[asterisk-users] IVR integration with third party application Help wanted

[Notify Me]

Hi list,

I hope this isn't in error but if it is I apologize.

I have a small project request on hand where the clients want their
customers to be able to dial in to conduct business over the phone in a
completely automated manner. From my limited understanding this looks a lot
like a call center where one has to build some sort of proxy that
understands their business logic and that can report stuff back to asterisk
which then reports it back to the customer.
I have little or no understanding of AGI or related architecture,  I just
know how to setup asterisk as a call manager.
if anyone would be willing to help me out to understand what needs doing
i'd be very grateful.

Thanks for listening,  and hope to hear from you soon!
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Re: [asterisk-users] IVR integration with third party application Help wanted

[Notify Me]

Hi and thanks for the response, much appreciated!
From what I'm being told, its some sort of pension (financial)
organization, customers are supposed to be able to manage their accounts
over the phone. That's all I know so far.
On Oct 20, 2013 5:57 PM, Steve Edwards asterisk@sedwards.com wrote:

 On Sun, 20 Oct 2013, Notify Me wrote:

  I have a small project request on hand where the clients want their
 customers to be able to dial in to conduct business over the phone in a
 completely automated manner. From my limited understanding this looks a lot
 like a call center where one has to build some sort of proxy that
 understands their business logic and that can report stuff back to asterisk
 which then reports it back to the customer.


 We need a lot more detail like what does 'conduct business over the phone
 in a completely automated manner' mean? Are customers calling in and
 ordering ink cartridges?

 To me, 'build some sort of proxy that understands their business logic'
 does not sound like a 'small project.'

 --
 Thanks in advance,
 --**--**
 -
 Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 Newline  Fax: +1-760-731-3000
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Re: [asterisk-users] Integration with skype

[me]

On Fri, 24 May 2013, Markus wrote:


Am 23.05.2013 16:04, schrieb Richard Kenner:

For voice, you can use SipToSis. Works flawlessly with Asterisk and the
best part, it's free. :)

www.mhspot.com/sts/
(site is down right now)


And that's related to the problem with it: it hasn't been maintained for
quite a while.


 If you know of another FREE alternative let me know.


While I agree with what others have said about Skype being evil, you can
find another alternative at http://nerdvittles.com/?p=5671

I do not use Skype but I use some of his other stuff and for the most part
it Just works

Hope this helps.

Regards,

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Re: [asterisk-users] Asterisk for Razberry Pi

[me]

On Wed, 2 Jan 2013, Robert Rawlinson wrote:


Has anyone ported Asterisk to the Razzberry Pi? If so could you point me
to info on doing so?


Maybe you will find this interesting:

http://nerdvittles.com/?p=3880

Regards,

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Re: [asterisk-users] AsteriskNOW x86_64 install GPT partitions

[me]

On Sat, 29 Sep 2012, Wrinkled Cheese wrote:


Hello everyone,

I'm having an issue installing AsteriskNOW 2.0.2 on a Dell server.  When I
go to install it, with BIOS legacy mode for partition tables, I get as far
as setting up the partition tables.  However, the installer then informs me
that GPT partition table schemes are required and that I have to resolve
the issue.

I changed from BIOS/MBR/Legacy mode to GPT/UEFI boot mode but then the
installer fails to install.  Upon investigation it seems that this is a
fault of the CentOS installer.


What makes you think Centos is at fault?


There must be a work around to this issue
since it seems that the partition tables require GPT mode but the installer
for the x86_64 disc don't support UEFI mode.o


How big are your partitions? The only reason Centos requires GPT is if you
have a partition larger than 2TB.


Does anyone know of a work around or solution to this problem?


If your partitions are not over 2 TB then you do not need GPT. The 2TB limit
is not a Centos limitation. It is a limit of the msdos boot block. This also
affects Windoze.

Regards,

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Re: [asterisk-users] asterisk distributions

[me]

On Thu, 1 Mar 2012, Ralph Green wrote:


Howdy,
 I have tried all of these and a few more.  PBXinaFlash gave me the
best results, by far.  AsteriskNow produced a basic working system.  I
could not get any of the others configured to work at all.  I should
tell you my restrictions.  I was evaluating these distros to see which
one I could use to teach at a local computer group.  I wanted to do
very little configuration through the command line, since my goal was
not just to get a working system, but to have something I could easily
show others how to setup.  And, I was using real phone hardware.  My
phone and line were driven from a Digium TDM400.  The AsteriskNow
system only worked because someone on IRC helped me find a couple of
obscure setting, but it does work.

So, it somewhat depends on your needs, but I'd go with PBXinaFlash.
And, I added the IncrediblePBX package.  It is not perfect.  I am now
trying to add IAX trunks, and the mysteries involved make that slower
than I would like.
Good luck


I too tried PIAF and while it worked, the big problem I had with it and
the reason I dumped it was because a lot of the scripts are compiled and
encrypted. This restricts what you can do with the system without reinventing
what they have already done. It is also possible this has changed as I have
not looked at PIAF in a couple of years. PIAF is also very attractive because
of the addons provided by Nerd Vittles and company. Some of them bolt onto
asteriskNow without much difficulty others not so easily.

I settled on AsteriskNow and have had it running for a couple of years.

It really just depends on what you want to do with the system. If you
do not mind the closed nature of the PIAF custom scripts than that can be
a good choice.

My next adventure is going to be testing the freepbx distro. If looks
like it should be easy to get going and support but I have not tried it.

Regards,

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Re: [asterisk-users] Safe to upgrade to Centos 5.6 now ???

[me]

On Thu, 14 Apr 2011, Vahan Yerkanian wrote:


On 4/14/11 1:04 AM, Shaun Ruffell wrote:

On Wed, Apr 13, 2011 at 01:00:36PM -0700, Jian Gao wrote:

Centos 5.6 came out. Any one tried to update to the 5.6 yet?

I am running Asterisk 1.8 and is there any risk to upgrade to Centos 5.6?


I'm not sure about Asterisk in general, but if you use DAHDI, please be 
sure

to install version 2.4.1.2.

http://lists.digium.com/pipermail/asterisk-announce/2011-April/000313.html



A word of notice: asterisk/digium yum repos xmls haven't been updated yet 
(properly):


Yes, I noticed that also. For some reason the latest Dahdi rpms are sitting in
the top level dir at http://packages.asterisk.org/centos/5/current/ but they are
not signed. They need to be signed and moved into the approiate arch directory
and the yum metadata rebuilt for them to be seen by yum.

In the mean time if you trust them, you could use wget to download the rpms
you need into a local directory and then do yum localupdate *.rpm to update
them.

Hope this helps.

Regards,


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Re: [asterisk-users] AsteriskNow updated to Centos 5.6 and DAHDI doesn't work

[me]

On Sun, 10 Apr 2011, Shaun Ruffell wrote:


On Sun, Apr 10, 2011 at 05:32:51PM +0200, bakko wrote:

this diff solve the problem:

--- include/dahdi/kernel.h 2010-08-19 20:03:25.0 +0200
+++ include/dahdi/kernel.h 2011-03-18 11:32:32.0 +0100
@@ -86,7 +86,9 @@
 #endif

 #if LINUX_VERSION_CODE  KERNEL_VERSION(2,6,26)
-#define dev_name(dev)  (dev)-bus_id
+#if RHEL_RELEASE_CODE  RHEL_RELEASE_VERSION(5,6)
+#define dev_name(dev) (dev)-bus_id
+#endif
 #define dev_set_name(dev, format, ...) \
  snprintf((dev)-bus_id, BUS_ID_SIZE, format, ## __VA_ARGS__);
 #endif



There is a patch for this attached on issue 18992 [1] that should work
regardless of which distribution may have back ported the dev_name
definition.  It will apply cleanly to both 2.4 and the current trunk.
For example, to apply it on top of 2.4.1.1:

]# svn co http://svn.asterisk.org/svn/dahdi/linux/tags/2.4.1.1 
dahdi-linux-2.4.1.1
]# cd dahdi-linux-2.4.1.1
]# wget 'https://issues.asterisk.org/file_download.phpfile_id=29097type=bug' 
-O - | patch -p1

[1] https://issues.asterisk.org/view.php?id=18992


Will this be added to the AsteriskNow rpms at some point or do I need to
build my own?

Also, does this mean that for RHEL-6 and its clones there is going to be an
issue?

Regards,

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[asterisk-users] asterisk-users@lists.digium.com Hello!

[Me]

Hi!
I saw your profile and would like to get to know you better.
I’m looking for open, adventurous people, in my area, but we can start here.
Email me back at maris...@email-chatting.com .
Muah!

Marishka ;-)


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Re: [Asterisk-Users] Help with JIAXClient

[me me]

I have already get a register, but I can't make a call.I had to setup a listener in order to get the register, but once the register is set I can't make a call in any way.Any hint with that??Thx in advance.Richard OSS [EMAIL PROTECTED] escribió: I think you have to set where to get the libraries (jiaxc*.jar files).Setup a webserver somewhere and put the jar files there.  Then in your code before initialize   client.setCodeBase("your URL to the jar files");HTH,richard   Enrique Sanchez [EMAIL PROTECTED] wrote:I'm trying to make a little example programfor register to an Asterisk PBX and dial a softphone, but i just can't register to the PBX.package iax;  import net.sourceforge.iaxclient.Call;import net.sourceforge.iaxclient.JIAXClient;import net.sourceforge.iaxclient.Registration;  public class TestIAX  { public static void main(String[] args) {   Registration registration; JIAXClient client = JIAXClient.getInstance(); client.initialize (1, 10); registration = client.register("kike", "elkike", "10.32.81.31:4569"); client.setCallerID("Kike", "1001");  client.call("1002"); System.out.println(registration);  }}  I'm frustrated because JIAX doesn't throw any exception, but the code is not working properly.  Greetings,  -- Enrique Sanchez ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options
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[Asterisk-Users] Plain Text Passwords for IAX and SIP

[Me]

Can someone tell me if passwords are sent in plain text when using IAX?

I have been told already that SIP automatically encrypts the password?

Anyone know of some good Asterisk security links, docs, articles?

Thanks!
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Re: [Asterisk-Users] Linksys WIP300 WiFi Phone

[Me]

Where are you folks getting the best deal on this phone right now?

After my experience with the last Wifi phone, I am a little gunshy at
the moment. I am not sure if I should wait to see how this product
plays out or not.



On 2/27/06, Philip Edelbrock [EMAIL PROTECTED] wrote:

 Omar A. Sabek wrote:
  Like BJ, I'm sorry you had bad luck Phil. I have been playing with
  this phone all weekend, and I have had minor problems. The voice
  quality is as good as my cisco and polycom sip phones. I asked a
  friend to guess what kind of phone I was talking on and he said it
  sounded like a regular home or office phone. I have been very happy
  with the voice quality.

 My first day was a huge disappointment.  Three crashes, calls wouldn't
 work over my work's wifi (eventhough it registered ok), short battery
 time, lost settings after a crash, etc.

 However, after I went in and cleared my settings back to default, the
 troubles went away!  I'm been using it for over three days without a glitch.

 So, I would recommend to anybody else who is getting one of these
 phones, to immediately set all settings back to 'default' (under the
 Tools menu) before spending too much time configuring it.

  I reported on the voip-info page dismal talk times but it must have
  been an anomoly. Today I spoke for over an hour on the phone and still
  had plenty of juice left.

 My battery life seems to have improved as well.  I don't know if that's
 was a glitch fixed by setting things back to the defaults, or if cycling
 the battery is helping.  I also have less of a tendency to play with the
 menus, and the backlight could be a power drainer (it is quite bright).

 
  All-in-all this phone is a winner. It works with Asterisk flawlessly.

 As long as my troubles don't come back, I would agree.  I think my phone
 was shipped to me in a funny state causing it not to work right. It's a
 winner now.

 There are some little things I would wish for, but I'm quite happy with it.


 Phil
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Re: [Asterisk-Users] Linksys WIP300 WiFi Phone

[Me]

How is the voice quality?

  I've just plugged mine back into the charger after having used it
 nearly all day. I didn't have any of the problems you've described.
 Sorry you're having such bad luck with it. I'm not certain what the
 phones are rated to do, but I probably got better than 3 hours talk
 time on it today which is definitely the best I've gotten with any of
 the WiFi phones up to this point.

  BJ

 --
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[Asterisk-Users] Debugging Realtime Asterisk

[Me]

Is there any way to get debug info on res_odbc?  I get
the following but this is the last I ever see of
anything ODBC related.  Obviously, my extensions are
not working from the database, but I can connect to
ODBC via isql and run queries just fine.


Jul 17 22:12:13 NOTICE[3923]: res_odbc.c:215
load_odbc_config: registered database handle
'asterisk' dsn-[asterisk]
Jul 17 22:12:13 NOTICE[3923]: res_odbc.c:473
odbc_obj_connect: Connecting asterisk
Jul 17 22:12:14 NOTICE[3923]: res_odbc.c:488
odbc_obj_connect: res_odbc: Connected to asterisk
[asterisk]
Jul 17 22:12:14 NOTICE[3923]: res_odbc.c:518
load_module: res_odbc loaded.
 [res_config_odbc.so] = (ODBC Configuration)
Jul 17 22:12:14 NOTICE[3923]: config.c:836
ast_config_engine_register: Registered Config Engine
odbc


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[Asterisk-Users] Outbound answer on TDM400P

[Me]

How come an outgoing call using my TDM400P immediately
say the call is answered?  I'd like to be able to
detect when the call is actually picked up, is this
possible?

If this is normal with analog cards, does the same
thing happen with T1 cards?

-L



 
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[Asterisk-Users] Bridging and unbridging channels

[Me]

Is is possible to initiate a call that is not bridged
to the current channel?

I'd like to initiate an unbridged DIAL, announce the
party that is calling, and then bridge the two calls
together.  Is this possible?

Thanks,
-L

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[Asterisk-Users] BN8S0 crash linux on connect

[me me]

I have installed a BN8S0 whith chan_misdn (snapshot
09_05_05) in a SuSE 9.0. I have updated the kernel to
2.6.9 in order to make chan_misdn works. And Asterisk
1.0.7 I use mISDN_for_PBX4Linux_2005_03_06 and
mISDNuser_for_PBX4Linux_2005_01_28

It works great, but today I have been doing a test
with the full eigth ports. I have 2 ISDN BOX from
TELCO1 and 3 ISDN from TELCO2 (TELCO1 and TELCO2 are
the telephonic providers of my country).

I have connected the first 5 ports of my BN8S0 to that
ISDN boxes. When I place a call from any of my ip
phones, the asterisk place the call and the called
answer. Exactly then, the whole linux crashes.

The only information I get from dmesg was this
messages:

Jun 2 12:43:24 kernel: dev_manager prim f1780 not
handled
Jun 2 12:43:24 kernel: unregister_instance: no layer
found
Jun 2 12:43:44 kernel: dev_manager prim f1780 not
handled
Jun 2 12:43:44 kernel: unregister_instance: no layer
found
Jun 2 12:44:54 kernel: dev_manager prim f1780 not
handled
Jun 2 12:44:54 kernel: unregister_instance: no layer
found
Jun 2 12:45:42 kernel: dev_manager prim f1780 not
handled
Jun 2 12:45:42 kernel: unregister_instance: no layer
found
Jun 2 12:46:21 kernel: dev_manager prim f1780 not
handled
Jun 2 12:46:21 kernel: unregister_instance: no layer
found
Jun 2 12:46:24 kernel: dev_manager prim f1780 not
handled
Jun 2 12:46:24 kernel: unregister_instance: no layer
found
Jun 2 12:48:07 kernel: dev_manager prim f1780 not
handled
Jun 2 12:48:07 kernel: unregister_instance: no layer
found
Jun 2 12:48:07 kernel: dev_manager prim f1780 not
handled
Jun 2 12:48:07 kernel: unregister_instance: no layer
found
Jun 2 12:48:58 kernel: dev_manager prim f1780 not
handled
Jun 2 12:48:58 kernel: unregister_instance: no layer
found
Jun 2 12:49:02 kernel: dev_manager prim f1780 not
handled
Jun 2 12:49:02 kernel: unregister_instance: no layer
found
Jun 2 12:49:11 kernel: hfcmulti_l1hw: unknown
PH_SIGNAL info 1308
Jun 2 12:49:14 kernel: hfcmulti_l1hw: unknown
PH_SIGNAL info 1308
Jun 2 12:49:29 kernel: MISDN free_device: entitylist
not empty
Jun 2 12:49:30 kernel: hfcmulti_l1hw: unknown
PH_SIGNAL info 1308
Jun 2 12:49:35 last message repeated 3 times
Jun 2 12:51:24 kernel: hfcmulti_l1hw: unknown
PH_SIGNAL info 1308

The asterisk log doesn't show anything that seems
relevant in that interval of time.

