Re: [asterisk-users] TDM400P won't ring GM phone of mere 0.1B

[Yuan LIU]

From: John Novack <[EMAIL PROTECTED]>

[EMAIL PROTECTED] wrote:
This is rather bizarre: My TDM11 (one FXS) rings a $10 passive phone with 
REN of 1.0B, a cheap speaker phone of 0.3B, and a cordless phone with 
marked REN of 0.0B.  But it couldn't properly ring this 27935GE3-B (FCC ID 
G9H2-7930) cordless phone rated at merely 0.1B.  Rarely, the phone will 
crack out an occasional weak and abrupt beap, but never a normal ring.  
Otherwise Asterisk and TDM400P works with this phone, dialing, voice, 
callerID and all that. (In fact, when the FXS "rings", the display lights 
up as it should.)


This is a dual-line 2.4 GHz cordless phone.  It rings normally when 
plugged into wall jack.


Any idea?  Though not described in the manual specifications, Table B1 
suggests that an FXS module (S110M?) supports at least REN 3.


Yuan Liu___


Have you tried toe boost ring voltage option then recompile Zaptel?
It is normally set to a fairly low voltage

John Novack


Thank you so much!  I googled a bit about how to change ring voltage and 
only found an old and suspended feature request from last year that 
concerned wcfxs.c, which is now superceded by wctdm.c.  Yet the same method 
applies.  So I changed the value of RING_X from 0x0160 to 0x023A as 
suggested for European countries (peak of 85V according to that document) 
but left frequency (RING_OSC) unchanged, like the following:


--- wctdm.c.1.2.10  2006-07-07 11:02:39.0 -0700
+++ wctdm.c 2006-12-13 22:04:28.862053256 -0800
@@ -81,7 +81,8 @@
{18,5,"OSC2Y",0x},
{19,6,"RING_V_OFF",0x},
{20,7,"RING_OSC",0x7EF0},
-{21,8,"RING_X",0x0160},
+// {21,8,"RING_X",0x0160},
+{21,8,"RING_X",0x023A},// ring voltage set higher
{22,9,"RING_Y",0x},
{23,255,"PULSE_ENVEL",0x2000},
{24,255,"PULSE_X",0x2000},

Now the GM phone actually rings, though still a little strangely.  Guess I 
just have to experiment a little to completely address this - but that 
feature request 0004542 should really be revived.


A configuration string "boostringer" was mentioned in several messages, 
including one concerning TDM400P, all without indicating the applicable 
configuration file.  This has no apparent effect on TDM400P wherever I 
tried.


BTW, I made some interesting tests - I'm relatively new to this, so bare 
with my learning curve.


Without an oscilloscope, I used a DT-830B multimeter to test three "lines": 
the Digium FXS, a Linksys WRTP54G FXS (yes, that's Vonage), and a PSTN land 
line (SBC).  This meter has a strange behaviour: it allows DC voltage to 
pass through when in AC mode if polarity is favourable.  Guess it's too 
cheap to contain a decent capacitor.  But this gives me an opportunity to 
observe the difference between SLIC in a CO and one in a home appliance.


With Digium and Linksys, if I change polarity of the meter to allow DC 
passthrough, idle AC would appear to be 108V, which drops to 60V during ring 
phase; but with SBC, the apparent "AC" would appear boosted to 161V.  Is 
this because SLIC in a home appliance outputs square waves instead of sine 
waves?


Linksys' AC output (without DC passthrough) measures about 61V, about the 
same as SBC, while Digium's measured about 45V.  Now Digium also measures 
about 61V, but there's still some strangeness.  Maybe an oscilloscope is 
really needed:-(



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Re: [asterisk-users] PRI to SIP

[Peder @ NetworkOblivion]

Virtually any Cisco device from a 2610 up will work.  2610, 2620, 2811, 
2821, 3640, 3700, 3800.  I have 2610 and 3640 in production for 2+ years 
with no issues.


