the guys working on it are (rightly so) waiting
to see how the new Macs pan out. In theory it should be cake to port the
driver to an Intel mac and hopefully you can take a stock card and plug it
in, but for now, only a SIP/FXS/FXO gateway is the most practical way. 

-Original Message-
From: Christian [mailto:[EMAIL PROTECTED]
Sent: Monday, March 06, 2006 9:30 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk on MacOS?


Hi,
I am just curious, does anyone know if I can run Asterisk on the Mac? I've 
read something that it should be possible, but cant find an eventual 
download page or what is supported. And also if the Zaptel driver is 
supported as well as Ztdummy.
Many thanks,
Christian 

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[Asterisk-Users]chan_zap.c:6570 handle_init_event error

2006-03-06 Thread asterisk183

I used quadBri Junghanns card and I config zaptel.conf: ZAPTEL.CONF loadzone=it defaultzone=it span=1,1,3,ccs,ami span=2,2,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami  bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12  ZAPATA.CONF [channels] language=it musiconhold=default switchtype = euroisdn  ; p2mp TE mode (for connecting ISDN lines in point-to-multipoint mode) signalling = bri_cpe_ptmp ; p2p TE mode (for connecting ISDN lines in point-to-point mode) ;signalling = bri_cpe ; p2mp NT mode (for connecting ISDN phones in point-to-multipoint mode) ;signalling = bri_net_ptmp ; p2p NT mode (for connecting an ISDN pbx in point-to-point mode) ;signalling = bri_net  pridialplan = local prilocaldialplan = local nationalprefix = 0 internationalprefix = 00  echocancel = yes  context=isdn_incoming group
 = 1 channel = 1-2  group = 2 channel = 4-5  group = 3 channel = 7-8  group = 4 channel = 10-11  But when I hangup the channel, Asterisk show this message:  Mar 6 17:31:20 WARNING[1437]: chan_zap.c:6570 handle_init_event: Detected alarm on channel 1: Red Alarm Mar 6 17:31:20 WARNING[1437]: chan_zap.c:1593 zt_disable_ec: Unable to disable echo cancellation on channel 1 Mar 6 17:31:20 WARNING[1437]: chan_zap.c:6570 handle_init_event: Detected alarm on channel 2: Red Alarm Mar 6 17:31:20 WARNING[1437]: chan_zap.c:1593 zt_disable_ec: Unable to disable echo cancellation on channel 2 Mar 6 17:31:20 NOTICE[1433]: chan_zap.c:8511 pri_dchannel: PRI got event: Alarm (4) on Primary D-channel of span 1 Mar 6 17:31:20 NOTICE[1433]: chan_zap.c:8518 pri_dchannel: pri_shutdown Mar 6 17:31:20 NOTICE[1437]: chan_zap.c:6565 handle_init_event: Alarm cleared on
  channel
 1 Mar 6 17:31:20 NOTICE[1437]: chan_zap.c:6565 handle_init_event: Alarm cleared on channel 2 Mar 6 17:31:20 NOTICE[1433]: chan_zap.c:8511 pri_dchannel: PRI got event: No more alarm (5) on Primary D-channel of span 1  Why? And What i can doing for solve this problem?  Thanks 
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[Asterisk-Users] Bad Meetme() Bug

Anyone seen this? If not I guess I'll have to post it as a bug.

Extensions.conf has this:
exten = 123,1,Meetme(|dMic|)

I dial 123, and enter my conference number. Asterisk asks me to enter my name. 
At this point I hang up. If I type at the Asterisk console 'meetme list 12345' 
it shows that I am a participant in the conference evenhough I hung up.

If I dial 123 again and this time do not hang up until after I have joined the 
conference, this does not occur. 'Meetme list' shows 0 participants. 

The fact that it works the second way and not the first would tend to indicate 
that it isn't a SIP messaging problem. If Asterisk gets the BYE while I'm in a 
conference, it should get it when I'm entering a conference.

Doug.
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[Asterisk-Users] Buddy watch?

2006-03-06 Thread rivy strauss

Hi,
I am using Polycom 501 and I came across a problem. As soon as I have
incominglimit=1 in sip.conf, which is necessary for buddy watching, I
cannot transfer calls.  On the console it tells me:
Call from user '3052' rejected due to usage limit of 1. Can someone
please tell me how to get around this problem?

(I don't know if this is relevant, but in the phone.cfg file, I have
reg.1.callsPerLineKey=1 to disable call waiting-- I need a busy signal
to be returned in the dialplan if the phone is busy.)

Thanks in advance for your help!
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Re: [Asterisk-Users] Meetme Participant Announcement

2006-03-06 Thread Doug Lytle


Douglas Garstang wrote:

I have the following in extensions.conf:

exten = 1000,1,Meetme(|dMic|)

According to the 'show application meetme' docs:

'i' - announce user join/leave (new in Asterisk 1.2) 



  

I use:

exten  = 4299,1,Meetme(|Msicp)

Seems to work ok for me.  But, I don't use the second pipe.

Doug


--
Ben Franklin quote:

Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety.


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RE: [Asterisk-Users] Buddy watch?

Why do you need to have to set incominglimit=1 for buddies to work? We've not 
had that requirement.

Doug.

-Original Message-
From: rivy strauss [mailto:[EMAIL PROTECTED]
Sent: Monday, March 06, 2006 9:46 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Buddy watch?

Hi,
I am using Polycom 501 and I came across a problem. As soon as I have
incominglimit=1 in sip.conf, which is necessary for buddy watching, I
cannot transfer calls.  On the console it tells me:
Call from user '3052' rejected due to usage limit of 1. Can someone
please tell me how to get around this problem?

(I don't know if this is relevant, but in the phone.cfg file, I have
reg.1.callsPerLineKey=1 to disable call waiting-- I need a busy signal
to be returned in the dialplan if the phone is busy.)

Thanks in advance for your help!
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[Asterisk-Users] Ringduration problem when calling out via Sip

2006-03-06 Thread Philipp Dreimann

Hello,

when I try to call someone via Sip, the called phone just rings about 25
seconds.

Here's my Outgoing-Context:
snip
exten = _X.,1,Dial(SIP/[EMAIL PROTECTED],120)
exten = s,1,Answer()
exten = s,2,Playback(invalid)
exten = s,3,Hangup()
exten = h,1,Hangup()
/snip

And here's a log that shows the problem. (I call from 11 to 12 via SIP. 12
is also a number for which my asterisk is responsible.)

snip
-- Accepting overlap voice call from '11' to '12' on channel 0/1, span 1
-- Starting simple switch on 'Zap/1-1'
-- Executing Dial(Zap/1-1, SIP/[EMAIL PROTECTED]|120) in new stack
-- Called [EMAIL PROTECTED]
-- Accepting voice call from '' to '12' on channel 0/1, span 2
-- Executing Dial(Zap/4-1, Zap/g1/12|120|rt) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/12
-- Zap/2-1 is proceeding passing it to Zap/4-1
-- SIP/freenet-56c5 is making progress passing it to Zap/1-1
-- Zap/2-1 is ringing
-- Channel 0/1, span 1 got hangup request
  == Spawn extension (extern, 12, 1) exited non-zero on 'Zap/1-1'
-- Executing Hangup(Zap/1-1, ) in new stack
  == Spawn extension (extern, h, 1) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
-- Channel 0/1, span 2 got hangup, cause 16
-- Hungup 'Zap/2-1'
  == Spawn extension (default, 12, 1) exited non-zero on 'Zap/4-1'
-- Executing Hangup(Zap/4-1, ) in new stack
  == Spawn extension (default, h, 1) exited non-zero on 'Zap/4-1'
-- Hungup 'Zap/4-1'
/snip

I'm using Asterisk Version 1.2.4-BRIstuffed-0.3.0-PRE-1k with spandsp
0.0.2pre25.

When I call 12 via my outgoing zap-interface the called phone rings about 2
minutes. It makes no difference if I call a number somewhere else or on my
system.

When the call is established within these 25 seconds everything works as is
should. 

Can someone please give me a hint?

Philipp


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Re: [Asterisk-Users] Asterisk on MacOS?

2006-03-06 Thread Martin Joseph



On Mar 6, 2006, at 8:39 AM, Colin Anderson wrote:

http://www.voip-info.org/tiki-index.php? 
page=Asterisk%20MacOSX%20Support


It works but it's bitchy as hell to run because of root issues in  
OSX.

I wonder what the above root issues means?
 I run it on my Mini. Zaptel is not supported. You have to use an  
external
gateway of some kind. Zaptel development support is stalled, most  
likely
because of the Intel thing, the guys working on it are (rightly so)  
waiting

to see how the new Macs pan out.

Actually they are working on a Unicall implementation instead.

In theory it should be cake to port the
driver to an Intel mac and hopefully you can take a stock card and  
plug it

in, but for now, only a SIP/FXS/FXO gateway is the most practical way.
Yes,  that's so,  although I personally think that might be preferable  
in general as external gateways eliminate many configuration issues and  
make the setup more swappable for service purposes. ie you don't have  
to power a server down to swap one...


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RE: [Asterisk-Users] Meetme Participant Announcement

Hi Doug. I worked it out. I had commented out chan_zap.so in modules.conf as I 
didn't think I needed it. It was doing weird stuff, including not playing the 
participants joining. Weird.