Jun 2 12:47:53 DEBUG[4419]: Scheduling timer at 160
sample intervals
Jun 2 12:47:53 DEBUG[4419]: Generator got voice,
switching to phase locked mode
Jun 2 12:47:53 DEBUG[4419]: Scheduling timer at 0
sample intervals
Jun 2 12:47:53 DEBUG[4419]: Dropping duplicate answer!
Jun 2 12:47:53 VERBOSE[4418]: --
misdn/g:octoBRI/911032070 answered SIP/120-35da
Jun 2 12:47:53 DEBUG[4383]: Stopping retransmission on
'[EMAIL PROTECTED]' of Response 1: Found
Jun 2 12:48:07 DEBUG[4418]: Didn't get a frame from
channel: SIP/120-35da
Jun 2 12:48:07 DEBUG[4418]: Bridge stops bridging
channels SIP/120-35da and misdn/g:octoBRI/911032070
Jun 2 12:48:07 DEBUG[4418]:
misdn_hangup(misdn/g:octoBRI/911032070)
Jun 2 12:48:07 DEBUG[4418]: Exiting with
DIALSTATUS=ANSWER.
Jun 2 12:48:07 VERBOSE[4418]: == Spawn extension
(from_sip, 0911032070, 1) exited non-zero on
'SIP/120-35da'
Jun 2 12:48:07 DEBUG[4418]: update_user_counter(120) -
decrement inUse counter
Jun 2 12:48:07 VERBOSE[4419]: -- Stopped music on hold
on mISDN/2/911032070
Jun 2 12:48:07 DEBUG[4419]: Scheduling timer at 0
sample intervals
Jun 2 12:48:07 VERBOSE[4419]: == Spawn extension
(Horario_Oficina, s, 5) exited non-zero on
'mISDN/2/911032070'
Jun 2 12:48:07 DEBUG[4419]:
misdn_hangup(mISDN/2/911032070)
Jun 2 12:48:40 DEBUG[4383]: Auto destroying call
'[EMAIL PROTECTED]'
Jun 2 12:48:58 VERBOSE[4393]: -- Stopped music on hold
on mISDN/1/916733232
Jun 2 12:48:58 DEBUG[4393]: Scheduling timer at 0
sample intervals
Jun 2 12:48:58 VERBOSE[4393]: == Spawn extension
(Horario_Oficina, s, 5) exited non-zero on
'mISDN/1/916733232'
Jun 2 12:48:58 DEBUG[4393]:
misdn_hangup(mISDN/1/916733232)
Jun 2 12:49:02 VERBOSE[4392]: -- Stopped music on hold
on mISDN/1/916733232
Jun 2 12:49:02 DEBUG[4392]: Scheduling timer at 0
sample intervals
Jun 2 12:49:02 VERBOSE[4392]: == Spawn extension
(Horario_Oficina, s, 5) exited non-zero on
'mISDN/1/916733232'
Jun 2 12:49:02 DEBUG[4392]:
misdn_hangup(mISDN/1/916733232)

Thanks.



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[Asterisk-Users] BN8S0 problems (was: chan_misdn problem)

[me me]

Ok, I have solved my problems by upgrading my
chan_misdn and downgrading my mISDNuser. Now I have
asterisk working with mISDN support.

My problem now is that no matter what I do always see
the link down.

I've plugged the BN8S0 adapter to get the 8 ports
working. When I plug to the ISDN box (using the
Beronet crossing map recomendations) I have no
response. I've tried to restart the port, restart the
asterisk, even reload the modules, but always the link
is down (on l1 and l2).

Any hint??

Thanks.





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RE: [Asterisk-Users] chan_misdn problem

[me me]

This is the /var/log/asterisk/full

May 27 06:51:08 VERBOSE[1107]:  [chan_misdn.so]May 27
06:51:08 VERBOSE[1107]:  [
chan_misdn.so] = (Channel driver for mISDN Support
(Bri/Pri))
May 27 06:51:08 VERBOSE[1107]:   == Parsing
'/etc/asterisk/misdn.conf': May 27 0
6:51:08 VERBOSE[1107]:   == Parsing
'/etc/asterisk/misdn.conf': Found
May 27 06:51:08 VERBOSE[1107]:   == Registered channel
type 'mISDN' (This driver
 enables the asterisk to use hardware which is
supported by the new )

I don't see any strange or wrong with that output.

I have tested two initd script, first I used one that
simple do

modprobe hfcmulti
layermask=0xf,0xf,0xf,0xf,0xf,0x0xf,f,0xf
protocol=0x22,0x22,0x22,0x22,0x22,0x22,0x22,0x22
type=0x08

modprobe mISDN_dsp

And i get that problem, then I begin to use that one
that I find in beronet web

MODPROBE=modprobe
RMMOD=rmmod
INSMOD=insmod

case $1 in
start|--start)
$MODPROBE mISDN_core
$MODPROBE mISDN_l1 debug=0
$MODPROBE mISDN_l2 debug=0
$MODPROBE l3udss1 
$MODPROBE mISDN_dsp debug=0
options=0x0

$MODPROBE hfcmulti type=0x08
layermask=0xf,0xf,0xf,0xf,0xf,0xf,0
xf,0xf
protocol=0x22,0x22,0x22,0x22,0x22,0x22,0x22,0x22

sleep 1
;;
stop|--stop)
$RMMOD hfcmulti
$RMMOD mISDN_dsp
$RMMOD l3udss1
$RMMOD mISDN_l2
$RMMOD mISDN_l1
$RMMOD mISDN_core
;;
restart|--restart)
sh $0 stop
sleep 2 # some phones will release tei
when layer 1 is down
sh $0 start
;;
help|--help)
echo Usage: $0
{start|stop|restart|help}
exit 0
;;
*)
echo Usage: $0
{start|stop|restart|help}
exit 2
;;

esac

but I get the same output from asterisk and die
exactly the same.

The BN8S0 have its 8 ports configured in TE mode.




--- David Phelan [EMAIL PROTECTED] escribió:
 Can you post the output of your asterisk log file
 and your initd script for
 starting mISDN.
 What versions of chan_misdn, ,mISDN and mISDNuser
 are you using.
 
 Also check to see that /dev/mISDN exists.
 
 Dave.
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On
 Behalf Of me me
 Sent: Thursday, 26 May 2005 9:18 PM
 To: Asterisk Users Mailing List
 Subject: [Asterisk-Users] chan_misdn problem
 
 I've installed asterisk 1.0.7 with linux kernel
 2.6.3 (patched for mISDN).
 
 I Compile mISDNuser and loaded de modules (hfcmulti,
 mISDNdsp) for my BN8S0 beronet card.
 
 I have installed chan_misdn-beta-0.0.3rc4 with no
 problems.
 
 I have configured my misdn.conf as follows:
 
 [general]
 context=default
 language=de
 debug=0
 immediate=no
 hold_allowed=yes
 
 [octoBRI]
 ports=1,8,2,7,3,6,4,5
 context=incoming
 msns=*
 
 when I start asterisk with asterisk -vvvc I
 get the following
 message and then asterisk dies:
 
 [chan_misdn.so] = (Channel driver for mISDN Support
 (Bri/Pri))
   == Parsing '/etc/asterisk/misdn.conf': Found
   == Registered channel type 'mISDN' (This driver
 enables the asterisk to
 use hardware which is supported by the new ) cannot
 request MGR_NEWENTITY
 from mISDN: Success Ouch ... error while writing
 audio data: : Broken pipe
 Warning, flexible rate not heavily tested!
 
 Can anyone help me??
 
 Thanks.
 
 
   
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[Asterisk-Users] chan_misdn problem

[me me]

I've installed asterisk 1.0.7 with linux kernel 2.6.3
(patched for mISDN).

I Compile mISDNuser and loaded de modules (hfcmulti,
mISDNdsp) for my BN8S0 beronet card.

I have installed chan_misdn-beta-0.0.3rc4 with no
problems.

I have configured my misdn.conf as follows:

[general] 
context=default
language=de
debug=0
immediate=no
hold_allowed=yes

[octoBRI]
ports=1,8,2,7,3,6,4,5
context=incoming
msns=*

when I start asterisk with asterisk -vvvc I
get the following message and then asterisk dies:

[chan_misdn.so] = (Channel driver for mISDN Support
(Bri/Pri))
  == Parsing '/etc/asterisk/misdn.conf': Found
  == Registered channel type 'mISDN' (This driver
enables the asterisk to use hardware which is
supported by the new )
cannot request MGR_NEWENTITY from mISDN: Success
Ouch ... error while writing audio data: : Broken pipe
Warning, flexible rate not heavily tested!

Can anyone help me??

Thanks.



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Re: [Asterisk-Users] txfax and Ghostscript 8.51

[Me]

If the problem is with libtiff, its a problem with every version i've
tried (3.5.7, 3.6.0,  3.6.1, 3.7.1 and 3.7.2)


On 4/30/05, Steve Underwood [EMAIL PROTECTED] wrote:
 Me wrote:
 
 Hi all,
 
 I'm trying to use spandsp and asterisk to send faxes. To do so I am
 creating tiffs with Ghostscript. When I use Ghostscript 6.50 it seems
 to work fine, but when I create the tiff using Ghostscript 8.51 (or
 7.06) txfax garbles the tiff and it comes through all messed up.
 First of all is this a known problem or is it just me. More
 importantly does anyone know of a way to fix this, I'd like to use
 8.51 instead of 6.50.
 
 By the way, if it makes a differnece i'm currently running
 [EMAIL PROTECTED] but I've encountered the same problem with all the other
 asterisk builds i've tried
 
 
 It is really a change to Ghostscript or a related change to libtiff
 causing you problems. Libtiff is the usual suspect when FAX images go wrong.
 
 Regards,
 Steve
 

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[Asterisk-Users] txfax and Ghostscript 8.51

[Me]

Hi all,

I'm trying to use spandsp and asterisk to send faxes. To do so I am
creating tiffs with Ghostscript. When I use Ghostscript 6.50 it seems
to work fine, but when I create the tiff using Ghostscript 8.51 (or
7.06) txfax garbles the tiff and it comes through all messed up. 
First of all is this a known problem or is it just me. More
importantly does anyone know of a way to fix this, I'd like to use
8.51 instead of 6.50.

By the way, if it makes a differnece i'm currently running
[EMAIL PROTECTED] but I've encountered the same problem with all the other
asterisk builds i've tried

thanks
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Re: [Asterisk-Users] voice pulse connect - no dtmf

[Me]

Ours just started working again..
- Original Message - 
From: Justin Richards [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Saturday, April 23, 2005 1:14 AM
Subject: Re: [Asterisk-Users] voice pulse connect - no dtmf

so how do we get this fixed, its happing to my one and only DID as well...
On 4/22/05, Me [EMAIL PROTECTED] wrote:
I had the same problem with another provider whom I got no response from 
as
usual..

We had 5 or 6 numbers that worked fine and one that just quit sending 
DTMF.


- Original Message - 
From: Doug Harris
To: [EMAIL PROTECTED] Digium. Com
Sent: Friday, April 22, 2005 11:52 AM
Subject: [Asterisk-Users] voice pulse connect - no dtmf

Hi,
I've got bunch of VP connect lines, and a day back two LA area numbers 
stop
sending DTMF.  They are IAX2.

So, simply my customers can dial in, it hit my IVR but when they punch-in
the number, my * running 1.0.7 cannot identify the dtmf. IAX debug does 
not
show dtmf being sent to me.

Just want to know whether any of you had this experience, and if so how 
that
was fixed. Funny thing is this happened on two dids and others are OK.

Cheers
DH

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Re: [Asterisk-Users] voice pulse connect - no dtmf

[Me]

I had the same problem with another provider whom I 
got no response from as usual..

We had 5 or 6 numbers that worked fine and one that 
just quit sending DTMF.



  - Original Message - 
  From: 
  Doug 
  Harris 
  To: [EMAIL PROTECTED] Digium. 
  Com 
  Sent: Friday, April 22, 2005 11:52 
  AM
  Subject: [Asterisk-Users] voice pulse 
  connect - no dtmf
  
  Hi,
  
  I've got bunch of 
  VP connect lines, and a day back two LA area numbers stop sending DTMF. 
  They are IAX2. 
  
  So, simply my 
  customers can dial in, it hit my IVR but when they punch-in the number, my * 
  running 1.0.7 cannot identify the dtmf. IAX debug does not show dtmf being 
  sent to me.
  
  Just want to know 
  whether any of you had this experience, and if so how that was fixed. Funny 
  thing is this happened on two dids and others are OK.
  
  Cheers
  
  DH
  
  

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Re: [Asterisk-Users] RE:Qwest opens 911 infrastructure to Vonage

[Me]

This is good but if your company name isn't Vonage, how do you get access?
- Original Message - 
From: Norm Zimon [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
asterisk-users@lists.digium.com
Sent: Friday, April 22, 2005 9:39 AM
Subject: [Asterisk-Users] RE:Qwest opens 911 infrastructure to Vonage


FYI!!!
Qwest opens 911 infrastructure to Vonage
Qwest Communications has agreed to give Vonage access to its 911
infrastructure. The deal will allow 911 calls from Vonage customers to
travel over Qwest's emergency calling infrastructure in 14 states,
enabling the calls to proceed directly to emergency dispatchers. News of
the arrangement appeared in a letter Vonage sent to the FCC earlier this
week. The agreement puts added pressure on the other Bell operating
phone companies--BellSouth, SBC Communications and Verizon
Communications--to open their 911 infrastructures to Vonage and other
VoIP services. Qwest, the smallest of the four Bell operating companies,
has trialed a Net-phone 911 service in Rhode Island and is promising to
launch a trial in New York City.

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[Asterisk-Users] RealTime ignoring switch = Realtime/context@realtime_ext

[Me]

OK, been messing with RealTime like a week off and on, I can safely say it's 
killing me!

I have dug and dug and dug to find what I am missing, no dice.
I am running the latest version of * from CVS as of about a week ago.
Call comes in from a PRI into the todd_test_1 extension, if I uncomment the 
lines for the _888 number directly in the extensions.conf file the call is 
answered without a problem. If I comment the lines and just leave the 
switch in place it's suppose to lookup the extensions from the mysql table 
from what I understand.

All I get when calling in from the PRI is this:
   -- Extension '8885551212' in context 'todd_test_1' from '2145551212' 
does not exist.  Rejecting call on channel 0/1, span 1

It appears that the switch command is totally being ignored. I also checked 
the MySQL logs to see if Asterisk/RealTime was even hitting it but I see 
nothing in the MySQL logs at all that would indicate Asterisk is talking to 
it.

My phone numbers/passwords etc. have been changed but most everything else 
in my configs are as is. Any help would be appreciated, I am sure I am just 
missing something really simple and I am gonna smack myself in the head when 
it's brought to my attention.


###   extensions.conf#
[todd_test_1]
switch = Realtime/[EMAIL PROTECTED]
;## New stuff for new system ##
;exten = _888NXX,1,Answer
;exten = _888NXX,2,Wait(1)
;exten = _888NXX,3,Playback(cannot-complete-as-dialed)
;exten = _888NXX,4,Playback(check-number-dial-again)
;exten = _888NXX,5,Hangup
#
---
##   extconfig.conf#
realtime_ext = mysql,mydbname,extensions_table
##

##   res_mysql.conf   #
[general]
dbhost = my.dbserver.com
dbname = mydbname
dbuser = mydbusername
dbpass = mydbpass
dbport = mydbport
dbsock = /tmp/mysql.sock
##
-
#   DB Schema   #
FieldTypeNullDefault
id  int(11) No
context  varchar(20) No
exten varchar(20) No
priority  tinyint(4)  No 0
app   varchar(20) No
appdata varchar(128)   No

1;todd_test_2;_888NXX;1;Wait;2
2;todd_test_2;_888NXX;2;SayNumber;102
3;todd_test_2;_888NXX;1;Playback;pbx-invalid
 

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Re: [Asterisk-Users] RealTime ignoring switch= Realtime/context@realtime_ext

[Me]

- Original Message - 
From: Ronald Wiplinger [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Wednesday, April 20, 2005 4:42 AM
Subject: Re: [Asterisk-Users] RealTime ignoring switch= 
Realtime/[EMAIL PROTECTED]


Me wrote:
OK, been messing with RealTime like a week off and on, I can safely say 
it's killing me!

I have dug and dug and dug to find what I am missing, no dice.
I am running the latest version of * from CVS as of about a week ago.
Call comes in from a PRI into the todd_test_1 extension, if I uncomment 
the lines for the _888 number directly in the extensions.conf file the 
call is answered without a problem. If I comment the lines and just leave 
the switch in place it's suppose to lookup the extensions from the 
mysql table from what I understand.

All I get when calling in from the PRI is this:
   -- Extension '8885551212' in context 'todd_test_1' from '2145551212' 
does not exist.  Rejecting call on channel 0/1, span 1

It appears that the switch command is totally being ignored. I also 
checked the MySQL logs to see if Asterisk/RealTime was even hitting it 
but I see nothing in the MySQL logs at all that would indicate Asterisk 
is talking to it.

My phone numbers/passwords etc. have been changed but most everything 
else in my configs are as is. Any help would be appreciated, I am sure I 
am just missing something really simple and I am gonna smack myself in 
the head when it's brought to my attention.


###   extensions.conf#
[todd_test_1]
switch = Realtime/[EMAIL PROTECTED]

shouldn't it be Realtime/[EMAIL PROTECTED]
or
[todd_test_1]
include = todd_test_2
[todd_test_2]
switch = Realtime/[EMAIL PROTECTED]
???
BTW, the numbering of the priorities should increase:
1;todd_test_2;_888NXX;1;Wait;2
2;todd_test_2;_888NXX;2;SayNumber;102
3;todd_test_2;_888NXX;1;Playback;pbx-invalid

bye
Ronald
Well, I am confused then about two things..
1- In switch = Realtime/[EMAIL PROTECTED] I am referring to 
todd_test_2 which is my context inside of the DB for the records I am 
referencing, I was not aware that this context also needed to exist within 
the text file extensions.conf.

2- Can I not have one context within the extensions.conf that has the switch 
command in it and then as many other context as I like within the database? 
I thought this was the whole idea, controlling the extensions from the DB 
which in my opinion includes using different context.

3- Someone mentioned to me the other day that I shouldn't have the same 
context in the DB as I have in the text file. For example, I think they told 
me it was a bad idea to have a context within the extensions.conf called 
todd_test_1 which had a switch command in it, then also have todd_test_1 
as the context in the DB. Maybe I totally misunderstood this person the 
other day regarding this. Basically this is why I now have two context 
todd_test_1 and todd_test_2.

