Re: [asterisk-users] Polycom IP 501 is displaying wrong time

[Crazy Boy]

Hi Noah,
 
 Thank you for your response. As you said, I tried to enter -18000 in GMT 
offset field. But, its not taking input from the phone dial pad or key board. 
Its giving chance to select the value from -12 to 12. I dont enter any SNTP 
Server. Is it must? How can I solve this problem? Can you tell me?
 
 Thank you.
 
 Regards,
 Chandra.

Noah Miller <[EMAIL PROTECTED]> wrote: Hi Chandra -

> This is Chandra. I have Polycom IP 501 phone. Its showing wrong time on the
> display screen. How can I set the "New York" time? What value I have to give
> to GMT offset value?

The GMT offset value is in seconds.  So, for example, the value to use
for EST is -18000, because EST is -5 hours from GMT (-5 x 3600 =
-18000).


- Noah
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RE: [asterisk-users] Polycom IP 501 is displaying wrong time

[Crazy Boy]

Hi,

Thank you for your response. As you said, I set it for -5. But, its displaying 
wrong time. I don't enter any SNTP Server. Is it must? How can I solve this 
problem? Can you tell me?

Thank you.

Regards,
Chandra.

Steve Totaro <[EMAIL PROTECTED]> wrote:v\:* 
{behavior:url(#default#VML);} o\:* {behavior:url(#default#VML);} w\:* 
{behavior:url(#default#VML);} .shape {behavior:url(#default#VML);}  
st1\:*{behavior:url(#default#ieooui) }   You can use the web interface 
and set it to -5 gmt.  Google for free NTP servers.  I used to use 
time.nist.gov and got mixed results.  I found another one that works almost all 
of the time.
   
Thanks,
 Steve Totaro
 http://www.asteriskhelpdesk.com
 KB3OPB
   
  

-
  
  From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Crazy Boy
 Sent: Thursday, April 19, 2007 7:17 AM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Polycom IP 501 is displaying wrong time
  
   
  Hi,
 
 This is Chandra. I have Polycom IP 501 phone. Its showing wrong time on the 
display screen. How can I set the "New   York" time? What value I have to give 
to GMT offset value?
 
 Look forward to your response. Thank you.
 
 Regards,
 Chandra.


-
  
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 Check out new cars at Yahoo! Autos. 
  
  
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Re: [asterisk-users] ztdummy does not load properly at server startup

[Theo Band]

Tzafrir Cohen wrote:

  On Thu, Apr 19, 2007 at 09:23:48PM +0200, Theo Band wrote:
  
  
Eric "ManxPower" Wieling wrote:


  In the zaptel source "make config" will install the zaptel init script
in /etc/rc.d/init.d for many distros.

  

Thanks. This was the missing configuration step. A manual start/stop
seems to work. I will try to reboot the machine tomorrow for the "final"
test but I feel this is going to work.

Stopping the zaptel drivers does not work properly, but I only need this
for rebooting...

  
  
You don't. You don't need to unload the zaptel modules before rebooting.
  

make config installs both a start and stop entry for me.
/etc/init.d/zaptel stop unload the kernel modules. I don't see the
point of trying to do that even if you switch runlevel.
The error just looks ugly when rebooting, but it's not really an issue
of course.

  
  
  
[EMAIL PROTECTED] zaptel-1.4.1]# service zaptel stop
Unloading zaptel hardware drivers: wcusb wctdm wcfxo wctdm24xxp wcte11xp
wct1xxp wct4xxp tor2.
Removing zaptel module:  ERROR: Module zaptel is in use by ztdummy
   [FAILED]

  
  
The unload_module function in the init.d script in the SVN should fix
this and get rid of that error.

  




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Re: [asterisk-users] HPEC audio clipping

[Olivier]

Kevin P. Fleming wrote:

> The code is done and initially tested; it is being reviewed internally
> and should be available on Friday or Monday.



Will this code be available in 1.2 and 1.4 versions alike ?
I can testify it's needed in 1.2.

Best regards
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Re: [asterisk-users] ztdummy does not load properly at server startup

[Theo Band]

Darryl Dunkin wrote:

  
  
  It's not playing a wav file at all, it
is mixing the live audio from all of the callers in that conference
room and sending it back out to them.
  
  
  

I understand. What I tried to say is that if a wav file can be played
at the correct speed, why would a conference application need a special
driver to achieve the same? I assume it is needed as part of the
hardware driver and that this application happens to use the timing
reference part of it.

Theo


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Re: [asterisk-users] 3rd T1 of quad card won't change signaling

[Tzafrir Cohen]

On Thu, Apr 19, 2007 at 02:44:21PM -0700, Jay Wilton wrote:
> Hello,
> 
> I'm trying to set the 3rd span of a new digium quad card as
> a E&M T1 for Faxes to a Hylafax server. The 1st and 2nd
> spans are working as PRIs. When I start asterisk, the logs
> show a signaling error and chan_zap.c dies. I also get an
> error that it can't read the gains but they are the
> standard shown below.
> 
> 2.6 kernel, Debian Stable, * 1.2 svn from feb 2007
> 
> my procedure:
> make changes to zaptel.conf zapata.conf
> rmmod wct4xxp
> modprobe wct4xxp
> ztcfg -vv #shows 1+2 span as PRI, 3rd span as E&M

And does not give any error?

> asterisk -vvc
> 
> ###Error log
> logger.c: -- Registered channel 47, PRI Signalling
> signalling
> chan_zap.c: Signalling requested on channel 49 is PRI
> Signalling but line is in E & M Immediate signalling
> chan_zap.c: Unable to register channel '49-72'
> loader.c: chan_zap.so: load_module failed, returning -1
> 
> --ZAPTEL.CONF---
> span=1,1,0,esf,b8zs
> bchan=1-23
> dchan=24
> 
> #span=2,2,1,esf,b8zs #have tried this way as well
> span=2,1,0,esf,b8zs
> bchan=25-47
> dchan=48
> 
> span=3,0,0,esf,b8zs
> e&m=49-72
> 
> ---ZAPATA.CONF
> [channels]
> language=en
> usecallerid=yes
> callerid=asreceived
> callwaiting=no
> relaxdtmf=no
> group=0
> callgroup=0
> faxdetect=no
> 
> rxgain=0
> txgain=0
> echocancel=yes
> echocancelwhenbridged=yes
> echotraining=600
> jitterbuffers=6
> 
> amaflags=billing
> context=from-pstn
> switchtype=national
> signalling=pri_cpe
> channel => 1-23
> 
> group=1
> channel => 25-47
> 
> ; NEW FAX t1
> group=3
> signaling=em_w
> context=from-internal
> channel => 49-72
> 
> 
> Thanks for any tips or glaring oversights on my part.
> JJ
> 
> 
> __
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Re: [asterisk-users] ZT_CHANCONFIG failed on channel 1: No suchdevice or address

[Tzafrir Cohen]

On Fri, Apr 20, 2007 at 06:54:22AM +1200, Cameron Beattie wrote:
> >
> >What is the output of:
> >
> >ls -l /sys/class/zaptel
> >
> ls -l /sys/class/zaptel
> total 0
> drwxr-xr-x 2 root root 0 Apr 19 20:05 zapchannel
> drwxr-xr-x 2 root root 0 Apr 19 20:05 zapctl
> drwxr-xr-x 2 root root 0 Apr 19 20:05 zappseudo
> drwxr-xr-x 2 root root 0 Apr 19 20:05 zaptimer
> drwxr-xr-x 2 root root 0 Apr 19 20:05 zaptranscode

So the udev configuration is *not* the issue. Maybe it is misconfigured,
but we haven't gotten to it yet ;-) .

The channel driver was not loaded or has failed to find hardware. What 
is the output of:

  lsmod | grep ^zaptel

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Re: [asterisk-users] ztdummy does not load properly at server startup

[Tzafrir Cohen]

On Thu, Apr 19, 2007 at 09:23:48PM +0200, Theo Band wrote:
> Eric "ManxPower" Wieling wrote:
> > In the zaptel source "make config" will install the zaptel init script
> > in /etc/rc.d/init.d for many distros.
> >
> Thanks. This was the missing configuration step. A manual start/stop
> seems to work. I will try to reboot the machine tomorrow for the "final"
> test but I feel this is going to work.
> 
> Stopping the zaptel drivers does not work properly, but I only need this
> for rebooting...

You don't. You don't need to unload the zaptel modules before rebooting.

> 
> [EMAIL PROTECTED] zaptel-1.4.1]# service zaptel stop
> Unloading zaptel hardware drivers: wcusb wctdm wcfxo wctdm24xxp wcte11xp
> wct1xxp wct4xxp tor2.
> Removing zaptel module:  ERROR: Module zaptel is in use by ztdummy
>[FAILED]

The unload_module function in the init.d script in the SVN should fix
this and get rid of that error.

-- 
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Re: [asterisk-users] ztdummy does not load properly at server startup

[Tzafrir Cohen]

On Thu, Apr 19, 2007 at 09:45:10PM +0300, Tzafrir Cohen wrote:
> On Thu, Apr 19, 2007 at 06:28:51PM +0200, Theo Band wrote:
> > Hi
> > 
> > I run asterisk 1.4.2 with zaptel 1.4.1.
> > Zaptel is only needed for the ztdummy driver to get the Meetme()
> > application to work. I don't have any specific hardware.
> > And it does work nicely. When I reboot the machine however I have to
> > manually load the driver like this:
> > 
> > [EMAIL PROTECTED] ~]# modprobe ztdummy
> > Notice: Configuration file is /etc/zaptel.conf
> > line 0: Unable to open master device '/dev/zap/ctl'
> > 
> > 1 error(s) detected
> 
> The error message is meaningless for ztdummy. Please test if you have a
> working timing source with:
> 
>   zttest -v
> 
> To get rid of this error, remove the line that runs an automatic ztcfg
> at the load of ztdummy:  sed -i '/ztdummy/d'

Sorry, my typo here:

On redhats:

  sed -i '/ztdummy/d' /etc/modprobe.conf

On Debians and probably most others:
  
  sed -i '/ztdummy/d' /etc/modprobe.d/zaptel

Basically: delete the line that runs 'ztcfg' automatically after the
modprobe of ztdummy. While for "real" hardware modules there is some
point in rnning this (and I still wouldn't recommend it), for zaptel,
ztdynamic, ztdummy and xpp_usb this is totally pointless and has indeed 
been removed  in the SVN versions.

-- 
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Re: [asterisk-users] Newbie Question about E1

[Paul Hales]

Your best bet is a dual port E1 card - set one side to pri_net and the
other to pri_cpe.

PaulH


On Fri, 2007-04-20 at 01:52 -0300, Luar Roji wrote:
> Hi everybody. I'm about to ask a newbie question, be warned!
> 
> I have a NEC 2000 IPS PBX connected to a E1. 
> 
> Now I want to set up an asterisk, with some digium card connected to that E1.
> (suggestions about the card? I'll have maybe another E1 more).
> 
> The "newbie question" is.. How can I connect the asterisk PC with the PBX? I
> don't know so much about E1, so I don't know if a E1 card in the PC can do
> as a telephone company, or you can't do that.. I'm clear with the question?
> 
> I'll make a little diagram:
> 
> This is what I have now.
> 
> 
> +---++---+ ---> phone 1
> | ANTEL ||NEC PBX| ---> phone 2
> +---+ E1 +---+ ---> ...
> 
> What I want is:
> 
> +---+++   +---+ ---> phone 1
> | ANTEL ||Asterisk box|---|NEC PBX| ---> phone 2
> +---+ E1 +| ? +---+ ---> ...
> 
> (ANTEL is my local phone company)
> 
> Thanks!
> 
> Another not very related question.. this NEC PBX says "Internet Protocol
> Server". Is there a way to connect it to the asterisk?
> 
> Thanks again!
> 
> --
> Luar Roji
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[asterisk-users] Cell phone that can be connected to standad phone switch network

[Joseph]

Are there any cell phone (gadgets) that can be connected to standard
switch phone network?  (ability to check email would be a plus).

Digium adapter S101i can be connected to any network and it allow a
standard phone to act as your local extension over the Internet (by
registering to asterisk), it works "almost" perfectly.
So it would be handy to have a cellular phone that can be connected to
standard switched phone network, are there any toys like this?

-- 
#Joseph
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Re: [asterisk-users] Asterisk 1.4.2 connection to Nortel CS1000M

[Denis Shaposhnikov]

On 04/19/07 23:07, Jerry Geis wrote:


I have asterisk 1.4.2 connected to a Nortel CS1000M using SIP.



The call gets placed, the phone rings, when I answer the phone it hangs up.


I have something like this for another hardware. Try to use 1.2 branch 
of Asterisk. It works for me.


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Re: [asterisk-users] MySQL Update from exten

[Benchev]

> Barton Fisher wrote:
> > I've tried every combination I could find on the net
> and so far there is
> > no joy
> > The thing is I can do this update from the command
> line:  Maybe some new
> > eyes might find the answer?
> >
> > exten => update,1,MYSQL(Connect connid localhost root
> password dax)
> > exten => update,n,MYSQL(QUERY resultid ${connid}
> UPDATE\ caller\ SET\
> >
>
lastcall=${LASTCALL}\,totalcalls=totalcalls+1\,currentcalls=currentcalls+1\
> > WHERE\ dnis=\'${IVR-Exten}\'\ AND\
> ani=\'${CALLERID(number)}\')
> > exten => update,n,MYSQL(Clear ${resultid})
> > exten => update,n,MYSQL(Disconnect ${connid})
> >
> > Asterisk logs says:
> > Apr 19 15:50:05 VERBOSE[19740] logger.c: -- Executing
> > MYSQL("SIP/5400-b7bbfaf0", "QUERY resultid 201 UPDATE
> caller SET
> > lastcall= 04/18/07 11:12:55, totalcalls= totalcalls+1,
> currentcalls=
> > currentcalls+1 WHERE dnis= '7690' AND ani= '5400'") in
> new stack
> > Apr 19 15:50:05 WARNING[18333] app_addon_sql_mysql.c:
> Identifier 200,
> > identifier_type 2 not found in identifier list
> > Apr 19 15:50:05 WARNING[18333] app_addon_sql_mysql.c:
> Invalid result
> > identifier 200 passed in aMYSQL_clear
> >

Probably

exten => update,n,MYSQL(QUERY resultid ${connid} UPDATE\
caller\ SET\
lastcall=${LASTCALL}\,totalcalls=${TOTALCALLS}+1\,currentcalls={CURRENTCALLS}+1\
 WHERE\ dnis=\'${IVR-Exten}\'\ AND\
 ani=\'${CALLERID(number)}\')

Benchev


-

Най-добрият начин да научаваш новините,
които те интересуват.Бързо лесно и безплатно!
новини с филтър
http://www.radar.bg/

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[asterisk-users] Newbie Question about E1

[Luar Roji]

Hi everybody. I'm about to ask a newbie question, be warned!

I have a NEC 2000 IPS PBX connected to a E1. 

Now I want to set up an asterisk, with some digium card connected to that E1.
(suggestions about the card? I'll have maybe another E1 more).

The "newbie question" is.. How can I connect the asterisk PC with the PBX? I
don't know so much about E1, so I don't know if a E1 card in the PC can do
as a telephone company, or you can't do that.. I'm clear with the question?

I'll make a little diagram:

This is what I have now.


+---++---+ ---> phone 1
| ANTEL ||NEC PBX| ---> phone 2
+---+ E1 +---+ ---> ...

What I want is:

+---+++   +---+ ---> phone 1
| ANTEL ||Asterisk box|---|NEC PBX| ---> phone 2
+---+ E1 +| ? +---+ ---> ...

(ANTEL is my local phone company)

Thanks!

Another not very related question.. this NEC PBX says "Internet Protocol
Server". Is there a way to connect it to the asterisk?

Thanks again!