-Original Message-
From: Doug Lytle [mailto:[EMAIL PROTECTED]
Sent: Monday, March 06, 2006 9:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Meetme Participant Announcement

Douglas Garstang wrote:
 I have the following in extensions.conf:

 exten = 1000,1,Meetme(|dMic|)

 According to the 'show application meetme' docs:

 'i' - announce user join/leave (new in Asterisk 1.2) 

I use:

exten  = 4299,1,Meetme(|Msicp)

Seems to work ok for me.  But, I don't use the second pipe.

Doug

-- 
Ben Franklin quote:

Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety.

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RE: [Asterisk-Users] Polycom 501 power over ethernet

2006-03-06 Thread The VoIP Connection

I've seen a lot of IP501 and I've never seen one with a power jack.
According to Polycom they all use the cable.  

Possibly it was an IP500? -Mike

Michael Crown
Managing Partner
www.thevoipconnection.com
321.989.6728 ext. 611
sip:[EMAIL PROTECTED]

 -Original Message-
 From: Douglas Garstang [mailto:[EMAIL PROTECTED] 
 Sent: Monday, March 06, 2006 10:13 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Polycom 501 power over ethernet
 
 No, some IP 501's have the inline cable and some have the power jack.
 
 -Original Message-
 From: Paul Hales [mailto:[EMAIL PROTECTED]
 Sent: Sunday, March 05, 2006 8:59 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Polycom 501 power over ethernet
 
 
 
 The IP300/301 has the power jack, the IP500/501 the inline cable.
 
 PaulH
 
 On Sun, 2006-03-05 at 20:56 -0700, Douglas Garstang wrote:
  Not true. Some do and some don't. Some have a place to plug 
 a separate DC adapter, and some have the inline power, where 
 the adapter plugs into the ethernet cable. Not sure which 
 ones are newer, and which are older.
  
  -Original Message- 
  From: Michael Welter [mailto:[EMAIL PROTECTED] 
  Sent: Sun 3/5/2006 6:50 PM 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Cc: 
  Subject: Re: [Asterisk-Users] Polycom 501 power over ethernet
  
  
  
  The IP501 does not have a power jack.  You'll need one 
 of the Polycom
  cables.
  
  William M Conlon wrote:
   My recollection of the marketing fluff was that we 
 would just use our
   legacy network (cables) and the devices at both ends 
 would figure out
   whether they were sourcing, sinking, or neither.  In 
 the case of the
   501, it's the special Polycom cable, either with or 
 without provision
   for an AC power adapter, that powers the phone.  
 That's what I meant by
   saying the '501' itself is not compliant with 802.3af 
 -- it needs a
   separate thingamajig [tech jargon :)]to be powered.
  
   Anyway I had hoped that I could just plug a CAT-5 
 patch cable from my
   RJ45 wall outlet into the phone.
  
   On Mar 5, 2006, at 5:17 PM, Michael Welter wrote:
  
   As I understand 802.3af, the phones go through a 
 negotiation with the
   unit supplying the power.  I don't think it's a 
 matter of -48VDC on a
   particular pair.  I remember a schematic from years 
 ago--it had each
   of the receive pair and the transmit pair going into 
 a transformer
   winding,  and that winding had a center tap for PoE. 
  This is not
   something that *I* am going to screw with.
  
   The IP501 telephone set is the same for both PoE and 
 local power. 
   With the PoE cable, the 802.3af electronics (the 
 negotiator) is a
   plastic thing in the cable.  For the local power, 
 there is a plastic
   thingie toward the wall end of the cable, and you 
 plug the wall wart
   into the plastic thingie.  Notice the advanced 
 technical jargon here
  
   With local power, there is still only one cable one 
 the desk--the
   power plugs into the cable towards the wall.  Except 
 for a power
   interruption, this has all the advantages of PoE.
  
  
  
   William M Conlon wrote:
   I saw that Polycom offered a cable (not stocked 
 anywhere), at $40 a
   pop for 802.3af connections.  That's what made me 
 think the phone
   itself is NOT 802.3af compliant.
   Presumably, for $40, there's more than a fuse in 
 that special cable.
   On Mar 5, 2006, at 4:31 PM, Paul Hales wrote:
   For Polycom IP500/501's and IP300/301's you need a 
 special polycom POE
   cable.
  
   When you buy Polycom phones you can usually 
 specify POE or powerpack.
  
   PaulH
  
   On Sun, 2006-03-05 at 16:23 -0800, William M Conlon wrote:
   When I bought two Polycom 501 SIP phones, I 
 naively thought they were
   Power-over-Ethernet (IEEE 802.3af) because they 
 were powered over
   ethernet.  Silly me.
  
   Polycom must have some odd voltage or funny way 
 of injecting the
   power, because the POE switch I bought for them 
 (Netgear [EMAIL PROTECTED])
   won't power them, though if I use the 
 Polycom-supplied AC adapter and
   ethernet power injector cable, they work with the 
 switch in either
   its powered or unpowered ports.
  
   Anyhow, I hadn't seen any mention of how people 
 power these phones,
   as I had planned on centralizing phone power on a 
 UPS to supply my
   Asterisk server and POE switch.  Now the question is:
  
   Can the Polycom AC-powered injector be used with 
 a standard ethernet
   patch cable:
  
   switch :: Polycom injector cable :: RJ45 
 coupler :: patch cable ::
   Polycom 501

Re: [Asterisk-Users] Two asterisks on one machine

2006-03-06 Thread Martin Joseph



On Mar 6, 2006, at 6:46 AM, [EMAIL PROTECTED] wrote:


Hi friend,
  I am running asterisk in production and it is being used by many 
people using h323. I cannot afford to change all their configurations. 
Also, the newer asterisk dosenot support inband for h323 properly. 
Thats why I want two asterisks one for backward compatibility and one 
for sip which I want to implement.



Getting a second development box for SIP sounds more sensible to me.  
After all, don't you want to leave the production box alone?



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Re: [Asterisk-Users] grandstream handytone 286 sometimes dials out wrong number

2006-03-06 Thread Martin Joseph



On Mar 6, 2006, at 6:46 AM, Giorgio Incantalupo wrote:


Hi,
I have an asterisk 1.2.1 (on a debian sarge) box with a TDM400P card. 
I connected the TDM400P to a grandstream 286 to use a VoIP provider.
It seems all right except for a little problem: one call every 30 is 
made to a wrong number.

Is there anybody who had the same problem and solved it?

Usually this is DTMF issue?  So make sure the extensions and the HT286 
have the correct DTMF config.  I have some experience with the HT-488 
FXS and that needed to have dtmfmode=rfc2833 in the extensions and the 
configuration on the HT-488 set the same.


Hope this helps,
Marty

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[Asterisk-Users] ring noise at the background

2006-03-06 Thread Baris Simsek


Hi,

While I am talking, if somebody call me, it is ringing at the background 
and I cannot hear well current peer.


Is there anyway to cancel new call notify?

regards,

- bs
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[Asterisk-Users] Polycom voice.gain.tx.analog.handset and asterisk echo

2006-03-06 Thread Wilson Pickett

While I'm asking about the Polycom ip500, the answers for all phones
where mic/handset/headset levels are adjustable would be of interest
to many I'm sure.

For the ip500, the default value for the handset seems to be
voice.gain.tx.analog.handset=3

I've noticed that echo all but goes away when one reduces the mic
volume on almost any phone. My question is, for you users that have
adjusted these levels for the purpose of echo reduction (or anything
else), what did you find optimal for you? I myself have a fairly loud
voice so I don't need any boost. I was considering lowering this value
but I'd love to hear what other have done (or not) and again, on any
SIP phones. Default or tinker?
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[Asterisk-Users] cdr records on transfer

2006-03-06 Thread Christian Benke

Hello!

i'm trying to set up transfer without using the respective
asterisk-function but with the built-in phone functions. my goal is to
have the first callleg billed to the caller and the second callleg to the
callee, who is responsible for the forward(and i can't bill a unknown
caller anyways)

so far it's working without problems, but my cdr's are messed. with the
help of the RDNIS-variable i've been able to set seperate records for each
call-leg with the correct accountcodes, but the billsec are still written
to the first callleg, the second callleg(originated by callee) receives 0
billsec, which is not what i want. the callee(the one who forwards the
call), should be billed.
since the local-channel is passed to the originating channel, it is clear
that the billsec are added to the callers record.
but is there any way to influence this??? since the phones have this
functionality built-in, why should i ask my clients to use some
keycombination to transfer calls and prevent transfer-by-button? As far as
i've understood, the /n-option for the local-channel would do the
behaviour i want - but how could i add it on a moved temporarily?

kind regards
christian
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[Asterisk-Users] call files and cdr I need src different from CallerID(number)

2006-03-06 Thread Thomas

Hi,

if I dial normal with the dial comman I have in my cdr file the peer-name as 
source and the CALLERID (number and name) as I have set it in the dialplan.

Now Iam using call files and Iam using in the file for example:
Callerid: name 333

333 will be used for the field src AND the CALLERID(number) in the cdr file.

So I dont have the choice to set CALLERID(number) different to the peer-name 
(src in the cdr file).

How this can be fixed.

best regards

Thomas




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Re: [Asterisk-Users] Polycom voice.gain.tx.analog.handset and asterisk echo

2006-03-06 Thread Doug Lytle


Wilson Pickett wrote:

While I'm asking about the Polycom ip500, the answers for all phones
where mic/handset/headset levels are adjustable would be of interest
to many I'm sure.

For the ip500, the default value for the handset seems to be
voice.gain.tx.analog.handset=3

I've noticed that echo all but goes away when one reduces the mic
volume on almost any phone. My question is, for you users that have
adjusted these levels for the purpose of echo reduction (or anything

  

I'd be interested in this myself.