Patrick Fortin wrote:

Hi

Can someone recommend a PRI to SIP Box that work well with asterisk

We are presently testing with a Patton Smartnode 2400 but we are unable 
to fax through it.


We don't want to use digium card in a linux box for the PRI connection.

Which Cisco box would work.

Thanks

Patrick

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--

Network stuff you didn't know
http://www.networkoblivion.com

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Re: [asterisk-users] TDM400P won't ring GM phone of mere 0.1B

[John Novack]

[EMAIL PROTECTED] wrote:

This is rather bizarre: My TDM11 (one FXS) rings a $10 passive phone with REN of 1.0B, a 
cheap speaker phone of 0.3B, and a cordless phone with marked REN of 0.0B.  But it 
couldn't properly ring this 27935GE3-B (FCC ID G9H2-7930) cordless phone rated at merely 
0.1B.  Rarely, the phone will crack out an occasional weak and abrupt beap, but never a 
normal ring.  Otherwise Asterisk and TDM400P works with this phone, dialing, voice, 
callerID and all that. (In fact, when the FXS "rings", the display lights up as 
it should.)

This is a dual-line 2.4 GHz cordless phone.  It rings normally when plugged 
into wall jack.

Any idea?  Though not described in the manual specifications, Table B1 suggests 
that an FXS module (S110M?) supports at least REN 3.

Yuan Liu___
  

Have you tried toe boost ring voltage option then recompile Zaptel?
It is normally set to a fairly low voltage

John Novack

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Re: [asterisk-users] Phone routing - curious what others are doing?

[John Novack]

Doug Crompton wrote:


 when I called Verizon to change (lower) my service it was a
bewildering spider web of rates structures just in the Philadelphia metropolitan area. 
  

And absolutely NO guarantee that the information was correct.
Perhaps if you called 5 times and averaged the answers . . .

It made me wonder why I send any of my calls to
Verizon!
Same here. We are in an area where pretty much every call is either 
local toll or toll, no detail billing and 5 bands of local toll.
We use the 500 minute Vonage plan, for better or worse. Quality and 
reliability is as good as VeriZon, and also Stanaphone and Gizmo.
I find Gizmo a little on the strange side. They send ringback during 
their rather lengthy call set up time, sometimes up to 3 rings, then far 
end ring or busy. Other than that it is also good.

 I was able to cut my Verizon cost down by about half.

  

We now have Verizon down to $14 bucks per month including taxes and fees.

I wonder if any others are splitting calls like this or just biting the bullet 
and going 100% voip???
  
No. I prefer to keep one ILEC line, as it is still more reliable. Even 
though our HSIA is 99%, it still is not 99.99%
all it takes is some geek 80 miles away to screw with the DNS, and 
everything is out for 10-20 minutes.
The modern computer industry still isn't up to the reliability standards 
of the telcos.



John Novack

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[asterisk-users] TDM400P won't ring GM phone of mere 0.1B

[yliu11]

This is rather bizarre: My TDM11 (one FXS) rings a $10 passive phone with REN 
of 1.0B, a cheap speaker phone of 0.3B, and a cordless phone with marked REN of 
0.0B.  But it couldn't properly ring this 27935GE3-B (FCC ID G9H2-7930) 
cordless phone rated at merely 0.1B.  Rarely, the phone will crack out an 
occasional weak and abrupt beap, but never a normal ring.  Otherwise Asterisk 
and TDM400P works with this phone, dialing, voice, callerID and all that. (In 
fact, when the FXS "rings", the display lights up as it should.)

This is a dual-line 2.4 GHz cordless phone.  It rings normally when plugged 
into wall jack.

Any idea?  Though not described in the manual specifications, Table B1 suggests 
that an FXS module (S110M?) supports at least REN 3.

Yuan Liu___
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[asterisk-users] PRI to SIP

[Patrick Fortin]

Hi

Can someone recommend a PRI to SIP Box that work well with asterisk

We are presently testing with a Patton Smartnode 2400 but we are unable to 
fax through it.


We don't want to use digium card in a linux box for the PRI connection.