Regarding my priority numbering, I know it was off but I am pretty sure that 
based on the error I am getting in the CLI when calling in as well as the 
fact that * never hits MySQL at all according to the logs, I would say the 
process never makes it to the database at all to even get to this error 
about the priority. But, thanks for letting me know, sometimes it's little 
things like this that can bugger you up along the way.

For your reference the error is below, this shows that it dies within 
todd_test_1:
   -- Extension '8885551212' in context 'todd_test_1' from '2145551212' 
does not exist.  Rejecting call on channel 0/1, span 1
Again, if I just add some lines to handle the call right under the switch 
command, all works well which tells me the switch command is likely being 
ignored totally.

FYI, I did install Asterisk-Addons, I am running the latest CVS as of a week 
or so ago, I do have the MySQL client and header libs installed. The MySQL 
server is on a box on the same LAN and is operational for other live 
services right now. I have double and triple checked my MySQL permissions, 
besides if it was rejected for permission reasons, I would show it in my 
MySQL logs.

I hate to be a ding bat here but, can someone tell me how to turn on Debug 
mode and where the debug logs show up? I am sure there is a Wiki page on 
this so a URL would be great, I will go dig for it some more now.

Thanks folks for all the help so far!
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Re: [Asterisk-Users] RealTime ignoringswitch=Realtime/context@realtime_ext

[Me]

Yes, I downloaded via CVS then ran make then make install.
In fact I did this again last night to be sure it was installed.
Maybe I downloaded the old add on package, and it didn't come with it. I 
have the latest version of Asterisk but I just pulled plain old 
Asterisk-Addons from CVS.

Do I need to pull a specific version?
- Original Message - 
From: Matthew Boehm [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, April 20, 2005 2:32 PM
Subject: Re: [Asterisk-Users] RealTime 
ignoringswitch=Realtime/[EMAIL PROTECTED]


chase1*CLI realtime mysql status
No such command 'realtime mysql' (type 'help' for help)
chase1*CLI
 This is your problem. You do not have res_config_mysql.so loaded.
 You said that you have downloaded the newest asterisk-addons. Did you 
compile
them? Did you install them?

-Matthew

This message was sent using IMP, the Internet Messaging Program.
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Re: [Asterisk-Users] RealTime ignoringswitch=Realtime/context@realtime_ext

[Me]

SUCCESS!!
OK, you were right on the money..
Problem was this..
The first time I installed it about a week ago, well... I don't think I did, 
I think I downloaded and forgot to install it. Then yesterday or today I 
installed again with no luck, then I realized that I only did a reload after 
the install. Of course the reload didn't load the new module!! So, I stopped 
and restarted Asterisk and now I get this at the CLI:

chase1*CLI realtime mysql status
Connected to [EMAIL PROTECTED], port 2891 with username ast_chase1 
for 2 minutes, 7 seconds.
Apr 20 21:15:14 DEBUG[15226]: res_config_mysql.c:605 mysql_reconnect: MySQL 
RealTime: Everything is fine.
chase1*CLI

Yeeehaww!!
Thanks a ton, now I can move on with the show.. Please let me know if there 
is anything I can do for you in return, I can't tell you how much I 
appreciate your help..

I can offer you at the very least some free domain registrations through 
Enom or a free dialup account for when you travel or??? Let me know how I 
can help..

Thanks again!
Todd Routhier
Lightwave Technologies, LLC.
- Original Message - 
From: Matthew Boehm [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, April 20, 2005 2:32 PM
Subject: Re: [Asterisk-Users] RealTime 
ignoringswitch=Realtime/[EMAIL PROTECTED]


chase1*CLI realtime mysql status
No such command 'realtime mysql' (type 'help' for help)
chase1*CLI
 This is your problem. You do not have res_config_mysql.so loaded.
 You said that you have downloaded the newest asterisk-addons. Did you 
compile
them? Did you install them?

-Matthew

This message was sent using IMP, the Internet Messaging Program.
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Re: [Asterisk-Users] Sipura SPA-841 Phone Review

[Me]

Well, I bought two of these when they were first released.. They seemed like 
VERY nice phones for the money except for the fact that the headset jacks 
did not work at all on either device. Tried multiple headsets none of them 
worked. I had to return the phones..

I also remember the buttons getting stuck down a lot..
It seems that there are LOTS of issues with the headset jack for folks. If 
Sipura could make the headset jack solid, it would be a great, affordable 
phone in my opinion.

SNIP 

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[Asterisk-Users] Firefly w/*?

[Me]

I have seen folks mention FireFly softphone on the list many times. I went 
to their website but could only find a version which connects directly to 
their service, it did not seem configurable to use with *.

Is FireFly in fact usable with *?
Thanks! 

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Re: [Asterisk-Users] Any work around for ISPs that block port 5060 and69

[Me]

Maybe they could start by finding the info on the lawsuit that was brought 
against the last ISP that tried this. They could then forward it to their 
ISP and see if that gets them anywhere.

I guess this could also get them disconnected from the only ISP available 
so... Don't listen to me... :)

- Original Message - 
From: Joel Jn-Francois [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, April 19, 2005 8:48 AM
Subject: [Asterisk-Users] Any work around for ISPs that block port 5060 
and69


I have a several friends registered on my asterisk box that are experience 
problems with their ISP blocking SIP default ports 5060 and tftp port 69. 
Is there any way around this problem or are they forever doomed to VOIP 
since their ISP is pretty much the only ISP company on that island.  So far 
I was able to have them change their default SIP port to 6070 and any 
packets coming in on that port on my asterisk box I would redirect to port 
5060.  That seem to be working fine, expect that they can make calls but 
cannot receive calls.  I think part of the problem might be that when 
asterisk tries to initiate a call to their sip phone it tries on port 5060 
instead of 6070 even though I have specified in the sip.conf that their 
port is 6070.  Has anyone else encountered this problem and was able to 
resolve it?  My other option is to change my asterisk box to work 
completely on a different port, but I am reluctant to do so since the 
majority of registered users for now do not have an issue with port 5060 
being blocked by their ISP.

Thanks for your help.
Joel
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[Asterisk-Users] RealTime Vs. AGI and PHP or MySQL calls within extensions.conf

[Me]

I may not understand fully how any of these three features work but...
Can someone tell me what benefit there is to using RealTime instead of say 
calling a MySQL database directly from the extensions.conf using the built 
in MySQL functionality? Also, it looks like I could use PHP via AGI to also 
lookup extensions?

Am I totally lost?
Thanks! 

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[Asterisk-Users] RealTime

[Me]

Is there any better docs or step by steps other than what's in the Wiki for 
Realtime setup?

We have been trying to get this running and it's driving us batty..
It seems that the switch command is totally being ignored as far as we can 
tell.

We are basically just getting an error telling us that the extension within 
default can't be found. We have the extensions in the table and have the 
switch command pointing out to RealTime.

If we put the extension in the text file it works, if we take it out of the 
text file it breaks.

We have searched and troubleshot all day, any other handy docs or step by 
steps out there?

We are using the latest * via CVS..
Thoughts?
Thanks.. 

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[Asterisk-Users] sip phones make connection but no-sound is heared

[me me]

This is the asterisk output:

 -- Executing Answer(SIP/202-8236, ) in new stack
-- Executing Dial(SIP/202-8236,
SIP/203|100|tTr) in new stack
-- Called 203
-- SIP/203-3c5d is ringing
-- SIP/203-3c5d answered SIP/202-8236
-- Attempting native bridge of SIP/202-8236 and
SIP/203-3c5d

It seems correct but no sound is heared on any phone,
any ideas??

Thx.



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[Asterisk-Users] New PRI install with new te110p

[Me]

Getting this error on a new install, I am lost since this is my first time 
messing with the te110p and my first PRI install.

I have signalling=pri_cpe as the Digium docs suggest, when I start Asterisk 
I get this over and over:

 == Primary D-Channel on span 1 down
Apr 13 01:30:08 WARNING[10128]: chan_zap.c:2054 pri_find_dchan: No 
D-channels available!  Using Primary on channel anyway 24!

If I change signalling to pri_net the errors go away, either way I can 
receive calls into Asterisk.

How should the signalling be set, to cpe or net?
Any idea what's causing this error?
I am not entirely sure my PRI is 100% up even, * seems to be talking to it 
because when I pull the cable it starts giving me alerts and such, the 
alerts go away when I plug the cable back in.

Of course the telco is waiting for me to call them so we can test the PRI 
against my equipment.. I guess they expect me to have known working 
equipment.. Well, it would help if I had a known working PRI to test and 
tweak my * box against..

SIGH..
Any help would be greatly appreciated!


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[Asterisk-Users] Beeps during Sip to Sip phone calls

[Me]

I keep hearing DTMF type beeps when on phone calls, I know this is some sort 
of trait of VOIP but it's driving me nuts..

I noticed that it happens MUCH more when I am on the phone with one 
particular person.

We are using SPA-2000's from Sipura on both ends.
Tonight I was looking at the CLI (*command line interface) while I was on 
the phone with this person.

Each time I heard a beep, I saw at EXACTLY the same time the following line:
   -- Attempting native bridge of SIP/206-5286 and SIP/109-fbf7
What's wierd is that we were already on the phone, I was 206 and he was 109. 
Does this give anyone a clue as to what might be happening here?

I also saw a bunch of these but not sure if it was related to our call or 
not.

Apr  6 21:16:03 NOTICE[3587]: rtp.c:430 ast_rtp_read: RTP: Received packet 
with bad UDP checksum
Apr  6 21:16:05 NOTICE[3587]: rtp.c:430 ast_rtp_read: RTP: Received packet 
with bad UDP checksum
Apr  6 21:16:39 NOTICE[3587]: rtp.c:430 ast_rtp_read: RTP: Received packet 
with bad UDP checksum
Apr  6 21:16:41 NOTICE[3587]: rtp.c:430 ast_rtp_read: RTP: Received packet 
with bad UDP checksum
Apr  6 21:16:44 NOTICE[3587]: rtp.c:430 ast_rtp_read: RTP: Received packet 
with bad UDP checksum
Apr  6 21:17:03 NOTICE[3587]: rtp.c:430 ast_rtp_read: RTP: Received packet 
with bad UDP checksum
Apr  6 21:17:10 NOTICE[3587]: rtp.c:430 ast_rtp_read: RTP: Received packet 
with bad UDP checksum

Thanks!
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Re: [Asterisk-Users] Beeps during Sip to Sip phone calls

[Me]

Anyone else using Sipura equipment and having excessive BEEPing?
Maybe a firmware upgrade would help?
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- Original Message - 
From: Rich Adamson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Thursday, April 07, 2005 12:52 AM
Subject: Re: [Asterisk-Users] Beeps during Sip to Sip phone calls


Inline...
I keep hearing DTMF type beeps when on phone calls, I know this is some 
sort
of trait of VOIP but it's driving me nuts..
Not really.
I noticed that it happens MUCH more when I am on the phone with one
particular person.
We are using SPA-2000's from Sipura on both ends.
I'm using a spa-3000 and have noticed the same thing. Some voices
trigger it, others don't. It hasn't happened often enough to cause
me to spend time on it. (I have the spa3000 configured so that
incoming fxo calls go directly to the fxs port (not through *), so
in my case the dtmf-like bursts have to be internal spa issues.
Since the spa2000 and 3000 share a lot of the same code base, the
tones you're hearing are likely internal spa issues as well.)
Tonight I was looking at the CLI (*command line interface) while I was on
the phone with this person.
Each time I heard a beep, I saw at EXACTLY the same time the following 
line:

-- Attempting native bridge of SIP/206-5286 and SIP/109-fbf7
What's wierd is that we were already on the phone, I was 206 and he was 
109.
Does this give anyone a clue as to what might be happening here?

I also saw a bunch of these but not sure if it was related to our call or
not.
Apr  6 21:16:03 NOTICE[3587]: rtp.c:430 ast_rtp_read: RTP: Received 
packet
with bad UDP checksum
Apr  6 21:16:05 NOTICE[3587]: rtp.c:430 ast_rtp_read: RTP: Received 
packet
with bad UDP checksum
Apr  6 21:16:39 NOTICE[3587]: rtp.c:430 ast_rtp_read: RTP: Received 
packet
with bad UDP checksum
Apr  6 21:16:41 NOTICE[3587]: rtp.c:430 ast_rtp_read: RTP: Received 
packet
with bad UDP checksum
Apr  6 21:16:44 NOTICE[3587]: rtp.c:430 ast_rtp_read: RTP: Received 
packet
with bad UDP checksum
Apr  6 21:17:03 NOTICE[3587]: rtp.c:430 ast_rtp_read: RTP: Received 
packet
with bad UDP checksum
Apr  6 21:17:10 NOTICE[3587]: rtp.c:430 ast_rtp_read: RTP: Received 
packet
with bad UDP checksum
I'd guess the above messages are simply damaged packets (eg, ethernet
collisions, broadband hits). Since there are multiple seconds between
most of those messages, I would doubt that you would actually notice
the hits in the audio.

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Re: [Asterisk-Users] Text Messaging or AIM

[Me]

I think I saw something a while back that would allow Asterisk to check AIM 
to see if a user of an extension was in front of their desk or not then send 
to VMail or whatever.

This may be a start for you but I can't recall the name of it or where the 
info is.

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- Original Message - 
From: Scheda [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
Asterisk-Users@lists.digium.com
Sent: Sunday, March 13, 2005 11:47 AM
Subject: [Asterisk-Users] Text Messaging or AIM


Does anyone know of a program/extention to asterisk that would allow
me to either text message my asterisk box or IM it from AIM on my cell
phone to allow it to call me? I've been looking with google yet can't
find anything. I don't code, so I'm SOL there, so I'm looking for
something premade. I plan on taking a class on perl during the fall
semester at my local community college, so if there isn't something
like this out there already, maybe I can get one out there.
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Re: [Asterisk-Users] Network Test Tool?

[Me]

Thanks, this looks like what I need. Setting it up looks like a career 
though but hey it's free so what can you do?

:)
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- Original Message - 
From: Florian Overkamp [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
asterisk-users@lists.digium.com
Sent: Tuesday, January 25, 2005 1:42 AM
Subject: RE: [Asterisk-Users] Network Test Tool?


Hi,
-Original Message-
We have been having WAY too many issues lately with our VOIP calls. I
suspect it may be the particular T1 we are pushing these
calls out through
from our office.
Is there a decent tool out there that I can stick on the
network that will
measure things like Jitter, ping times and overall network
quality for say a
24 hour period and stick it in a human readable report.
How about smokeping ? www.smokeping.org
Florian
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[Asterisk-Users] Ordering a Voice PRI for Asterisk

[Me]

We are in the process of ordering a Voice PRI to plug into Asterisk. Of 
course we will be buying a card from Digium for this.

Question is this, there seem to be MANY options technically when ordering 
this PRI (in the US) but since this is the first time ordering a voice 
circuit I am clueless as to what options we need.

Any clues would be helpful or maybe something has already been written about 
this?

Thanks in advance!
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[Asterisk-Users] Big Increase in SPAM over the last few weeks

[Me]

We have been seeing tons of additional SPAM coming through our Modus 4 
server, mostly medical stuff.

Is anyone else seeing a big increase lately? I have not seen the list for a 
bit seems I was unsubscribed somehow.

Thanks!
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[Asterisk-Users] RE: Big increase in SPAM lately

[Me]

Doh!
Wrong list, please ignore..
Sorry.. 30 lashes for me..
Todd
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[Asterisk-Users] Get SPA-2000 to dial out on one * and get calls in from a different *?

[Me]

I have my main * box setup for all incoming and outgoing calls to and from 
our SPA-2000's. I have now setup another * box in a different location and I 
would like the SPA's to send all outgoing calls out through the new * server 
but continue registering with the old * server so all incoming calls will 
still be routed through the old server to our SPA's.

In the SPA-2000 config screens under the Line 1 tab, I see a few entries 
of interest. I originally had them set like this:

Proxy: old server IP set here
Outbound Proxy: old server IP set here too
Use Outbound Proxy: YES
Use OB Proxy In Dialog: YES  --- no clue what this is for but it's set to 
yes and has always worked with the one server setup

Now I figured that I could just change the outbound proxy IP address to that 
of the new server and viola, all outgoing calls would go through the new * 
server! Well, they did but the problem is that my SPA-2000 stops registering 
with the server which owns the IP address in the Proxy field so no 
incoming calls get to the SPA anymore.

It seems that the Outbound Proxy field overrides the Proxy field, are they 
one in the same? If so, what's the purpose? Any help on getting this to work 
would be greatly appreciated.

Thanks,
Todd
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[Asterisk-Users] Making two * servers share same dial plan?

[Me]

Can someone point me to some docs that explain this or give me a direction 
to go in. I have seen docs on this in the past but can't seem to dig em up 
now when I need them.

Basically I want one Asterisk server to be the traffic cop and send some 
calls directly to ATA's and some calls to another Asterisk server, the other 
Asterisk server will then direct the calls to the end users ATA on their 
desk or to vmail etc.

Thanks..
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Re: [Asterisk-Users] Weird Issue: Call will not go into VM

[Me]

What is your setup? Zap, ATA's etc?
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- Original Message - 
From: [EMAIL PROTECTED]; [EMAIL PROTECTED]:Go Technology 
Management LLC [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, February 24, 2005 2:36 PM
Subject: [Asterisk-Users] Weird Issue: Call will not go into VM


Weird Problem:  I have 2 EXT. One will ring and NOT go into VM (eventually 
call will timeout/hang up), the other EXT goes into VM when the call is 
not answered like it should. If I enable DND, then the call will go 
directly in VM as it should.