Doug


--
Ben Franklin quote:

Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety.


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[Asterisk-Users] PLEASE respond: how to get Asterisk to change coders on RTP handoff??

2006-03-06 Thread Dan Miller




I have a hardwareFXO/FXS which handle my voip calls, and they support 
G723 internally. Asterisk hands off these calls just fine, and everything 
works, as long as I don't wantPBX menues available... The 
problem is, once I want it to return messages, it will only return them as 
GSM... which is fine, since my FXO/FXS support multiple coders. However, 
even though Asterisk lets me specify a list of valid coders, it will only use 
one...

I want Ast to use GSM to playback messages, then when it hands off the call 
to the endpoints, it should tell them to use G723 in the RE-INVITE messages... I 
don't see any way to get it to do this; *is* there some way??

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Re: [Asterisk-Users] Preferred editor(s) dialplan coding?

2006-03-06 Thread David McNett

On 04-Mar-2006, Pete Barnwell wrote:
 Emacs...

On Sat, 2006-03-04 at 01:35 +0100, adibar wrote:
 Vim forever ;-)

http://unix.rulez.org/~calver/pictures/curves.jpg
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[Asterisk-Users] Initiate and monitor multiple calls?

2006-03-06 Thread Ken D'Ambrosio

I'd like to set up a sort-of follow-me: on a call to a given extension,
I'd like to simultaneously call several different numbers, play them all a
prompt upon answering, and monitor for DTMF digit 1.  I know how to get
Dial() to dial multiple numbers, and I know how to play prompts and
monitor for digits... but I don't know how to mix it all together.  Any
pointers on where to start looking?

Thanks!

-Ken

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[Asterisk-Users] Upgrading AAH

2006-03-06 Thread Rolf Brusletto

All - I've a new system, that since it's been in production, has seen a 
few issues, that look like they should be fixed by upgrading asterisk @ 
home to the latest version. I was curious if anybody out there can tell 
me their experiences with this, and what to expect.


Thanks,

Rolf Brusletto
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RE: [Asterisk-Users] NEWS: SIP Firmware Available for Cisco 7970

2006-03-06 Thread Nabeel Jafferali

 I've just recieved a copy of the new SIP firmware for the Cisco 7970,
 those of you with Cisco accounts may wish to try it (shock horror I'm
 sticking with SCCP).

I have a service contract for my 7960 but I don't see 8.x SIP firmware for
it at http://www.cisco.com/cgi-bin/tablebuild.pl/sip-ip-phone7960.

I do see a .cop file for the 7941/7961 8.x SIP load, but nothing for the
7960.

Nabeel

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Re: [Asterisk-Users] Upgrading AAH

2006-03-06 Thread john

With pen in hand, Rolf Brusletto succussfully stormed bulwarks which
others armed with sword and excommunication have been repulsed, and said
...
 All - I've a new system, that since it's been in production, has seen a
 few issues, that look like they should be fixed by upgrading asterisk @
 home to the latest version. I was curious if anybody out there can tell
 me their experiences with this, and what to expect.


Rolf,

I upgraded from 2.2 to 2.4 with only minor issues aferwards.

Back up your /etc/asterisk directory before you do anything, of course,
then untar asteriskathome.tar.gz distro in /var/aah_load directory and run
the install.sh script.

If you use sugar crm then back this up too because it overwrites
everything. You will also have to reset all the passwords as these are set
back to the standard initial passwords that [EMAIL PROTECTED] sets up.

After you're done with the upgrade, just diff all the etc/asteisk conf
files and also check, through the amp portal, all your settings. A few
parts on mine disappeared, but I had saved all the configs so I think my
total time to rebuild and reset everything was under an hour.

Of course, I forgot about sugar, which I use, but not enough to have
remembered to back it up, but that's another issue.

Hope this helps answers your question.

Regards,

John C.


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RE: [Asterisk-Users] NEWS: SIP Firmware Available for Cisco 7970

On Mon, 2006-03-06 at 12:38, Nabeel Jafferali wrote:

 I have a service contract for my 7960 but I don't see 8.x SIP firmware for
 it at http://www.cisco.com/cgi-bin/tablebuild.pl/sip-ip-phone7960.
 
 I do see a .cop file for the 7941/7961 8.x SIP load, but nothing for the
 7960.
 

You have to have developer support contracts to currently get to them.

-Greg

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Re: [Asterisk-Users] NEWS: SIP Firmware Available for Cisco 7970

On 13:38, Mon 06 Mar 06, Nabeel Jafferali wrote:
  I've just recieved a copy of the new SIP firmware for the Cisco 7970,
  those of you with Cisco accounts may wish to try it (shock horror I'm
  sticking with SCCP).
 
 I have a service contract for my 7960 but I don't see 8.x SIP firmware for
 it at http://www.cisco.com/cgi-bin/tablebuild.pl/sip-ip-phone7960.
 
 I do see a .cop file for the 7941/7961 8.x SIP load, but nothing for the
 7960.

The 7960 and the 7970 are 2 different phones...
-- 
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.info
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D

Why is it drug addicts and computer afficionados are both called users?

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[Asterisk-Users] No ring when doing blind transfer.

2006-03-06 Thread Chuck Bunn


Hi,

I have an odd problem when doing a blind transfer. The transfer is 
intiated and the transferred caller hears nothing until the timeout. I 
have tried setting the 'r' and the 'm' variables in the dial command. 
Nothing happens when I use the 'r' variable when I use the 'm' variable 
I briefly hear music on hold and then it stops until the timeout for no 
answer is reached. When the timeout is reached and no on answers the 
system does go to voice mail as expected. I have also tried it without 
either the 'r' or 'm' variables and I get the same results no ring. I am 
using asterisk 1.2.4 with zaptel 1.2.3.


Here are my files:

extensions.conf **

[general]
#include macros.incl
#include outgoing.incl
#include extensions-home.incl
#include menu.incl

[globals]
OUTBOUNDTRUNK=Zap/g1
PSTN1=Zap/1
PSTN2=Zap/2
PSTN3=Zap/5
PSTN4=Zap/6
PHONE1=Zap/3
PHONE2=Zap/4
***

macros.incl ***

[macro-stdexten]
exten = s,1,Set(DYNAMIC_FEATURES=automon)
exten = s,2,Dial(${ARG2},20,Ttw)
exten = s,3,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail([EMAIL PROTECTED])
exten = s-NOANSWER,2,Playback(thank-you-for-callinggoodbye)
exten = s-NOANSWER,3,Hangup
exten = s-BUSY,1,Voicemail([EMAIL PROTECTED])
exten = s-BUSY,2,Playback(thank-you-for-callinggoodbye)
exten = s-BUSY,3,Hangup
exten = s-CHANUNAVAIL,1,Voicemail([EMAIL PROTECTED])
exten = s-CHANUNAVAIL,2,Playback(thank-you-for-callinggoodbye)
exten = s-CHANUNAVAIL,3,Hangup
exten = _s-.,1,Goto(s-NOANSWER,1)

[macro-menuexten]
exten = s,1,Set(DYNAMIC_FEATURES=automon)
exten = s,2,Dial(${ARG2},20,Ttmw)
exten = s,3,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail([EMAIL PROTECTED])
exten = s-NOANSWER,2,Playback(thank-you-for-callinggoodbye)
exten = s-NOANSWER,3,Hangup
exten = s-BUSY,1,Voicemail([EMAIL PROTECTED])
exten = s-BUSY,2,Playback(thank-you-for-callinggoodbye)
exten = s-BUSY,3,Hangup
exten = s-CHANUNAVAIL,1,Voicemail([EMAIL PROTECTED])
exten = s-CHANUNAVAIL,2,Playback(thank-you-for-callinggoodbye)
exten = s-CHANUNAVAIL,3,Hangup
exten = _s-.,1,Goto(s-NOANSWER,1)


[macro-novmail]
exten = s,1,Dial(${ARG2},20,Ttw)
exten = s,2,Playback(thank-you-for-callinggoodbye)
exten = s,3,Hangup
exten = s,102,Playback(thank-you-for-callinggoodbye)
exten = s,103,Hangup
**

extensions-home.incl ***

[default]
;Operator queue, Operator Console, and Receptionist Phone
exten = s,1,Answer()
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout(5)
exten = s,4,ResponseTimeout(30)
exten = s,5,GotoIfTime(8:00-21:00|*|*|*?default,s,7)
exten = s,6,Goto(mainmenu,s,1)
exten = s,7,Queue(extensions-home|tn|||25)
exten = s,8,Goto(mainmenu,s,1)

include = mainmenu

;Ageless
exten = _400,1,Voicemail([EMAIL PROTECTED][EMAIL PROTECTED])
exten = _405,1,Voicemail([EMAIL PROTECTED][EMAIL PROTECTED])
exten = _41[0-3],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _499,1,Macro(novmail,${EXTEN},SIP/${EXTEN})

;Spa Personnel
exten = _500,1,Voicemail([EMAIL PROTECTED][EMAIL PROTECTED])
exten = _51[0],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _590,1,Macro(novmail,${EXTEN},ZAP/3)

;Chicken
;exten = _60[0],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})

;Resedential
;exten = _70[0-3],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})

;Voicemail Main
exten = 800,1,Answer
exten = 800,2,VoicemailMain(@default)

;Agent Login
exten = 801,1,AgentCallbackLogin(||@default)