Which Cisco box would work.

Thanks

Patrick

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Re: [asterisk-users] Measuring VoIP latency and packet loss

[Mochamad Susantok]

I have already use smokeping, and great for measure latency and packet
loss, but not voip packet especialy, or you has been modified smokeping ?
> I like smokeping, as it gives a good sense of the quality of the route
> over time.
>
> --
> Chris Mason
> (264) 497-5670 Fax: (264) 497-8463
> Int:  (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271
> Cell: 264-235-5670
> Yahoo IM: [EMAIL PROTECTED]
>
>
> --
> This message has been scanned for viruses and
> dangerous content by MailScanner, and is
> believed to be clean.
>
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This email was sent using Student EEPIS-Webmail.
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[asterisk-users] Phone routing - curious what others are doing?

[Doug Crompton]

I just went through an exercise of writing a Perl script called from my
Asterisk dialplan to look at a list of area codes and exchanges to
determine which ones are local (no or little cost) under my current
Verizon plan. I route calls outside of my local limits to Gizmo. It works
fine but when I called Verizon to change (lower) my service it was a
bewildering spider web of rates structures just in the Philadelphia
metropolitan area. It made me wonder why I send any of my calls to
Verizon! I was able to cut my Verizon cost down by about half.

I wonder if any others are splitting calls like this or just biting the
bullet and going 100% voip???

With Asterisk/Gizmo I have a local DID for $30/year plus I put $10 credit
on callout last June and I still have $3 left. I prefer pay as you go
rather than flat rate which at $20 or more a month would (for me) be a
$150/year waste! When you have a Gizmo DID the callout CID is
automatically the DID number. You can request a different number though as
long as you have control of it.

Doug

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[asterisk-users] Asterisk, Bluetooth, and wireless phone

[Doug Crompton]

With most of the new wireless phones now Bluetooth is anyone interfacing
(pairing) them to Asterisk? It would be nice to just plop the phone down
near the computer and have home phone access to it. I would be interested
in hardware that might be used for this? I have a Bluetooth phone but not
an interface for the computer.

Doug

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Re: [asterisk-users] ssh access using zaptel channel to dial in.

[Andrew Joakimsen]

The issue is that modem generator, first it would not even be reliable
unless it was a TDM connection, if you are using analog lines why would you
do that when you can use a regular modem? If you are using ISDN then the
issue is where do you get the modem from? I wonder if perhaps spandsp could
support this?

On 12/13/06, Jordan Novak <[EMAIL PROTECTED]> wrote:


 Has anyone done this, or have a thought on how to do it.

I forsee it working like this...

Dial in to a main greeting, dial an extension using a modem string like
782-,,,##409*. The extension would some kind of modem emulator. I know
this compromises security. I was hoping to use an authenticate app in there
as well. My main question is using the zap hardware and some kind of
dialplan app to accomplish this

 Jordan Novak

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Re: [asterisk-users] Searchable Archives of this list

[cb]

On Dec 13, 2006, at 7:42 PM, Hadley Rich wrote:


Google does :)

http://www.google.com/search?q=something+site:lists.digium.com


Sweet... I live off of google, and for some reason trying a site  
specific search from google just didn't cross my mind.


Thanks!

-chris



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Re: [asterisk-users] Asterisk to a Huawei softX3000 problem has already been solved ！

[Josué Conti]

Kevin, contributes with the list, somebody can have this problem and you it
can help. The list is here for helping, but also we must contribute with it.
:)
Best Regards

Josue

2006/12/13, kevinho <[EMAIL PROTECTED]>:



Asterisk to a Huawei softX3000 problem has already been solved ！

msn:[EMAIL PROTECTED]
_
Windows Live Safety Center 为您的计算机提供免费的安全扫描服务。
http://safety.live.com/site/ZH-CN/default.htm
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Re: [asterisk-users] Searchable Archives of this list

[Hadley Rich]

On Thursday 14 December 2006 13:31, cb wrote:
> Is there a searchable archive of this list? Did I overlook something
> obvious? I can find the archives, but short of downloading all the
> monthly gzips and building my own searchable database, it seems my
> only other option is to go month by month looking at subjects and
> hope to stumble on what I'm after.
>
> Does anyone maintain a public searchable version of the archives?