--
Luar Roji
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[asterisk-users] SLA with SIP only configuration

[Yehavi Bourvine +972-8-9489444]

Hello,

  I would like to understand how SLA is working, but all the examples are using
trunk of ZAP type and stations of SIP. However, in my case the stations are SIP
and the outgoing connection to PSTN is SIP also (link to a Cisco gateway).

  can anyone send me a simple configuration of how to make SLA between two SIP
phones without dependency of anything else? It seems I lack some basic
understanding of thios feature and maybe such an example can help.

  Thanks! __Yehavi:
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Re: [asterisk-users] users.conf SIP registration fails

[0xception]

OHH that makes a lot more sense... thanks a lot... i can just stick w/ real
time SIP/IAX configuration then and forget about the users.conf file

On 4/19/07, dave cantera <[EMAIL PROTECTED]> wrote:


0xception,
yes, I suspect this is the reason.  the users.conf file may not used as
you expect.  the users.conf file in 1.4 is a source file for generating
the dialplan on-the-fly by the gui...   if you hand edit it, the gui
doesn't regenerate it...  one way to verify this is to, at the CLI>, type
 dialplan show [EMAIL PROTECTED]
substitute the extens and context to suite your needs... I see that
these should be 6058 exten and context=default in the conf you provided..
daveC


0xception wrote:
> I recently upgraded from asterisk 1.2.13 to 1.4.2 and am looking at
> using the users.conf file to setup my users, before i was using real
> time SIP which worked fine. However when i create a user in users.conf
> i am unable to register the user form a softphone, however that same
> softphone can still register a different the users i currently have
> setup form the sip.conf from real time. i've recieved different error
> messages at different times but none of them stick and i can't seem to
> reproduce much of them. (for example for a while i was getting
> host=dynamic not found/known errors. then i was getting something
> about disallow errors, now i dont receive any errors but when i turn
> on sip debugging it just say unauthorized.
>
> below is my users.conf file
>
> [6058]
> allow = all
> callwaiting = yes
> cid_number = 6058
> context = default
> email = [EMAIL PROTECTED] 
> fullname = Bob smith
> group =
> hasagent = yes
> hasdirectory = yes
> hasiax = no
> hasmanager = no
> hassip = yes
> hasvoicemail = yes
> host=dynamic
> mailbox = 6058
> secret = myPassword
> threewaycalling = yes
> vmsecret = 1337
> zapchan =
> registeriax = no
> registersip = yes
>
>
> Any idea why this doesn't work but registering normal SIP configured
> accounts does?
> 
>
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> 
>
> No virus found in this incoming message.
> Checked by AVG Free Edition.
> Version: 7.5.446 / Virus Database: 269.5.1/765 - Release Date:
04/17/2007 05:20 PM
>

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Re: [asterisk-users] users.conf SIP registration fails

[dave cantera]

0xception,
yes, I suspect this is the reason.  the users.conf file may not used as 
you expect.  the users.conf file in 1.4 is a source file for generating 
the dialplan on-the-fly by the gui...   if you hand edit it, the gui 
doesn't regenerate it...  one way to verify this is to, at the CLI>, type

dialplan show [EMAIL PROTECTED]
substitute the extens and context to suite your needs... I see that 
these should be 6058 exten and context=default in the conf you provided..

daveC


0xception wrote:
I recently upgraded from asterisk 1.2.13 to 1.4.2 and am looking at 
using the users.conf file to setup my users, before i was using real 
time SIP which worked fine. However when i create a user in users.conf 
i am unable to register the user form a softphone, however that same 
softphone can still register a different the users i currently have 
setup form the sip.conf from real time. i've recieved different error 
messages at different times but none of them stick and i can't seem to 
reproduce much of them. (for example for a while i was getting 
host=dynamic not found/known errors. then i was getting something 
about disallow errors, now i dont receive any errors but when i turn 
on sip debugging it just say unauthorized.


below is my users.conf file

[6058]
allow = all
callwaiting = yes
cid_number = 6058
context = default
email = [EMAIL PROTECTED] 
fullname = Bob smith
group =
hasagent = yes
hasdirectory = yes
hasiax = no
hasmanager = no
hassip = yes
hasvoicemail = yes
host=dynamic
mailbox = 6058
secret = myPassword
threewaycalling = yes
vmsecret = 1337
zapchan =
registeriax = no
registersip = yes


Any idea why this doesn't work but registering normal SIP configured 
accounts does?



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No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.5.446 / Virus Database: 269.5.1/765 - Release Date: 04/17/2007 05:20 
PM
  


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Re: [asterisk-users] Failed to authenticate on INVITE

[dave cantera]

ango,
can you provide some sip.conf and extens.conf info?
daveC

Rilawich Ango wrote:

hi,
 I have 2 asterisks with the following configuration.
asterisk server 1 (S1) has an user 9002
asterisk server 2 (S2) has an user 9003
Both users can make call to each other without problem.
Now I add both users to both servers, i.e.
asterisk server 1 (S1) has users 9002,9003
asterisk server 2 (S2) has users 9002,9003
When 9002 dials 9003, Dial(SIP/[EMAIL PROTECTED]) or visa versa.  Both processes
failed to make call with the following error.
Apr 16 11:55:41 NOTICE[19658]: chan_sip.c:9802 handle_response_invite:
Failed to authenticate on INVITE to '"9002"
;tag=as2ff0c493'
Any solution to let them call each others?
ango
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[asterisk-users] Failed to authenticate on INVITE

[Rilawich Ango]

hi,
 I have 2 asterisks with the following configuration.
asterisk server 1 (S1) has an user 9002
asterisk server 2 (S2) has an user 9003
Both users can make call to each other without problem.
Now I add both users to both servers, i.e.
asterisk server 1 (S1) has users 9002,9003
asterisk server 2 (S2) has users 9002,9003
When 9002 dials 9003, Dial(SIP/[EMAIL PROTECTED]) or visa versa.  Both processes
failed to make call with the following error.
Apr 16 11:55:41 NOTICE[19658]: chan_sip.c:9802 handle_response_invite:
Failed to authenticate on INVITE to '"9002"
;tag=as2ff0c493'
Any solution to let them call each others?
ango
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Re: [asterisk-users] MySQL Update from exten

[Lee Jenkins]

Barton Fisher wrote:
I've tried every combination I could find on the net and so far there is 
no joy
The thing is I can do this update from the command line:  Maybe some new 
eyes might find the answer?


exten => update,1,MYSQL(Connect connid localhost root password dax)
exten => update,n,MYSQL(QUERY resultid ${connid} UPDATE\ caller\ SET\ 
lastcall=${LASTCALL}\,totalcalls=totalcalls+1\,currentcalls=currentcalls+1\ 
WHERE\ dnis=\'${IVR-Exten}\'\ AND\ ani=\'${CALLERID(number)}\')

exten => update,n,MYSQL(Clear ${resultid})
exten => update,n,MYSQL(Disconnect ${connid})

Asterisk logs says:
Apr 19 15:50:05 VERBOSE[19740] logger.c: -- Executing 
MYSQL("SIP/5400-b7bbfaf0", "QUERY resultid 201 UPDATE caller SET 
lastcall= 04/18/07 11:12:55, totalcalls= totalcalls+1, currentcalls= 
currentcalls+1 WHERE dnis= '7690' AND ani= '5400'") in new stack
Apr 19 15:50:05 WARNING[18333] app_addon_sql_mysql.c: Identifier 200, 
identifier_type 2 not found in identifier list
Apr 19 15:50:05 WARNING[18333] app_addon_sql_mysql.c: Invalid result 
identifier 200 passed in aMYSQL_clear


I understand what the warning message is really saying

Bart


I don't use mySQL much, but if lastcall is a Timestamp field, don't you 
have to quote it out?


--

Warm Regards,

Lee



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Re: [asterisk-users] Asterisk 1.4.2 + Cisco 7960G not registering

[Jason Howk]

Not that it's probably going to help a whole heck of a lot, but...

Cisco7960> show reg

LINE REGISTRATION TABLE
Proxy Registration: ENABLED, state: REGISTERED
line  APR  state  timer   expires proxy:port
  ---  -  --  --

1 111  REGISTERED 3595954 192.168.0.222:5060
2 111  REGISTERED 3595957 192.168.0.222:5060
3 111  REGISTERED 3595958 192.168.0.222:5060
4 ...  NONE   0   0   undefined:0
5 ...  NONE   0   0   undefined:0
6 ...  NONE   0   0   undefined:0
1-BU  111  REGISTERED 3595944 192.168.0.222:5060

Note: APR is Authenticated, Provisioned, Registered

We don't have any firewall issues as everything behind ours, so I can't
help there.  If you want/need anything, config files, commands run, just
let me know.  I'll be glad to help.

--Jason.

David Olsen wrote:
> On 2007-04-19 at 13:09:51, Doug Lytle <[EMAIL PROTECTED]> wrote:
>> Do you see anything weird when logging (telnet to the ip) into the phone 
>> and doing a show register?
> 
> I see as follows:
> 
> cisco-7960> show reg
> 
> LINE REGISTRATION TABLE
> Proxy Registration: ENABLED, state: REGISTERING
> line  APR  state  timer   expires proxy:port
>   ---  -  --  --  
> 1 .1x  REGISTERING120 17  10.2.5.11:5060
> 2 .1x  REGISTERING120 17  10.2.5.11:5060
> 3 ...  NONE   0   0   undefined:0
> 4 ...  NONE   0   0   undefined:0
> 5 ...  NONE   0   0   undefined:0
> 6 ...  NONE   0   0   undefined:0
> 1-BU  .1x  REGISTERING120 16  10.2.5.11:5060
> 
> Note: APR is Authenticated, Provisioned, Registered
> 
> -d
> 
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RE: [asterisk-users] Re: Trigger for unavailable SIP peer

[Yuan LIU]

From: "C F" <[EMAIL PROTECTED]>
Date: Thu, 19 Apr 2007 09:35:13 -0400

Thank you all for your response, but it appears that some of you
didn't understand my question. I know I can schedule a cron to check
the status (I can even use asterisk -rx "sip show peers" | grep
UNREACHABLE if I use a cron) but that is not what I want. I want
either a way that just as asterisk prints to the CLI  the following:
Peer '120' is now UNREACHABLE!  Last qualify: 118
it should also be able to trigger whatever action from a conf file or the 
like.


I think you can start a dial plan loop from a call file upon asterisk start 
just for this purpose.  Then you should be able to use dial plan logic to 
take action.  Still not out-of-box, but adds a little more flexibility than 
cron (in the sense of less programming, not in ultimate control).


Yuan Liu


Or if there is an available solution even that involves a cron job but
already has all the options, so I don't have to reinvent the wheel.


On 4/18/07, C F <[EMAIL PROTECTED]> wrote:

I use qualify in sip.conf and need to setup a trigger when asterisk
sees it as unreachable, so that I can either drop a call file, or send
an email, or both. How can I do that?

Thank you



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Re: [asterisk-users] 3rd T1 of quad card won't change signaling

[Richard Lyman]

try using this in zaptel.conf

span=3,0,0,d4,ami

*snipped


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Re: [asterisk-users] 3rd T1 of quad card won't change signaling

[Barton Fisher]

Looks like:

amaflags=billing
switchtype=national

is being carry-over from prior PRI.. (All PRI stuff) Try moving below 
before the first PRI?


; NEW FAX t1
group=3
signaling=em_w
context=from-internal
channel => 49-72



Bart

Jay Wilton wrote:

Hello,

I'm trying to set the 3rd span of a new digium quad card as
a E&M T1 for Faxes to a Hylafax server. The 1st and 2nd
spans are working as PRIs. When I start asterisk, the logs
show a signaling error and chan_zap.c dies. I also get an
error that it can't read the gains but they are the
standard shown below.

2.6 kernel, Debian Stable, * 1.2 svn from feb 2007

my procedure:
make changes to zaptel.conf zapata.conf
rmmod wct4xxp
modprobe wct4xxp
ztcfg -vv #shows 1+2 span as PRI, 3rd span as E&M
asterisk -vvc

###Error log
logger.c: -- Registered channel 47, PRI Signalling
signalling
chan_zap.c: Signalling requested on channel 49 is PRI
Signalling but line is in E & M Immediate signalling
chan_zap.c: Unable to register channel '49-72'
loader.c: chan_zap.so: load_module failed, returning -1

--ZAPTEL.CONF---
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24

#span=2,2,1,esf,b8zs #have tried this way as well
span=2,1,0,esf,b8zs
bchan=25-47
dchan=48

span=3,0,0,esf,b8zs
e&m=49-72

---ZAPATA.CONF
[channels]
language=en
usecallerid=yes
callerid=asreceived
callwaiting=no
relaxdtmf=no
group=0
callgroup=0
faxdetect=no

rxgain=0
txgain=0
echocancel=yes
echocancelwhenbridged=yes
echotraining=600
jitterbuffers=6

amaflags=billing
context=from-pstn
switchtype=national
signalling=pri_cpe
channel => 1-23

group=1
channel => 25-47

; NEW FAX t1
group=3
signaling=em_w
context=from-internal
channel => 49-72


Thanks for any tips or glaring oversights on my part.
JJ


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[asterisk-users] users.conf SIP registration fails

[0xception]

I recently upgraded from asterisk 1.2.13 to 1.4.2 and am looking at using
the users.conf file to setup my users, before i was using real time SIP
which worked fine. However when i create a user in users.conf i am unable to
register the user form a softphone, however that same softphone can still
register a different the users i currently have setup form the sip.conf from
real time. i've recieved different error messages at different times but
none of them stick and i can't seem to reproduce much of them. (for example
for a while i was getting host=dynamic not found/known errors. then i was
getting something about disallow errors, now i dont receive any errors but
when i turn on sip debugging it just say unauthorized.

below is my users.conf file

[6058]
allow = all
callwaiting = yes
cid_number = 6058
context = default
email = [EMAIL PROTECTED]
fullname = Bob smith
group =
hasagent = yes
hasdirectory = yes
hasiax = no
hasmanager = no
hassip = yes
hasvoicemail = yes
host=dynamic
mailbox = 6058
secret = myPassword
threewaycalling = yes
vmsecret = 1337
zapchan =
registeriax = no
registersip = yes


Any idea why this doesn't work but registering normal SIP configured
accounts does?
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[asterisk-users] MySQL Update from exten

[Barton Fisher]

I've tried every combination I could find on the net and so far there is 
no joy
The thing is I can do this update from the command line:  Maybe some new 
eyes might find the answer?


exten => update,1,MYSQL(Connect connid localhost root password dax)
exten => update,n,MYSQL(QUERY resultid ${connid} UPDATE\ caller\ SET\ 
lastcall=${LASTCALL}\,totalcalls=totalcalls+1\,currentcalls=currentcalls+1\ 
WHERE\ dnis=\'${IVR-Exten}\'\ AND\ ani=\'${CALLERID(number)}\')

exten => update,n,MYSQL(Clear ${resultid})
exten => update,n,MYSQL(Disconnect ${connid})

Asterisk logs says:
Apr 19 15:50:05 VERBOSE[19740] logger.c: -- Executing 
MYSQL("SIP/5400-b7bbfaf0", "QUERY resultid 201 UPDATE caller SET 
lastcall= 04/18/07 11:12:55, totalcalls= totalcalls+1, currentcalls= 
currentcalls+1 WHERE dnis= '7690' AND ani= '5400'") in new stack
Apr 19 15:50:05 WARNING[18333] app_addon_sql_mysql.c: Identifier 200, 
identifier_type 2 not found in identifier list
Apr 19 15:50:05 WARNING[18333] app_addon_sql_mysql.c: Invalid result 
identifier 200 passed in aMYSQL_clear


I understand what the warning message is really saying

Bart


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RE: [asterisk-users] incoming SIP call

[Bala Neelakantan]

Well, for outbound calls, the SIP Server challenges the INVITE with 401/407.
Then Re-INVITE is sent which explains why outgoing call works.  It is
possible that the SIP Server doesn’t check to see whether the caller is
Registered.

 

For inbound call, the SIP server needs to know the gateway contact
information, and it is obtained only through REGISTER (if not statically
configured in the SIP server, which is very unlikely).