Any ideas what it might be?
Thank you,
Jake
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Re: [Asterisk-Users] Damn DTMF Beeps on my calls

[Me]

Well this happens a LOT when I call one particular person, not so much when 
I call others.

Both sides of the call are running Sipura ATA's with * in the middle, no 
termination or Zap in between at all.

It seems that when I call this person from my home address it occurs a LOT 
like 1 or 2 times a minute or more at times. When I call from another 
location, same ATA type but different building it doesn't happen (I don't 
think).

The other caller does not here it at all, he only hears silence when I hear 
the beep. It sounds EXACTLY like a key being pressed on my phone. It's not 
just a beep, to answer the other posters question.

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- Original Message - 
From: Steven Critchfield [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Monday, January 24, 2005 2:38 PM
Subject: Re: [Asterisk-Users] Damn DTMF Beeps on my calls


On Mon, 2005-01-24 at 13:38 -0600, Me wrote:
Can someone give me a clue as to why I keep hearing DTMF type beeps on my
phone calls. It sounds exactly like someone on the other end is pushing a
key on their phone but they are not!
Has anyone ever heard of this before? It use to happen once in a while,
today it's been happening a LOT and it's driving me batty..
As usual, if you want to ask a smart question you need to add more
details.
DTMF can be caused by talk off. Essentially a voice pattern that
triggered the DTMF detection. Now for the part that would have been
smart, identifying the location your DTMF is being detected. If it where
all zap, then it is the DTMF routines in asterisk/zapata, but as you
didn't bother to expound what is going on, it could be SIP hardware
phones acting up on you.
More details please before you go batty.
--
Steven Critchfield [EMAIL PROTECTED]
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[Asterisk-Users] Damn DTMF Beeps on my calls

[Me]

Can someone give me a clue as to why I keep hearing DTMF type beeps on my 
phone calls. It sounds exactly like someone on the other end is pushing a 
key on their phone but they are not!

Has anyone ever heard of this before? It use to happen once in a while, 
today it's been happening a LOT and it's driving me batty..

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[Asterisk-Users] Network Test Tool?

[Me]

We have been having WAY too many issues lately with our VOIP calls. I 
suspect it may be the particular T1 we are pushing these calls out through 
from our office.

Is there a decent tool out there that I can stick on the network that will 
measure things like Jitter, ping times and overall network quality for say a 
24 hour period and stick it in a human readable report.

Thanks!
--
Start Your Own ISP!
http://www.YourOwnISP.com 

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[Asterisk-Users] modprobe: Can't locate module wctdm

[Me]

After getting zaptel from the CVS server, compiling and installing it I
type:

modprobe zaptel

and all is well. Then I type:

modprobe wctdm

and I get this:

modprobe: Can't locate module wctdm

Any idea why?

I did this yesterday but with the CVS head of Asterisk and I got by this
part without a problem. I reinstalled it all today because I wanted the
stable release on our server since we use it daily for calls in our office.

I am stuck.. Any help?

I am running Fedora 1 FYI.. Also, I compiled and installed Asterisk this
time before Zaptel, is that bad?

Thanks,
 Todd


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Re: [Asterisk-Users] modprobe: Can't locate module wctdm

[Me]

I get the same error with modprobe wcfxs.

It's weird, yesterday I installed CVS Head and the latest Zaptel and did not
have these problems..

I tried updated Zaptel via CVS then ran make clean; make install. When I
tried modprobe wctdm it still flaked and when I tried to start asterisk it
totally just blew up..

Since I have a little time at the moment and I am trying to learn from this,
I started over and formatted :)

Now I am wondering which version of Asterisk and which version of Zaptel I
should get this time... I want a stable release of Asteisk, not the latest
CVS but I am not sure if I need a matching version of Zaptel or if I can and
should get the latest version of Zaptel for this newer analog card I have.

Does anyone know:

1- If the version of Zaptel and the version of Asterisk MUST be the same?
2- If I need the latest version of Zaptel to run the TDM400 card?

Thanks!

- Original Message - 
From: Michael Graves [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, January 05, 2005 7:43 PM
Subject: Re: [Asterisk-Users] modprobe: Can't locate module wctdm


 On Wed, 5 Jan 2005 19:09:00 -0600, Me wrote:

 After getting zaptel from the CVS server, compiling and installing it I
 type:
 
 modprobe zaptel
 
 and all is well. Then I type:
 
 modprobe wctdm
 
 and I get this:
 
 modprobe: Can't locate module wctdm
 
 Any idea why?
 
 I did this yesterday but with the CVS head of Asterisk and I got by this
 part without a problem. I reinstalled it all today because I wanted the
 stable release on our server since we use it daily for calls in our
office.
 
 I am stuck.. Any help?
 
 I am running Fedora 1 FYI.. Also, I compiled and installed Asterisk this
 time before Zaptel, is that bad?

 I had this happen myself just recently. Moreover, I've never been able
 to get modprobe wctdm to work. I'm running v1.0.3 on FC1.

 I end up doing

 modprobe zaptel
 modprobe wcfxs
 ztcfg -vv

 which makes it all work.

 However, on Monday I restarted my server and modprobe could not find
 zaptel at all. I ended up doing a cvs update of zaptel and it finally
 started ok.

 Michael

 --
 Michael Graves   [EMAIL PROTECTED]
 Sr. Product Specialist  www.pixelpower.com
 Pixel Power Inc. [EMAIL PROTECTED]

 o713-861-4005
 o800-905-6412
 c713-201-1262



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Re: [Asterisk-Users] modprobe: Can't locate module wctdm

[Me]

Also, I wonder if there is some sort of issue with the fact that I compiled
and installed Asterisk before Zaptel?

??
- Original Message - 
From: Michael Graves [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, January 05, 2005 7:43 PM
Subject: Re: [Asterisk-Users] modprobe: Can't locate module wctdm


 On Wed, 5 Jan 2005 19:09:00 -0600, Me wrote:

 After getting zaptel from the CVS server, compiling and installing it I
 type:
 
 modprobe zaptel
 
 and all is well. Then I type:
 
 modprobe wctdm
 
 and I get this:
 
 modprobe: Can't locate module wctdm
 
 Any idea why?
 
 I did this yesterday but with the CVS head of Asterisk and I got by this
 part without a problem. I reinstalled it all today because I wanted the
 stable release on our server since we use it daily for calls in our
office.
 
 I am stuck.. Any help?
 
 I am running Fedora 1 FYI.. Also, I compiled and installed Asterisk this
 time before Zaptel, is that bad?

 I had this happen myself just recently. Moreover, I've never been able
 to get modprobe wctdm to work. I'm running v1.0.3 on FC1.

 I end up doing

 modprobe zaptel
 modprobe wcfxs
 ztcfg -vv

 which makes it all work.

 However, on Monday I restarted my server and modprobe could not find
 zaptel at all. I ended up doing a cvs update of zaptel and it finally
 started ok.

 Michael

 --
 Michael Graves   [EMAIL PROTECTED]
 Sr. Product Specialist  www.pixelpower.com
 Pixel Power Inc. [EMAIL PROTECTED]

 o713-861-4005
 o800-905-6412
 c713-201-1262



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[Asterisk-Users] Ouch... Error while writing audio data

[Me]

After installing the stable version of * and the Zaptel drivers with a
TDM400 card using 1 FXO module on port 4, I start Asterisk and get this
rolling up my screen thousands of times:

Ouch... Error while writing audio data
Ouch... Error while writing audio data
Ouch... Error while writing audio data
Ouch... Error while writing audio data
Ouch... Error while writing audio data
Ouch... Error while writing audio data

Can't break it with control C or anything, I have to kill the box and
restart..

Help!


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Re: [Asterisk-Users] Ouch... Error while writing audio data

[Me]

Firstly, if you are having problems with asterisk, don't start it with 
safe_asterisk.  Start it using asterisk -vvvgc
I didn't I started it with /usr/sbin/asterisk -cvvv

Secondly, you're probably going to need to kill the version that you may 
have set up to start automatically.
I do not have anything set to start automatically yet.
I fixed the issue, I think the problem was in my zapata.conf where I was 
referencing channel one on a TDM card that only had a module on channel 4.

Thanks!
--
Start Your Own ISP!
http://www.YourOwnISP.com
- Original Message - 
From: Matt Riddell [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Wednesday, January 05, 2005 9:11 PM
Subject: Re: [Asterisk-Users] Ouch... Error while writing audio data


Me wrote:
After installing the stable version of * and the Zaptel drivers with a
TDM400 card using 1 FXO module on port 4, I start Asterisk and get this
rolling up my screen thousands of times:
Ouch... Error while writing audio data
Ouch... Error while writing audio data
Ouch... Error while writing audio data
Ouch... Error while writing audio data
Ouch... Error while writing audio data
Ouch... Error while writing audio data
Can't break it with control C or anything, I have to kill the box and
restart..
Firstly, if you are having problems with asterisk, don't start it with 
safe_asterisk.  Start it using asterisk -vvvgc

This will mean that you get more detailed information and it will not 
restart.

Secondly, you're probably going to need to kill the version that you may 
have set up to start automatically.  Just find the process in ps and then 
kill it (kill 1234 - where 1234 is the process id).

--
Cheers,
Matt Riddell
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Re: [Asterisk-Users] modprobe: Can't locate module wctdm

[Me]

I had a little time and I wanted to see what would happen if I wiped the 
drive and started over but in a different order and with some different 
versions.

I did the following:
1- Installed FC1
2- yum update
3- Downloaded the latest Zaptel via CVS, compiled and installed
4- Configured zaptel.conf and zapata.conf
5- modprobe zapata
6- modprobe wctdm
7- /sbin/ztcfg -vv
8- Downloaded stable 1-0 version of asterisk via CVS, compiled and installed
9- Fixed the music on hold problem with Fedora and mpg123 by wget 
http://www.mpg123.de/mpg123/precompiled/mpg123-0.59q-1.i386.rpm; and then 
rpm -ivh mpg123-0.59q-1.i386.rpm
10- Ran cat /proc/interrupts to make sure my card was not sharing an 
interrupt with with any other hardware
11- Pulled my config files in from my other * box
12- Started * with /usr/sbin/asterisk -cvvv

and VIOLA!
--
Start Your Own ISP!
http://www.YourOwnISP.com
- Original Message - 
From: Michael Graves [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Wednesday, January 05, 2005 8:43 PM
Subject: Re: [Asterisk-Users] modprobe: Can't locate module wctdm


Yes, I believe that this is a problem. Everything I've read says you
compile and install zaptel first...then asterisk. On Monday I rebooted
my server again, the just did a CVS update of zaptel. That was all the
was required.
Michael
On Wed, 5 Jan 2005 20:10:27 -0600, Me wrote:
Also, I wonder if there is some sort of issue with the fact that I 
compiled
and installed Asterisk before Zaptel?

??
- Original Message - 
From: Michael Graves [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, January 05, 2005 7:43 PM
Subject: Re: [Asterisk-Users] modprobe: Can't locate module wctdm


On Wed, 5 Jan 2005 19:09:00 -0600, Me wrote:
After getting zaptel from the CVS server, compiling and installing it I
type:

modprobe zaptel

and all is well. Then I type:

modprobe wctdm

and I get this:

modprobe: Can't locate module wctdm

Any idea why?

I did this yesterday but with the CVS head of Asterisk and I got by 
this
part without a problem. I reinstalled it all today because I wanted the
stable release on our server since we use it daily for calls in our
office.

I am stuck.. Any help?

I am running Fedora 1 FYI.. Also, I compiled and installed Asterisk 
this
time before Zaptel, is that bad?

I had this happen myself just recently. Moreover, I've never been able
to get modprobe wctdm to work. I'm running v1.0.3 on FC1.
I end up doing
modprobe zaptel
modprobe wcfxs
ztcfg -vv
which makes it all work.
However, on Monday I restarted my server and modprobe could not find
zaptel at all. I ended up doing a cvs update of zaptel and it finally
started ok.
Michael
--
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]
o713-861-4005
o800-905-6412
c713-201-1262

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--
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]
o713-861-4005
o800-905-6412
c713-201-1262

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Re: [Asterisk-Users] modprobe: Can't locate module wctdm

[Me]

Yes, try moving the card to different slots until you see wctdm as the 
only item listed next to an IRQ number. If that doesn't work there are other 
ways to skin that beast..

Another thing I highly recommend is to turn off all devices that are not 
needed in your bios. For example, on board sound, USB ports, Serial Ports, 
LPT port and so on. Doing this will free up IRQ's and you will have less of 
a chance that the IRQ will be shared with something else.

Todd
--
Start Your Own ISP!
http://www.YourOwnISP.com
- Original Message - 
From: Kristian Kielhofner [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Thursday, January 06, 2005 12:26 AM
Subject: Re: [Asterisk-Users] modprobe: Can't locate module wctdm


Ronald Wiplinger wrote:
Me wrote:
10- Ran cat /proc/interrupts to make sure my card was not sharing an 
interrupt with with any other hardware

Can you interprete it for my situation ?
 CPU0   CPU10: 1469462555 1466917080IO-APIC-edge 
timer
 1: 153130 186604IO-APIC-edge  keyboard
 2:  0  0  XT-PIC  cascade
 8:  2  0IO-APIC-edge  rtc
 9:  0  0   IO-APIC-level  acpi
12:  35521  0IO-APIC-edge  PS/2 Mouse
14:98386327431613IO-APIC-edge  ide0
16: 14  0   IO-APIC-level  eth1, usb-uhci, usb-uhci
17: 16  0   IO-APIC-level  eth2, ohci1394
18: 1714185828 1225735285   IO-APIC-level  eth3, usb-uhci, wctdm
19:   22648117   12403171   IO-APIC-level  eth0, usb-uhci
23:  0  0   IO-APIC-level  ehci_hcd
NMI:  0  0 LOC: 2936506117 2936506698 ERR:  0
MIS:  0

wctdm is sharing IRQ 18 with eth3, and your USB controller.  That's a bad 
thing!

--
Kristian Kielhofner
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Re: [Asterisk-Users] Inbound Calls

[Me]

We will need more info on your setup.
When people call into your Asterisk system what device will they be calling 
in on?

Will they call a number provided by a termination/origination provider which 
is then fed into your Asterisk server using IAX or SIP?

Will they call a TDM card attached to your Asterisk system that has a PSTN 
(copper) phone line plugged into it?

Basically if the call comes in from say an IAX provider then the call starts 
in the iax.conf file then you push it to your dial plan in extensions.conf. 
If the call comes in through a card installed in your system then the call 
starts in the zapata.conf and is pushed into your extensions.conf file to 
the context you specify. Once the call is in your extensions.conf file you 
tell it what to do next. For example, answer it :) then decide if you will 
forward the call to another phone number or extension or maybe send it 
directly to voicemail etc.

--
Start Your Own ISP!
http://www.YourOwnISP.com
- Original Message - 
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, January 06, 2005 12:56 AM
Subject: [Asterisk-Users] Inbound Calls


Hey peoples,
I am in the midst of getting my server running and have gotten everything 
to work but the ability to take inbound calls. Any ideas?

Thanks!
~Dan
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Re: [Asterisk-Users] Sending call to analog then to Vmailaftertimeout?

[Me]

Thanks for the example! I was using something similar to this that I found
in the Wiki but the problem I ran into was the Record() part. Each time *
got to the record part I got some error saying, can't remember what it was,
I will dig it up and post it in a reply.

Start Your Own Internet Service!
http://www.YourOwnISP.com

- Original Message - 
From: C F [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, December 29, 2004 9:41 AM
Subject: Re: [Asterisk-Users] Sending call to analog then to
Vmailaftertimeout?


 [macro-stdcs]
 ;
 ;; Call a device with cs;;
 ;; Takes 2 arguments  ;;
 ;; arg1 exten   ;;
 ;; arg2 device   ;;
 ;; tnen goes to vm;;
 ;
 ;screen-record: Please record your name press pound when finished.
 ;screen-from: You have a call from
 ;screen-accept: Press 1 to accept 2 to reject, and 3 to transfer.
 exten = s,1,Wait(0.2)
 exten = s,2,Playback(vm-rec-name)
 exten = s,3,SetVar(SCREEN_FILE=/tmp/${CALLERIDNUM}-${EPOCH})
 exten = s,4,Record(${SCREEN_FILE}.gsm|2|4)
 exten = s,5,Playback(pls-wait-connect-call)
 exten = s,6,Dial(${ARG2},30,mtM(screen^${SCREEN_FILE}))
 exten = s,7,Goto(17);VM
 'I always leaeve room for more in case the dial plan changes
 exten = s,17,Voicemail(u${ARG1})
 exten = s,18,Playback(goodbye)
 exten = s,19,Hangup
 exten = s,107,Goto(17)

 exten = h,1,System(/bin/rm ${ARG1}.gsm)

  [macro-screen]
 ;this is called in the Dial statement using M
 ;ARG1 recorded name to play back
 ;TODO: add a response timeout, after which the message is repeated
 (needed for outgoing zap fxo channels) and absolute timeout, after
 which VM is used
 exten = s,1,noop(${ARG1})
 exten = s,2,Playback(custom/screen-from) ;you have an incoming call from:
 exten = s,3,Playback(${ARG1})
 ;press 1 to accept 2 to reject 3 to transfer
 exten = s,4,Read(ACCEPT|custom/screnn-accept|1)
 exten = s,5,Gotoif($[${ACCEPT} = 1] ?50) ;connect
 exten = s,6,Gotoif($[${ACCEPT} = 2] ?30) ;reject to vm
 exten = s,7,Gotoif($[${ACCEPT} = 3] ?40) ;TRANSFER
 exten = s,8,Gotoif($[${ACCEPT} = 4] ?30:30) ;any thing else vm

 exten = s,30,SetVar(MACRO_RESULT=CONTINUE)
 exten = s,31,Goto(50)

 exten = s,40,Read(TEXTEN|custom/screen-exten|3)
 ;ask for extension then set macro to goto that and continue
 exten = s,41,Gotoif($[${LEN(${TEXTEN})} = 3]?42:45)
 exten = s,42,SetVar(MACRO_RESULT=GOTO:internaldial^${TEXTEN}^1)
 exten = s,43,Goto(50)
 exten = s,45,Gotoif($[${TEXTEN} = 0] ?46:46)
 ;the logic is here to allow transfer to operator, i just didn't imlepent
it yet
 exten = s,46,SetVar(MACRO_RESULT=CONTINUE)
 exten = s,47,Goto(50)

 exten = s,50,System(/bin/rm ${ARG1}.gsm)

 exten = h,1,System(/bin/rm ${ARG1}.gsm)




 On Wed, 29 Dec 2004 00:35:34 -0600, Me [EMAIL PROTECTED] wrote:
  Nevermind, it looks like Asterisk cmd Read is my lucky command :)
 
  Thanks!
 