Google does :)

http://www.google.com/search?q=something+site:lists.digium.com

-- 
http://nicegear.co.nz
New Zealand's VoIP supplier
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[asterisk-users] Re: Realtime +Mysql +Failover

[JR Richardson]

> Can Realtime be set up with a secondary mysql server to get its data
> from. We can set up mysql to sync with its fellow server, and maybe when
> it goes down, it couldn't make any changes (write), but either way, you
> could still get the extension info, etc, so your phones would still
> survive a mysql outage.


I run MySQL Replication from a Master MySQL Server (only runs the
database function).  Apply patch
http://bugs.digium.com/view.php?id=5881to the various asterisk servers
and configure them as slaves to the master.  Write only to the Master
database and all the slaves update automatically, in realtime.  Have
the PBX's write to the Master and read from 127.0.0.1(localhost), if a
PBX goes down, it doesn't affect database operations with the rest of
the cluster.  Also if the Master goes down, all the PBX's can still
read from their local database.

snip from res_mysql.conf

[general]
dbname=asterisk
dbuser=asterisk
dbpass=asterisk
dbport = 3306
dbsock = /tmp/mysql.sock

[read]
dbhost = 127.0.0.1

[write]
dbhost = 10.10.10.10


For MySQL redundancy, have a second MySQL server running in a
Master-to-Master replication with the first Master.   There is your
backup if one goes down, switch IP or run LVS or Heatbeat between the
two Master MySQL servers.

This setup, IMHO, is less costly than a MySQL Cluster (due to less
servers needed to implement), is easier to configure and get running,
easier to recover during a failure, greatly increases realtime
performance on the PBX (reading from localhost than from across a
network), scales very well (15-20 slaves for each Master).  Also MySQL
replication is more mature that MySQL clustering (I'm using 3.23, not
5.0 which is needed for clustering) so the software footprint is
smaller comparatively).

Hope this helps.

JR Richardson
Engineering for the Masses
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[asterisk-users] Asterisk to a Huawei softX3000 problem has already been solved ！

[kevinho]

Asterisk to a Huawei softX3000 problem has already been solved ！

msn:[EMAIL PROTECTED]
_
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[asterisk-users] Searchable Archives of this list

[cb]

Is there a searchable archive of this list? Did I overlook something  
obvious? I can find the archives, but short of downloading all the  
monthly gzips and building my own searchable database, it seems my  
only other option is to go month by month looking at subjects and  
hope to stumble on what I'm after.


Does anyone maintain a public searchable version of the archives?  
I've got tons of questions brewing, but I can't believe I'd be the  
first to ask any of them, so I'd really rather search thru old posts  
for answers before asking something that has likely been asked a  
dozen times before.


-chris



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Re: [asterisk-users] CallerID Issue (asterisk newbie)

[Anselm Martin Hoffmeister]

Am Mittwoch, den 13.12.2006, 20:56 +0100 schrieb Sven Beisiegel:
> Hi everybody...
> 
> I have a similar problem... I don't get the ID of the person that i
> called on my phone... Does anyone know something about this problem?
> 
> greets,
> Sven

Well, this sounds as specific as "my computer is broken...". To get
people to helping you, giving lots of details of your problem is
generally a good idea(tm). Something like:

When someone from SIP|PSTN|IAX calls ME(SIP|IAX|whatever), CALLERID(num)
contains BLAH although it should contain BLUBB. My display shows ERNIE
instead of BERT. The relevant part of my sip.conf|zapata.conf|
extensions.conf|logfile is .