 

Thanks,

Neel

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jean Marc Le
Fevre
Sent: Thursday, April 19, 2007 1:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] incoming SIP call

 

Hello and thanks for answering,

 

As I just answer to Yuan LIU, what I don't understand, is that I can place
an outbound call from asterisk to a gsm at the same time I can't get
asterisk thought a inbound call. But I'll try what you advice me.

I'll tell you the result of it 

 

Jean-Marc LE FEVRE

 

 

 

Le 19 avr. 07 à 20:08, Bala Neelakantan a écrit :





If your SIP server loses REGISTERs then it cant place an inbound SIP call.
Try changing the REGI STER frequency to lower value.

 

When you see incoming SIP call fail, you might want to check whether the
REGISTERs are working.

 

Thanks,

Neel

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jean Marc Le
Fevre
Sent: Wednesday, April 18, 2007 11:15 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] incoming SIP call

 

Hello all, 

 

 

I'm having a quite simple configuration like: 

 

SIP provider <=> asterisk SIP <=> lan

 

Everythings works fine but sometime I can't get incoming call.

 

here are some of the logs from set debug 25 set verbosity 25 sip show debug
and sip.conf and a part of extension.conf

thanks in advance

 

 

Reliably Transmitting (NAT) to 212.27.52.5:5060:

OPTIONS sip:freephonie.net SIP/2.0

Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;branch= z9hG4bK67c2df66;rport

From: "asterisk" ;tag=as01265eaf

To: & lt;sip:freephonie.net>

Contact: 

Call-ID: [EMAIL PROTECTED]

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Wed, 18 Apr 2007 13:57:55 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Length: 0

 

 

---

12 headers, 0 lines 

Reliably Transmitting (NAT) to 212.27.52.5:5060:

OPTIONS sip:freephonie.net SIP/2.0

Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;b ranch=z9hG4bK253c1a3d;rport

From: "asterisk" ;tag=as372da2cb

To: 

Contact: 

Call-ID: [EMAIL PROTECTED]

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Wed, 18 Apr 2007 13:57:55 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Length: 0

 

 

---

Zpro*CLI> 

<-- S IP read from 212.27.52.5:5060: 

SIP/2.0 403 not registered

Call-ID: 7263e88c20c9f3

[EMAIL PROTECTED]

CSeq: 102 OPTIONS

From: "asterisk" ;tag=as01265eaf

To: ;tag=00-31057-001dc 208-591e1ca81

Via: SIP/2.0/UDP
82.XXX.XXX.XXX:5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK67c2df6
6

Content-Length: 0

 

 

--- (7 headers 0 lines) ---

Destroying call '[EMAIL PROTECTED]
 '

Zpro*CLI> 

<-- SIP read from 212.27.52.5:5060: 

SIP/2.0 403 not registered

Call-ID: [EMAIL PROTECTED]

CSeq: 102 OPTIONS

From: "asteris k" ;tag=as372da2cb

To: ;tag=00-32700-001dc209-6fc2b3303

Via: SIP/2.0/UDP
82.XXX.XXX.XXX:5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK253c1a3
d

Content-Length: 0

 

--- (7 headers 0 lines) ---

Destroying call '[EMAIL PROTECTED]'

 

 

sip.conf

 

[general]

context=incoming

realm=etatcritik.dyndns.org

bindport=5060

bindaddr=0.0.0.0

srvlookup=no

maxexpiry=3600

defaultexpiry=1800

videosupport=yes

disallow=all

all ow=ulaw

allow=ilbc

allow=alaw

allow=gsm

musicclass=default

language=fr

useragent=Asterisk PBX 

dtmfmode = auto

register => 09:[EMAIL PROTECTED]

registertimeout=40

externip = 82.XXX.XXX.XXX

localnet=10.XXX.XXX.XXX/255.255.255.0

qualify=6

nat = yes

[test]

type=friend

username=test

secret=test

host=dynamic

context=home

callerid =test <>

dmtfmode=rfc2833

authuser=test

fromuser=test

allow=all

[freephonie_outbound]

type=peer

allow=all

host=freephonie.net

secret=SECRET

fromuser=09XXX

username=09XXX

dtmfmode=inband

quali fy=6

fromdomain=freephonie.net

[freep honie_inbound]

type=peer

context=incoming

host=freephonie.net

qualify=6

allow=all< /P> 

deny=0.0.0.0/0..0.0.0

permit=212.27.52.5/255.255.255.255 ; ip de freephonie.net

 

etension.conf

 

 

...

[incoming]

exten => s,1,Ringing

exten => s,2,Noop(I receive a sip call);

exten => s,n,Goto(home,1000,1)

exten => s,n,Congestion

;

...

 

 

 

 

 

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Re: [asterisk-users] RE: OT (a little): IPV6 Ramifications Article

[Remco Post]

Hans Witvliet wrote:

> The only obstacles currently, are the ISP's.

Any decent ISP (eg. XS4All.nl) will give you an ipv6 address as well as
an ipv4 address.

> afaik, all dsl-modems currently can only work with v4. 
> (correct me if i'm wrong)
> 

So, let the modem be a modem, not a router, and do ip where it belongs,
on the host/router. Now, for the more intresting questions, where to
find a decent ipv6 firewall (yes bleeding edge linux kernels have
one, sort of)

> Only option currently is: tunnelbroker.net 
> 
> Perhaps if we actually get directly fibre or ethernet to our home, 
> ipv6 will get mainstream.
> 
> hw
> (Well, if * would support it, without any extra patches, would help)
> 

And that is a problem, far to many new applications still get developed
for ipv4 only, rather than using the generic interfaces available


-- 
Met vriendelijke groeten,

Remco Post

SARA - Reken- en Netwerkdiensten  http://www.sara.nl
High Performance Computing  Tel. +31 20 592 3000Fax. +31 20 668 3167
PGP Key fingerprint = 6367 DFE9 5CBC 0737 7D16  B3F6 048A 02BF DC93 94EC

"I really didn't foresee the Internet. But then, neither did the
computer industry. Not that that tells us very much of course - the
computer industry didn't even foresee that the century was going to
end." -- Douglas Adams
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Re: [asterisk-users] CallerID masking

[Edoardo Serra]

Hello Rob,
  try to set che MONITOR_FILENAME as something containing the internal 
extension befor emasking the CID


hope it helps

Edoardo

Rob Schall ha scritto:

Hello all,

I currently have all outgoing calls set to mask the caller id so it will
always appear to be coming from our main number. The problem I'm having
though, is with both the call detail in mysql and with the automon
(recording) feature. It shows the originating number as the number I
masked it to, rather than the actual person calling. How can I go about
having both the destination see our main number, but our internally
logging see the correct #?


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--
Ing. Edoardo Serra
WeBRainstorm S.r.l.
Via Pio Foà 83/C
10126 - Torino

Tel: +39 011 678 100
Fax: +39 011 678 275

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Re: [asterisk-users] Transfercapability DIGITAL

[robert boardman]

Hi Chris

I'm using Zap hardware , the second leg is always speech, and the far 
end anwsers and sets up a data call but there is no data transfered back 
so the call is dropped


Regards
Robb

Christoph Fürstaller wrote:

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi robb,

Have you just seen the bearer capability in asterisk or is the call nat
working? I've seen that a digital call shows up as speech.

You are using Zap? Or are you using mISDN? Cause there you have to set
an extra parameter in the dial statement.

chris...

robert boardman schrieb:
  

yes and it is still set to speech

I've even tried to port the old patch here
http://bugs.digium.com/view.php?id=6251 to the system with no luck

robb



Melcon Moraes wrote:


Have you tried:

exten => s,n,SetTransferCapability(DIGITAL)

?

[]'s
MM

 -Original Message-
From:   robert boardman <[EMAIL PROTECTED]>
To: Asterisk Users Mailing List - Non-Commercial Discussion

Cc: Sent:  Tue, 17 Apr 2007 23:17:13 +0100
Delivered:  Tue,  17 Apr 2007 19:15:09 Subject:[asterisk-users]
Transfercapability DIGITAL

Hi

I have a requirement to bridge Digital ISDN call through an asterisk
box but no matter what I setup in the dial plan the second leg of the
zap bridge is always set to Transfer Capability of SPEECH, I wondered
if any one has come across this and managed to fix it?

Thanks in advance for your help

Robb
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RE: [asterisk-users] ztdummy does not load properly at server startup

[Darryl Dunkin]

It's not playing a wav file at all, it is mixing the live audio from all
of the callers in that conference room and sending it back out to them.



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Theo Band
Sent: Thursday, April 19, 2007 13:36
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ztdummy does not load properly at server
startup



Or add /dev/null

Why would one application need a special driver? What so
different about the Meetme() application? Playing a wav file doesn't
need a special timing source for instance. But, I'm just a simple end
user of course, not understanding all the complex details of a PBX :-)


Theo 

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[asterisk-users] 3rd T1 of quad card won't change signaling

[Jay Wilton]

Hello,

I'm trying to set the 3rd span of a new digium quad card as
a E&M T1 for Faxes to a Hylafax server. The 1st and 2nd
spans are working as PRIs. When I start asterisk, the logs
show a signaling error and chan_zap.c dies. I also get an
error that it can't read the gains but they are the
standard shown below.

2.6 kernel, Debian Stable, * 1.2 svn from feb 2007

my procedure:
make changes to zaptel.conf zapata.conf
rmmod wct4xxp
modprobe wct4xxp
ztcfg -vv #shows 1+2 span as PRI, 3rd span as E&M
asterisk -vvc

###Error log
logger.c: -- Registered channel 47, PRI Signalling
signalling
chan_zap.c: Signalling requested on channel 49 is PRI
Signalling but line is in E & M Immediate signalling
chan_zap.c: Unable to register channel '49-72'
loader.c: chan_zap.so: load_module failed, returning -1

--ZAPTEL.CONF---
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24

#span=2,2,1,esf,b8zs #have tried this way as well
span=2,1,0,esf,b8zs
bchan=25-47
dchan=48

span=3,0,0,esf,b8zs
e&m=49-72

---ZAPATA.CONF
[channels]
language=en
usecallerid=yes
callerid=asreceived
callwaiting=no
relaxdtmf=no
group=0
callgroup=0
faxdetect=no

rxgain=0
txgain=0
echocancel=yes
echocancelwhenbridged=yes
echotraining=600
jitterbuffers=6

amaflags=billing
context=from-pstn
switchtype=national
signalling=pri_cpe
channel => 1-23

group=1
channel => 25-47

; NEW FAX t1
group=3
signaling=em_w
context=from-internal
channel => 49-72


Thanks for any tips or glaring oversights on my part.
JJ


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Re: [asterisk-users] CallerID masking

[Alex Balashov]

On Thu, 19 Apr 2007, Rob Schall said something to this effect:


Hello all,

I currently have all outgoing calls set to mask the caller id so it will
always appear to be coming from our main number. The problem I'm having
though, is with both the call detail in mysql and with the automon
(recording) feature. It shows the originating number as the number I
masked it to, rather than the actual person calling. How can I go about
having both the destination see our main number, but our internally
logging see the correct #?


  I haven't tried this one, but you could try assign the source caller ID 
number to a temporary variable, mask the caller ID, then Set() one of

the CDR variables (${CDR(src)} I presume) to the value of the temporary
variable... in the dialplan.

-- Alex

--
Alex Balashov <[EMAIL PROTECTED]>
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Re: [asterisk-users] RE: OT (a little): IPV6 Ramifications Article

[Hans Witvliet]

On Wed, 2007-04-18 at 17:11 -0400, Dean Collins wrote:
> Hi guys,
> 
> I know it’s a little off topic but……Wondering if you can help.
> My wife has been asked to find a writer to produce a story on “The
> dramatic ramifications of IPV6 on commercial businesses and how it
> will change the product designs for ordinary household/commercial use
> in a 5-10 year time frame”

Ordinary household equipment
Fridge (sending snmp traps if a dork leaves the door open ;)
radio/tv/vcr (obviously)
central heating system
airco
security
 

> 
> So her company hired someone who should have been able to deliver the
> goods (ex magazine editor – maybe a little too ‘ex’….)
> 
>  
> 
> He has come back with the story angle that is boring (and just plain
> wrong) that says; 
> 
>  
> 
> IPV6 is a big cost to companies like the Y2k bug was.
afaik, all modern equipment supports v6
> 
> That it will stop spam (hmmm Cringley you have a lot to answer for)
(some people probably wont't get it working, so they will be off the
Net ;)
> 
> - That Asia is leading the way but we can ignore it as the USA
> have many many IPV4 addresses to use for the future.
(sleep well)
>  
> 
>  
> 
> So now my wife has egg on her face and her boss thinks that IPV6 is of
> no interest to anyone in their customers companies, apart from the CIO
> who needs to implement it, when I’m telling her that there are
> dramatic applications; eg.
> 
>  
> 
> - That Ford needs to consider how your car having an IP
> addresses changes the way they should be building cars (oh and the
> streetlights have one as well).
> 
> - That Sharp needs to consider what your TV having an IP
> address means (and your set top box and your front door bell as well)
> 
> - That Verizon needs to consider what every mobile phone
> having an IP address means (and your desk phone and your office phone)
> 
> - That Chase needs to consider what IPV6 means to your wallet,
> the ATM and the POS cash registers.
> 
 
I don't think that for john doe much will change (v4 => v6)
All unices support v6, and even vista has it (unlike previous products,
IPv6 is ON by default and can not be turned off ;)

In the good old days, everybody got a fixed ip by default, and some
euro's extra you got four or eight addrresses. Now you are lucky to get
one fixed address. Natting is very nice, but you're out-of-luck when
dealing with multiple ssl-sites (forward/backward name-resolving breaks
it)
All those troubles are over with v6

Best part however, is the build-in support for vpn, native encryption.
And QoS, which seems to be very nice for seperation voip-traffic from
torrent-traffic.

The only obstacles currently, are the ISP's.
afaik, all dsl-modems currently can only work with v4. 
(correct me if i'm wrong)

Only option currently is: tunnelbroker.net 

Perhaps if we actually get directly fibre or ethernet to our home, 
ipv6 will get mainstream.

hw
(Well, if * would support it, without any extra patches, would help)

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Re: [asterisk-users] [OT] Nokia E60 firmware update break SIP

[Gordon Henderson]

On Thu, 19 Apr 2007, Remco Barendse wrote:


On Mon, 16 Apr 2007, Martin Joseph wrote:

Just a warning for you all that are using Nokia series E phones for SIP 
function.


I updated my phones firmware today using the Nokia Updater,  and now the 
SIP functionality, which previously worked pretty well is completely 
broken.


The phone no longer registers with asterisk, although it displays the 
little icon as though it has, and it doesn't even seem to try to pass calls 
to asterisk...


So,  I would avoid 3.06330904 20-11-06 RM-49


Where did you find this version?  My Nokia updater only offers an update to 
2.0something  (my phone had 1.0something)


In a similar vein, there's been some (bad) press in the UK recently about 
mobile companys removing the VoIP software from some phones. Eg.


  http://www.theregister.co.uk/2007/04/19/vodafone_explains/

Gordon
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[asterisk-users] Polycom SIP Phones On LAN can't register without WAN (Internet) Access

[Shawn Kelley]

We are having an issue that I have been unable to figure out how to resolve.
I think its related to the Polycom Phones and not the Asterisk
configuration, but I'm not positive.

We have several Polycom 500/501/601's on both a LAN and at employee homes.

The problem we are having is if our internet connection goes down the Local
LAN phones loose their connection to the Asterisk Server.
I've checked everything I can think of but can't figure out why its
happening.
I believe since the Asterisk Box is on the LAN and the phones are on the
same LAN then it shouldn't need internet to function.

The closest I have narrowed this down is to the DNS area. I found that if I
block access to our ISP's DNS that the phones aren't able to register with
asterisk.

This baffles me because the phone has the LAN address for the Asterisk
server so I don't know why it's doing DNS lookups.