  Start Your Own Internet Service!
  http://www.YourOwnISP.com
 
  - Original Message -
  From: Me [EMAIL PROTECTED]
  To: C F [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial
  Discussion asterisk-users@lists.digium.com
  Sent: Wednesday, December 29, 2004 12:19 AM
  Subject: Re: [Asterisk-Users] Sending call to analog then to
  Vmailaftertimeout?
 
   I was trying this logic before, I got as far as going into the Macro,
   playing a message and then.. Well... I got lost, I am not sure how to
go
   about require them to press a button. Normally I can make someone
press an
   extension but from what I read about Macros in * you have to stay
within
  the
   s extension.
  
   Any idea where I can find an example of this sort of thing?
  
   Thanks!
  
   Start Your Own Internet Service!
   http://www.YourOwnISP.com
   - Original Message -
   From: C F [EMAIL PROTECTED]
   To: Asterisk Users Mailing List - Non-Commercial Discussion
   asterisk-users@lists.digium.com
   Sent: Tuesday, December 28, 2004 11:34 PM
   Subject: Re: [Asterisk-Users] Sending call to analog then to
   Vmailaftertimeout?
  
  
-- Forwarded message --
From: C F [EMAIL PROTECTED]
Date: Wed, 29 Dec 2004 00:34:28 -0500
Subject: Re: [Asterisk-Users] Sending call to analog then to Vmail
   aftertimeout?
To: Me [EMAIL PROTECTED]
   
   
try the M option which will do a macro and will not connect the
caller
unless s/he presses some button. and if no button is pressed then it
goes to VM. now remember to replay the message (to press the button)
a
few times b4 going to VM otherwise they will never hear it, since *
considers it answered .
http://www.voip-info.org/wiki-Asterisk+cmd+dial
   
   
On Tue, 28 Dec 2004 23:29:54 -0600, Me [EMAIL PROTECTED]
wrote:
 I was aware of the c option but it's a pain for people to have
to
   press
 the # sign plus they have to know they are suppose

Re: [Asterisk-Users] IP Phone recommendations?

[Me]

Why not use ATA adapters? This way you can use just about any phone you
want.


Start Your Own Internet Service!
http://www.YourOwnISP.com

- Original Message - 
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, December 29, 2004 10:28 AM
Subject: RE: [Asterisk-Users] IP Phone recommendations?


Okay,
  I'm feeling a little stupid here But I'm gonna ask anyway.

You mention support and firmware on the Ci$co phones.

I understand the support item.
I guess it makes sense that the phones have firmware.  Does it have to
be updated or changed or messed with that often?

If there is an article somewhere that covers this I'd love to read it.

It seems like most of the VOIP marketing-speak is aimed at companies
with mega$$$ who want to spend $500/head on it.  We're a tad smaller and
we have $ to spend not $$ or $$$ or .  :)  Worse yet, we need $ to
go find and bring back it's friends.  :)  Anyhow, I haven't seen
anything that really tackles moving from a CISC Nortel Meridian KSU to a
IP based system.  I'm guessing that this is Nortel's absolute worst
nightmare.  It seems like they trickle down the technology from the
large switches to the micro PBX systems.


Brian Greul
Texas Shirt Company
www.txshirts.com
713-802-0369 / 713-861-6261 (fax)

-Original Message-
From: Damon Estep [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 29, 2004 10:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] IP Phone recommendations?

Many use cisco IP phones, almost any model. Support and firmware access
has a fee.
SNOM 190 works well, free firmware, good community support.
Lots of reports of good luck with Polycom phones (IP500), but they wont
provide any support when used with * and you have to get your firmware
from the net, not from polycom, even if are willing to pay.

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
 Sent: Wednesday, December 29, 2004 8:51 AM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] IP Phone recommendations?

 Hey gang,
   I'm looking at escaping from a Nortel Meridian CISC system to
 Asterisk/Digium/SIP phones.  I'm currently in the testing and proof of

 concept phase.  I'm going to need a SIP phone and don't want to
 re-purchase and have orphans around.

 We currently run Nortel 7310 phones and they work great.
 I'm sort of overwhelmed by all of the different IP phones.  I was
hoping
 some folks would share what they have found.
 My primary goal is to replicate the 7310's features and to allow room
 for growth in the future with telephony applications.

 Our primary driver is configurability and features that we can get in
 Asterisk, that we can get without a lot of money from Nortel.

 Namely-
 Voicemail, telecommuting workers on the pbx, better call handling,
 better automation.
 I'd like to be able to integrate smart features like directory and
call
 handling to the handset, but I'll freely admit I'm just starting out.
 My initial goal is to just to get onto Asterisk and get it working.
 I'll worry about cool stuff later.

 Our integration and migration plan is as follows:  If anyone has some
 suggestions or pointers I'd love to hear them.

 1. Test and evaluate Asterisk with TDM400 with 1 FXO/FSO port each.
 2. Configure Asterisk to be the primary PBX and slave the Nortel
 Meridian system to it using a second TDM400.  This avoids immediate
 replacement of all handsets.  Will allow immediate access to features
 such as Voicemail.
 3. Overtime, upgrade desk phones to IP phones.  When all phones are
 replaced, decommission Nortel and sell on Ebay.  :)

 Cold turkey option is to spend the extra $ and buy the handsets
upfront
 and just ditch nortel without a transition period.

 We currently have 4 pbx lines and 1 dedicated fax/credit card line.
 We have 10 handsets.

 Thanks,

 Brian Greul
 Texas Shirt Company
 www.txshirts.com
 713-802-0369 / 713-861-6261 (fax)

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To 

[Asterisk-Users] Sending call to analog then to Vmail after timeout?

[Me]

I have one analog line hooked in my Asterisk box 
using an x100p (I think that's the model number).

When I do this in my extensions.conf:

exten = 
1200,1,playback(pls-wait-connect-call)exten = 
1200,2,Dial(Zap/1/551212,20,rTt)exten = 1200,3,VoiceMail([EMAIL PROTECTED])exten = 
1200,4,Goto,t|1

The phone rings beyond the 20 second timeout and 
never really goes to the * voicemail. I can't seem to get it to timeout 
regardless of how many seconds I set it to.

I assume this has something to do with the fact 
that * considers the call answered as soon as the zap channel picks it up, 
right?

Anyhow, is there a way to make the above config 
work and go to the * voicemail after 20 seconds if the called party does not 
answer after 20 seconds? Also, what happens if the called party's line is busy, 
have not run into this yet so I am curious.

Thanks!


--
Start Your Own Internet Service!
http://www.YourOwnISP.com


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[Asterisk-Users] Hardware opinions?

[Me]

Hello, I am trying to build up a pretty meaty 
Asterisk box after doing our initial testing and playing on a 1ghz 
system.

Right now I have decided on a prebuilt system which 
I normally don't do but thought it seemed like a good deal.

I have included the initial specs below, I will be 
adding another 1 GB of RAM for a total of 2 GB. 

My first question is regarding the serial ATA 
drives... I will be using Fedora and considering FC1 seems to be the smartest of 
the builds when it comes to the digium hardware, will I have to scrap the SATA 
drives because FC1 doesn't support them or do I have bad 
information?

If I need to scrap the SATA drives and let's say I 
didn't care about the Raid functionality, would you folks think that IDE drives 
would be fine or would the speed of SCSI really make much of a difference when 
it comes to Asterisk? If speed of drives does matter, can someone tell me why 
Asterisk might need fast drives vs. say 7200 IDE drives?

Next and last question is, how many simultaneous 
calls do you folks figure I can run on this in the following two 
scenarios:

1- All clients would be using SIP devices like 
SPA-2000's and all calls would originate/terminate using an IAX termination 
partner.
2- All clients would be using IAX like Asterisk or 
an IAXy and all calls would originate/terminate using an IAX termination 
partner.


Here are the specs:


  Dual (2) Intel Xeon 2.66GHz CPUs w/ HT Technology
  533MHz Front Side Bus 
  1GB PC2100 DDR ECC Registered Memory
  Two (2) 120GB SATA 7200RPM Hard Drives with 8MB Cache 
  52X CD-RW Drive w/Burning Software 
  3.5" 1.44MB Floppy Drive 
  ATI Rage XL with 8MB Onboard
  Onboard RAID controller 
  (2) Intel Ethernet Controllers (1x1000BT Gigabit  1x10/100) 
  2U Rackmount Chassis w/ 500-Watt Power Supply 
Thanks!

--
Start Your Own ISP!
http://www.YourOwnISP.com

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Re: [Asterisk-Users] Sending call to analog then to Vmail after timeout?

[Me]

Sorry about the HTML emails, on my laptop and forgot to change the sending
format from the default.


- Original Message - 
From: Me
To: asterisk-users@lists.digium.com
Sent: Tuesday, December 28, 2004 2:01 PM
Subject: [Asterisk-Users] Sending call to analog then to Vmail after
timeout?


I have one analog line hooked in my Asterisk box using an x100p (I think
that's the model number).

When I do this in my extensions.conf:

exten = 1200,1,playback(pls-wait-connect-call)
exten = 1200,2,Dial(Zap/1/551212,20,rTt)
exten = 1200,3,VoiceMail([EMAIL PROTECTED])
exten = 1200,4,Goto,t|1

The phone rings beyond the 20 second timeout and never really goes to the *
voicemail. I can't seem to get it to timeout regardless of how many seconds
I set it to.

I assume this has something to do with the fact that * considers the call
answered as soon as the zap channel picks it up, right?

Anyhow, is there a way to make the above config work and go to the *
voicemail after 20 seconds if the called party does not answer after 20
seconds? Also, what happens if the called party's line is busy, have not run
into this yet so I am curious.

Thanks!


--
Start Your Own Internet Service!
http://www.YourOwnISP.com






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Re: [Asterisk-Users] Hardware opinions?

[Me]

Dorn,

 Can you give me some details on this linux md driver you mentioned?

Also, you say not to scrap the SATA drives, is this because you think I can
use them with FC1 or because you think I should try Debian? I really don't
want to venture away from Fedora at the moment for a few reasons.

Thanks!


- Original Message - 
From: Dorn Hetzel [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, December 28, 2004 4:07 PM
Subject: Re: [Asterisk-Users] Hardware opinions?


 On Tue, Dec 28, 2004 at 02:17:47PM -0600, Me wrote:
  Hello, I am trying to build up a pretty meaty Asterisk box after doing
our initial testing and playing on a 1ghz system.
 
  Right now I have decided on a prebuilt system which I normally don't do
but thought it seemed like a good deal.
 
  I have included the initial specs below, I will be adding another 1 GB
of RAM for a total of 2 GB.
 
  My first question is regarding the serial ATA drives... I will be using
Fedora and considering FC1 seems to be the smartest of the builds when it
comes to the digium hardware, will I have to scrap the SATA drives because
FC1 doesn't support them or do I have bad information?
 
  If I need to scrap the SATA drives and let's say I didn't care about the
Raid functionality, would you folks think that IDE drives would be fine or
would the speed of SCSI really make much of a difference when it comes to
Asterisk? If speed of drives does matter, can someone tell me why Asterisk
might need fast drives vs. say 7200 IDE drives?
 
  Next and last question is, how many simultaneous calls do you folks
figure I can run on this in the following two scenarios:
 
  1- All clients would be using SIP devices like SPA-2000's and all calls
would originate/terminate using an IAX termination partner.
  2- All clients would be using IAX like Asterisk or an IAXy and all calls
would originate/terminate using an IAX termination partner.
 
 
  Here are the specs:
 
a.. Dual (2) Intel Xeon 2.66GHz CPUs w/ HT Technology
b.. 533MHz Front Side Bus
c.. 1GB PC2100 DDR ECC Registered Memory
d.. Two (2) 120GB SATA 7200RPM Hard Drives with 8MB Cache
e.. 52X CD-RW Drive w/Burning Software
f.. 3.5 1.44MB Floppy Drive
g.. ATI Rage XL with 8MB Onboard
h.. Onboard RAID controller
i.. (2) Intel Ethernet Controllers (1x1000BT Gigabit  1x10/100)
j.. 2U Rackmount Chassis w/ 500-Watt Power Supply
 
  Thanks!
 

 You don't need to scrap the SATA drives, they are very nice.
 You might want to give Debian a try.
 Don't use the RAID mode on the motherboard as it's likely
 fake raid, instead use linux md driver for software raid,
 it's smoking fast for SATA drives and on my 3.2ghz box
 barely scratches the CPU resyncing.

 -Dorn

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Re: [Asterisk-Users] Hardware opinions?

[Me]

So are you saying that if I have one of the supported controllers, FC1 will
work out of the box with the SATA drives attached?

Also, what about FC2 or 3?

Is there a patch for any of these three builds that will support the SATA
controllers?

Thanks!

--
Start Your Own ISP!
http://www.YourOwnISP.com

- Original Message - 
From: Sean Cook [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, December 27, 2004 7:31 PM
Subject: Re: [Asterisk-Users] Hardware opinions?


 On Tue, 2004-12-28 at 16:12 -0600, Me wrote:
  Dorn,
 
   Can you give me some details on this linux md driver you mentioned?
 
  Also, you say not to scrap the SATA drives, is this because you think I
can
  use them with FC1 or because you think I should try Debian? I really
don't
  want to venture away from Fedora at the moment for a few reasons.
 

 FC1 does support SATA drives, however it is dependant upon the sata
 controller.  The intel sata driver is supported, adaptec, 3ware 7xxx and
 8xxx controllers are also supported.

  Thanks!
 
 
  - Original Message - 
  From: Dorn Hetzel [EMAIL PROTECTED]
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  asterisk-users@lists.digium.com
  Sent: Tuesday, December 28, 2004 4:07 PM
  Subject: Re: [Asterisk-Users] Hardware opinions?
 
 
   On Tue, Dec 28, 2004 at 02:17:47PM -0600, Me wrote:
Hello, I am trying to build up a pretty meaty Asterisk box after
doing
  our initial testing and playing on a 1ghz system.
   
Right now I have decided on a prebuilt system which I normally don't
do
  but thought it seemed like a good deal.
   
I have included the initial specs below, I will be adding another 1
GB
  of RAM for a total of 2 GB.
   
My first question is regarding the serial ATA drives... I will be
using
  Fedora and considering FC1 seems to be the smartest of the builds when
it
  comes to the digium hardware, will I have to scrap the SATA drives
because
  FC1 doesn't support them or do I have bad information?
   
If I need to scrap the SATA drives and let's say I didn't care about
the
  Raid functionality, would you folks think that IDE drives would be fine
or
  would the speed of SCSI really make much of a difference when it comes
to
  Asterisk? If speed of drives does matter, can someone tell me why
Asterisk
  might need fast drives vs. say 7200 IDE drives?
   
Next and last question is, how many simultaneous calls do you folks
  figure I can run on this in the following two scenarios:
   
1- All clients would be using SIP devices like SPA-2000's and all
calls
  would originate/terminate using an IAX termination partner.
2- All clients would be using IAX like Asterisk or an IAXy and all
calls
  would originate/terminate using an IAX termination partner.
   
   
Here are the specs:
   
  a.. Dual (2) Intel Xeon 2.66GHz CPUs w/ HT Technology
  b.. 533MHz Front Side Bus
  c.. 1GB PC2100 DDR ECC Registered Memory
  d.. Two (2) 120GB SATA 7200RPM Hard Drives with 8MB Cache
  e.. 52X CD-RW Drive w/Burning Software
  f.. 3.5 1.44MB Floppy Drive
  g.. ATI Rage XL with 8MB Onboard
  h.. Onboard RAID controller
  i.. (2) Intel Ethernet Controllers (1x1000BT Gigabit  1x10/100)
  j.. 2U Rackmount Chassis w/ 500-Watt Power Supply
   
Thanks!
   
  
   You don't need to scrap the SATA drives, they are very nice.
   You might want to give Debian a try.
   Don't use the RAID mode on the motherboard as it's likely
   fake raid, instead use linux md driver for software raid,
   it's smoking fast for SATA drives and on my 3.2ghz box
   barely scratches the CPU resyncing.
  
   -Dorn
  
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Re: [Asterisk-Users] Sending call to analog then to Vmail aftertimeout?

[Me]

I was aware of the c option but it's a pain for people to have to press
the # sign plus they have to know they are suppose to do that. In addition,
I tried to use the A option to play a sound to them when they answer
reminding them to press pound at the end of the message but the sound
doesn't play until they press pound :)

So.. It appears I am still stuck with * considering the call answered when
the Zap channels grabs it and connects the other leg of the call. Hopefully
there is some other way to make this happen.