;Recording Interface
exten = 820,1,Goto(phrase,s,1)

;Voice Conferencing
exten = _85X,1,Answer
exten = _85X,2,MeetMe(${EXTEN})

;Music on Hold
exten = 870,1,Answer
exten = 870,2,SetMusicOnHold(default)
exten = 870,3,WaitMusicOnHold(420)
exten = 870,4,Hangup


Thanks
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Re: [Asterisk-Users] seg fault when skinny phone answers

2006-03-06 Thread Ryan Laginski

Thanks Michiel. I haven't tried chan_sccp in awhile. This weekend, I installed 1.2.5 with the latest sccp. Asterisk no longer cores when the 12 SP, however, there is no audio in either direction. There is one way audio if I dialout from the device, but internal call to call does not work, nor does receiving a call from external zap.
I changed the earlyrtp=ringout as per a mailing list thread, and viewed the debug output on setting 10. Nothing obvious stood out.Regards,-RyanOn 3/5/06, 
Michiel van Baak [EMAIL PROTECTED] wrote:
On 20:19, Sat 04 Mar 06, Ryan Laginski wrote: Downgrade to 1.0.10. I was unable to get the 12sp+ to work reliably in 1.2.0-1.2.4 and had the same problem.You could try the 
chan-sccp.org driver for skinny/sccpThe 12SP+ is listed as supported device.--Michiel van Baak[EMAIL PROTECTED]
http://michiel.vanbaak.infoGnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2DWhy is it drug addicts and computer afficionados are both called users?
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Re: [Asterisk-Users] seg fault when skinny phone answers

On 14:23, Mon 06 Mar 06, Ryan Laginski wrote:
 Thanks Michiel. I haven't tried chan_sccp in awhile. This weekend, I
 installed 1.2.5 with the latest sccp. Asterisk no longer cores when the 12

That's good.

 SP, however, there is no audio in either direction. There is one way audio
 if I dialout from the device, but internal call to call does not work, nor
 does receiving a call from external zap.

That isn't ;)

 
 I changed the earlyrtp=ringout as per a mailing list thread, and viewed the
 debug output on setting 10. Nothing obvious stood out.
 Regards,
 -Ryan

I think your best bet is to ask on the chan_sccp
mailinglist. I know from previous posts ppl use those
devices with succes.
I don't own a 12SP, so can't really help you more.

-- 
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.info
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D

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[Asterisk-Users] agi channel status

2006-03-06 Thread Danish Samad

Hi,

I have developed a custom agi and connect to it by placing a call
through a sip phone. The agi issues the STREAM FILE command from a
number of places in code to play out prerecorded messages. The problem
is if the agi tries to play a file, using the STREAM FILE command,
after the caller has dropped the call, the agi crashes midway.

After issuing a agi debug command on the console here is what I observed.
1. Whenever a caller drops a call no interrupt is fired from asterisk
to the agi to notify a hangup event. This prevents me from taking
precautionary measures before issuing any command.

2. Even after the caller has dropped the call, I still see an
active channel (verified using the show channels command). This
is probably due to the fact that the agi is still connected.

3. When I send the STREAM FILE command from my agi this is what is shown on the asterisk console:

AGI Rx STREAM FILE file1 # 0
Mar 6 23:50:22 WARNING[5978]: file.c:583 ast_readaudio_callback: Failed to write frame
AGI Tx 200 result=-1 endpos=6400
Spawn extension exited non-zero on 'SIP/101-6600'

As I understand, when asterisk receives the STREAM FILE command its has
no channel to play it to and hence the warning is issued.
Asterisk does send a -1 result but straight away kills the agi process
( I deduce this from the last line) which causes the agi to crash.

Now I am totally clueless on how to handle such erroneous conditions. Any help will be appreciated.

Thanks,
Danish

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[Asterisk-Users] Asterisk and CISCO 7970 color

2006-03-06 Thread Diego Mariano Velo









Hi All,



Have you any idea to
configure Cisco 7970 with Asterisk. Please if any of you have the phone
configured, send me any instructions.



Thanks in advance.



Diego.






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[Asterisk-Users] Re: One Extension - Two Calls?

2006-03-06 Thread Bromont -


On my 4-line IP phones I can have 4 simutaneous calls come in with only the 1 
registration. When a second call comes in you push line 2 and Asterisk starts 
music-on-hold on line 1. What kind of IP phones are you using.

I'm trying to figure out how to allow an extension to register more than 
once.  For instance, I have all of these 4 line IP phones that I use with 
Asterisk and I would like to have a persons extension (say 101) ring at all 
four lines so that if the person is on the phone they can take another call, 
but it appears as though if you try to register the same extension more than 
once then the most recent registration is the only one that works (this 
determined by calling that extension and seeing which 'line' rings).

This would also be handy for those working from home, this way their 
extension follows them whereever they are.

Any thoughts on this?  From what I have read so far it appears as though 
asterisk cannot do this and I wondered if anyone else had done something 
similar.


Thank You!

Craig Shortreed


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Re: [Asterisk-Users] Asterisk and CISCO 7970 color

On 17:00, Mon 06 Mar 06, Diego Mariano Velo wrote:
 Hi All,
  
 Have you any idea to configure Cisco 7970 with Asterisk. Please if any
 of you have the phone configured, send me any instructions.

SIP or SCCP ?

-- 
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.info
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D

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[Asterisk-Users] Music on hold volume too high - using built in music on hold.

2006-03-06 Thread Chuck Bunn


Hi,

I saw this problem mentioned before but the user appeared to be using 
the MP3 software with asterisk. I am using the native music on hold 
player in asterisk 1.2 and I too have a volume problem with music on 
hold. Is this controllable through the 'indications.conf'? I know this 
file controls frequency range for various sounds might it also control 
sound level or am I barking up the wrong tree?


Thanks
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Re: [Asterisk-Users] Polling Asterisk for Life

2006-03-06 Thread Geoff Karl

On 3/2/06, Matt Riddell [NZ] [EMAIL PROTECTED] wrote:
 Matt wrote:
  Yup.. that's the exact problem I'm having.   I really can't explain
  what happens.  If I don't restart asterisk it seems to happen after
  about 2 days.   So I restart asterisk once a day at 3am.  And it still
  goes down about once a month...

 Are you guys perchance using Local/[EMAIL PROTECTED] in your installations?

 --
 Cheers,

 Matt Riddell
 ___


Is there a known issue when using the Local/[EMAIL PROTECTED]

thanks,

Geoff
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Re: [Asterisk-Users] NEWS: SIP Firmware Available for Cisco 7970

2006-03-06 Thread asterisk


On Tue, 7 Mar 2006, Julien Goodwin wrote:

I've just recieved a copy of the new SIP firmware for the Cisco 7970,
those of you with Cisco accounts may wish to try it (shock horror I'm
sticking with SCCP).
This coincides with the release of v8 firmware for all Cisco phones (and
for those of you running Sergio's chan_sccp v8 works fine)
The firmware is now also (and for the 7970 SIP, only) distributed in
.cop files, these are actually just tarballs (.tar.gz) with a new
name. The names are mangled, but relativly easy to figure out.
Please note that I will not give this firmware out, nor point people to
places where they may pirate it.


Looks like regular smartnet customers do not get access to 7970 SIP.

The only thing regular smartnet customers get for the 7970 is sccp 8.0.

-Dan
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[Asterisk-Users] call manager integration

2006-03-06 Thread Jerry Geis


I am getting this error from call manager (4.0) and asterisk 1.2.4

I have canreinvite=yes on the call manager setup.

I can call into the asterisk box from call manager. THat seems to work.
When I am calling out of the box using a call file I see 
this entry from call manager...


What might be the problem with my setup?

THanks,

JErry





Date03/06/2006 13:58:36.374/Date 
 ClusterCO-CCMPUB-01-Cluster/Cluster 
 CMHost10.101.66.10/CMHost 
 TraceTypeTrace/TraceType 
 CTag2,100,114,1.347/CTag 
 SrcDev10.66.101.10/SrcDev 
 SrcIpINVITE/SrcIp 
 CTMapKey / 
 CTMapVal / 
 infoCisco CallManagerDigit analysis: wait_DaReq - cepn=[] BlockFlag=[1]/info 
 /trace

- trace
 Date03/06/2006 13:58:36.374/Date 
 ClusterCO-CCMPUB-01-Cluster/Cluster 
 CMHost10.101.66.10/CMHost 
 TraceTypeTrace/TraceType 
 CTag2,100,114,1.347/CTag 
 SrcDev10.66.101.10/SrcDev 
 SrcIpINVITE/SrcIp 
 CTMapKey / 
 CTMapVal / 


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Re: [Asterisk-Users] NEWS: SIP Firmware Available for Cisco 7970

2006-03-06 Thread Aaron Daniel

Ok, so, we've got the 7970 SIP Firmware now, but their readme is a 
little sparse... Anyone have any clue as to the upgrade procedure for a 
non-ccm5 system?  (i.e. asterisk ;))


Aaron

Julien Goodwin wrote:

I've just recieved a copy of the new SIP firmware for the Cisco 7970,
those of you with Cisco accounts may wish to try it (shock horror I'm
sticking with SCCP).

This coincides with the release of v8 firmware for all Cisco phones (and
for those of you running Sergio's chan_sccp v8 works fine)

The firmware is now also (and for the 7970 SIP, only) distributed in
.cop files, these are actually just tarballs (.tar.gz) with a new
name. The names are mangled, but relativly easy to figure out.