Phone Example: (SIP 600)
Phone's Config File:
reg.1.address="254"
reg.1.label="x204"
reg.1.type="private"
reg.1.thirdPartyName=""
reg.1.auth.userId="254"
reg.1.auth.password="mypass"
reg.1.server.1.address="192.168.0.5"
reg.1.server.1.port=""
reg.1.server.1.transport="DNSnaptr"
reg.1.server.2.transport="DNSnaptr"
reg.1.server.1.expires=""
reg.1.server.1.register="1"
reg.1.server.1.retryTimeOut=""
reg.1.server.1.retryMaxCount=""
reg.1.server.1.expires.lineSeize=""

Phone's IP Info:
IP is 192.168.0.160
IP Gateway: 192.168.0.1
SNTP: 192.168.0.5
DNS Server: 192.168.0.1
DNS Alternate: 192.168.0.5

Phone's Software:
BootBlock: 2.4.0
Bootrom: 3.2.1.0012
SIP: 1.6.7.0098

The Asterisk Box has2 Network Cards.
1 on the LAN with IP of 192.168.0.5
1 on the WAN with a public IP


Does anyone have any idea on what direction I need to go to resolve this?
Let me know if you need to know any further details about the configuration.
Also, FYI: when this occurs the logs show no activity on the Asterisk
server. No attempted subscribes/logins/registers.

Thanks in advance!!
--Shawn
[EMAIL PROTECTED]



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RE: [asterisk-users] AudioCodes MP-104 MGCP?

[J. David Bavousett]

Fair enough concerns, Andrew, but...

1)  This particular device doesn't have SIP on any of its' menus.  Is
there a firmware upgrade that will get it there?  MGCP and MEGACO seem
to be the only choices I have.

2)  I'm trying to preserve existing investment; we're a non-profit, and
can't afford to go tossing money around.  I'd love to dump the silly
thing in the trash and get something from someone who's more open-source
friendly, but that's not in my playbook at this point.  

Anyone know of a *cheap* interface for four POTS lines, doesn't use a
slot on the servers--our systems are very thin rackmounts--that does
support SIP and is ideologically sound?

--David

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Joakimsen
Sent: Thursday, April 19, 2007 2:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] AudioCodes MP-104 MGCP?

1) dont use MGCP -- SIP is better supported
2) Don't use Audiocodes, they blatantly ignore the GPL license.

On 4/19/07, J. David Bavousett <[EMAIL PROTECTED]> wrote:
>
>
> Greetings;
>
> We are trying to get Asterisk up and happy at our site-we tried VOIP
> using Sphere about a year ago, spent a *boodle* on expensive hardware
> and services from a local "expert", but it never was happy.
>
> I'm brand-spanking new at VOIP, and I've learned a *ton* getting
> Asterisk breathing in the last couple of days.  I have three Polycom
> Soundpoint IP 500 SIP phones, which are all working wonderfully, when
> talking to each other.  Voicemail seems happy, tooNow to get it
> talking to the outside world.
>
> In our previous trial, we purchased an AudioCodes MP-104 MGCP gateway.
> I'm supposing that something needs to be set up in mgcp.conf for it,
but
> what I've tried just hasn't worked.  Here's what I have:
>
> -
>
> [general]
> port = 2427
> bindaddr = 0.0.0.0
>
> [mp104-1]
> host = 192.168.10.179
> callgroup=1
> context=default
> callerid="Incoming Line 1"
> line => aaln/1
>
> -
>
> In the Asterisk CLI, I'm getting "NOTICE[12321]: chan_mgcp.c:1656
> find_subchannel_and_lock: Gateway 'mp104-1' (and thus its endpoint
'*')
> does not exist"...several of them, every few minutes.
>
> Is there someone out there who wouldn't mind sharing relevant sections
> of the .conf files to get this thing *basically* working?  Incoming
> calls should fall on a specific extension (199), outgoing should be
> accessible by keying 9+the number.  All our long distance, 1-800
> outbound, etc, all go out the same four lines...this is a *simple*
> setup.
>
> Apologies if this is something that's been answered before, or if I've
> just missed the link I need, and thanks in advance for any assistance
> you can give.  Our old PBX has perished, and I need to get this
> breathing *quickly*.
>
> J. David Bavousett
> System Administrator
> Abilene Library Consortium
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[asterisk-users] Problem with TDM2400 and Polycom 501... Voice Quality Lost...

[Arturo Ochoa]

Hi List...

I have an asterisk box with one TDM2400, it has 8 FXS's and 12 FXO's, 
and it also has the echo canceller...
I'm running Asterisk 1.2.13 and Zaptel 1.2.12 with the Kernel 
2.6.9-34.0.2.EL

I'm using Polycom's  501 with the SIP 1.6.2.0041

The problem is when someone dials to or from the PSTN through the 
TDM2400, the voice quality is crappy...Instead of hearing:


Hello, this is John

You hear..

He  o, th  s   J hn

I already tried with the fxotune utility, also using G711 or G729, 
dealing with the gains... but I can't see the light...



Any Ideas ?

--
Ing. Arturo Ochoa N
Network Administrator
Electrosystems,

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Re: [asterisk-users] gxp2000 expansion module blf leds not working

[Zoilo Gomez]

Am I the only one using the GXP2000 expansion module?

Thanks,

Zoilo.


Zoilo Gomez wrote:

Today a 56-button expansion module for the GXP2000 came in.

When I program the buttons+leds on the expansion module for BLF, then 
speed-dial works fine: when I press the button the programmed ext 
number is called properly.


However the LEDs are always off: neither green nor red  They are 
not broken, because on reboot the LEDs flash red!


On the GXP2000 itself, this function works fine, with LEDs being green 
when the ext is free, or red whenever it is busy.


Does anybody know this problem?

Or can anyone confirm that the LEDs on the GXP2000 expansion module 
should be working properly?


Thanks,

Z.
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[asterisk-users] CLI Dialplan options...

[Carlos Chavez]

I have a very strange problem.  I have two Asterisk servers running
1.4.2.  On the first one I have the following options:

Connected to Asterisk 1.4.2 currently running on pbxoficina (pid = 7057)
Verbosity is at least 3
pbxoficina*CLI> help dialplan
   dialplan add extension  Add new extension into context
   dialplan add ignorepat  Add new ignore pattern
 dialplan add include  Include context in other context
  dialplan reload  Reload extensions and *only* extensions
dialplan remove extension  Remove a specified extension
dialplan remove ignorepat  Remove ignore pattern from context
  dialplan remove include  Remove a specified include from context
dialplan save  Save dialplan
dialplan show  Show dialplan


On the second one:

Connected to Asterisk 1.4.2 currently running on pbxskandiamty2 (pid =
3873)
-- Remote UNIX connection
Verbosity is at least 3
pbxskandiamty2*CLI> help dialplan
dialplan show  Show dialplan


Why are all the other dialplan commands missing?

-- 
Telecomunicaciones Abiertas de Mexico S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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Re: [asterisk-users] ztdummy does not load properly at server startup

[Theo Band]

Tzafrir Cohen wrote:

  On Thu, Apr 19, 2007 at 06:28:51PM +0200, Theo Band wrote:
  
  
Hi

I run asterisk 1.4.2 with zaptel 1.4.1.
Zaptel is only needed for the ztdummy driver to get the Meetme()
application to work. I don't have any specific hardware.
And it does work nicely. When I reboot the machine however I have to
manually load the driver like this:

[EMAIL PROTECTED] ~]# modprobe ztdummy
Notice: Configuration file is /etc/zaptel.conf
line 0: Unable to open master device '/dev/zap/ctl'

1 error(s) detected

  
  
The error message is meaningless for ztdummy. Please test if you have a
working timing source with:

  zttest -v
  

--- Results after 28 passes ---
Best: 98.156738 -- Worst: 97.900391 -- Average: 98.024205

2% too slow? Doesn't look very accurate for a timing reference. On the
other end Meetme() work like a charm.

  
To get rid of this error, remove the line that runs an automatic ztcfg
at the load of ztdummy:  sed -i '/ztdummy/d'

  

Or add /dev/null

Why would one application need a special driver? What so different
about the Meetme() application? Playing a wav file doesn't need a
special timing source for instance. But, I'm just a simple end user of
course, not understanding all the complex details of a PBX :-)


Theo


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Re: [asterisk-users] Asterisk Queue Call Transfer

[lenz]

Why don't you simply "pause" them when they are unavailable?
l.

In data Thu, 19 Apr 2007 14:33:52 +0200, Arun Kumar <[EMAIL PROTECTED]>  
ha scritto:



Hi

I've configured the queue on my asterisk box and everything is working  
fine.
In my queue I've 3 agents logged in the queue. When call comes they are  
able

to receive the calls without any problem. But some time they are on break
and there extension rings and no one is there to answer the call (we  
don't
want them to log off from the queue) but we have one normal user in the  
same
asterisk box registered so I want he dial some thing from his phone and  
that

call should come to that normal user. Please advice.


thanks

arun




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Re: [asterisk-users] Setup Asterisk configuration

[Tim Verscheure]

2007/4/19, Noah Miller <[EMAIL PROTECTED]>:

Hi Tim -

> I'm new to this list. For the last couple of days I was searching for
> a good solution using AsteriskNOW. I noticed that in the configuration
> steps of the server, they asked for a service provider. We don't
> really need one.
>
> We had something in mind like installing two Asterisk servers and make
> a connection between them. It's just to connect two buildings. Would
> this be a right approach? And when this is done, the one server could
> be used as a service provider (SIP server) for the other one and vice
> versa?

Yep, you can definitely do that with Asterisk(NOW).  Unless you have
other plans for voicemail, meetme conferences, or other fancy asterisk
things, it might be a little overkill, though.  You could also do the
same thing without asterisk using sip clients connecting direct to one
another.


Can you explain that last thing a bit more? Thanks in advance for the
quick answer...



As far as the AsteriskNOW setup: You could put in dummy values for the
provider, and figure out your site to site connection settings later.



But the concept of my idea could work? As a final result when
everythings ready use one Asterisk server as provider for the other
one?


Tim
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Re: [asterisk-users] Re: Trigger for unavailable SIP peer

[0xception]

My company also uses zabbix to monitor our networks, and is the project that
gave me the idea for creating the triggers.pl add-on i mentioned in my
previous post... This addon works in basically the same way as zabbix
triggers/actions work. i haven't released the code yet but i can email it to
anyone interested...

for more details: http://eagle.skyhighspeed.com/blogs/cwenman/?p=21

On 4/19/07, Theo Band <[EMAIL PROTECTED]> wrote:


Edoardo Serra wrote:
> I'm using zabbix (http://www.zabbix.com/) as a complete monitoring
> solution
>
> zabbix agent has the possibility to specify custom checks that are run
> as often as you wish
> (maybe an "asterisk -rx "sip show peers" | grep UNREACHABLE | wc -l")
> the output of the script is sent to zabbix server which can fire
> actions (email, sms, etc)
> in a very flexible manner
This works only if you run zabbix as a privileged user (the same as runs
asterisk, so probably root).
But this is still a polling solution, not event based as the OP wants.
I tried this approach with zabbix to get an overview of the amount of
active lines versus time. I did not succeed because of the permission
problem. I can only achieve it using a cron job and have the result read
by zabbix, which is not very elegant.

My 1 cent :-)
>
> My 2 cents
>
> Regards
>
> C F ha scritto:
>> Thank you all for your response, but it appears that some of you
>> didn't understand my question. I know I can schedule a cron to check
>> the status (I can even use asterisk -rx "sip show peers" | grep
>> UNREACHABLE if I use a cron) but that is not what I want. I want
>> either a way that just as asterisk prints to the CLI  the following:
>> Peer '120' is now UNREACHABLE!  Last qualify: 118
>> it should also be able to trigger whatever action from a conf file or
>> the like.
>> Or if there is an available solution even that involves a cron job but
>> already has all the options, so I don't have to reinvent the wheel.

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[asterisk-users] CallerID masking

[Rob Schall]

Hello all,

I currently have all outgoing calls set to mask the caller id so it will
always appear to be coming from our main number. The problem I'm having
though, is with both the call detail in mysql and with the automon
(recording) feature. It shows the originating number as the number I
masked it to, rather than the actual person calling. How can I go about
having both the destination see our main number, but our internally
logging see the correct #?


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[asterisk-users] Asterisk 1.4.2 connection to Nortel CS1000M -followup with log

[Jerry Geis]

Here is the CS1000 log. Again, the CS1000 using SIP accepts incoming calls just fine. 
However, using outgoing call files the CS1000 is hanging up after I answer the call.


I dont know why?

Thanks, for any assistance.

Jerry

my sip.conf entry is:
   [Nortel]
   type=friend
   dtmfmode=rfc2833
   username=X
   disallow=all
   allow=ulaw
   allow=alaw
   context=nortel
   host=XXX
   canreinvite=yes
   qualify=yes
usereqphone=yes


-

Use 'exit' when done

Asterisk 1.4.2, Copyright (C) 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer <[EMAIL PROTECTED]>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for 
details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=
 == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Connected to Asterisk 1.4.2 currently running on hfemsrv (pid = 18420)
hfemsrv*CLI> 
Verbosity is at least 5


hfemsrv*CLI> sip debug
hfemsrv*CLI> 
SIP Debugging enabled

The 'sip debug' command is deprecated and will be removed in a future release. 
Please use 'sip set debug' instead.

hfemsrv*CLI> 
Reliably Transmitting (no NAT) to 192.168.45.129:5060:

OPTIONS sip:192.168.45.129 SIP/2.0
Via: SIP/2.0/UDP 161.49.142.250:5060;branch=z9hG4bK2508d83c;rport
From: "asterisk" ;tag=as2cc96e52
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 19 Apr 2007 19:25:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
?
hfemsrv*CLI> 
<--- SIP read from 192.168.45.129:5060 --->

SIP/2.0 200 OK
From: "asterisk";tag=as2cc96e52
To: ;tag=812da8c0-13c4-46277c06-279cd106-42ff
Call-ID: [EMAIL PROTECTED]
CSeq: 102 OPTIONS
Allow: 
INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE
Via: SIP/2.0/UDP 161.49.142.250:5060;rport=5060;branch=z9hG4bK2508d83c
Supported: 100rel,sipvc,replaces
User-Agent: Nortel CS1000 SIP GW release_4.0 version_sse-4.00.55
Content-Length: 0


<->
?--- (10 headers 0 lines) ---
?
hfemsrv*CLI> 
Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS

?
hfemsrv*CLI> 
   -- Attempting call on SIP/QuadNortel/7113 for [EMAIL PROTECTED]:1 (Retry 1)

?
hfemsrv*CLI> 
Audio is at 161.49.142.250 port 1

?
hfemsrv*CLI> 
Adding codec 0x4 (ulaw) to SDP

?Adding codec 0x8 (alaw) to SDP
?
hfemsrv*CLI> 
Reliably Transmitting (no NAT) to 192.168.45.129:5060:

INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 161.49.142.250:5060;branch=z9hG4bK11268a7d;rport
From: "Admin System 34" ;tag=as4e5a553d
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 19 Apr 2007 19:25:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 212

v=0
o=root 18420 18420 IN IP4 161.49.142.250
s=session
c=IN IP4 161.49.142.250
t=0 0
m=audio 1 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
?
hfemsrv*CLI> 
<--- SIP read from 192.168.45.129:5060 --->

SIP/2.0 100 Trying
From: "Admin System 34";tag=as4e5a553d
To: ;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 161.49.142.250:5060;rport=5060;branch=z9hG4bK11268a7d
Supported: 100rel,sipvc,replaces
User-Agent: Nortel CS1000 SIP GW release_4.0 version_sse-4.00.55
Contact: 
Allow: 
INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE
Content-Length: 0


<->
?--- (11 headers 0 lines) ---
?
hfemsrv*CLI> 
<--- SIP read from 192.168.45.129:5060 --->

SIP/2.0 180 Ringing
From: "Admin System 34";tag=as4e5a553d
To: ;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 161.49.142.250:5060;rport=5060;branch=z9hG4bK11268a7d
Supported: 100rel,sipvc,replaces
User-Agent: Nortel CS1000 SIP GW release_4.0 version_sse-4.00.55
Contact: 
Allow: 
INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE
Content-Length: 0


<->
?--- (11 headers 0 lines) ---
?
hfemsrv*CLI> 
<--- SIP read from 192.168.45.129:5060 --->

SIP/2.0 200 OK
From: "Admin System 34";tag=as4e5a553d
To: ;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 161.49.142.250:5060;rport=5060;branch=z9hG4bK11268a7d
Supported: 100rel,sipvc,replaces
User-Agent: Nortel CS1000 SIP GW release_4.0 version_sse-4.00.55
P-Asserted-Identity: 
Privacy: none
Contact: 
Allow: 
IN

Re: [asterisk-users] Outgoing CallerID

[Forrest Beck]

Thanks.  I will just use the asterisk database.