Thanks for the feedback though.

Start Your Own Internet Service!
http://www.YourOwnISP.com

- Original Message - 
From: C F [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, December 28, 2004 6:26 PM
Subject: Re: [Asterisk-Users] Sending call to analog then to Vmail
aftertimeout?


 Follow these:
 http://www.voip-info.org/wiki-Asterisk+zap+channels
 looks like this would work:
  exten = 1200,1,playback(pls-wait-connect-call)
  exten = 1200,2,Dial(Zap/1c/551212,20,rTt) ;note the c after the
 channel number
  exten = 1200,3,VoiceMail([EMAIL PROTECTED])
  exten = 1200,4,Goto,t|1


 On Tue, 28 Dec 2004 14:20:02 -0600, Me [EMAIL PROTECTED] wrote:
  Sorry about the HTML emails, on my laptop and forgot to change the
sending
  format from the default.
 
 
  - Original Message -
  From: Me
  To: asterisk-users@lists.digium.com
  Sent: Tuesday, December 28, 2004 2:01 PM
  Subject: [Asterisk-Users] Sending call to analog then to Vmail after
  timeout?
 
  I have one analog line hooked in my Asterisk box using an x100p (I think
  that's the model number).
 
  When I do this in my extensions.conf:
 
  exten = 1200,1,playback(pls-wait-connect-call)
  exten = 1200,2,Dial(Zap/1/551212,20,rTt)
  exten = 1200,3,VoiceMail([EMAIL PROTECTED])
  exten = 1200,4,Goto,t|1
 
  The phone rings beyond the 20 second timeout and never really goes to
the *
  voicemail. I can't seem to get it to timeout regardless of how many
seconds
  I set it to.
 
  I assume this has something to do with the fact that * considers the
call
  answered as soon as the zap channel picks it up, right?
 
  Anyhow, is there a way to make the above config work and go to the *
  voicemail after 20 seconds if the called party does not answer after 20
  seconds? Also, what happens if the called party's line is busy, have not
run
  into this yet so I am curious.
 
  Thanks!
 
  --
  Start Your Own Internet Service!
  http://www.YourOwnISP.com
 
 
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 http://lists.digium.com/mailman/listinfo/asterisk-users
 
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Re: [Asterisk-Users] Hardware opinions?

[Me]

 What sort of chipset is your SATA controller interface?  Intel
 ICH6R?

Adaptec ICH5R SATA controller according to SuperMicro which makes the Mobo.
The board has an Intel® E7501 main chipset.


Start Your Own Internet Service!
http://www.YourOwnISP.com
- Original Message - 
From: Dorn Hetzel [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, December 28, 2004 7:00 PM
Subject: Re: [Asterisk-Users] Hardware opinions?


 On Tue, Dec 28, 2004 at 04:12:26PM -0600, Me wrote:
  Dorn,
 
   Can you give me some details on this linux md driver you mentioned?
 
  Also, you say not to scrap the SATA drives, is this because you think I
can
  use them with FC1 or because you think I should try Debian? I really
don't
  want to venture away from Fedora at the moment for a few reasons.
 
 It's likely you can make the SATA drives work with Fedora, I just
 can't say from personal experience.  The md driver is a software
 raid implementation.  check out mdadm (the setup command) man pages
 for more info.  I'm using three different flavors on the last
 server I built, raid0 for speed /tmp type space, raid5 for speed
 and security, and a triple-copy raid1 for really important stuff.

 What sort of chipset is your SATA controller interface?  Intel
 ICH6R?

 -Dorn

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Re: [Asterisk-Users] Sending call to analog then to Vmailaftertimeout?

[Me]

I was trying this logic before, I got as far as going into the Macro,
playing a message and then.. Well... I got lost, I am not sure how to go
about require them to press a button. Normally I can make someone press an
extension but from what I read about Macros in * you have to stay within the
s extension.

Any idea where I can find an example of this sort of thing?

Thanks!

Start Your Own Internet Service!
http://www.YourOwnISP.com
- Original Message - 
From: C F [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, December 28, 2004 11:34 PM
Subject: Re: [Asterisk-Users] Sending call to analog then to
Vmailaftertimeout?


 -- Forwarded message --
 From: C F [EMAIL PROTECTED]
 Date: Wed, 29 Dec 2004 00:34:28 -0500
 Subject: Re: [Asterisk-Users] Sending call to analog then to Vmail
aftertimeout?
 To: Me [EMAIL PROTECTED]


 try the M option which will do a macro and will not connect the caller
 unless s/he presses some button. and if no button is pressed then it
 goes to VM. now remember to replay the message (to press the button) a
 few times b4 going to VM otherwise they will never hear it, since *
 considers it answered .
 http://www.voip-info.org/wiki-Asterisk+cmd+dial


 On Tue, 28 Dec 2004 23:29:54 -0600, Me [EMAIL PROTECTED] wrote:
  I was aware of the c option but it's a pain for people to have to
press
  the # sign plus they have to know they are suppose to do that. In
addition,
  I tried to use the A option to play a sound to them when they answer
  reminding them to press pound at the end of the message but the sound
  doesn't play until they press pound :)
 
  So.. It appears I am still stuck with * considering the call answered
when
  the Zap channels grabs it and connects the other leg of the call.
Hopefully
  there is some other way to make this happen.
 
  Thanks for the feedback though.
 
  Start Your Own Internet Service!
  http://www.YourOwnISP.com
 
  - Original Message -
  From: C F [EMAIL PROTECTED]
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  asterisk-users@lists.digium.com
  Sent: Tuesday, December 28, 2004 6:26 PM
  Subject: Re: [Asterisk-Users] Sending call to analog then to Vmail
  aftertimeout?
 
   Follow these:
   http://www.voip-info.org/wiki-Asterisk+zap+channels
   looks like this would work:
exten = 1200,1,playback(pls-wait-connect-call)
exten = 1200,2,Dial(Zap/1c/551212,20,rTt) ;note the c after the
   channel number
exten = 1200,3,VoiceMail([EMAIL PROTECTED])
exten = 1200,4,Goto,t|1
  
  
   On Tue, 28 Dec 2004 14:20:02 -0600, Me [EMAIL PROTECTED]
wrote:
Sorry about the HTML emails, on my laptop and forgot to change the
  sending
format from the default.
   
   
- Original Message -
From: Me
To: asterisk-users@lists.digium.com
Sent: Tuesday, December 28, 2004 2:01 PM
Subject: [Asterisk-Users] Sending call to analog then to Vmail after
timeout?
   
I have one analog line hooked in my Asterisk box using an x100p (I
think
that's the model number).
   
When I do this in my extensions.conf:
   
exten = 1200,1,playback(pls-wait-connect-call)
exten = 1200,2,Dial(Zap/1/551212,20,rTt)
exten = 1200,3,VoiceMail([EMAIL PROTECTED])
exten = 1200,4,Goto,t|1
   
The phone rings beyond the 20 second timeout and never really goes
to
  the *
voicemail. I can't seem to get it to timeout regardless of how many
  seconds
I set it to.
   
I assume this has something to do with the fact that * considers the
  call
answered as soon as the zap channel picks it up, right?
   
Anyhow, is there a way to make the above config work and go to the *
voicemail after 20 seconds if the called party does not answer after
20
seconds? Also, what happens if the called party's line is busy, have
not
  run
into this yet so I am curious.
   
Thanks!
   
--
Start Your Own Internet Service!
http://www.YourOwnISP.com
   
   
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Re: [Asterisk-Users] Sending call to analog then to Vmailaftertimeout?

[Me]

Nevermind, it looks like Asterisk cmd Read is my lucky command :)

Thanks!

Start Your Own Internet Service!
http://www.YourOwnISP.com

- Original Message - 
From: Me [EMAIL PROTECTED]
To: C F [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Sent: Wednesday, December 29, 2004 12:19 AM
Subject: Re: [Asterisk-Users] Sending call to analog then to
Vmailaftertimeout?


 I was trying this logic before, I got as far as going into the Macro,
 playing a message and then.. Well... I got lost, I am not sure how to go
 about require them to press a button. Normally I can make someone press an
 extension but from what I read about Macros in * you have to stay within
the
 s extension.

 Any idea where I can find an example of this sort of thing?

 Thanks!

 Start Your Own Internet Service!
 http://www.YourOwnISP.com
 - Original Message - 
 From: C F [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Tuesday, December 28, 2004 11:34 PM
 Subject: Re: [Asterisk-Users] Sending call to analog then to
 Vmailaftertimeout?


  -- Forwarded message --
  From: C F [EMAIL PROTECTED]
  Date: Wed, 29 Dec 2004 00:34:28 -0500
  Subject: Re: [Asterisk-Users] Sending call to analog then to Vmail
 aftertimeout?
  To: Me [EMAIL PROTECTED]
 
 
  try the M option which will do a macro and will not connect the caller
  unless s/he presses some button. and if no button is pressed then it
  goes to VM. now remember to replay the message (to press the button) a
  few times b4 going to VM otherwise they will never hear it, since *
  considers it answered .
  http://www.voip-info.org/wiki-Asterisk+cmd+dial
 
 
  On Tue, 28 Dec 2004 23:29:54 -0600, Me [EMAIL PROTECTED] wrote:
   I was aware of the c option but it's a pain for people to have to
 press
   the # sign plus they have to know they are suppose to do that. In
 addition,
   I tried to use the A option to play a sound to them when they answer
   reminding them to press pound at the end of the message but the sound
   doesn't play until they press pound :)
  
   So.. It appears I am still stuck with * considering the call answered
 when
   the Zap channels grabs it and connects the other leg of the call.
 Hopefully
   there is some other way to make this happen.
  
   Thanks for the feedback though.
  
   Start Your Own Internet Service!
   http://www.YourOwnISP.com
  
   - Original Message -
   From: C F [EMAIL PROTECTED]
   To: Asterisk Users Mailing List - Non-Commercial Discussion
   asterisk-users@lists.digium.com
   Sent: Tuesday, December 28, 2004 6:26 PM
   Subject: Re: [Asterisk-Users] Sending call to analog then to Vmail
   aftertimeout?
  
Follow these:
http://www.voip-info.org/wiki-Asterisk+zap+channels
looks like this would work:
 exten = 1200,1,playback(pls-wait-connect-call)
 exten = 1200,2,Dial(Zap/1c/551212,20,rTt) ;note the c after
the
channel number
 exten = 1200,3,VoiceMail([EMAIL PROTECTED])
 exten = 1200,4,Goto,t|1
   
   
On Tue, 28 Dec 2004 14:20:02 -0600, Me [EMAIL PROTECTED]
 wrote:
 Sorry about the HTML emails, on my laptop and forgot to change the
   sending
 format from the default.


 - Original Message -
 From: Me
 To: asterisk-users@lists.digium.com
 Sent: Tuesday, December 28, 2004 2:01 PM
 Subject: [Asterisk-Users] Sending call to analog then to Vmail
after
 timeout?

 I have one analog line hooked in my Asterisk box using an x100p (I
 think
 that's the model number).

 When I do this in my extensions.conf:

 exten = 1200,1,playback(pls-wait-connect-call)
 exten = 1200,2,Dial(Zap/1/551212,20,rTt)
 exten = 1200,3,VoiceMail([EMAIL PROTECTED])
 exten = 1200,4,Goto,t|1

 The phone rings beyond the 20 second timeout and never really goes
 to
   the *
 voicemail. I can't seem to get it to timeout regardless of how
many
   seconds
 I set it to.

 I assume this has something to do with the fact that * considers
the
   call
 answered as soon as the zap channel picks it up, right?

 Anyhow, is there a way to make the above config work and go to the
*
 voicemail after 20 seconds if the called party does not answer
after
 20
 seconds? Also, what happens if the called party's line is busy,
have
 not
   run
 into this yet so I am curious.

 Thanks!

 --
 Start Your Own Internet Service!
 http://www.YourOwnISP.com


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Re: [Asterisk-Users] Record() problem

[Me]

That is what I used :) except I got it from the another page in the Wiki I
think.. I just changed the sound file references to a sound file that
existed on my side.

After using this example I got the error when * gets to the record line of
extensions.conf:

WARNING[3293201]: app_record.c:117 record_exec: No extension found

Thanks!

--
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- Original Message - 
From: Brian West [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Friday, December 24, 2004 10:38 PM
Subject: RE: [Asterisk-Users] Record() problem


 http://bugs.digium.com/bug_view_page.php?bug_id=0002905

 Refer to my example on that bug note.

 bkw

  -Original Message-
  From: [EMAIL PROTECTED] [mailto:asterisk-users-
  [EMAIL PROTECTED] On Behalf Of Me
  Sent: Friday, December 24, 2004 11:06 AM
  To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-
  Commercial Discussion
  Subject: Re: [Asterisk-Users] Record() problem
 
  It was executed from the dial plan within extensions.conf and I did not
  hard
  code the IAX2/[EMAIL PROTECTED]/5 in there. I will paste the exact text
  below
  from my extensions.conf which I really should have done the first time
:)
  sorry..
 
  I didn't include the Macro but that's not where it's blowing up. Any
help
  would be appreciated.
 
  Happy Holidays to all!
 
  *From extensions.conf*
 
  ; 1100 - Test call whisper type thing
  ;exten = 1100,1,Wait(0.2)
  ;exten = 1100,2,Playback(say-name)
  ;exten = 1100,3,SetVar(SCREEN_FILE=/tmp/${CALLERIDNUM}-${EPOCH})
  ;exten = 1100,4,Record(${SCREEN_FILE}.gsm,6,25)
  ;exten = 1100,5,Dial(SIP/1100,60,gM(screen^${SCREEN_FILE}))
  ;exten = 1100,6,Voicemail([EMAIL PROTECTED])
 
  ***End
  --
  Start Your Own ISP!
  http://www.YourOwnISP.com
  - Original Message -
  From: Bill Seddon [EMAIL PROTECTED]
  To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
  asterisk-users@lists.digium.com
  Sent: Friday, December 24, 2004 3:00 AM
  Subject: RE: [Asterisk-Users] Record() problem
 
 
   You syntax for the command is incorrect.  See
   http://www.voip-info.org/wiki-Asterisk+cmd+record.
  
   Record is an application to be executed from within the dialplan.  So
  the
   channel it will record is implicit and cannot be explicitly stated as
  one
   of
   the parameters.
  
   If you want to originate and record a call automatically, you will
have
  to
   do this via AGI.
  
   Bill Seddon
  
   -Original Message-
   From: [EMAIL PROTECTED]
   [mailto:[EMAIL PROTECTED] On Behalf Of Me
   Sent: December 24, 2004 6:38 AM
   To: asterisk-users@lists.digium.com
   Subject: [Asterisk-Users] Record() problem
  
   Any idea why this:
   Record(IAX2/[EMAIL PROTECTED]/5, /tmp/whatever.gsm|6|25)
  
   Would result in this:
   WARNING[3293201]: app_record.c:117 record_exec: No extension found
  
   Thanks!
  
   --
   Start Your Own ISP!
   http://www.YourOwnISP.com
  
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Re: [Asterisk-Users] Record() problem

[Me]

It was executed from the dial plan within extensions.conf and I did not hard 
code the IAX2/[EMAIL PROTECTED]/5 in there. I will paste the exact text below 
from my extensions.conf which I really should have done the first time :) 
sorry..

I didn't include the Macro but that's not where it's blowing up. Any help 
would be appreciated.

Happy Holidays to all!
*From extensions.conf*
; 1100 - Test call whisper type thing
;exten = 1100,1,Wait(0.2)
;exten = 1100,2,Playback(say-name)
;exten = 1100,3,SetVar(SCREEN_FILE=/tmp/${CALLERIDNUM}-${EPOCH})
;exten = 1100,4,Record(${SCREEN_FILE}.gsm,6,25)
;exten = 1100,5,Dial(SIP/1100,60,gM(screen^${SCREEN_FILE}))
;exten = 1100,6,Voicemail([EMAIL PROTECTED])
***End
--
Start Your Own ISP!
http://www.YourOwnISP.com
- Original Message - 
From: Bill Seddon [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
asterisk-users@lists.digium.com
Sent: Friday, December 24, 2004 3:00 AM
Subject: RE: [Asterisk-Users] Record() problem


You syntax for the command is incorrect.  See
http://www.voip-info.org/wiki-Asterisk+cmd+record.
Record is an application to be executed from within the dialplan.  So the
channel it will record is implicit and cannot be explicitly stated as one 
of
the parameters.

If you want to originate and record a call automatically, you will have to
do this via AGI.
Bill Seddon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Me
Sent: December 24, 2004 6:38 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Record() problem
Any idea why this:
Record(IAX2/[EMAIL PROTECTED]/5, /tmp/whatever.gsm|6|25)
Would result in this:
WARNING[3293201]: app_record.c:117 record_exec: No extension found
Thanks!
--
Start Your Own ISP!
http://www.YourOwnISP.com
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[Asterisk-Users] Asked to transmit frame type 2, while native formats is 4???

[Me]

Anyone know what this error message means?
**
Dec 23 23:12:31 WARNING[3031057]: chan_sip.c:1874 sip_write: Asked to 
transmit frame type 2, while native formats is 4 (read/write = 4/4)
**

I see this in my CLI when I call into Asterisk and press * which should hang 
up the call since I have the h option in my dial string.

My CLI goes nuts, the call connects fine but I get hundreds of lines of 
these rolling up my screen until I hang up the call.

Any ideas?
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Re: [Asterisk-Users] What does t mean in a CDR entry?