Please note that I will not give this firmware out, nor point people to
places where they may pirate it.

Thanks,
Julien




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RE: [Asterisk-Users] NEWS: SIP Firmware Available for Cisco 7970

2006-03-06 Thread Darren Wright

OK.

 I've got the COP SIP filehow do we use this thing on the 7970?

-Darren


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Re: [Asterisk-Users] call manager integration

On Mon, 2006-03-06 at 15:00, Jerry Geis wrote:
 I am getting this error from call manager (4.0) and asterisk 1.2.4
 
 I have canreinvite=yes on the call manager setup.
 
 I can call into the asterisk box from call manager. THat seems to work.
 When I am calling out of the box using a call file I see 
 this entry from call manager...
 
 What might be the problem with my setup?
 

What is the output on the console with sip debug turned on?

-Greg

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Re: [Asterisk-Users] Polycom 501 power over ethernet

2006-03-06 Thread pdhales

I have installed several hundred polycom's, and I have never seen a 500/501
with a power jack.
All with the inline cable, as you mention.

Of course, if someone can provide photo evidence I will stand corrected.

PaulH

- Original Message - 
From: The VoIP Connection [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Tuesday, March 07, 2006 4:26 AM
Subject: RE: [Asterisk-Users] Polycom 501 power over ethernet


 I've seen a lot of IP501 and I've never seen one with a power jack.
 According to Polycom they all use the cable.

 Possibly it was an IP500? -Mike

 Michael Crown
 Managing Partner
 www.thevoipconnection.com
 321.989.6728 ext. 611
 sip:[EMAIL PROTECTED]

  -Original Message-
  From: Douglas Garstang [mailto:[EMAIL PROTECTED]
  Sent: Monday, March 06, 2006 10:13 AM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: RE: [Asterisk-Users] Polycom 501 power over ethernet
 
  No, some IP 501's have the inline cable and some have the power jack.
 
  -Original Message-
  From: Paul Hales [mailto:[EMAIL PROTECTED]
  Sent: Sunday, March 05, 2006 8:59 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: RE: [Asterisk-Users] Polycom 501 power over ethernet
 
 
 
  The IP300/301 has the power jack, the IP500/501 the inline cable.
 
  PaulH
 
  On Sun, 2006-03-05 at 20:56 -0700, Douglas Garstang wrote:
   Not true. Some do and some don't. Some have a place to plug
  a separate DC adapter, and some have the inline power, where
  the adapter plugs into the ethernet cable. Not sure which
  ones are newer, and which are older.
  
   -Original Message- 
   From: Michael Welter [mailto:[EMAIL PROTECTED]
   Sent: Sun 3/5/2006 6:50 PM
   To: Asterisk Users Mailing List - Non-Commercial Discussion
   Cc:
   Subject: Re: [Asterisk-Users] Polycom 501 power over ethernet
  
  
  
   The IP501 does not have a power jack.  You'll need one
  of the Polycom
   cables.
  
   William M Conlon wrote:
My recollection of the marketing fluff was that we
  would just use our
legacy network (cables) and the devices at both ends
  would figure out
whether they were sourcing, sinking, or neither.  In
  the case of the
501, it's the special Polycom cable, either with or
  without provision
for an AC power adapter, that powers the phone.
  That's what I meant by
saying the '501' itself is not compliant with 802.3af
  -- it needs a
separate thingamajig [tech jargon :)]to be powered.
   
Anyway I had hoped that I could just plug a CAT-5
  patch cable from my
RJ45 wall outlet into the phone.
   
On Mar 5, 2006, at 5:17 PM, Michael Welter wrote:
   
As I understand 802.3af, the phones go through a
  negotiation with the
unit supplying the power.  I don't think it's a
  matter of -48VDC on a
particular pair.  I remember a schematic from years
  ago--it had each
of the receive pair and the transmit pair going into
  a transformer
winding,  and that winding had a center tap for PoE.
   This is not
something that *I* am going to screw with.
   
The IP501 telephone set is the same for both PoE and
  local power.
With the PoE cable, the 802.3af electronics (the
  negotiator) is a
plastic thing in the cable.  For the local power,
  there is a plastic
thingie toward the wall end of the cable, and you
  plug the wall wart
into the plastic thingie.  Notice the advanced
  technical jargon here
   
With local power, there is still only one cable one
  the desk--the
power plugs into the cable towards the wall.  Except
  for a power
interruption, this has all the advantages of PoE.
   
   
   
William M Conlon wrote:
I saw that Polycom offered a cable (not stocked
  anywhere), at $40 a
pop for 802.3af connections.  That's what made me
  think the phone
itself is NOT 802.3af compliant.
Presumably, for $40, there's more than a fuse in
  that special cable.
On Mar 5, 2006, at 4:31 PM, Paul Hales wrote:
For Polycom IP500/501's and IP300/301's you need a
  special polycom POE
cable.
   
When you buy Polycom phones you can usually
  specify POE or powerpack.
   
PaulH
   
On Sun, 2006-03-05 at 16:23 -0800, William M Conlon wrote:
When I bought two Polycom 501 SIP phones, I
  naively thought they were
Power-over-Ethernet (IEEE 802.3af) because they
  were powered over
ethernet.  Silly me.
   
Polycom must have some odd voltage or funny way
  of injecting the
power, because the POE switch I bought for them
  (Netgear [EMAIL PROTECTED])
won't power them, though if I use the
  Polycom-supplied AC adapter and
ethernet power injector cable, they work with the
  switch in either
its powered or unpowered ports.
   
Anyhow, I hadn't seen any mention of how people
  power these phones,
as I had planned on centralizing phone power on a
  UPS to supply my

Re: [Asterisk-Users] Polycom 501 power over ethernet

2006-03-06 Thread pdhales

Can you provide a photo of this?

I am interested in seeing it!

PaulH

- Original Message - 
From: Douglas Garstang [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, March 07, 2006 2:13 AM
Subject: RE: [Asterisk-Users] Polycom 501 power over ethernet

 No, some IP 501's have the inline cable and some have the power jack.

 -Original Message-
 From: Paul Hales [mailto:[EMAIL PROTECTED]
 Sent: Sunday, March 05, 2006 8:59 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Polycom 501 power over ethernet

 The IP300/301 has the power jack, the IP500/501 the inline cable.

 PaulH

 On Sun, 2006-03-05 at 20:56 -0700, Douglas Garstang wrote:
  Not true. Some do and some don't. Some have a place to plug a separate
DC adapter, and some have the inline power, where the adapter plugs into the
ethernet cable. Not sure which ones are newer, and which are older.

  -Original Message- 
  From: Michael Welter [mailto:[EMAIL PROTECTED]
  Sent: Sun 3/5/2006 6:50 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Cc:
  Subject: Re: [Asterisk-Users] Polycom 501 power over ethernet

  The IP501 does not have a power jack.  You'll need one of the Polycom
  cables.

  William M Conlon wrote:
   My recollection of the marketing fluff was that we would just use our
   legacy network (cables) and the devices at both ends would figure out
   whether they were sourcing, sinking, or neither.  In the case of the
   501, it's the special Polycom cable, either with or without provision
   for an AC power adapter, that powers the phone.  That's what I meant
by
   saying the '501' itself is not compliant with 802.3af -- it needs a
   separate thingamajig [tech jargon :)]to be powered.

   Anyway I had hoped that I could just plug a CAT-5 patch cable from my
   RJ45 wall outlet into the phone.

   On Mar 5, 2006, at 5:17 PM, Michael Welter wrote:

   As I understand 802.3af, the phones go through a negotiation with the
   unit supplying the power.  I don't think it's a matter of -48VDC on a
   particular pair.  I remember a schematic from years ago--it had each
   of the receive pair and the transmit pair going into a transformer
   winding,  and that winding had a center tap for PoE.  This is not
   something that *I* am going to screw with.

   The IP501 telephone set is the same for both PoE and local power.
   With the PoE cable, the 802.3af electronics (the negotiator) is a
   plastic thing in the cable.  For the local power, there is a plastic
   thingie toward the wall end of the cable, and you plug the wall wart
   into the plastic thingie.  Notice the advanced technical jargon
here

   With local power, there is still only one cable one the desk--the
   power plugs into the cable towards the wall.  Except for a power
   interruption, this has all the advantages of PoE.

   William M Conlon wrote:
   I saw that Polycom offered a cable (not stocked anywhere), at $40 a
   pop for 802.3af connections.  That's what made me think the phone
   itself is NOT 802.3af compliant.
   Presumably, for $40, there's more than a fuse in that special cable.
   On Mar 5, 2006, at 4:31 PM, Paul Hales wrote:
   For Polycom IP500/501's and IP300/301's you need a special polycom
POE
   cable.

   When you buy Polycom phones you can usually specify POE or
powerpack.

   PaulH

   On Sun, 2006-03-05 at 16:23 -0800, William M Conlon wrote:
   When I bought two Polycom 501 SIP phones, I naively thought they
were
   Power-over-Ethernet (IEEE 802.3af) because they were powered over
   ethernet.  Silly me.

   Polycom must have some odd voltage or funny way of injecting the
   power, because the POE switch I bought for them (Netgear [EMAIL 
   PROTECTED])
   won't power them, though if I use the Polycom-supplied AC adapter
and
   ethernet power injector cable, they work with the switch in either
   its powered or unpowered ports.