This brought up a new question though.

This is what I am using to dial out.  If a key for the phone exist in
the db it will assign it the did specified.  If not, just assign the
main incoming operator number.

I have a new family in the astdb named external_did.  Some entries
look like this:

external_did/2503/9195551212
external_did/2505/9195551213

[macro-dialoutpstn]
exten => s,1,Set(curextension=${CALLERID(num)})
exten => s,2,Set(dbdid="${DB(external_did/${curextension})}")
exten => s,3,NoOp(${curextension})
exten => s,4,NoOp(${dbdid})
exten => s,4,GoToIf($["${dbdid}" = ""]?5:8)
exten => s,5,Set(CALLERID(num)=9195559595)
exten => s,6,Dial(SIP/mspri/${MACRO_EXTEN:1},300)
exten => s,7,Hanup()
exten => s,8,Set(CALLERID(num)=${DB(external_did/${curextension})})
exten => s,9,NoOp(${CALLERID(num)})
exten => s,10,Dial(SIP/mspri/${MACRO_EXTEN:1},300)
exten => s,11,Hangup()

This works just fine.

Now what about my incoming calls.  My incoming calls will be sent from
the telco to asterisk as the seven digit number that was dialed.  So
if I have _X. in my context it will be processed as extension
9195551212.

So is there a way to lookup in the asterisk database a value and
return the key it belongs to?  Because I already have the phone number
in the asterisk database set to each extension.

I know I could just create a new family and add the keys there, like so

incoming_did/9195551212/2503
incoming_did/9195551213/2504

I was just looking to see if I could save myself a step.

This may be where I will need to switch to MySQL.



On 4/19/07, Alex Balashov <[EMAIL PROTECTED]> wrote:


On Thu, 19 Apr 2007, Forrest Beck said something to this effect:

> I thought of maybe adding a key for each extension to the astdb and
> have a Macro query the astdb.  Any other ideas?

   That would work, and is certainly the easiest, since you can bulk-load
the DID -> extension maps via external CLI commands with a simple script.

   You could also have Asterisk do MySQL dips for this information, if the
desire is to administer it from a web-based front-end.  Or if there is
some sort of mathematical relationship between the extension and the DID
range, the dialplan interpreter itself is capable of fairly sophisticated
mathematical extrapolations.

-- Alex

--
Alex Balashov <[EMAIL PROTECTED]>
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--
***
Forrest Beck
IAXTEL: 17002871718
[EMAIL PROTECTED]
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Re: [asterisk-users] ztdummy does not load properly at server startup

[Theo Band]

Eric "ManxPower" Wieling wrote:
> In the zaptel source "make config" will install the zaptel init script
> in /etc/rc.d/init.d for many distros.
>
Thanks. This was the missing configuration step. A manual start/stop
seems to work. I will try to reboot the machine tomorrow for the "final"
test but I feel this is going to work.

Stopping the zaptel drivers does not work properly, but I only need this
for rebooting...

[EMAIL PROTECTED] zaptel-1.4.1]# service zaptel stop
Unloading zaptel hardware drivers: wcusb wctdm wcfxo wctdm24xxp wcte11xp
wct1xxp wct4xxp tor2.
Removing zaptel module:  ERROR: Module zaptel is in use by ztdummy
   [FAILED]

Cheers,
Theo
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Re: [asterisk-users] ztdummy does not load properly at server startup

[Theo Band]

Matthew J. Roth wrote:
> Theo,
>
> Unless things have changed significantly in the newer releases, you
> must load zaptel prior to loading ztdummy.  Additionally, the zaptel
> devices are not created instantly, so after you load zaptel you must
> wait a few seconds before loading ztdummy.  You can perform some sort
> of polling if you want to script this, but a less sophisticated method
> is just to sleep for 10 or 15 seconds between the calls to modprobe.
>
> If your goal is to start Asterisk automatically at boot, some init
> scripts for different distributions are available at
> . 
> I'm using Fedora, so I installed 'rc.redhat.asterisk' with chkconfig
> as follows:
>
> # install -m 755 ./rc.redhat.asterisk /etc/rc.d/init.d/asterisk
> # chkconfig --add asterisk
> # chkconfig --list asterisk
> asterisk0:off   1:off   2:on3:on4:on5:on6:off
The make install already did that for me .
>
>
> Note that I made the following customizations to the script prior to
> installing it:
>
> * I don't want to run safe_asterisk, so I comment out all of the lines
> that reference the SAFE_ASTERISK variable.
> * I want to load ztdummy and raise the open file limit, so I add the
> following lines to the start() function immediately prior to the
> 'daemon' statement:
> modprobe zaptel > /dev/null 2> /dev/null
> sleep 15
> modprobe ztdummy > /dev/null 2> /dev/null
> ztcfg > /dev/null 2> /dev/null
> ulimit -n 65536 > /dev/null 2> /dev/null
> * And add the following lines to the stop() function, immediately
> after the 'RETVAL=$?' line:
> rmmod ztdummy > /dev/null 2> /dev/null
> rmmod zaptel > /dev/null 2> /dev/null
Yes it works indeed thanks. It turns out that (for me) the "modprobe
ztdummy" alone works as well. I do get an error message as shown before,
but both ztummy and zaptel get loaded and Meetme() works. The 15 seconds
delay would add to the boot time
I also tried the suggestion of Eric Wieling. Appearantly make config is
what I had not done. This creates the needed module load scripts. The
module remove scripts do not work properly, but who cares if the machine
is going down anyway?
>
> Things will differ depending on your distribution, but that should be
> enough to get you going in the right direction.
Yep it did :-)

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Re: [asterisk-users] AudioCodes MP-104 MGCP?

[Andrew Joakimsen]

1) dont use MGCP -- SIP is better supported
2) Don't use Audiocodes, they blatantly ignore the GPL license.

On 4/19/07, J. David Bavousett <[EMAIL PROTECTED]> wrote:



Greetings;

We are trying to get Asterisk up and happy at our site-we tried VOIP
using Sphere about a year ago, spent a *boodle* on expensive hardware
and services from a local "expert", but it never was happy.

I'm brand-spanking new at VOIP, and I've learned a *ton* getting
Asterisk breathing in the last couple of days.  I have three Polycom
Soundpoint IP 500 SIP phones, which are all working wonderfully, when
talking to each other.  Voicemail seems happy, tooNow to get it
talking to the outside world.

In our previous trial, we purchased an AudioCodes MP-104 MGCP gateway.
I'm supposing that something needs to be set up in mgcp.conf for it, but
what I've tried just hasn't worked.  Here's what I have:

-

[general]
port = 2427
bindaddr = 0.0.0.0

[mp104-1]
host = 192.168.10.179
callgroup=1
context=default
callerid="Incoming Line 1"
line => aaln/1

-

In the Asterisk CLI, I'm getting "NOTICE[12321]: chan_mgcp.c:1656
find_subchannel_and_lock: Gateway 'mp104-1' (and thus its endpoint '*')
does not exist"...several of them, every few minutes.

Is there someone out there who wouldn't mind sharing relevant sections
of the .conf files to get this thing *basically* working?  Incoming
calls should fall on a specific extension (199), outgoing should be
accessible by keying 9+the number.  All our long distance, 1-800
outbound, etc, all go out the same four lines...this is a *simple*
setup.

Apologies if this is something that's been answered before, or if I've
just missed the link I need, and thanks in advance for any assistance
you can give.  Our old PBX has perished, and I need to get this
breathing *quickly*.

J. David Bavousett
System Administrator
Abilene Library Consortium
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[asterisk-users] Asterisk 1.4.2 connection to Nortel CS1000M

[Jerry Geis]

I have asterisk 1.4.2 connected to a Nortel CS1000M using SIP.
Version 4.00T.

I can call into the connection just fine. I get my voice prompts and I 
thought
everything was working. Then I tried a call out using the outgoing spool 
directory.

The call gets placed, the phone rings, when I answer the phone it hangs up.

Anyone know what might be of issue with this? At the moment I dont have 
a log...

I was hoping it might be something simple.

THanks,

Jerry
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Re: [asterisk-users] Re: Can I add distinctive ring with asterisk and TDM400?

[Brian McEntire]

Maybe not, but I got it working "good enough" and time is scarce these
days so I didn't mess with it after that.

On 4/19/07, Stephen Bosch <[EMAIL PROTECTED]> wrote:

Brian McEntire wrote:
> A follow-up with the solution in case anyone else is looking for this
> answer:
>
> I created two contexts in my zapata.conf file, since each VOIP line is
> terminated by a VOIP adapter and then just comes in "hardwired" to the
> TDM400 via RJ11 line, I know which VOIP number is connected to which
> Wildcard port.
>
> In Zapata.conf:
>
> usedistinctiveringdetection=yes

Is this line even necessary? You're sending distinctive ring, not
receiving it.

-Stephen-

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Re: [asterisk-users] ZT_CHANCONFIG failed on channel 1: No such device or address

[kjcsb]

>What is the output of:
>ls -l /sys/class/zaptel

ls -l /sys/class/zaptel
total 0
drwxr-xr-x 2 root root 0 Apr 19 20:05 zapchannel
drwxr-xr-x 2 root root 0 Apr 19 20:05 zapctl
drwxr-xr-x 2 root root 0 Apr 19 20:05 zappseudo
drwxr-xr-x 2 root root 0 Apr 19 20:05 zaptimer
drwxr-xr-x 2 root root 0 Apr 19 20:05 zaptranscode


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Re: [asterisk-users] Asterisk 1.4.2 + Cisco 7960G not registering

[David Olsen]

On 2007-04-19 at 14:16:49, Doug Lytle <[EMAIL PROTECTED]> wrote:
> This what mine looked like before the firmware downgrade.  Except the 
> timer and expires were huge numbers.
> 
> Firewall issue?
> 
> I've also had issues when compiling my own kernel and thinking it was a 
> good thing to enable 'SIP Protocol Support' in the Netfilter section;  
> It wasn't.

Definitely not. The packet dumps reveal an active refusal from the server,
just continued 401/Unauth'd responses. 

-d

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Re: [asterisk-users] ZT_CHANCONFIG failed on channel 1: No suchdevice or address

[Cameron Beattie]

What is the output of:

ls -l /sys/class/zaptel


ls -l /sys/class/zaptel
total 0
drwxr-xr-x 2 root root 0 Apr 19 20:05 zapchannel
drwxr-xr-x 2 root root 0 Apr 19 20:05 zapctl
drwxr-xr-x 2 root root 0 Apr 19 20:05 zappseudo
drwxr-xr-x 2 root root 0 Apr 19 20:05 zaptimer
drwxr-xr-x 2 root root 0 Apr 19 20:05 zaptranscode

Regards

Cameron
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[asterisk-users] aastra phones with asterisk 1.2.17 - hangup after 20 seconds

[Jeronimo Romero]

Running asterisk 1.2.7 with latest zaptel on centos4.4. with Aastra 55i phones. 
Local outbound calling works fine, but ATT requires clients enter 7 digit code 
for long distance. All calls with 7 digit code are lost within 20 seconds of 
the call. This is the message I’m getting: 

Apr 19 12:38:16 WARNING[9615]: chan_sip.c:1228 retrans_pkt: Maximum retries 
exceeded on transmission [EMAIL PROTECTED] for seqno 783509378 (Critical 
Response)
Apr 19 12:38:16 WARNING[9615]: chan_sip.c:1245 retrans_pkt: Hanging up call 
[EMAIL PROTECTED] - no reply to our critical packet.

The strange thing is that when I use an xten softphone this issue does not 
occur. Is this a SIP signaling issue? Any help would be appreciated.  This 
issue does not occurr with any other ip phone on our network. ONLY THE AASTRAs. 
any ideas?
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Re: [asterisk-users] Asterisk - Cisco Call Manager Express Trunk

[Noah Miller]

Hi Diego -


I want to make a SIP trunk between a Cisco 2811 router and a Asterisk.
Both 2811 and Asterisk are working fine (2811 has 1XX and asterisk
2XX). Now I want to configure a trunk so that 2811 users can call *
users. I've been reading a lot but I'm still confused.


I don't know if you've seen this page on the WIKI yet, but it does
have a section for Call Manger Express:

http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Integration


- Noah
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Re: [asterisk-users] ztdummy does not load properly at server startup

[Tzafrir Cohen]

On Thu, Apr 19, 2007 at 06:28:51PM +0200, Theo Band wrote:
> Hi
> 
> I run asterisk 1.4.2 with zaptel 1.4.1.
> Zaptel is only needed for the ztdummy driver to get the Meetme()
> application to work. I don't have any specific hardware.
> And it does work nicely. When I reboot the machine however I have to
> manually load the driver like this:
> 
> [EMAIL PROTECTED] ~]# modprobe ztdummy
> Notice: Configuration file is /etc/zaptel.conf
> line 0: Unable to open master device '/dev/zap/ctl'
> 
> 1 error(s) detected

The error message is meaningless for ztdummy. Please test if you have a
working timing source with:

  zttest -v

To get rid of this error, remove the line that runs an automatic ztcfg
at the load of ztdummy:  sed -i '/ztdummy/d'

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Outgoing CallerID

[David Gomillion]

On 4/19/07, Forrest Beck <[EMAIL PROTECTED]> wrote:



I thought of maybe adding a key for each extension to the astdb and
have a Macro query the astdb.  Any other ideas?




That's how we do it. We created a MySQL DB that maps DIDs to extensions, and
a php script to write our configuration files for us (a file called did.conf,
which is #include'd into extensions.conf), as well as push the DID into the
Asterisk DB. Actually the DB holds all of the information for our phones,
and all of the files we need are generated each night, including sip
configs, provisioning files for our Polycoms,  the dhcpd configurations to
give "static" addresses, and a few other miscellaneous files. And it creates
our phone list. The nice thing about building the DB yourself is that you
can do anything you want with it.

On one of our boxes, I went a step further and created individual outgoing
contexts, one for each device. The context set the caller ID. But it was
definite overkill; I haven't done that since.


Thanks.


--
***
Forrest Beck
IAXTEL: 17002871718
[EMAIL PROTECTED]
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Re: [asterisk-users] Hardware suggestions for 8-10 lines in the UK

[Stephen Bosch]

Chris Bagnall wrote:
>> However, my experience hasn't been that VoiP is as reliable as
>> copper lines and so unless you can tolerate the odd outage once per
>>  month or two then you might want to stick to copper for the main 
>> carrier?  Does this match with the experience from others?
> 
> Until recently, I'd have agreed entirely with this statement.
> However, recently one of our clients (in a shopping centre)
> encountered a scenario where a contractor had chopped through a load
> of PSTN lines whilst enlarging the car park. Their calls were routed
> via an ADSL connection which came in on an alternate location. For
> the better part of a week, they were the only shop in the building
> able to make & receive calls.

I'd say it's only a miracle that the ADSL connection wasn't entering
through the same cable bundle.