[Me]

Can you give me an example of how a call would end up in the timeout ext?
--
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- Original Message - 
From: Seth Remington [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Tuesday, December 21, 2004 7:33 AM
Subject: Re: [Asterisk-Users] What does t mean in a CDR entry?


On Mon, 2004-12-20 at 13:45, Me wrote:
What does t mean in a CDR entry?
The 't' probably means that the call ended up in the timeout extension.
-Seth
--
Seth Remington
SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (330)336-8559
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[Asterisk-Users] Record() problem

[Me]

Any idea why this:
Record(IAX2/[EMAIL PROTECTED]/5, /tmp/whatever.gsm|6|25) 

Would result in this:
WARNING[3293201]: app_record.c:117 record_exec: No extension found
Thanks!
--
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[Asterisk-Users] Call dies in 180 seconds exactly

[Me]

I have been seeing some strange problems with our in-house Asterisk system. 
Each of them have slightly different circumstances but I want to focus on 
one in particular.

Here is how the call flowed:
1- Came in via iax.cc from our DID with them to our Asterisk system
2- The caller dialed Zero for an operator
3- Operator answered and transferred the call to one of our internal 
extensions 106
4- Person at 106 answered and started the conversation, voice quality was 
well then after just a few minutes (3 I think) the call hung up!

Person called back, hit Zero again then was transferred to 106 again. Same 
thing call dies after 3 minutes or so.

I looked at the CDR entries for these two calls and two fields/columns of 
the entries got me pretty curious. For BOTH calls under the billsec and 
callduration fields the value was exactly 180 seconds.

This leads me to believe that somewhere in our system or with iax.cc (our 
origination for the DID) there is some sort of 180 timer set that killed the 
calls.

Any ideas? Any help would be appreciated!
--
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Re: [Asterisk-Users] Call dies in 180 seconds exactly

[Me]

Nope, I searched the extensions.conf, sip.conf and iax.conf for 180 and
found nothing.

--
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- Original Message - 
From: Wilson Pickett [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, December 22, 2004 10:20 AM
Subject: Re: [Asterisk-Users] Call dies in 180 seconds exactly


  thing call dies after 3 minutes or so.

 Any AbsoluteTimeout(180) lines in extensions.conf ?
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[Asterisk-Users] What does t mean in a CDR entry?

[Me]

What does t mean in a CDR entry? This is in place of where the number that 
was dialed normally goes. For one IAX termination provider it always has a t 
instead of the number dialed. Also, we always see the word hunguup in the 
same record entry. This is the provider we have set to our secondary not 
primary. Is it transfer of some sort? I don't think there was a transfer on 
the calls I am seeing this on, they were just outgoing IAX calls to a VOIP 
termination provider via IAX.

Also, what does it mean when I see hangup in the CRD entry? Does it mean 
that our caller (our extension) hungup or that the called party hung up or??

Sound like stupid questions, I know but none the less I would like to know 
the answers :)

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Re: [Asterisk-Users] Problem using SPA-2000 behind NAT

[Me]

I have lots of these working and at least two behind NATs..
Start by setting your SPA-2000's IP address as the DMZ address on your 
router. If everything works all of a sudden then that's a good start. I did 
this and it least it told me that all was well with the adapter itself.

What type of NAT router are you using? I have been successful so far with a 
Linksys, Dlink and Airlink Plus and the SPA-2000. I did have trouble with 
the phone not ringing after a few minutes of sitting after power up, I 
solved this eventually by enabling NAT Keep Alive in the Sipura settings.

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- Original Message - 
From: Patrick Conroy [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Monday, December 20, 2004 8:03 PM
Subject: Re: [Asterisk-Users] Problem using SPA-2000 behind NAT


 Enable STUN on the SPA-2000 and use (for example) stun.fwdnet.net:3478
 as the STUN server. I have a SPA-3000 working perfectly like that.
I just tried this and unfortunately it didn't help.
I did a side-by-side comparison of an X-Lite client and the SPA both
behind the same NAT.  The X-PRO works perfectly in both directions.
The SPA works only if it initiates the calls, otherwise no audio in
either direction.  In an example of the same SIP message sent to both
clients, it looks like message to the SPA has the private IP in the
Via header and the public IP in the To header, while they are
swapped (which I assume is correct) in the message to the X-PRO.  Any
ideas what would cause this?
James,  it might be helpful if you could walk me through what you did
to set up your SPA-2000.  Any help I can get to get this thing
working, would be very much appreciated.
Thanks,
Patrick
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[Asterisk-Users] Per extension/user CDR?

[Me]

It seems that all my CDR is dumping into the Master.csv file. There is a way 
to create per user/extension CDR but I have looked endlessly in the Wiki, 
docs, README.CDR, mailing list archives etc.. I can't seem to find a way to 
do this..

Any help would be appreciated.
Thanks!
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[Asterisk-Users] Call confirmation on NON Zap channels

[Me]

I would like to setup call confirmation so that the called party has to 
press a key to accept the call. There seems to be an Asterisk feature to do 
this with Zap channels where you place a c in the dial string. I want to 
do the same thing without re-inventing the wheel with IAX and SIP channels.

Right now the best I can do is play a sound to the called party, I have also 
figured out how to run a Macro when the called party answers but not sure 
what to dump into the Macro to make the conformation work.

Any help would be appreciated!
Thanks..
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Re: [Asterisk-Users] VOIP Phone Suggestions

[Me]

I have one but never was able to get it to ring 
with or without a NAT in front of it. Calls out worked fine.

There seemed to be only one person supporting this 
product at Uniden, he was very nice but after 4 or 5 calls I just gave up. The 
phone now collects dust on one of the desk in the office.

Also, I was told several times that the phone will 
not work at all behind a NAT. I tried it at the office where there is no NAT in 
between the phone and the * box but still could not get it to ring.

Todd

--Start Your Own ISP!http://www.YourOwnISP.com

  - Original Message - 
  From: 
  Kevin 
  Curtis 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Wednesday, December 15, 2004 10:53 
  PM
  Subject: Re: [Asterisk-Users] VOIP Phone 
  Suggestions
  
  I would recommend Uniden UIP200 phones. Great 
  sound quality with inbuilt phone book, call logs etc works great with 
  asterisk. I recently purchased from www.qualvoip.com (they also provided me 
  sample configuration files for asterisk).
  
  Kevin
  
- Original Message - 
From: 
Shawn 
Dillon 
To: [EMAIL PROTECTED] 

Sent: Wednesday, December 15, 2004 8:39 
PM
Subject: [Asterisk-Users] VOIP Phone 
Suggestions

We are in the final stage of a 
rollout of Asterisk in our company. We had some Polycom IP 600 , a Snom 220 
, a Grandstream 102 and recently a Sayson 480i phone. I am interested in 
anyones opinions in the phone they suggest to implement. I must admit I am a 
little partial to the Sayson 480i , but if there are convincing arguments 
with regards to other models I would like to hear them.

If anyone has had more experience with the 
Sayson please let me know. There is a company in Vancouver that deals in 
them , call NetVoice. As a newbie in the market , they ( George) gave great 
service and advice. Even called me to see how the Snom 220 was working out ( 
Great customer service!!).

Anyways , your feedback is 
appreciated.

Shawn



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Re: [Asterisk-Users] VoIP Termination

[Me]

I have been most impressed with iax.cc lately.. Only been with them a few 
days but so far, so good!

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- Original Message - 
From: Mike Diehl (Encrypted email preferred) [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
[EMAIL PROTECTED]
Sent: Wednesday, December 15, 2004 11:03 PM
Subject: [Asterisk-Users] VoIP Termination


Hi all.
I'm looking to change from a standard telephone line to a VoIP phone line 
at
home.  I'm looking for recommendations for VoIP providers that I can use 
with
Asterisk.

One of the catches is that I often telecommute and sometimes I do some 
side
business; these practices violate many provider's acceptable use policies.
So, I need a provider who doesn't care how I use the phone, and one that
works well with Asterisk.

Any comments welcome.
Thanx,
--
Mike
gpg key: http://diehlnet.com/~mdiehl/mdiehl.asc
83AD D927 758D 4BFC A800 0277 4B26 75A4 F0D1 C7EB
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Re: [Asterisk-Users] QOS Device?

[Me]

Nate, this is a piece of software?
Any idea of the cost?
--
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- Original Message - 
From: Nathan C. Smith [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
[EMAIL PROTECTED]
Sent: Wednesday, December 15, 2004 11:12 PM
Subject: RE: [Asterisk-Users] QOS Device?



Here is the situation:
A T1 router going into an office which then plugs into the firewall box 
then

into the switch.
None of these devices support QOS..
Is there some sort of box/device that I can place between the T1 router 
and
the firewall box which will allow me to prioritize voice traffic on this
link?

I can't change the T1 router to something that supports QOS because it has
certain redundant features with an ISDN line which are needed.
No commercial interest, just a satisfied customer. . . .
NetEqualizer from APConnections http://www.netequalizer.com/
-Nate
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[Asterisk-Users] QOS Device?

[Me]

Here is the situation:
A T1 router going into an office which then plugs into the firewall box then 
into the switch.

None of these devices support QOS..
Is there some sort of box/device that I can place between the T1 router and 
the firewall box which will allow me to prioritize voice traffic on this 
link?

I can't change the T1 router to something that supports QOS because it has 
certain redundant features with an ISDN line which are needed.

Any help here would be appreciated!
--
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Re: [Asterisk-Users] Follow Me Music on hold

[Me]

Thanks but I am aware of this method, I am trying to get the sequential 
method to work.

--
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- Original Message - 
From: Kristian Kielhofner [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
[EMAIL PROTECTED]
Sent: Monday, December 13, 2004 1:56 AM
Subject: Re: [Asterisk-Users] Follow Me  Music on hold


Me wrote:
OK, I have an extension setup with a follow me like so:
;Operator Going to Sue first, then Mary
exten = 0,1,playback(pls-wait-connect-call)
exten = 0,2,Dial(SIP/103,20,mTt)
exten = 0,3,Dial(SIP/102,20,mTt)
exten = 0,4,VoiceMail([EMAIL PROTECTED])
exten = 0,5,Goto,t|1
This works well except for the fact that the music on hold stops after 
the first timeout and starts over at the beginning of the next line. What 
I mean is that the music sort of skips a beat (so to speak) when * stops 
ring extension 103 and starts ringing extension 102.

Can someone suggest a better/smoother way to do this so the music just 
continues to play until both extensions timeout?

--
Start Your Own ISP!
http://www.YourOwnISP.com
What about calling them both at the same time, not sequentially:
exten = 0,1,playback(pls-wait-connect-call)
exten = 0,2,Dial(SIP/103SIP/102,20,mTt)
exten = 0,3,VoiceMail([EMAIL PROTECTED])
exten = 0,4,Goto,t|1
asterisk -rx show application Dial
would have told you this!
--
Kristian Kielhofner
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[Asterisk-Users] IAX.cc / Sixtel?

[Me]

Anyone using IAX.cc / Sixtel? Would love to hear experiences good or bad.
Thanks!
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Re: [Asterisk-Users] Sipura SPA-2000 won't ring

[Me]

It seems that this is now fixed!
Looks like it was the NAT Keep Alive setting which needed to be set to 
yes in my case.

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- Original Message - 
From: Me [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, December 12, 2004 10:45 PM
Subject: [Asterisk-Users] Sipura SPA-2000 won't ring


I had a Grandstream 286 at my home hitting my Asterisk box at the office,
all worked well and I received phone calls fine until the device just up 
and
died.

I replaced this unit with an SPA-2000 because I have been impressed with 
the
Sipura devices and decided to use them for most of my needs in the future.

Problem is that my phone attached to the device rings shortly after power 
up
of the device but seems to lose it's head after a period of time and stops
ringing until I power cycle the unit or reboot it.

My Asterisk config is the same regarding NAT for this extension and I have
the Sipura registering with * so I am at a loss as to why Asterisk loses 
or
stops ringing this device.

I have dug around and can't seem to solve this issue so far, any help 
would
be appreciated.

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[Asterisk-Users] Follow Me Music on hold

[Me]

OK, I have an extension setup with a follow me like so:
;Operator Going to Sue first, then Mary
exten = 0,1,playback(pls-wait-connect-call)
exten = 0,2,Dial(SIP/103,20,mTt)
exten = 0,3,Dial(SIP/102,20,mTt)
exten = 0,4,VoiceMail([EMAIL PROTECTED])
exten = 0,5,Goto,t|1
This works well except for the fact that the music on hold stops after the 
first timeout and starts over at the beginning of the next line. What I mean 
is that the music sort of skips a beat (so to speak) when * stops ring 
extension 103 and starts ringing extension 102.

Can someone suggest a better/smoother way to do this so the music just 
continues to play until both extensions timeout?

--
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http://www.YourOwnISP.com 

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[Asterisk-Users] Sipura SPA-2000 won't ring

[Me]

I had a Grandstream 286 at my home hitting my Asterisk box at the office,
all worked well and I received phone calls fine until the device just up and
died.
I replaced this unit with an SPA-2000 because I have been impressed with the
Sipura devices and decided to use them for most of my needs in the future.
Problem is that my phone attached to the device rings shortly after power up
of the device but seems to lose it's head after a period of time and stops
ringing until I power cycle the unit or reboot it.
My Asterisk config is the same regarding NAT for this extension and I have
the Sipura registering with * so I am at a loss as to why Asterisk loses or
stops ringing this device.
I have dug around and can't seem to solve this issue so far, any help would
be appreciated.
--
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http://www.YourOwnISP.com 

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Re: [Asterisk-Users] Best option for FSX: IAXy or TDM400P or Voip phone?

[Me]

Personally I find the ATA adapters to be the most versatile, your mileage 
may vary though. When you need more extensions you just buy more ATA's, no 
need to tear up the * box or take it down etc.

Buying IP phones is OK but you are limited to IP Phones only. With the ATA's 
you can buy ANY phone at the local store etc..

Just my opinion of course :)
--
Start Your Own ISP!
http://www.YourOwnISP.com
- Original Message - 
From: Humberto Aicardi [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
[EMAIL PROTECTED]
Sent: Saturday, December 11, 2004 4:50 PM
Subject: [Asterisk-Users] Best option for FSX: IAXy or TDM400P or Voip 
phone?

Hi,
I currently have a * server with a IAXy adapter and a Voip phone. My
doubt is: which is the best option? I personally find IAXy to be very
effective, except from the fact that they don't support G729. The other
option would be to use the TDM400P, which I have heard that it has some
problems with echo, is this true? And finally to use a VOIP phone which look
good and includes several extra features. Oops, I forgot there's still the
gateway option, including ATA186, VoicePlanet, Mediatrix and so on. The
problem is that they are expensive compared to prior options, except the
VOIP phone.
What I really need is a solution that works without the usual * echo
problems. The major issue with IAXy is the price at US$99. I can buy for US$
75 a Grandstream BT102.
Can anyone share their experience with the above solutions?
Thanks in advance,
Humberto
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[Asterisk-Users] SPA-2000 NAT Problems

[Me]

I had a Grandstream 286 at my home hitting my Asterisk box at the office, 
all worked well and I received phone calls fine until the device just up and 
died.

I replaced this unit with an SPA-2000 because I have been impressed with the 
Sipura devices and decided to use them for most of my needs in the future.

Problem is that my phone attached to the device rings shortly after power up 
of the device but seems to lose it's head after a period of time and stops 
ringing until I power cycle the unit or reboot it.

My Asterisk config is the same regarding NAT for this extension and I have 
the Sipura registering with * so I am at a loss as to why Asterisk loses or 
stops ringing this device.

I have dug around and can't seem to solve this issue so far, any help would 
be appreciated.

--
Start Your Own ISP!
http://www.YourOwnISP.com 

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Re: [Asterisk-Users] Experiences with Termination Providers?

[Me]

Good point ;)
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- Original Message - 
From: Linus Surguy [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
[EMAIL PROTECTED]
Sent: Wednesday, December 01, 2004 7:43 AM
Subject: Re: [Asterisk-Users] Experiences with Termination Providers?


Indeed they do - but if you want numbers, you need to say where you are - 
there is no point our company supplying you with UK numbers or toll free, 
if you actually US people to call them!

- Original Message - 
From: Me [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
[EMAIL PROTECTED]
Sent: Tuesday, November 30, 2004 11:48 PM
Subject: Re: [Asterisk-Users] Experiences with Termination Providers?


Mostly interested in US to US for now but interested in all areas, I was 
not aware I was restricted to looking for a provider in only certain 
areas. Most of the termination providers I have dealt with so far offer 
calling worldwide.

Thanks,
Todd
--
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- Original Message - 
From: Linus Surguy [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
[EMAIL PROTECTED]
Sent: Sunday, November 28, 2004 9:34 AM
Subject: Re: [Asterisk-Users] Experiences with Termination Providers?



I hope this is an appropriate question for the list..
I am looking for a VOIP termination provider who can offer the 
following:

-Flat Rate DID's in lots of areas
-GOOD customer service/support with quick response times
-Toll Free DID's at a reasonable rate
-Reliable/Redundant network and availability etc.
It is an appropriate question - but I think the 'Welcome to the mailing 
list' message should point out that this is not a USA only list - anyone 
who posts this type of message should really say where they want service 
to and from!

Linus
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Re: [Asterisk-Users] Experiences with Termination Providers?

[Me]

I saw them too and they looked pretty good. I assume you can buy the minutes 
and use them for whatever you want.

Only issue I have with them at the moment is that their ping times don't 
seem great from where I will be setting up our initial server.

I may setup an account with them for testing purposes.
--
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- Original Message - 
From: Greg - Cirelle Enterprises [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
[EMAIL PROTECTED]
Sent: Wednesday, December 01, 2004 2:55 PM
Subject: Re: [Asterisk-Users] Experiences with Termination Providers?