   Anyhow, I hadn't seen any mention of how people power these
phones,
   as I had planned on centralizing phone power on a UPS to supply my
   Asterisk server and POE switch.  Now the question is:

   Can the Polycom AC-powered injector be used with a standard
ethernet
   patch cable:

   switch :: Polycom injector cable :: RJ45 coupler :: patch
cable ::
   Polycom 501

   which would allow me to power the Polycom AC adapters by my UPS.
Or
   do I need to provide a UPS at each phone and run the ethernet like

   switch :: patch cable :: RJ45 coupler :: Polycom injector
cable ::
   Polycom 501

   thanks.
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Re: [Asterisk-Users] Polycom 501 power over ethernet

2006-03-06 Thread Ken D'Ambrosio

On Mon, March 6, 2006 4:19 pm, [EMAIL PROTECTED] wrote:
 I have installed several hundred polycom's, and I have never seen a
 500/501
 with a power jack. All with the inline cable, as you mention.

 Of course, if someone can provide photo evidence I will stand corrected.

I think the confusion here is the different *ways* the 300/500/600 do PoE:

301 has a power brick, just like (say) a Grandstream.
501 has _almost_ PoE: the cable is (as noted above) in-line, but this
might confuse someone differentiating with the 301.
601 has true PoE, where you've got your PoE switch, a stock Ethernet
cable, and the phone -- nothing else, and no special cabling required.

-Ken (purveyor of fine differentiations)


 PaulH


 - Original Message -
 From: The VoIP Connection [EMAIL PROTECTED]
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 asterisk-users@lists.digium.com
 Sent: Tuesday, March 07, 2006 4:26 AM
 Subject: RE: [Asterisk-Users] Polycom 501 power over ethernet



 I've seen a lot of IP501 and I've never seen one with a power jack.
 According to Polycom they all use the cable.


 Possibly it was an IP500? -Mike


 Michael Crown
 Managing Partner
 www.thevoipconnection.com 321.989.6728 ext. 611
 sip:[EMAIL PROTECTED]


 -Original Message-
 From: Douglas Garstang [mailto:[EMAIL PROTECTED]
 Sent: Monday, March 06, 2006 10:13 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Polycom 501 power over ethernet


 No, some IP 501's have the inline cable and some have the power jack.


 -Original Message-
 From: Paul Hales [mailto:[EMAIL PROTECTED]
 Sent: Sunday, March 05, 2006 8:59 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Polycom 501 power over ethernet




 The IP300/301 has the power jack, the IP500/501 the inline cable.


 PaulH


 On Sun, 2006-03-05 at 20:56 -0700, Douglas Garstang wrote:

 Not true. Some do and some don't. Some have a place to plug

 a separate DC adapter, and some have the inline power, where the
 adapter plugs into the ethernet cable. Not sure which ones are newer,
 and which are older.

 -Original Message-
 From: Michael Welter [mailto:[EMAIL PROTECTED]
 Sent: Sun 3/5/2006 6:50 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Cc:
 Subject: Re: [Asterisk-Users] Polycom 501 power over ethernet




 The IP501 does not have a power jack.  You'll need one

 of the Polycom
 cables.

 William M Conlon wrote:

 My recollection of the marketing fluff was that we

 would just use our
 legacy network (cables) and the devices at both ends
 would figure out
 whether they were sourcing, sinking, or neither.  In
 the case of the
 501, it's the special Polycom cable, either with or

 without provision
 for an AC power adapter, that powers the phone.
 That's what I meant by

 saying the '501' itself is not compliant with 802.3af
 -- it needs a

 separate thingamajig [tech jargon :)]to be powered.

 Anyway I had hoped that I could just plug a CAT-5

 patch cable from my
 RJ45 wall outlet into the phone.


 On Mar 5, 2006, at 5:17 PM, Michael Welter wrote:


 As I understand 802.3af, the phones go through a

 negotiation with the
 unit supplying the power.  I don't think it's a
 matter of -48VDC on a
 particular pair.  I remember a schematic from years
 ago--it had each
 of the receive pair and the transmit pair going into
 a transformer
 winding,  and that winding had a center tap for PoE.
 This is not

 something that *I* am going to screw with.

 The IP501 telephone set is the same for both PoE and

 local power.
 With the PoE cable, the 802.3af electronics (the

 negotiator) is a
 plastic thing in the cable.  For the local power,
 there is a plastic
 thingie toward the wall end of the cable, and you
 plug the wall wart
 into the plastic thingie.  Notice the advanced
 technical jargon here

 With local power, there is still only one cable one

 the desk--the
 power plugs into the cable towards the wall.  Except
 for a power
 interruption, this has all the advantages of PoE.



 William M Conlon wrote:

 I saw that Polycom offered a cable (not stocked

 anywhere), at $40 a
 pop for 802.3af connections.  That's what made me
 think the phone
 itself is NOT 802.3af compliant. Presumably, for $40, there's
 more than a fuse in
 that special cable.
 On Mar 5, 2006, at 4:31 PM, Paul Hales wrote:

 For Polycom IP500/501's and IP300/301's you need a

 special polycom POE
 cable.

 When you buy Polycom phones you can usually

 specify POE or powerpack.

 PaulH


 On Sun, 2006-03-05 at 16:23 -0800, William M Conlon wrote:

 When I bought two Polycom 501 SIP phones, I

 naively thought they were
 Power-over-Ethernet (IEEE 802.3af) because they

 were powered over
 ethernet.  Silly me.

 Polycom must have some odd voltage or funny way

 of injecting the
 power, because the POE switch I bought for them
 (Netgear [EMAIL PROTECTED])

 won't power them, though if I use

[Asterisk-Users] Problem getting two x200p cards working on 1.2.4

2006-03-06 Thread Guillermo Salas M

Hi, I using asterisk 1.2.4 on a CentOS with Linux 2.6.9-22.0.2.ELsmp
 kernel. 

I've two x100p cards connected, only one card is reconigzed by asterisk.

02:01.0 Communication controller: Tiger Jet Network Inc. Tiger3XX
Modem/ISDN interface
02:02.0 Ethernet controller: Davicom Semiconductor, Inc. 21x4x DEC-Tulip
compatible 10/100 Ethernet (rev 31)
02:03.0 Communication controller: Tiger Jet Network Inc. Tiger3XX
Modem/ISDN interface

This is the cli output for zap show channels :

My /etc/zaptel.conf :

# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#

# It must be in the module loading order


# Span 1: WCFXO/1 Generic Clone Board 2
fxsks=1

# Span 2: ZTDUMMY/1 ZTDUMMY/1 1

# Global data

loadzone= us
defaultzone = us


My /etc/asterisk/zapata-auto.conf

; Zaptel Channels Configurations (zapata.conf)
;
; This is not intended to be a complete zapata.conf. Rather, it is
intended
; to be #include-d by /etc/zapata.conf that will include the global
settings
;
callerid=asreceived

; Span 1: WCFXO/1 Generic Clone Board 2
signalling=fxs_ks
; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 1
context=from-pstn
group=0
channel = 1


; Span 2: ZTDUMMY/1 ZTDUMMY/1 1


This is the corresponding 'lspci -vv -n' for my two cards:

02:01.0 Class 0780: e159:0001
Subsystem: 8086:0003
Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop-
ParErr- Stepping- SERR+ FastB2B-
Status: Cap+ 66Mhz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort-
TAbort- MAbort- SERR- PERR-
Latency: 32 (250ns min, 32000ns max)
Interrupt: pin A routed to IRQ 201
Region 0: I/O ports at b800 [size=256]
Region 1: Memory at feaff000 (32-bit, non-prefetchable)
[size=4K]
Capabilities: [40] Power Management version 2
Flags: PMEClk- DSI+ D1- D2+ AuxCurrent=55mA PME(D0
+,D1-,D2+,D3hot+,D3cold+)
Status: D0 PME-Enable- DSel=0 DScale=0 PME-


02:03.0 Class 0780: e159:0001
Subsystem: 8086:0003
Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop-
ParErr- Stepping- SERR+ FastB2B-
Status: Cap+ 66Mhz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort-
TAbort- MAbort- SERR- PERR-
Latency: 32 (250ns min, 32000ns max)
Interrupt: pin A routed to IRQ 177
Region 0: I/O ports at b000 [size=256]
Region 1: Memory at feafd000 (32-bit, non-prefetchable)
[size=4K]
Capabilities: [40] Power Management version 2
Flags: PMEClk- DSI+ D1- D2+ AuxCurrent=55mA PME(D0
+,D1-,D2+,D3hot+,D3cold+)
Status: D0 PME-Enable- DSel=0 DScale=0 PME-



And dmesg shows:

NET: Registered protocol family 10
Disabled Privacy Extensions on device c0340020(lo)
IPv6 over IPv4 tunneling driver
divert: not allocating divert_blk for non-ethernet device sit0
eth0: no IPv6 routers present
Freed a Wildcard
Unregistered Tormenta2
Zapata Telephony Interface Unloaded
Zapata Telephony Interface Registered on major 196
Zaptel Version:  Echo Canceller: KB1
Registered Tormenta2 PCI
Registered tone zone 0 (United States / North America)
Registered tone zone 0 (United States / North America)
Registered tone zone 0 (United States / North America)
Registered tone zone 0 (United States / North America)
ACPI: PCI interrupt :02:01.0[A] - GSI 22 (level, low) - IRQ 201
Failed to initailize DAA, giving up...
wcfxo: probe of :02:01.0 failed with error -5
ACPI: PCI interrupt :02:03.0[A] - GSI 19 (level, low) - IRQ 177
wcfxo: DAA mode is 'FCC'
Found a Wildcard FXO: Generic Clone
Registered tone zone 0 (United States / North America)
Registered tone zone 0 (United States / North America)
Registered tone zone 0 (United States / North America)
Registered tone zone 0 (United States / North America)


Any ideas?