Feel free to be more elaborate than that ;-)

Best regards,
Anselm

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RE: [asterisk-users] Stress test

[Cory Andrews]

Empirix has products and services for stress testing
http://www.empirix.com/default.asp?action=category&ID=12
 
Cory Andrews



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dovid B
Sent: Wednesday, December 13, 2006 7:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Stress test


A backwards way may be to set up a diffrent asterisk server and call in
to your main box. Keep calling untill the box that your testing acts up.
 

- Original Message - 
From: Andre Luiz Martins Rodrigues
  
To: asterisk-users@lists.digium.com 
Sent: Wednesday, December 13, 2006 3:55 PM
Subject: [asterisk-users] Stress test

Hello peoples,


I need to do a test of urgent stress.  It know as much as
connections simultaneous my equipment is going to do passing codec g729
and g723.  Someone knows say me as obtain does him?  


Andre Luiz Martins 
mailto:[EMAIL PROTECTED]










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Re: [asterisk-users] Playing a sound file on handset pickup

[Dovid B]

I would do the same. Basicly each time they call out check to see if it is 
enabled or not. If it isnt then send out the call. If it is then play 
reminder and then send out the call.



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


Sent: Wednesday, December 13, 2006 10:41 PM
Subject: Re: [asterisk-users] Playing a sound file on handset pickup



How about put it in the dial plan?  So anytime you try to make an
outbound call it would play a reminder saying that the alternate
greeting is enabled.   You could just use a DB variable.

On 12/13/06, Mailinglisten <[EMAIL PROTECTED]> wrote:

John French wrote:
> I've added the ability for a user to record a custom message associated
> with a special IVR menu for occasions when business will be closed for
> some non-standard amount of time (Maybe 4 days at Christmas...)   They
> just dial 800, record the message then hang up and dial 801 to enable
> it.  Presumably, when they return after the holiday, they should dial
> 802 to disable it and return to the normally scheduled menus.  But they
> will most likely forget so I'd like to set up some type of reminder
> functionality; perhaps playing a message back to them stating that the
> custom message is still enabled before giving them dialtone or 
> something

> to the same effect.  Is this possible and can anyone offer
> recommendations?
>
> Thanks.
>
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>
Why not just add that functionality to the s extension? If no extension
is given, they will end up there, won't they? So if that "I'm not here"
message is set up, and the client picks up the phone, we assume that
he/she is back and thus delete the notification without notice.

- Fabian Foerster
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Re: [asterisk-users] Need help getting started with asterisk

[Dovid B]

- Original Message - 
From: "Michael Sullivan" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Wednesday, December 13, 2006 3:35 PM
Subject: Re: [asterisk-users] Need help getting started with asterisk



On Wed, 2006-12-13 at 15:53 +1100, Paul Hales wrote:

What does zttool show?

And after you 'modprobe wctdm' what does your dmesg
read? /var/log/messages?

You should see something about a card being recognised

PaulH


After I modprobe wctdm, nothing new shows up in /var/log/messages and
dmesg is just notices about my firewall.  zttool doesn't show much of
anything... :(

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The X100P clone cards arent to expensive. Try looking for one on ebay. 



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Re: [asterisk-users] Stress test

[Dovid B]

A backwards way may be to set up a diffrent asterisk server and call in to your 
main box. Keep calling untill the box that your testing acts up.

  - Original Message - 
  From: Andre Luiz Martins Rodrigues 
  To: asterisk-users@lists.digium.com 
  Sent: Wednesday, December 13, 2006 3:55 PM
  Subject: [asterisk-users] Stress test


  Hello peoples,


  I need to do a test of urgent stress.  It know as much as connections 
simultaneous my equipment is going to do passing codec g729 and g723.  Someone 
knows say me as obtain does him?  


  Andre Luiz Martins 
  mailto:[EMAIL PROTECTED]





--


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Re: [asterisk-users] sip help for newbie

[Dovid B]

You need port 5060 as well as 1-2 UDP open to the server. Also is the 
server behind NAT at all ?