VoIP connections are no less vulnerable to contractor mistakes -- what
you describe is really a matter of redundancy; while it may be hard to
get redundant PSTN connections through different physical feeds, it's
not impossible, and the advantages you describe are not inherent to VoIP
-- they're inherent to redundant transports :)

> By having two independent net connections plus a "true" PSTN backup
> you've got 3 levels of redundancy. Short of spending a fortune on
> "guaranteed fix times" from the telco

...which are the stuff of myth, anyway. If the service goes down it goes
down -- and all the SLA in the world won't protect you. You can pound
your fist on the table and turn blue if you want -- the best you'll do
is get your money refunded.

You are better off investing your money in as much vendor-separate
redundancy as you can afford. Then, when the bulldozer goes through the
underground cable, cross your fingers.

I still wouldn't go without PSTN.

-Stephen-

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Re: [asterisk-users] incoming SIP call

[Jean Marc Le Fevre]

Hello and thanks for answering,

As I just answer to Yuan LIU, what I don't understand, is that I can  
place an outbound call from asterisk to a gsm at the same time I  
can't get asterisk thought a inbound call. But I'll try what you  
advice me.

I'll tell you the result of it

Jean-Marc LE FEVRE



Le 19 avr. 07 à 20:08, Bala Neelakantan a écrit :

If your SIP server loses REGISTERs then it cant place an inbound  
SIP call.  Try changing the REGISTER frequency to lower value.



When you see incoming SIP call fail, you might want to check  
whether the REGISTERs are working.



Thanks,

Neel


-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk- 
[EMAIL PROTECTED] On Behalf Of Jean Marc Le Fevre

Sent: Wednesday, April 18, 2007 11:15 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] incoming SIP call


Hello all,



I'm having a quite simple configuration like:


SIP provider <=> asterisk SIP <=> lan


Everythings works fine but sometime I can't get incoming call.


here are some of the logs from set debug 25 set verbosity 25 sip  
show debug and sip.conf and a part of extension.conf


thanks in advance



Reliably Transmitting (NAT) to 212.27.52.5:5060:

OPTIONS sip:freephonie.net SIP/2.0

Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;branch= z9hG4bK67c2df66;rport

From: "asterisk" ;tag=as01265eaf

To: 

Contact: 

Call-ID: [EMAIL PROTECTED]

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Wed, 18 Apr 2007 13:57:55 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Length: 0



---

12 headers, 0 lines

Reliably Transmitting (NAT) to 212.27.52.5:5060:

OPTIONS sip:freephonie.net SIP/2.0

Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;b ranch=z9hG4bK253c1a3d;rport

From: "asterisk" ;tag=as372da2cb

To: 

Contact: 

Call-ID: [EMAIL PROTECTED]

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Wed, 18 Apr 2007 13:57:55 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Length: 0



---

Zpro*CLI>

<-- SIP read from 212.27.52.5:5060:

SIP/2.0 403 not registered

Call-ID: 7263e88c20c9f3 [EMAIL PROTECTED]

CSeq: 102 OPTIONS

From: "asterisk" ;tag=as01265eaf

To: ;tag=00-31057-001dc208-591e1ca81

Via: SIP/2.0/UDP 82.XXX.XXX.XXX: 
5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK67c2df66


Content-Length: 0



--- (7 headers 0 lines) ---

Destroying call '[EMAIL PROTECTED]'

Zpro*CLI>

<-- SIP read from 212.27.52.5:5060:

SIP/2.0 403 not registered

Call-ID: [EMAIL PROTECTED]

CSeq: 102 OPTIONS

From: "asteris k" ;tag=as372da2cb

To: ;tag=00-32700-001dc209-6fc2b3303

Via: SIP/2.0/UDP 82.XXX.XXX.XXX: 
5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK253c1a3d


Content-Length: 0


--- (7 headers 0 lines) ---

Destroying call '[EMAIL PROTECTED]'



sip.conf


[general]

context=incoming

realm=etatcritik.dyndns.org

bindport=5060

bindaddr=0.0.0.0

srvlookup=no

maxexpiry=3600

defaultexpiry=1800

videosupport=yes

disallow=all

all ow=ulaw

allow=ilbc

allow=alaw

allow=gsm

musicclass=default

language=fr

useragent=Asterisk PBX

dtmfmode = auto

register => 09:[EMAIL PROTECTED]

registertimeout=40

externip = 82.XXX.XXX.XXX

localnet=10.XXX.XXX.XXX/255.255.255.0

qualify=6

nat = yes

[test]

type=friend

username=test

secret=test

host=dynamic

context=home

callerid =test <>

dmtfmode=rfc2833

authuser=test

fromuser=test

allow=all

[freephonie_outbound]

type=peer

allow=all

host=freephonie.net

secret=SECRET

fromuser=09XXX

username=09XXX

dtmfmode=inband

qualify=6

fromdomain=freephonie.net

[freep honie_inbound]

type=peer

context=incoming

host=freephonie.net

qualify=6

allow=all

deny=0.0.0.0/0..0.0.0

permit=212.27.52.5/255.255.255.255 ; ip de freephonie.net


etension.conf



...

[incoming]

exten => s,1,Ringing

exten => s,2,Noop(I receive a sip call);

exten => s,n,Goto(home,1000,1)

exten => s,n,Congestion

;

...










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!DSPAM:4627b30550701698699180!




!DSPAM:4627b7bb50703422486060!
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Re: [asterisk-users] [OT] OMG Verizon is terrible

[Noah Miller]

> Had an appointment for these schmoes to come out and install another
> line.  Was supposed to be 8-12.  Its now 6PM and not even call.
Missed
> 3 sales calls waiting on these jerks.
>
> No wonder customers were jumping ship to Vonage.


I once had to oversee Verizon install a PRI line in Manhattan.  I live
2.5 hours away, but we made the appointment, and I was there, but the
Verizon tech never showed.  I made another appointment, and it
happened again, and again, and again.  I don't even remember how many
times it finally took, but it was ridiculous.  The techs were even
lying and saying they came and there was no one there to let them in.
They seem to have gotten better in recent years, but they own all the
physical lines, and they know it.


- Noah


On 4/19/07, Steve Totaro <[EMAIL PROTECTED]> wrote:

They are all terrible in their own way.  Don't you have someone you can
delegate the Verizon babysitting responsibility to?  I would consider
sales calls a little more important than being a babysitter.

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Lee Jenkins
> Sent: Wednesday, April 18, 2007 6:07 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] [OT] OMG Verizon is terrible
>
>
> Had an appointment for these schmoes to come out and install another
> line.  Was supposed to be 8-12.  Its now 6PM and not even call.
Missed
> 3 sales calls waiting on these jerks.
>
> No wonder customers were jumping ship to Vonage.
>
> --
>
> Warm Regards,
>
> Lee
>
>
>
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Re: [asterisk-users] incoming SIP call

[Jean Marc Le Fevre]

Well thanks for answering,

When I test, I use my GSM and call the number my provider gives me.
How often it works or not, I didn't make test like 10 calls per hour  
for a pretty long time so I can't exactly tell. When I test, well  
sometimes it works great, sometime, the incoming call is redirected  
to an phone that is connected on my DSL box.
I didn't see the error message SIP/2.0 403 not registered, but in  
that case:
1) I can make a call from asterisk to a gsm call (so It goes IAX  
phone => asterisk => SIP provider => GSM.
2) if I do show sip register in asterisk CLI, I can see I'm  
registered (or I may be misinterpretting this command.


What can I do to investigate this registration message ? Is there an  
special debug command ?


thanks :)


From: Jean Marc Le Fevre <[EMAIL PROTECTED]>
Date: Wed, 18 Apr 2007 18:14:41 +0200

Hello all,

I'm having a quite simple configuration like:

SIP provider <=> asterisk SIP <=> lan

Everythings works fine but sometime I can't get incoming call.


Define "sometimes" and from where the income call you can't get?

here are some of the logs from set debug 25 set verbosity 25 sip  
show  debug and sip.conf and a part of extension.conf

thanks in advance


[good stuff sniffed]
Where do you suspect the error message is?


---
Zpro*CLI>
<-- SIP read from 212.27.52.5:5060:
SIP/2.0 403 not registered


Does this message make sense, "not registered"?

Yuan Liu


Call-ID: [EMAIL PROTECTED]
CSeq: 102 OPTIONS
From: "asterisk" ;tag=as01265eaf
To: ;tag=00-31057-001dc208-591e1ca81
Via: SIP/2.0/UDP 82.XXX.XXX.XXX:  
5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK67c2df66

Content-Length: 0


--- (7 headers 0 lines) ---
Destroying call '[EMAIL PROTECTED]'
Zpro*CLI>
<-- SIP read from 212.27.52.5:5060:
SIP/2.0 403 not registered
Call-ID: [EMAIL PROTECTED]
CSeq: 102 OPTIONS
From: "asterisk" ;tag=as372da2cb
To: ;tag=00-32700-001dc209-6fc2b3303
Via: SIP/2.0/UDP 82.XXX.XXX.XXX:  
5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK253c1a3d

Content-Length: 0

--- (7 headers 0 lines) ---
Destroying call '[EMAIL PROTECTED]'


sip.conf

[general]
context=incoming
realm=etatcritik.dyndns.org
bindport=5060
bindaddr=0.0.0.0
srvlookup=no
maxexpiry=3600
defaultexpiry=1800
videosupport=yes
disallow=all
allow=ulaw
allow=ilbc
allow=alaw
allow=gsm
musicclass=default
language=fr
useragent=Asterisk PBX
dtmfmode = auto
register => 09:[EMAIL PROTECTED]
registertimeout=40
externip = 82.XXX.XXX.XXX
localnet=10.XXX.XXX.XXX/255.255.255.0
qualify=6
nat = yes
[test]
type=friend
username=test
secret=test
host=dynamic
context=home
callerid =test <>
dmtfmode=rfc2833
authuser=test
fromuser=test
allow=all
[freephonie_outbound]
type=peer
allow=all
host=freephonie.net
secret=SECRET
fromuser=09XXX
username=09XXX
dtmfmode=inband
qualify=6
fromdomain=freephonie.net
[freephonie_inbound]
type=peer
context=incoming
host=freephonie.net
qualify=6
allow=all
deny=0.0.0.0/0.0.0.0
permit=212.27.52.5/255.255.255.255  ; ip de freephonie.net

etension.conf


...
[incoming]
exten => s,1,Ringing
exten => s,2,Noop(I receive a sip call);
exten => s,n,Goto(home,1000,1)
exten => s,n,Congestion
;
...













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Re: [asterisk-users] Setup Asterisk configuration

[Noah Miller]

Hi Tim -


I'm new to this list. For the last couple of days I was searching for
a good solution using AsteriskNOW. I noticed that in the configuration
steps of the server, they asked for a service provider. We don't
really need one.

We had something in mind like installing two Asterisk servers and make
a connection between them. It's just to connect two buildings. Would
this be a right approach? And when this is done, the one server could
be used as a service provider (SIP server) for the other one and vice
versa?


Yep, you can definitely do that with Asterisk(NOW).  Unless you have
other plans for voicemail, meetme conferences, or other fancy asterisk
things, it might be a little overkill, though.  You could also do the
same thing without asterisk using sip clients connecting direct to one
another.

As far as the AsteriskNOW setup: You could put in dummy values for the
provider, and figure out your site to site connection settings later.


- Noah
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RE: [asterisk-users] RE: OT (a little): IPV6 Ramifications Article

[Dean Collins]

Maybe that's a little over the top but Minotaur as a broadband provider
could offer a new range of electronic door bell devices that also
control access to key strike locking plates.

As part of selling customers their broadband IPV6 package they are given
free access to this device (or minimal cost of hardware as a way of
preventing exploitation without implementation).

The door bell/lock device is directly connected to an ASP web service
that controls access to the property for residents, time of day access
for cleaners etc, remote access for allowing remote unlock for one off
repair people etc.

By Minotaur offering this device for the life of the broadband
connection it reduces the churn rate for your business taking pressure
off looking for customers seeking the lowest margin solution making you
far more profitable than you otherwise would be.

This is not a technical discussion but more of a business development
product design commercial discussion.

It comes back to something I have been saying about asterisk for a long
time. I want to go to Astricon one year and have an Asterisk System
Integrator tell me that he was involved in a deal competing against
Cisco and they won the deal even though the Asterisk installation was
more expensive than the Cisco quote.

The reason they won it was because Asterisk has far more additional
functionality like Mexuar Click-to-Talk or the Iotum Relevance Engine or
the Portal Blue Dashboard display. Building services is what is going to
make Asterisk profitable and successful not discounts.




 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Chris Bagnall
> Sent: Thursday, 19 April 2007 11:00 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [asterisk-users] RE: OT (a little): IPV6 Ramifications
Article
> 
> > If there was something useful in ones kettle having an ethernet
> > connection, it would probably already have it.  After all, with
NAT'ing
> > there's no real shortage of IP-addresses.  And perhaps we would
already
> > have K2K networks, with K2K proxies etc.
> 
> It'd be great if I could get my kettle to generate MRTG or Cacti
graphs for number
> of times boiled in a period, amount of water in it at the time, amount
of water
> poured from it, thus being able to work out power efficiency, etc.
etc. ;-)
> 
> Regards,
> 
> Chris
> --
> C.M. Bagnall, Director, Minotaur I.T. Limited
> For full contact details visit http://www.minotaur.it/chris.html
> This email is made from 100% recycled electrons
> 
> 
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Re: [asterisk-users] Asterisk 1.4.2 + Cisco 7960G not registering

[Doug Lytle]

David Olsen wrote:

On 2007-04-19 at 13:09:51, Doug Lytle <[EMAIL PROTECTED]> wrote:
  
Do you see anything weird when logging (telnet to the ip) into the phone 
and doing a show register?



I see as follows:

cisco-7960> show reg

LINE REGISTRATION TABLE
Proxy Registration: ENABLED, state: REGISTERING
line  APR  state  timer   expires proxy:port
  ---  -  --  --  
1 .1x  REGISTERING120 17  10.2.5.11:5060
2 .1x  REGISTERING120 17  10.2.5.11:5060
3 ...  NONE   0   0   undefined:0

  
This what mine looked like before the firmware downgrade.  Except the 
timer and expires were huge numbers.


Firewall issue?

I've also had issues when compiling my own kernel and thinking it was a 
good thing to enable 'SIP Protocol Support' in the Netfilter section;  
It wasn't.


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] incoming SIP call

[Bala Neelakantan]

If your SIP server loses REGISTERs then it cant place an inbound SIP call.
Try changing the REGISTER frequency to lower value.

 

When you see incoming SIP call fail, you might want to check whether the
REGISTERs are working.

 

Thanks,

Neel

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jean Marc Le
Fevre
Sent: Wednesday, April 18, 2007 11:15 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] incoming SIP call

 

Hello all, 

 

 

I'm having a quite simple configuration like: 

 

SIP provider <=> asterisk SIP <=> lan

 

Everythings works fine but sometime I can't get incoming call.