-- 
Guillermo Salas M.
Telconet S.A. Manta
Calle 15 y Av. 24 Esq.
Phone : 593 5 262 8071
Mobile: 593 9 985 5138
SIP   : [EMAIL PROTECTED]
e-mail: [EMAIL PROTECTED]
www   : http://www.telconet.net
http://www.telcocarrier.net

Linux User: 255902
Soporte en Linea en http://www.manta.telconet.net

Please avoid sending me Word or PowerPoint attachments.
See http://www.fsf.org/philosophy/no-word-attachments.html

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Re: [Asterisk-Users] Polycom 501 power over ethernet

2006-03-06 Thread pdhales

Totally correct - according to me at least.

PaulH

- Original Message - 
From: Ken D'Ambrosio [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users@lists.digium.com
Sent: Tuesday, March 07, 2006 8:25 AM
Subject: Re: [Asterisk-Users] Polycom 501 power over ethernet

 On Mon, March 6, 2006 4:19 pm, [EMAIL PROTECTED] wrote:
  I have installed several hundred polycom's, and I have never seen a
  500/501
  with a power jack. All with the inline cable, as you mention.

  Of course, if someone can provide photo evidence I will stand corrected.

 I think the confusion here is the different *ways* the 300/500/600 do PoE:

 301 has a power brick, just like (say) a Grandstream.
 501 has _almost_ PoE: the cable is (as noted above) in-line, but this
 might confuse someone differentiating with the 301.
 601 has true PoE, where you've got your PoE switch, a stock Ethernet
 cable, and the phone -- nothing else, and no special cabling required.

 -Ken (purveyor of fine differentiations)

  PaulH

  - Original Message -
  From: The VoIP Connection [EMAIL PROTECTED]
  To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
  asterisk-users@lists.digium.com
  Sent: Tuesday, March 07, 2006 4:26 AM
  Subject: RE: [Asterisk-Users] Polycom 501 power over ethernet

  I've seen a lot of IP501 and I've never seen one with a power jack.
  According to Polycom they all use the cable.

  Possibly it was an IP500? -Mike

  Michael Crown
  Managing Partner
  www.thevoipconnection.com 321.989.6728 ext. 611
  sip:[EMAIL PROTECTED]

  -Original Message-
  From: Douglas Garstang [mailto:[EMAIL PROTECTED]
  Sent: Monday, March 06, 2006 10:13 AM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: RE: [Asterisk-Users] Polycom 501 power over ethernet

  No, some IP 501's have the inline cable and some have the power jack.

  -Original Message-
  From: Paul Hales [mailto:[EMAIL PROTECTED]
  Sent: Sunday, March 05, 2006 8:59 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: RE: [Asterisk-Users] Polycom 501 power over ethernet

  The IP300/301 has the power jack, the IP500/501 the inline cable.

  PaulH

  On Sun, 2006-03-05 at 20:56 -0700, Douglas Garstang wrote:

  Not true. Some do and some don't. Some have a place to plug

  a separate DC adapter, and some have the inline power, where the
  adapter plugs into the ethernet cable. Not sure which ones are newer,
  and which are older.

  -Original Message-
  From: Michael Welter [mailto:[EMAIL PROTECTED]
  Sent: Sun 3/5/2006 6:50 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Cc:
  Subject: Re: [Asterisk-Users] Polycom 501 power over ethernet

  The IP501 does not have a power jack.  You'll need one

  of the Polycom
  cables.

  William M Conlon wrote:

  My recollection of the marketing fluff was that we

  would just use our
  legacy network (cables) and the devices at both ends
  would figure out
  whether they were sourcing, sinking, or neither.  In
  the case of the
  501, it's the special Polycom cable, either with or

  without provision
  for an AC power adapter, that powers the phone.
  That's what I meant by

  saying the '501' itself is not compliant with 802.3af
  -- it needs a

  separate thingamajig [tech jargon :)]to be powered.

  Anyway I had hoped that I could just plug a CAT-5

  patch cable from my
  RJ45 wall outlet into the phone.

  On Mar 5, 2006, at 5:17 PM, Michael Welter wrote:

  As I understand 802.3af, the phones go through a

  negotiation with the
  unit supplying the power.  I don't think it's a
  matter of -48VDC on a
  particular pair.  I remember a schematic from years
  ago--it had each
  of the receive pair and the transmit pair going into
  a transformer
  winding,  and that winding had a center tap for PoE.
  This is not

  something that *I* am going to screw with.

  The IP501 telephone set is the same for both PoE and

  local power.
  With the PoE cable, the 802.3af electronics (the

  negotiator) is a
  plastic thing in the cable.  For the local power,
  there is a plastic
  thingie toward the wall end of the cable, and you
  plug the wall wart
  into the plastic thingie.  Notice the advanced
  technical jargon here

  With local power, there is still only one cable one

  the desk--the
  power plugs into the cable towards the wall.  Except
  for a power
  interruption, this has all the advantages of PoE.

  William M Conlon wrote:

  I saw that Polycom offered a cable (not stocked

  anywhere), at $40 a
  pop for 802.3af connections.  That's what made me
  think the phone
  itself is NOT 802.3af compliant. Presumably, for $40, there's
  more than a fuse in
  that special cable.
  On Mar 5, 2006, at 4:31 PM, Paul

[Asterisk-Users] call manager integration

2006-03-06 Thread Jerry Geis


On Mon, 2006-03-06 at 15:00, Jerry Geis wrote:
/ I am getting this error from call manager (4.0) and asterisk 1.2.4
// 
// I have canreinvite=yes on the call manager setup.
// 
// I can call into the asterisk box from call manager. THat seems to work.
// When I am calling out of the box using a call file I see 
// this entry from call manager...
// 
// What might be the problem with my setup?
// 
/

What is the output on the console with sip debug turned on?



-Greg




greg

here is some of the output. I am no longer the to spcifically do sip 
debug but this is what I have.

along with my sip.conf snip.

The call to extension 3726 never rings. so it never gets answered.


co-drpage-01*CLI -- Attempting call on SIP/CallManager//3726 for 
[EMAIL PROTECTED]:1 
mailto:[EMAIL PROTECTED]:1 (Retry 1)

   Channel SIP/CallManager-03a0 was never answered.
co-drpage-01*CLI Mar  6 13:57:49 NOTICE[4283]: pbx_spool.c:270 
attempt_thread: Call failed to go through, reason 8
co-drpage-01*CLI Mar  6 13:58:24 WARNING[4298]: cdr.c:548 
ast_cdr_disposition: Cause not handled
   -- Executing AGI(OutgoingSpoolFailed, smvoice|-digium_failed) in 
new stack

   -- Launched AGI Script /var/lib/asterisk/agi-bin/smvoice
co-drpage-01*CLI -- Attempting call on SIP/CallManager//3726 for 
[EMAIL PROTECTED]:1 
mailto:[EMAIL PROTECTED]:1 (Retry 1)
co-drpage-01*CLI   == Spawn extension (smvoice-dialout, failed, 1) 
exited non-zero on 'OutgoingSpoolFailed'
Mar  6 13:58:25 NOTICE[4298]: pbx_spool.c:270 attempt_thread: Call 
failed to go through, reason 8

co-drpage-01*CLI Channel SIP/CallManager-9209 was never answered.
co-drpage-01*CLI co-drpage-01*CLI
co-drpage-01*CLI Mar  6 13:58:00 WARNING[4290]: cdr.c:548 
ast_cdr_disposition: Cause not handled
co-drpage-01*CLI co-drpage-01*CLI -- Executing 
AGI(OutgoingSpoolFailed, smvoice|-digium_failed) in new stack
co-drpage-01*CLI co-drpage-01*CLI -- Launched AGI Script 
/var/lib/asterisk/agi-bin/smvoice
co-drpage-01*CLI co-drpage-01*CLI   == Spawn extension 
(smvoice-dialout, failed, 1) exited non-zero on 'OutgoingSpoolFailed'
co-drpage-01*CLI co-drpage-01*CLI Mar  6 13:58:01 NOTICE[4290]: 
pbx_spool.c:270 attempt_thread: Call failed to go through, reason 8

co-drpage-01*CLI co-drpage-01*CLI co-drpage-01*CLI

co-drpage-01*CLI -- Attempting call on SIP/CallManager//3726 for 
[EMAIL PROTECTED]:1 
mailto:[EMAIL PROTECTED]:1 (Retry 1)

co-drpage-01*CLI Channel SIP/CallManager-11f2 was never answered.
co-drpage-01*CLI Mar  6 13:58:12 WARNING[4294]: cdr.c:548 
ast_cdr_disposition: Cause not handled
co-drpage-01*CLI -- Executing AGI(OutgoingSpoolFailed, 
smvoice|-digium_failed) in new stack
co-drpage-01*CLI -- Launched AGI Script 
/var/lib/asterisk/agi-bin/smvoice
co-drpage-01*CLI   == Spawn extension (smvoice-dialout, failed, 1) 
exited non-zero on 'OutgoingSpoolFailed'
co-drpage-01*CLI Mar  6 13:58:13 NOTICE[4294]: pbx_spool.c:270 
attempt_thread: Call failed to go through, reason 8
co-drpage-01*CLI -- Attempting call on SIP/CallManager//3726 for 
[EMAIL PROTECTED]:1 
mailto:[EMAIL PROTECTED]:1 (Retry 1)

co-drpage-01*CLI Channel SIP/CallManager-03a0 was never answered.
co-drpage-01*CLI Mar  6 13:58:24 WARNING[4298]: cdr.c:548 
ast_cdr_disposition: Cause not handled
co-drpage-01*CLI -- Executing AGI(OutgoingSpoolFailed, 
smvoice|-digium_failed) in new stack
co-drpage-01*CLI -- Launched AGI Script 
/var/lib/asterisk/agi-bin/smvoice
co-drpage-01*CLI   == Spawn extension (smvoice-dialout, failed, 1) 
exited non-zero on 'OutgoingSpoolFailed'
co-drpage-01*CLI Mar  6 13:58:25 NOTICE[4298]: pbx_spool.c:270 
attempt_thread: Call failed to go through, reason 8

co-drpage-01*CLI co-drpage-01*CLI

   -- Attempting call on SIP/CallManager//3726 for 
[EMAIL PROTECTED]:1 
mailto:[EMAIL PROTECTED]:1 (Retry 1)