At 02:07 AM 11/28/04, you wrote:
I hope this is an appropriate question for the list..
I am looking for a VOIP termination provider who can offer the following:
-Flat Rate DID's in lots of areas
-GOOD customer service/support with quick response times
-Toll Free DID's at a reasonable rate
-Reliable/Redundant network and availability etc.
So far I have tested 4 providers which I will not mention here. I have 
found two of them to be offer a quality service with most of the features 
I want but horrible customer service/support and response times to my 
questions etc. The other two seem to respond quickly and have great 
customer service but have awful connections to the web and basically 
unusable services.

Can someone recommend a termination partner for our VOIP Venture that can 
provide reliable services, good features/DID's and GOOD customer service?


Have you had experience with livevoip?
I saw their rates 1 - 10 minutes etc but their
TOS says only residential... very little market there
But I tried emailing them with regards to reselling
their service, so far no response.

Regards
Greg Cirino
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Re: SV: [Asterisk-Users] www.voip-info.org

[Me]

Dead for me too.. I am in the US..
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- Original Message - 
From: David Uzzell [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
[EMAIL PROTECTED]
Sent: Thursday, December 02, 2004 12:41 AM
Subject: Re: SV: [Asterisk-Users] www.voip-info.org


Thorben G. Jensen wrote:
It dead from Denmark too :-(
Well I think yes it is! :(
All I get on traceroute from me!
traceroute to www.voip-info.org (66.151.54.101), 30 hops max, 38 byte 
packets
 1  192.168.2.1 (192.168.2.1)  0.377 ms  0.366 ms  0.189 ms
 2  rns02-kent-syd.comindico.com.au (203.194.30.201)  30.771 ms  25.099 ms 
26.116 ms

 snip to save bandwidth
14  unknown.Level3.net (63.208.234.134)  178.782 ms  180.669 ms  179.576 
ms
15  border17.ge3-0-bbnet2.lax.pnap.net (216.52.255.85)  178.842 ms 180.493 
ms  180.459 ms
16  commp-2.border17.lax.pnap.net (216.52.253.50)  185.969 ms  190.263 ms 
185.715 ms
17  * * *
18  * * *
19  * * *

Me hopes it is not down to long am in the middle on tring to config my now 
working * server :(

Oh well at least it is not just me.
Thanks
David

-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] På vegne af David Uzzell
Sendt: 2. december 2004 07:32
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: [Asterisk-Users] www.voip-info.org

Has the wiki died or is it just my routing to the wiki from Australia?
I have not been able to connect to it for the last hour or more :(
David
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Re: [Asterisk-Users] Experiences with Termination Providers?

[Me]

Hmm, interesting.. I guess my company is the Unicorn of wholesale dialup.. 
We are not a huge company but we do offer reliable services and the BEST 
customer service/tech support in the industry..

Guess I was silly for thinking I could find the same in the VOIP game.. 
Thanks for your input.

Todd Routhier
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- Original Message - 
From: Eric Wieling aka ManxPower [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
[EMAIL PROTECTED]
Sent: Sunday, November 28, 2004 2:44 AM
Subject: Re: [Asterisk-Users] Experiences with Termination Providers?


Me wrote:
I hope this is an appropriate question for the list..
I am looking for a VOIP termination provider who can offer the following:
-Flat Rate DID's in lots of areas
-GOOD customer service/support with quick response times
-Toll Free DID's at a reasonable rate
-Reliable/Redundant network and availability etc.
So far I have tested 4 providers which I will not mention here. I have 
found two of them to be offer a quality service with most of the features 
I want but horrible customer service/support and response times to my 
questions etc. The other two seem to respond quickly and have great 
customer service but have awful connections to the web and basically 
unusable services.

Can someone recommend a termination partner for our VOIP Venture that can 
provide reliable services, good features/DID's and GOOD customer service?

Price is important as well but comes last in line after the items 
mentioned above.
As far as I can tell there are no providers that match your requirements. 
It's the typical growth pattern.  Tiny companies have less reliable 
service, but great customer service.  Larger companies have more reliable 
service, but crappy customer service.  If you ever do fine a unicorn, let 
the rest of us know.

--Eric
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Re: [Asterisk-Users] Experiences with Termination Providers?

[Me]

Mostly interested in US to US for now but interested in all areas, I was not 
aware I was restricted to looking for a provider in only certain areas. Most 
of the termination providers I have dealt with so far offer calling 
worldwide.

Thanks,
Todd
--
Start Your Own ISP!
http://www.YourOwnISP.com
- Original Message - 
From: Linus Surguy [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
[EMAIL PROTECTED]
Sent: Sunday, November 28, 2004 9:34 AM
Subject: Re: [Asterisk-Users] Experiences with Termination Providers?



I hope this is an appropriate question for the list..
I am looking for a VOIP termination provider who can offer the following:
-Flat Rate DID's in lots of areas
-GOOD customer service/support with quick response times
-Toll Free DID's at a reasonable rate
-Reliable/Redundant network and availability etc.
It is an appropriate question - but I think the 'Welcome to the mailing 
list' message should point out that this is not a USA only list - anyone 
who posts this type of message should really say where they want service 
to and from!

Linus
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[Asterisk-Users] Experiences with Termination Providers?

[Me]

I hope this is an appropriate question for the list..
I am looking for a VOIP termination provider who can offer the following:
-Flat Rate DID's in lots of areas
-GOOD customer service/support with quick response times
-Toll Free DID's at a reasonable rate
-Reliable/Redundant network and availability etc.
So far I have tested 4 providers which I will not mention here. I have found 
two of them to be offer a quality service with most of the features I want 
but horrible customer service/support and response times to my questions 
etc. The other two seem to respond quickly and have great customer service 
but have awful connections to the web and basically unusable services.

Can someone recommend a termination partner for our VOIP Venture that can 
provide reliable services, good features/DID's and GOOD customer service?

Price is important as well but comes last in line after the items mentioned 
above.

Thanks!
--
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[Asterisk-Users] Answer Confirmation c

[Me]

On this page in the Wiki:
http://www.voip-info.org/wiki-Asterisk+ZAP+Channels
This text exist:
*
If the letter c follows, then Answer Confirmation is requested, in which 
the call is not considered answered until the called user presses #.
*

Question:
From what I understand you can only use the 'c' option on Zap channels. Is 
there something similar that would allow answer confirmation on NON Zap 
channels?

Thanks!
--
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http://www.YourOwnISP.com 

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Re: [Asterisk-Users] Linux and Windows

[Me]

The thing is, why run it on Windows.. Even though there is a Windows version 
now it's not really a Windows version is a Linux version running on a 
version of Linux that will run on Windows.. YUCK.. That's like taking a 
Cadillac engine and putting in a Yugo just because you feel more comfortable 
driving your Yugo. Just jump in the Cadillac and enjoy the full power of the 
engine without your Yugo wheels falling off along the way.

No point to it, just load up a free version of Linux on an separate PC and 
you are off. Fedora is so easy to install these days it's not like it's 
reserved for just the super geeks.

--
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http://www.YourOwnISP.com
- Original Message - 
From: Tim Donahue [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
[EMAIL PROTECTED]
Sent: Monday, November 01, 2004 8:12 AM
Subject: Re: [Asterisk-Users] Linux and Windows


But don't forget one important point, at this point there is absolutly
NO hardware support for running Asterisk under Windows.  If you need
hardware support (TDM Cards, etc) you will need to run Asterisk on a
Linux based server.
Tim Donahue
On Mon, 2004-11-01 at 01:47, [EMAIL PROTECTED] wrote:
I saw something on the Digium site a few days ago that Asterisk was 
available
for MS based platforms.  Its called AstWind.

http://www.digium.com/index.php?menu=astwind
Cheers,
Sahil
Quoting Bilal Ghayad [EMAIL PROTECTED]:
 Asterisk is working only in Linux? Can not work in Windows 2000?

 Please advise.
 Regards
 Bilal

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Re: [Asterisk-Users] Transfer caller

[Me]

Give us your extensions.conf and we may be able to help you
___


Not sure if you wanted all of it but here it is with my ID's and domains 
changed of course.

*
[general]
static=yes
writeprotect=no

[globals]

[incoming]
exten = s,1,Answer
exten = s,2,Background(ext-or-zero)
exten = s,3,DigitTimeout,3
exten = s,4,ResponseTimeout,30

;Operator Going to Dale for now
exten = 0,1,playback(pls-wait-connect-call)
exten = 0,2,Dial(SIP/102,25,mTt)
exten = 0,3,VoiceMail([EMAIL PROTECTED])
exten = 0,4,Goto,t|1
; 8000 - Get to Vmail
exten = 8000,1,playback(pls-wait-connect-call)
exten = 8000,2,VoiceMailMain(@mydomain.com)
exten = 8000,3,Goto,t|1

; 100 - Todd Office
exten = 100,1,playback(pls-wait-connect-call)
exten = 100,2,Dial(SIP/100,25,mTt)
exten = 100,3,VoiceMail([EMAIL PROTECTED])
exten = 100,4,Goto,t|1

; 1100 - Todd Home
exten = 1100,1,playback(pls-wait-connect-call)
exten = 1100,2,Dial(SIP/1100,25,mTt)
exten = 1100,3,VoiceMail([EMAIL PROTECTED])
exten = 1100,4,Goto,t|1

; 101 - Lewis
exten = 101,1,playback(pls-wait-connect-call)
exten = 101,2,Dial(SIP/101,25,mTt)
exten = 101,3,VoiceMail([EMAIL PROTECTED])
exten = 101,4,Goto,t|1

; 102 - Dale
exten = 102,1,playback(pls-wait-connect-call)
exten = 102,2,Dial(SIP/102,25,mTt)
exten = 102,3,VoiceMail([EMAIL PROTECTED])
exten = 102,4,Goto,t|1

; 103 - Maria
exten = 103,1,playback(pls-wait-connect-call)
exten = 103,2,Dial(SIP/103,25,mTt)
exten = 103,3,VoiceMail([EMAIL PROTECTED])
exten = 103,4,Goto,t|1
; 104 - Jim
exten = 104,1,playback(pls-wait-connect-call)
exten = 104,2,Dial(SIP/104,25,mTt)
exten = 104,3,VoiceMail([EMAIL PROTECTED])
exten = 104,4,Goto,t|1

exten = t,1,Hangup

[outgoing]
; 8000 - Get to Vmail
exten = 8000,1,playback(pls-wait-connect-call)
exten = 8000,2,VoiceMailMain(@mydomain.com)
exten = 8000,3,Goto,t|1
; 100 - Todd
exten = 100,1,playback(pls-wait-connect-call)
exten = 100,2,Dial(SIP/100,25,mTt)
exten = 100,3,VoiceMail([EMAIL PROTECTED])
exten = 100,4,Goto,t|1
; 1100 - Todd Home
exten = 1100,1,playback(pls-wait-connect-call)
exten = 1100,2,Dial(SIP/1100,25,mTt)
exten = 1100,3,VoiceMail([EMAIL PROTECTED])
exten = 1100,4,Goto,t|1

; 101 - Lewis
exten = 101,1,playback(pls-wait-connect-call)
exten = 101,2,Dial(SIP/101,25,mTt)
exten = 101,3,VoiceMail([EMAIL PROTECTED])
exten = 101,4,Goto,t|1

; 102 - Dale
exten = 102,1,playback(pls-wait-connect-call)
exten = 102,2,Dial(SIP/102,25,mTt)
exten = 102,3,VoiceMail([EMAIL PROTECTED])
exten = 102,4,Goto,t|1
; 103 - Maria
exten = 103,1,playback(pls-wait-connect-call)
exten = 103,2,Dial(SIP/103,25,mTt)
exten = 103,3,VoiceMail([EMAIL PROTECTED])
exten = 103,4,Goto,t|1
; 104 - Jim
exten = 104,1,playback(pls-wait-connect-call)
exten = 104,2,Dial(SIP/104,25,mTt)
exten = 104,3,VoiceMail([EMAIL PROTECTED])
exten = 104,4,Goto,t|1

;VoicePulse1
exten = 
_1NXXNXX,1,Dial(IAX2/MyUID:[EMAIL PROTECTED]/${EXTEN})

;VoicePulse2
exten = 
_1NXXNXX,1,Dial(IAX2/MyUID:[EMAIL PROTECTED]/${EXTEN})

;Local on copper line when not dialing a 1
exten = _NXXNXX,2,Dial(Zap/1/${EXTEN})
;Long distance on copper line
exten = _1NXXNXX,2,Dial(Zap/1/${EXTEN})

exten = t,1,Hangup
*
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Re: [Asterisk-Users] Transfer caller

[Me]

Any ideas on this folks? I am kinda stuck without it..
Thanks for any help you can provide..
Todd
--
Start Your Own ISP!
http://www.YourOwnISP.com
- Original Message - 
From: Me [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
[EMAIL PROTECTED]
Sent: Thursday, October 28, 2004 1:51 AM
Subject: Re: [Asterisk-Users] Transfer caller


Give us your extensions.conf and we may be able to help you
___


Not sure if you wanted all of it but here it is with my ID's and domains 
changed of course.

*
[general]
static=yes
writeprotect=no

[globals]

[incoming]
exten = s,1,Answer
exten = s,2,Background(ext-or-zero)
exten = s,3,DigitTimeout,3
exten = s,4,ResponseTimeout,30

;Operator Going to Dale for now
exten = 0,1,playback(pls-wait-connect-call)
exten = 0,2,Dial(SIP/102,25,mTt)
exten = 0,3,VoiceMail([EMAIL PROTECTED])
exten = 0,4,Goto,t|1
; 8000 - Get to Vmail
exten = 8000,1,playback(pls-wait-connect-call)
exten = 8000,2,VoiceMailMain(@mydomain.com)
exten = 8000,3,Goto,t|1

; 100 - Todd Office
exten = 100,1,playback(pls-wait-connect-call)
exten = 100,2,Dial(SIP/100,25,mTt)
exten = 100,3,VoiceMail([EMAIL PROTECTED])
exten = 100,4,Goto,t|1

; 1100 - Todd Home
exten = 1100,1,playback(pls-wait-connect-call)
exten = 1100,2,Dial(SIP/1100,25,mTt)
exten = 1100,3,VoiceMail([EMAIL PROTECTED])
exten = 1100,4,Goto,t|1

; 101 - Lewis
exten = 101,1,playback(pls-wait-connect-call)
exten = 101,2,Dial(SIP/101,25,mTt)
exten = 101,3,VoiceMail([EMAIL PROTECTED])
exten = 101,4,Goto,t|1

; 102 - Dale
exten = 102,1,playback(pls-wait-connect-call)
exten = 102,2,Dial(SIP/102,25,mTt)
exten = 102,3,VoiceMail([EMAIL PROTECTED])
exten = 102,4,Goto,t|1

; 103 - Maria
exten = 103,1,playback(pls-wait-connect-call)
exten = 103,2,Dial(SIP/103,25,mTt)
exten = 103,3,VoiceMail([EMAIL PROTECTED])
exten = 103,4,Goto,t|1
; 104 - Jim
exten = 104,1,playback(pls-wait-connect-call)
exten = 104,2,Dial(SIP/104,25,mTt)
exten = 104,3,VoiceMail([EMAIL PROTECTED])
exten = 104,4,Goto,t|1

exten = t,1,Hangup

[outgoing]
; 8000 - Get to Vmail
exten = 8000,1,playback(pls-wait-connect-call)
exten = 8000,2,VoiceMailMain(@mydomain.com)
exten = 8000,3,Goto,t|1
; 100 - Todd
exten = 100,1,playback(pls-wait-connect-call)
exten = 100,2,Dial(SIP/100,25,mTt)
exten = 100,3,VoiceMail([EMAIL PROTECTED])
exten = 100,4,Goto,t|1
; 1100 - Todd Home
exten = 1100,1,playback(pls-wait-connect-call)
exten = 1100,2,Dial(SIP/1100,25,mTt)
exten = 1100,3,VoiceMail([EMAIL PROTECTED])
exten = 1100,4,Goto,t|1

; 101 - Lewis
exten = 101,1,playback(pls-wait-connect-call)
exten = 101,2,Dial(SIP/101,25,mTt)
exten = 101,3,VoiceMail([EMAIL PROTECTED])
exten = 101,4,Goto,t|1

; 102 - Dale
exten = 102,1,playback(pls-wait-connect-call)
exten = 102,2,Dial(SIP/102,25,mTt)
exten = 102,3,VoiceMail([EMAIL PROTECTED])
exten = 102,4,Goto,t|1
; 103 - Maria
exten = 103,1,playback(pls-wait-connect-call)
exten = 103,2,Dial(SIP/103,25,mTt)
exten = 103,3,VoiceMail([EMAIL PROTECTED])
exten = 103,4,Goto,t|1
; 104 - Jim
exten = 104,1,playback(pls-wait-connect-call)
exten = 104,2,Dial(SIP/104,25,mTt)
exten = 104,3,VoiceMail([EMAIL PROTECTED])
exten = 104,4,Goto,t|1

;VoicePulse1
exten = 
_1NXXNXX,1,Dial(IAX2/MyUID:[EMAIL PROTECTED]/${EXTEN})

;VoicePulse2
exten = 
_1NXXNXX,1,Dial(IAX2/MyUID:[EMAIL PROTECTED]/${EXTEN})

;Local on copper line when not dialing a 1
exten = _NXXNXX,2,Dial(Zap/1/${EXTEN})
;Long distance on copper line
exten = _1NXXNXX,2,Dial(Zap/1/${EXTEN})

exten = t,1,Hangup
*
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