 

here are some of the logs from set debug 25 set verbosity 25 sip show debug
and sip.conf and a part of extension.conf

thanks in advance

 

 

Reliably Transmitting (NAT) to 212.27.52.5:5060:

OPTIONS sip:freephonie.net SIP/2.0

Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;branch= z9hG4bK67c2df66;rport

From: "asterisk" ;tag=as01265eaf

To: 

Contact: 

Call-ID: [EMAIL PROTECTED]

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Wed, 18 Apr 2007 13:57:55 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Length: 0

 

 

---

12 headers, 0 lines

Reliably Transmitting (NAT) to 212.27.52.5:5060:

OPTIONS sip:freephonie.net SIP/2.0

Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;b ranch=z9hG4bK253c1a3d;rport

From: "asterisk" ;tag=as372da2cb

To: 

Contact: 

Call-ID: [EMAIL PROTECTED]

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Wed, 18 Apr 2007 13:57:55 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Length: 0

 

 

---

Zpro*CLI> 

<-- SIP read from 212.27.52.5:5060: 

SIP/2.0 403 not registered

Call-ID: 7263e88c20c9f3

[EMAIL PROTECTED]

CSeq: 102 OPTIONS

From: "asterisk" ;tag=as01265eaf

To: ;tag=00-31057-001dc208-591e1ca81

Via: SIP/2.0/UDP
82.XXX.XXX.XXX:5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK67c2df6
6

Content-Length: 0

 

 

--- (7 headers 0 lines) ---

Destroying call '[EMAIL PROTECTED]'

Zpro*CLI> 

<-- SIP read from 212.27.52.5:5060: 

SIP/2.0 403 not registered

Call-ID: [EMAIL PROTECTED]

CSeq: 102 OPTIONS

From: "asteris k" ;tag=as372da2cb

To: ;tag=00-32700-001dc209-6fc2b3303

Via: SIP/2.0/UDP
82.XXX.XXX.XXX:5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK253c1a3
d

Content-Length: 0

 

--- (7 headers 0 lines) ---

Destroying call '[EMAIL PROTECTED]'

 

 

sip.conf

 

[general]

context=incoming

realm=etatcritik.dyndns.org

bindport=5060

bindaddr=0.0.0.0

srvlookup=no

maxexpiry=3600

defaultexpiry=1800

videosupport=yes

disallow=all

all ow=ulaw

allow=ilbc

allow=alaw

allow=gsm

musicclass=default

language=fr

useragent=Asterisk PBX 

dtmfmode = auto

register => 09:[EMAIL PROTECTED]

registertimeout=40

externip = 82.XXX.XXX.XXX

localnet=10.XXX.XXX.XXX/255.255.255.0

qualify=6

nat = yes

[test]

type=friend

username=test

secret=test

host=dynamic

context=home

callerid =test <>

dmtfmode=rfc2833

authuser=test

fromuser=test

allow=all

[freephonie_outbound]

type=peer

allow=all

host=freephonie.net

secret=SECRET

fromuser=09XXX

username=09XXX

dtmfmode=inband

qualify=6

fromdomain=freephonie.net

[freep honie_inbound]

type=peer

context=incoming

host=freephonie.net

qualify=6

allow=all

deny=0.0.0.0/0..0.0.0

permit=212.27.52.5/255.255.255.255 ; ip de freephonie.net

 

etension.conf

 

 

...

[incoming]

exten => s,1,Ringing

exten => s,2,Noop(I receive a sip call);

exten => s,n,Goto(home,1000,1)

exten => s,n,Congestion

;

...

 

 

 






!DSPAM:462643f450705772331342! 

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Re: [asterisk-users] Monitor application inestability and high load

[Matthew J. Roth]

Edgar A. Luna Diaz wrote:


I'm having high load, choppy sound and slow responsives with an 
asterisk server (version 1.2.12.1) that make a peak of 90 channels 
(around 60 phones calling at max, isn't necessary to reach this peak 
to get the problem). All the traffic is SIP, with recording for every 
call.


What codecs are you using?  Are you performing any transcoding?  What 
format are you using for the recordings?


The problems are detected in the high count of asterisk processes and 
sh wrappers to soxmix which could be as old as 1 hour in the server 
without a reason to stay idle, but for some unknow reason this "sh" 
don't die fast. This is when the dialplan calls Monitor obviously. I 
already tried to switch to MixMonitor but yesterday users reported 
that in some calls the recording isn't complete. Which is similar to a 
bug that is mentioned in mantis but for versions prior to 1.2.7. The 
asterisk logs don't show any particular message in verbose level 3. 
Apart from the recording, I have a high use of Manager and the mysql 
is used for some bussines logic but I think that nothing to high load, 
indeed mysql never is the most important part in processor, memmory 
and disk access statistics.


What do your disk access statistics look like?  In my experience, your 
call quality will begin to seriously deteriorate as you approach 60 
simultaneous recordings.  This is because the Monitor() code places a 
disk write in the code path that bridges channels.  If it's possible to 
disable recordings for a while, you can confirm whether or not this is 
the source of your problem.


Any knows a solution to this problem? or has an explanation for it?

In general, you should try to offload as many processes from the 
Asterisk server as possible.  MySQL is a good candidate for that, but 
the big one is soxmix.  Transcoding audio files is CPU intensive and I 
wouldn't be surprised if it impacted your call quality.


I've overcome the Monitor() problem by writing to a RAM disk.  At the 
end of each call, the recordings are moved from the RAM disk to a remote 
machine via NFS where they are mixed and indexed for retrieval.  My 
documentation of this process is available here:




and here:



Note that we are overconfigured for RAM.  You could probably get away 
with a 2 GB RAM disk.  I've also made some refinements to the setup that 
increased reliability.  If you choose to go down this path, I'll be 
happy to discuss them with you.


Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer


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[asterisk-users] Setup Asterisk configuration

[Tim Verscheure]

Hi,

I'm new to this list. For the last couple of days I was searching for
a good solution using AsteriskNOW. I noticed that in the configuration
steps of the server, they asked for a service provider. We don't
really need one.

We had something in mind like installing two Asterisk servers and make
a connection between them. It's just to connect two buildings. Would
this be a right approach? And when this is done, the one server could
be used as a service provider (SIP server) for the other one and vice
versa?

I'm pretty new to the whole Asterisk environment so better solutions
or configurations are welcome.


Tim
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Re: [asterisk-users] Re: Trigger for unavailable SIP peer

[Theo Band]

Edoardo Serra wrote:
> I'm using zabbix (http://www.zabbix.com/) as a complete monitoring
> solution
>
> zabbix agent has the possibility to specify custom checks that are run
> as often as you wish
> (maybe an "asterisk -rx "sip show peers" | grep UNREACHABLE | wc -l")
> the output of the script is sent to zabbix server which can fire
> actions (email, sms, etc)
> in a very flexible manner
This works only if you run zabbix as a privileged user (the same as runs
asterisk, so probably root).
But this is still a polling solution, not event based as the OP wants.
I tried this approach with zabbix to get an overview of the amount of
active lines versus time. I did not succeed because of the permission
problem. I can only achieve it using a cron job and have the result read
by zabbix, which is not very elegant.

My 1 cent :-)
>
> My 2 cents
>
> Regards
>
> C F ha scritto:
>> Thank you all for your response, but it appears that some of you
>> didn't understand my question. I know I can schedule a cron to check
>> the status (I can even use asterisk -rx "sip show peers" | grep
>> UNREACHABLE if I use a cron) but that is not what I want. I want
>> either a way that just as asterisk prints to the CLI  the following:
>> Peer '120' is now UNREACHABLE!  Last qualify: 118
>> it should also be able to trigger whatever action from a conf file or
>> the like.
>> Or if there is an available solution even that involves a cron job but
>> already has all the options, so I don't have to reinvent the wheel.

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Re: [asterisk-users] ztdummy does not load properly at server startup

[Eric \"ManxPower\" Wieling]

In the zaptel source "make config" will install the zaptel init script 
in /etc/rc.d/init.d for many distros.


Matthew J. Roth wrote:

Theo,

Unless things have changed significantly in the newer releases, you must 
load zaptel prior to loading ztdummy.  Additionally, the zaptel devices 
are not created instantly, so after you load zaptel you must wait a few 
seconds before loading ztdummy.  You can perform some sort of polling if 
you want to script this, but a less sophisticated method is just to 
sleep for 10 or 15 seconds between the calls to modprobe.


If your goal is to start Asterisk automatically at boot, some init 
scripts for different distributions are available at 
.  I'm 
using Fedora, so I installed 'rc.redhat.asterisk' with chkconfig as 
follows:


# install -m 755 ./rc.redhat.asterisk /etc/rc.d/init.d/asterisk
# chkconfig --add asterisk
# chkconfig --list asterisk
asterisk0:off   1:off   2:on3:on4:on5:on6:off

Note that I made the following customizations to the script prior to 
installing it:


* I don't want to run safe_asterisk, so I comment out all of the lines 
that reference the SAFE_ASTERISK variable.
* I want to load ztdummy and raise the open file limit, so I add the 
following lines to the start() function immediately prior to the 
'daemon' statement:

modprobe zaptel > /dev/null 2> /dev/null
sleep 15
modprobe ztdummy > /dev/null 2> /dev/null
ztcfg > /dev/null 2> /dev/null
ulimit -n 65536 > /dev/null 2> /dev/null
* And add the following lines to the stop() function, immediately after 
the 'RETVAL=$?' line:

rmmod ztdummy > /dev/null 2> /dev/null
rmmod zaptel > /dev/null 2> /dev/null

Things will differ depending on your distribution, but that should be 
enough to get you going in the right direction.


Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

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RE: [asterisk-users] Hardware suggestions for 8-10 lines in the UK

[Chris Bagnall]

> However, my experience hasn't been that VoiP is as reliable
> as copper lines and so unless you can tolerate the odd outage once per
> month or two then you might want to stick to copper for the main
> carrier?  Does this match with the experience from others?

Until recently, I'd have agreed entirely with this statement. However, recently 
one of our clients (in a shopping centre) encountered a scenario where a 
contractor had chopped through a load of PSTN lines whilst enlarging the car 
park. Their calls were routed via an ADSL connection which came in on an 
alternate location. For the better part of a week, they were the only shop in 
the building able to make & receive calls.

By having two independent net connections plus a "true" PSTN backup you've got 
3 levels of redundancy. Short of spending a fortune on "guaranteed fix times" 
from the telco, you're unlikely to do much better than that.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit http://www.minotaur.it/chris.html
This email is made from 100% recycled electrons


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Re: [asterisk-users] Asterisk 1.4.2 + Cisco 7960G not registering

[David Olsen]

On 2007-04-19 at 13:09:51, Doug Lytle <[EMAIL PROTECTED]> wrote:
> Do you see anything weird when logging (telnet to the ip) into the phone 
> and doing a show register?

I see as follows:

cisco-7960> show reg

LINE REGISTRATION TABLE
Proxy Registration: ENABLED, state: REGISTERING
line  APR  state  timer   expires proxy:port
  ---  -  --  --  
1 .1x  REGISTERING120 17  10.2.5.11:5060
2 .1x  REGISTERING120 17  10.2.5.11:5060
3 ...  NONE   0   0   undefined:0
4 ...  NONE   0   0   undefined:0
5 ...  NONE   0   0   undefined:0
6 ...  NONE   0   0   undefined:0
1-BU  .1x  REGISTERING120 16  10.2.5.11:5060

Note: APR is Authenticated, Provisioned, Registered

-d

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RE: [asterisk-users] Asterisk 1.2 and mixmonitor stopping short

[Edgar A. Luna Diaz]

I don't have any usefull information to add, just that as can be see in a mail 
from yesterday I have the same result, some files are being shorter than its 
must be.

Regards,


-Original Message-
From: [EMAIL PROTECTED] on behalf of Garth van Sittert
Sent: Jue 19/04/2007 12:04 p.m.
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk 1.2 and mixmonitor stopping short
 
Hi All

According to http://bugs.digium.com/view.php?id=6457 this has been 
resolved since 04-11-2006 and I have seen mentioned since 1.2.7.  I have 
tried using mixmonitor on asterisk 1.2.13 and 1.2.17 with the exact same 
results.  The WAV files are recorded but are cut short.  I am using a 
b410p card on the box.  When using HFC based cards I have no problem 
with the recording.

Does anyone have any ideas?  Is it possible to reopen this bug on the 
old 1.2 code?

Garth

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Re: [asterisk-users] ztdummy does not load properly at server startup

[Matthew J. Roth]

Theo,

Unless things have changed significantly in the newer releases, you must 
load zaptel prior to loading ztdummy.  Additionally, the zaptel devices 
are not created instantly, so after you load zaptel you must wait a few 
seconds before loading ztdummy.  You can perform some sort of polling if 
you want to script this, but a less sophisticated method is just to 
sleep for 10 or 15 seconds between the calls to modprobe.


If your goal is to start Asterisk automatically at boot, some init 
scripts for different distributions are available at 
.  I'm 
using Fedora, so I installed 'rc.redhat.asterisk' with chkconfig as follows:


# install -m 755 ./rc.redhat.asterisk /etc/rc.d/init.d/asterisk
# chkconfig --add asterisk
# chkconfig --list asterisk
asterisk0:off   1:off   2:on3:on4:on5:on6:off

Note that I made the following customizations to the script prior to 
installing it:


* I don't want to run safe_asterisk, so I comment out all of the lines 
that reference the SAFE_ASTERISK variable.
* I want to load ztdummy and raise the open file limit, so I add the 
following lines to the start() function immediately prior to the 
'daemon' statement:

modprobe zaptel > /dev/null 2> /dev/null
sleep 15
modprobe ztdummy > /dev/null 2> /dev/null
ztcfg > /dev/null 2> /dev/null
ulimit -n 65536 > /dev/null 2> /dev/null
* And add the following lines to the stop() function, immediately after 
the 'RETVAL=$?' line:

rmmod ztdummy > /dev/null 2> /dev/null
rmmod zaptel > /dev/null 2> /dev/null

Things will differ depending on your distribution, but that should be 
enough to get you going in the right direction.


Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

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RE: [asterisk-users] SIP kpml DTMF support in *

[Dan Austin]

Grigoriy wrote:
> Dan Austin wrote:
>> If you are running the phone loads that shipped with CCM5,
>> then your skinny phones have 'support' for RFC2833.  CCM
>> figures out during the call if the call will traverse a
>> SIP trunk and instruct the phone to use RFC2833 for DTMF
>> I have a CCM5<->Asterisk trunk setup for MeetMe conferencing
>> with NO MTP and DTMF works fine.
>>
   
> Can you specify the version of the loads?
Not specifically.  I am already up on CCM 5.1 which ships
with 8.0(4) for 7940/7960 phones.  I seem to recall that
CCM 5.0 had 8.0(1), but could be wrong.  In any case I
was using SIP trunks without MTP and with G729 25 days after
5.0 was released (I managed to get Cisco to actually release
it to me when they announced it as available instead of the
more normal 90 days after announcement)

Dan
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Re: [asterisk-users] Re: Trigger for unavailable SIP peer

[Edoardo Serra]

I'm using zabbix (http://www.zabbix.com/) as a complete monitoring solution

zabbix agent has the possibility to specify custom checks that are run 
as often as you wish

(maybe an "asterisk -rx "sip show peers" | grep UNREACHABLE | wc -l")
the output of the script is sent to zabbix server which can fire actions 
(email, sms, etc)

in a very flexible manner

My 2 cents

Regards

C F ha scritto:

Thank you all for your response, but it appears that some of you
didn't understand my question. I know I can schedule a cron to check
the status (I can even use asterisk -rx "sip show peers" | grep
UNREACHABLE if I use a cron) but that is not what I want. I want
either a way that just as asterisk prints to the CLI  the following:
Peer '120' is now UNREACHABLE!  Last qualify: 118
it should also be able to trigger whatever action from a conf file or 
the like.

Or if there is an available solution even that involves a cron job but
already has all the options, so I don't have to reinvent the wheel.


On 4/18/07, C F <[EMAIL PROTECTED]> wrote:

I use qualify in sip.conf and need to setup a trigger when asterisk
sees it as unreachable, so that I can either drop a call file, or send
an email, or both. How can I do that?

Thank you


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--
Ing. Edoardo Serra
WeBRainstorm S.r.l.
Via Pio Foà 83/C
10126 - Torino

Tel: +39 011 678 100
Fax: +39 011 678 275

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Re: [asterisk-users] Asterisk 1.4.2 + Cisco 7960G not registering

[Doug Lytle]

David Olsen wrote:

On 2007-04-19 at 12:29:50, Doug Lytle <[EMAIL PROTECTED]> wrote:
  

I was mistaken.  I just checked the phone, it's 7.4



Odd. No dice either there. I must be doing something else wrong. 

  


Do you see anything weird when logging (telnet to the ip) into the phone 
and doing a show register?