   Channel SIP/CallManager-7dcc was never answered.
Mar  6 13:58:36 WARNING[4302]: cdr.c:548 ast_cdr_disposition: Cause not 
handled
   -- Executing AGI(OutgoingSpoolFailed, smvoice|-digium_failed) in 
new stack

   -- Launched AGI Script /var/lib/asterisk/agi-bin/smvoice
 == Spawn extension (smvoice-dialout, failed, 1) exited non-zero on 
'OutgoingSpoolFailed'
Mar  6 13:58:37 NOTICE[4302]: pbx_spool.c:270 attempt_thread: Call 
failed to go through, reason 8

co-drpage-01*CLI


sip.conf
   [CallManager]
   type=friend
   host=10.101.66.10
   context=from_call_manager
   disallow=all
   allow=alaw
   allow=ulaw
   allow=gsm
   dtmfmode=rfc2833
   canreinvite=yes


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Re: [Asterisk-Users] Polycom 501 power over ethernet

2006-03-06 Thread William M Conlon

Maybe this can conclude the thread.  This powering arrangement works  
for me:


Netgear FS108 :: Polycom injector cable :: RJ45 coupler :: patch  
cable :: Polycom 501


Some notes:
1.  The Polycom injector cable should be plugged into a POE port on  
the switch (the Netgear FS108 switch has both powered and unpowered  
ports), or the Polycom injector will not source power.

2.  The Netgear FS108 is NOT sourcing power.
3.  The patch cable is a 50-foot CAT5.
3.  To beat a dead horse, the Polycom 501 itself, is NOT a POE phone,  
IMHO.  Caveat emptor.


bill

On Mar 6, 2006, at 1:32 PM, [EMAIL PROTECTED] wrote:


Totally correct - according to me at least.

PaulH

- Original Message -
From: Ken D'Ambrosio [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc: [EMAIL PROTECTED]; Asterisk Users Mailing  
List -

Non-Commercial Discussion asterisk-users@lists.digium.com
Sent: Tuesday, March 07, 2006 8:25 AM
Subject: Re: [Asterisk-Users] Polycom 501 power over ethernet



On Mon, March 6, 2006 4:19 pm, [EMAIL PROTECTED] wrote:

I have installed several hundred polycom's, and I have never seen a
500/501
with a power jack. All with the inline cable, as you mention.

Of course, if someone can provide photo evidence I will stand  
corrected.


I think the confusion here is the different *ways* the 300/500/600  
do PoE:


301 has a power brick, just like (say) a Grandstream.
501 has _almost_ PoE: the cable is (as noted above) in-line, but this
might confuse someone differentiating with the 301.
601 has true PoE, where you've got your PoE switch, a stock  
Ethernet
cable, and the phone -- nothing else, and no special cabling  
required.


-Ken (purveyor of fine differentiations)



PaulH


- Original Message -
From: The VoIP Connection [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Tuesday, March 07, 2006 4:26 AM
Subject: RE: [Asterisk-Users] Polycom 501 power over ethernet




I've seen a lot of IP501 and I've never seen one with a power jack.
According to Polycom they all use the cable.


Possibly it was an IP500? -Mike


Michael Crown
Managing Partner
www.thevoipconnection.com 321.989.6728 ext. 611
sip:[EMAIL PROTECTED]



-Original Message-
From: Douglas Garstang [mailto:[EMAIL PROTECTED]
Sent: Monday, March 06, 2006 10:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Polycom 501 power over ethernet


No, some IP 501's have the inline cable and some have the power  
jack.



-Original Message-
From: Paul Hales [mailto:[EMAIL PROTECTED]
Sent: Sunday, March 05, 2006 8:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Polycom 501 power over ethernet




The IP300/301 has the power jack, the IP500/501 the inline cable.


PaulH


On Sun, 2006-03-05 at 20:56 -0700, Douglas Garstang wrote:


Not true. Some do and some don't. Some have a place to plug


a separate DC adapter, and some have the inline power, where the
adapter plugs into the ethernet cable. Not sure which ones are  
newer,

and which are older.


-Original Message-
From: Michael Welter [mailto:[EMAIL PROTECTED]
Sent: Sun 3/5/2006 6:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: Re: [Asterisk-Users] Polycom 501 power over ethernet




The IP501 does not have a power jack.  You'll need one


of the Polycom

cables.

William M Conlon wrote:


My recollection of the marketing fluff was that we


would just use our

legacy network (cables) and the devices at both ends

would figure out

whether they were sourcing, sinking, or neither.  In

the case of the

501, it's the special Polycom cable, either with or


without provision

for an AC power adapter, that powers the phone.

That's what I meant by


saying the '501' itself is not compliant with 802.3af

-- it needs a


separate thingamajig [tech jargon :)]to be powered.

Anyway I had hoped that I could just plug a CAT-5


patch cable from my

RJ45 wall outlet into the phone.


On Mar 5, 2006, at 5:17 PM, Michael Welter wrote:



As I understand 802.3af, the phones go through a


negotiation with the

unit supplying the power.  I don't think it's a

matter of -48VDC on a

particular pair.  I remember a schematic from years

ago--it had each

of the receive pair and the transmit pair going into

a transformer

winding,  and that winding had a center tap for PoE.

This is not


something that *I* am going to screw with.

The IP501 telephone set is the same for both PoE and


local power.

With the PoE cable, the 802.3af electronics (the


negotiator) is a

plastic thing in the cable.  For the local power,

there is a plastic

thingie toward the wall end of the cable, and you

plug the wall wart

into the plastic thingie.  Notice the advanced

technical jargon here


With local power, there is still only one cable one


the

Re: [Asterisk-Users] NEWS: SIP Firmware Available for Cisco 7970

2006-03-06 Thread Mailing List

tar zxfv *.cop

- Original Message - 
From: Aaron Daniel [EMAIL PROTECTED]

To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Monday, March 06, 2006 4:00 PM
Subject: Re: [Asterisk-Users] NEWS: SIP Firmware Available for Cisco 7970

Ok, so, we've got the 7970 SIP Firmware now, but their readme is a 
little sparse... Anyone have any clue as to the upgrade procedure for a 
non-ccm5 system?  (i.e. asterisk ;))

Aaron

Julien Goodwin wrote:

I've just recieved a copy of the new SIP firmware for the Cisco 7970,
those of you with Cisco accounts may wish to try it (shock horror I'm
sticking with SCCP).

This coincides with the release of v8 firmware for all Cisco phones (and
for those of you running Sergio's chan_sccp v8 works fine)

The firmware is now also (and for the 7970 SIP, only) distributed in
.cop files, these are actually just tarballs (.tar.gz) with a new
name. The names are mangled, but relativly easy to figure out.

Please note that I will not give this firmware out, nor point people to
places where they may pirate it.

Thanks,
Julien

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Re: [Asterisk-Users] NEWS: SIP Firmware Available for Cisco 7970

2006-03-06 Thread Mailing List


It's just a tarball, extract it
tar zxfv *.cop

_
Mobilcom
http://www.mobilcom.net

- Original Message - 
From: Darren Wright [EMAIL PROTECTED]

To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Monday, March 06, 2006 4:03 PM
Subject: RE: [Asterisk-Users] NEWS: SIP Firmware Available for Cisco 7970


OK.

I've got the COP SIP filehow do we use this thing on the 7970?

-Darren


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Re: [Asterisk-Users] call manager integration

On Mon, 2006-03-06 at 15:42, Jerry Geis wrote:

 here is some of the output. I am no longer the to spcifically do sip 
 debug but this is what I have.
 along with my sip.conf snip.
 
 The call to extension 3726 never rings. so it never gets answered.
 

Are you sure your sip trunk and route pattern are in the same
partition/CSS by chance?

Without more info (AGI script and SIP debug), I really can't be much
more help.  Your sip.conf entry is good though.

Your callmanager context from extensions.conf will help as well.

-Greg

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Re: [Asterisk-Users] Weird DTMF issue

2006-03-06 Thread Andrew Kohlsmith

On Monday 27 February 2006 19:36, Joshua M Thompson wrote:
 1.2.4 now. It was on 1.2.3 but upgrading asterisk and zaptel was the
 first thing I tried when we noticed the problem this morning.

So you were on 1.2.3, it worked and you went to 1.2.4 and it didn't?

-A.
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Re: [Asterisk-Users] NEWS: SIP Firmware Available for Cisco 7970