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] Asterisk 1.2 and mixmonitor stopping short

[Garth van Sittert]

Hi All

According to http://bugs.digium.com/view.php?id=6457 this has been 
resolved since 04-11-2006 and I have seen mentioned since 1.2.7.  I have 
tried using mixmonitor on asterisk 1.2.13 and 1.2.17 with the exact same 
results.  The WAV files are recorded but are cut short.  I am using a 
b410p card on the box.  When using HFC based cards I have no problem 
with the recording.


Does anyone have any ideas?  Is it possible to reopen this bug on the 
old 1.2 code?


Garth

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Re: [asterisk-users] Asterisk 1.4.2 + Cisco 7960G not registering

[David Olsen]

On 2007-04-19 at 12:29:50, Doug Lytle <[EMAIL PROTECTED]> wrote:
> I was mistaken.  I just checked the phone, it's 7.4

Odd. No dice either there. I must be doing something else wrong. 

-d

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Re: [asterisk-users] MeetMe Error

[Theo Band]

Manolet Gmail wrote:
> I use modprobe ztdummy, next i restart asterisk and now works fine,
> modprobe is to load the driver rigth? what i need to do in order to
> load automatically, not at the boot time but when asterisk start?
>
Funny. I just posted exactly the same question. How someone has an
answer :-)

Theo
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Re: [asterisk-users] SIP kpml DTMF support in *

[Grigoriy Puzankin]

Dan Austin wrote:
> If you are running the phone loads that shipped with CCM5,
> then your skinny phones have 'support' for RFC2833.  CCM
> figures out during the call if the call will traverse a
> SIP trunk and instruct the phone to use RFC2833 for DTMF
> I have a CCM5<->Asterisk trunk setup for MeetMe conferencing
> with NO MTP and DTMF works fine.
>   
Can you specify the version of the loads?
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Re: [asterisk-users] [OT] Nokia E60 firmware update break SIP

[Remco Barendse]

On Mon, 16 Apr 2007, Martin Joseph wrote:

Just a warning for you all that are using Nokia series E phones for SIP 
function.


I updated my phones firmware today using the Nokia Updater,  and now the SIP 
functionality, which previously worked pretty well is completely broken.


The phone no longer registers with asterisk, although it displays the little 
icon as though it has, and it doesn't even seem to try to pass calls to 
asterisk...


So,  I would avoid 3.06330904 20-11-06 RM-49


Where did you find this version?  My Nokia updater only offers an update 
to 2.0something  (my phone had 1.0something)


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Re: [asterisk-users] Asterisk 1.4.2 + Cisco 7960G not registering

[Doug Lytle]

David Olsen wrote:

On 2007-04-18 at 22:57:39, Steve Finkelstein <[EMAIL PROTECTED]> wrote:
  

I was only able to get a stable setup after I moved my 7940 back to SIP
version 7.5
  


I'm working with Steve on this issue, and the phone was running 8.6 when we
originally tried it. Just for kicks i downgraded it to 7.5 this morning, only
to uncover the same issue with that version.

  


I was mistaken.  I just checked the phone, it's 7.4

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] AudioCodes MP-104 MGCP?

[J. David Bavousett]

Greetings;

We are trying to get Asterisk up and happy at our site-we tried VOIP
using Sphere about a year ago, spent a *boodle* on expensive hardware
and services from a local "expert", but it never was happy. 

I'm brand-spanking new at VOIP, and I've learned a *ton* getting
Asterisk breathing in the last couple of days.  I have three Polycom
Soundpoint IP 500 SIP phones, which are all working wonderfully, when
talking to each other.  Voicemail seems happy, tooNow to get it
talking to the outside world.

In our previous trial, we purchased an AudioCodes MP-104 MGCP gateway.
I'm supposing that something needs to be set up in mgcp.conf for it, but
what I've tried just hasn't worked.  Here's what I have:

-

[general]
port = 2427
bindaddr = 0.0.0.0

[mp104-1]
host = 192.168.10.179
callgroup=1
context=default
callerid="Incoming Line 1"
line => aaln/1

-

In the Asterisk CLI, I'm getting "NOTICE[12321]: chan_mgcp.c:1656
find_subchannel_and_lock: Gateway 'mp104-1' (and thus its endpoint '*')
does not exist"...several of them, every few minutes.  

Is there someone out there who wouldn't mind sharing relevant sections
of the .conf files to get this thing *basically* working?  Incoming
calls should fall on a specific extension (199), outgoing should be
accessible by keying 9+the number.  All our long distance, 1-800
outbound, etc, all go out the same four lines...this is a *simple*
setup.

Apologies if this is something that's been answered before, or if I've
just missed the link I need, and thanks in advance for any assistance
you can give.  Our old PBX has perished, and I need to get this
breathing *quickly*.

J. David Bavousett
System Administrator
Abilene Library Consortium 
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[asterisk-users] ztdummy does not load properly at server startup

[Theo Band]

Hi

I run asterisk 1.4.2 with zaptel 1.4.1.
Zaptel is only needed for the ztdummy driver to get the Meetme()
application to work. I don't have any specific hardware.
And it does work nicely. When I reboot the machine however I have to
manually load the driver like this:

[EMAIL PROTECTED] ~]# modprobe ztdummy
Notice: Configuration file is /etc/zaptel.conf
line 0: Unable to open master device '/dev/zap/ctl'

1 error(s) detected

FATAL: Error running install command for ztdummy
[EMAIL PROTECTED] ~]# modprobe -r ztdummy
[EMAIL PROTECTED] ~]# modprobe ztdummy
Notice: Configuration file is /etc/zaptel.conf
line 0: Unable to open master device '/dev/zap/ctl'

1 error(s) detected

FATAL: Error running install command for ztdummy

[EMAIL PROTECTED] ~]# modprobe ztdummy

So a simple load of the driver does not seem to be enough. I need to
remove then wait and then load again.

My modprobe.conf contains (amongst others) this line:
install ztdummy /sbin/modprobe --ignore-install ztdummy  && /sbin/ztcfg

What is the proper way to get this driver loaded at startup? Have I
missed a configuration somewhere?

Thanks for any help,
Theo

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RE: [asterisk-users] SIP kpml DTMF support in *

[Dan Austin]

Grigoriy wrote:

> I'm trying to connect Asterisk 1.4 and Cisco CallManager 5 
> using SIP Trunk without MTP (media termination point). 
> Howerver, Cisco 79xx phones do not support RFC2833, they 
> always notify CCM5 via SKINNY channel no matter where they
> send RTP to.
If you are running the phone loads that shipped with CCM5,
then your skinny phones have 'support' for RFC2833.  CCM
figures out during the call if the call will traverse a
SIP trunk and instruct the phone to use RFC2833 for DTMF
I have a CCM5<->Asterisk trunk setup for MeetMe conferencing
with NO MTP and DTMF works fine.

> For non-MTP trunk there's Out-of-band DTMF support in CCM5 
> called "kpml". I wonder if Asterisk can support it.
Interesting, will look it up...

> I found an intertnet-draft for kpml:
> http://tools.ietf.org/id/draft-ietf-sipping-kpml-07.txt, but
> it seems to be very old - "Expires June 25, 2005".

> I know that using MTP in SIP Trunk at CCM5 makes DTMF work 
> in RFC2833, but MTP resource is very limited and I don't want 
> to proxy RTP via CCM5.
I don't blame you, nut again as of CCM5 you are no longer
required to use an MTP for SIP trunks.

> Please, do not offer to use H.323.
OK, not an offer, but I have found that even as of the latest
CCM5 release, the SIP stack is 'quirky'.  I also maintain
a H323 trunk between the same CCM cluster and Asterisk and
in general it is much better behaved (using chan_ooh323).
Either will work

Dan
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[asterisk-users] CDR(dst) != CALLERID(dnid)

[Rizwan Hisham]

Hi guys,
i just came to know that CDR(dst) field is set to current extension instead
of the dialed no. i need to set it to DNID because our every user has 5 dids
and i want to show the caller at the end of the month which numbers he
dialed for every call, along with other cdr info. Our rating depends on the
dialed number also. here is my extensions.conf

exten=> 1212,1,Dial(SIP/rizwan) ;Primary did for user rizwan
exten=> 1212,2,Hangup

exten=> 1714,1,Goto(,1212,1) ;Secondary did for user rizwan

here if the caller dials 1714 the call is jumped over to 1212, and the
CDR(dst) field after hangup is set to 1212. i need to set it to 1714 even if
the caller is on extension 1212 after hangup. so is there anyway to do this?

i dont want to do this for every secondary did
exten=> 1212,1,Dial(SIP/rizwan) ;Primary did for user rizwan
exten=> 1212,2,Hangup

exten=> 1714,1,Dial(SIP/rizwan) ;Secondary did for user rizwan
exten=> 1714,2,Hangup


--
Regards
Rizwan Hisham
Software Engineer
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[asterisk-users] Asterisk - Cisco Call Manager Express Trunk

[Diego Quintana Cruz]

Hi all,
I want to make a SIP trunk between a Cisco 2811 router and a Asterisk.
Both 2811 and Asterisk are working fine (2811 has 1XX and asterisk
2XX). Now I want to configure a trunk so that 2811 users can call *
users. I've been reading a lot but I'm still confused.

Hope you can help me,
--
Diego Quintana a.k.a. RouterMaN
Ingeniero de las Telecomunicaciones
Linux Registered User #382615 - http://counter.li.org/
SIP # 1-747-633-6676 Ext. 1011
FWD # 764839 Ext. 1011
http://routerman.blogsome.com
http://gst.telecom.pucp.edu.pe
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Re: [asterisk-users] Polycom IP 501 is displaying wrong time

[Steve Totaro]

Stephen Bosch wrote:

Steve Totaro wrote:
  

You can use the web interface and set it to -5 gmt.  Google for free NTP
servers.  I used to use time.nist.gov and got mixed results.  I found
another one that works almost all of the time.



If you use pool.ntp.org (or a regional variant thereof, such as
ca.pool.ntp.org) it will automatically load balance and you're certain
to get a working NTP source.

-Stephen-

  


Thanks for the tip.  I think that might be what I used for my last 
deployment but could not remember for sure (funny how I remember 
complicated things but something so obvious as ntp.org for NTP slips my 
mind)


Thanks,
Steve Totaro
www.asteriskhelpdesk.com
KB3OPB

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Re: [asterisk-users] Dial plans

[Steve Totaro]

[EMAIL PROTECTED] wrote:

How can I add extra digits to go through different carriers?

If it is long distance, but not a toll free number, then add "10 15
xxx".

  

Here are a couple of good reads for you
http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Planning
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial


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Re: [asterisk-users] Hardware suggestions for 8-10 lines in the UK

[Stephen Bosch]

Hi, Ed:

Ed W wrote:
> Agreed.  My experience is that quality is higher on Voip than it is via
> a TDM400p.  However, my experience hasn't been that VoiP is as reliable
> as copper lines and so unless you can tolerate the odd outage once per
> month or two then you might want to stick to copper for the main
> carrier?  Does this match with the experience from others?

Oh, yes. When I read Chris' message I thought -- "good thing he's still
got some PSTN lines." I would never, ever suggest that anybody go all
VoIP. The reliability is a long way from being where it needs to be. As
an adjunct, it's great, but I'd never replace PSTN service across the board.

Obviously I can't speak for BT and the UK circumstance, but once the
card a TDM400P card is tuned properly (which, I admit, can take some
effort), the call quality is excellent. The biggest hurdle, of course,
is echo, and SIP calls aren't (normally) afflicted with that because the
echo cancellation is present along the entire path.

-Stephen-

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Re: [asterisk-users] Dial plans

[Alex Balashov]

On Thu, 19 Apr 2007, [EMAIL PROTECTED] said something to this effect:


How can I add extra digits to go through different carriers?

If it is long distance, but not a toll free number, then add "10 15
xxx".


  Use IF conditionals in the dial plan to manipulate strings and/or create 
new dial strings based on ${EXTEN}?


--
Alex Balashov <[EMAIL PROTECTED]>
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Re: [asterisk-users] Polycom IP 501 is displaying wrong time

[Stephen Bosch]

Steve Totaro wrote:
> You can use the web interface and set it to -5 gmt.  Google for free NTP
> servers.  I used to use time.nist.gov and got mixed results.  I found
> another one that works almost all of the time.

If you use pool.ntp.org (or a regional variant thereof, such as
ca.pool.ntp.org) it will automatically load balance and you're certain
to get a working NTP source.

-Stephen-

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Re: [asterisk-users] Re: Can I add distinctive ring with asterisk and TDM400?

[Stephen Bosch]

Brian McEntire wrote:
> A follow-up with the solution in case anyone else is looking for this
> answer:
> 
> I created two contexts in my zapata.conf file, since each VOIP line is
> terminated by a VOIP adapter and then just comes in "hardwired" to the
> TDM400 via RJ11 line, I know which VOIP number is connected to which
> Wildcard port.
> 
> In Zapata.conf:
> 
> usedistinctiveringdetection=yes

Is this line even necessary? You're sending distinctive ring, not
receiving it.

-Stephen-

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RE: [asterisk-users] RE: OT (a little): IPV6 Ramifications Article

[Chris Bagnall]

> If there was something useful in ones kettle having an ethernet
> connection, it would probably already have it.  After all, with NAT'ing
> there's no real shortage of IP-addresses.  And perhaps we would already
> have K2K networks, with K2K proxies etc.

It'd be great if I could get my kettle to generate MRTG or Cacti graphs for 
number of times boiled in a period, amount of water in it at the time, amount 
of water poured from it, thus being able to work out power efficiency, etc. 
etc. ;-)

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit http://www.minotaur.it/chris.html
This email is made from 100% recycled electrons


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Re: [asterisk-users] Outgoing CallerID

[Alex Balashov]

On Thu, 19 Apr 2007, Forrest Beck said something to this effect:


I thought of maybe adding a key for each extension to the astdb and
have a Macro query the astdb.  Any other ideas?


  That would work, and is certainly the easiest, since you can bulk-load
the DID -> extension maps via external CLI commands with a simple script.

  You could also have Asterisk do MySQL dips for this information, if the
desire is to administer it from a web-based front-end.  Or if there is
some sort of mathematical relationship between the extension and the DID
range, the dialplan interpreter itself is capable of fairly sophisticated
mathematical extrapolations.

-- Alex

--
Alex Balashov <[EMAIL PROTECTED]>
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Re: [asterisk-users] Polycom IP 501 is displaying wrong time

[Dave Miller]

Chris Mason (Lists) wrote on 4/19/07 6:10 AM:
> If your phone is getting its parameters by DHCP from a linux server, add
> the NTP server option  to that server:
> in /etc/dhcpd.conf
> option time-servers 192.168.0.3;
> 
> If your phone is getting an NTP server setting by DHCP server, you can't
> override that from any setting. I came across this where a polycome 501
> was connected to the internet directly and comcast was setting NTP to
> 10.10.x.x, which was ridiculous. Their tech support could never
> understand why this was a problem and would not address the problem
> despite repeated calls.

Also of note is that the time zone can also be set via DHCP, and if it
is, that can't be overridden in the phone, either.

-- 
Dave Miller   http://www.justdave.net/
System Administrator, Mozilla Corporation  http://www.mozilla.com/
Project Leader, Bugzilla Bug Tracking System  http://www.bugzilla.org/
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[asterisk-users] Dial plans

[ctotos]

How can I add extra digits to go through different carriers?

If it is long distance, but not a toll free number, then add "10 15
xxx".

-- 
Thanks
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[asterisk-users] Outgoing CallerID

[Forrest Beck]

I am not sure of the best way to do this, so I thought I would query the list.

I have about 100 internal extensions ranging from 2000 - 2100.  Each
internal extension has a external DID number.  For example: 2001 =
5552871620.  As you can see the internal externsion and DID don't
match in any way.  What would be the best way to set the DID for when
a extension dials out on the PRI?  In sip.conf I am using CallerID as
their internal number.

I thought of maybe adding a key for each extension to the astdb and
have a Macro query the astdb.  Any other ideas?

Thanks.

--
***
Forrest Beck
IAXTEL: 17002871718
[EMAIL PROTECTED]
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