  - Original Message - 
  From: blackwater dev 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Sent: Wednesday, December 13, 2006 5:14 AM
  Subject: Re: [asterisk-users] sip help for newbie


  Thanks for the info, I've gone through the tutorial and followed it and 
asterisk is running but I just can't seem to log in.  The xten phone just tells 
me connection timed out.  I'm simply running asterisk on a webserver that is 
also running apache and service content.  I simply pinged the box to get the ip 
to plug into the softphone.  Do I need to open a port or something? 


  On 12/12/06, Forrest Beck <[EMAIL PROTECTED]> wrote:
www.asteriskguru.com


On 12/12/06, blackwater dev <[EMAIL PROTECTED]> wrote:
> Does anyone know of any good step by step tutorials on getting sip set 
up? 
> I have asterisk installed but can't seem to figure out how to get an 
account
> set up and connect from my xTen phone so I can try the demo.  The 
tutorials
> I read online seem to go into voicepulse stuff and all and I don't have 
an 
> account there so am a bit lost.
>
> Thanks!
>
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Re: [asterisk-users] MFC/R2 on chan_zap

[Moises Silva]

And unicall is failing because have you read the error message?

it says fails to open a channel, so is highly probable that the
problem is your configuration, or zaptel driver module not properly
loaded.

regards

On 12/13/06, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:

On Wed, Dec 13, 2006 at 09:12:13AM -0500, Alejandro Rios Peña wrote:
> Hello.
>
> I'm trying to setup MFC/R2 signaling, but chan_unicall fails at loading
> with this message:
>
> ---
> Unable to load module chan_unicall.so
>  Loaded /usr/lib/asterisk/modules/chan_unicall.so => (Unified call
> processing (UniCall))
>   == Parsing '/etc/asterisk/unicall.conf': Found
> 061213-075938 ERROR[11454]: chan_unicall.c:3361 mkintf: Unable to open
> channel 1: Success
> here = 0, tmp->channel = 0, channel = 1
> 061213-075938 ERROR[11454]: chan_unicall.c:4081 setup_unicall: Unable to
> register channel '1-15'
> 061213-075938 WARNING[11454]: loader.c:414 __load_resource:
> chan_unicall.so: load_module failed, returning -1
> ---
>
>
> I saw there is some MFC/R2 code on chan_zap.c of asterisk version
> 1.2.13, has anyone tried it?

It was removed for being non-functional.

--
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icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Need help getting started with asterisk

[Paul Hales]

Sounds like you may have an unsupported card. :(

PaulH


On Wed, 2006-12-13 at 07:35 -0600, Michael Sullivan wrote:
> On Wed, 2006-12-13 at 15:53 +1100, Paul Hales wrote:
> > What does zttool show?
> > 
> > And after you 'modprobe wctdm' what does your dmesg
> > read? /var/log/messages?
> > 
> > You should see something about a card being recognised
> > 
> > PaulH
> 
> After I modprobe wctdm, nothing new shows up in /var/log/messages and
> dmesg is just notices about my firewall.  zttool doesn't show much of
> anything... :(
> 
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RE: [asterisk-users] Polycom MyStat

[Douglas Garstang]

DND is different. When Asterisk sends an INVITE to a phone with DND set, the 
phone responds to the invite with a 'Busy'... or it could be 'Declined' 
something like that.

-Original Message-
From: Brian Roy [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 13, 2006 3:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom MyStat




On 12/13/06, LST < [EMAIL PROTECTED]> wrote: 



 



I think that is strictly a Polycom to Polycom thing (Buddywatch).  I do not 
believe it affects Asterisk (i.e. Busy = DND).  With that being said, I don't 
think it works very well even with all Polycom phones.  I can change my status 
to Busy and look at the other Polycom Phones and they still show me as Online.  
(Yes, I have bw set to 1.) 

 
 
Does that mean that DND doesn't show up as a hint either? 
 
-Brian

 

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Re: [asterisk-users] Polycom MyStat

[Brian Roy]

On 12/13/06, LST <[EMAIL PROTECTED]> wrote:






I think that is strictly a Polycom to Polycom thing (Buddywatch).  I do
not believe it affects Asterisk (i.e. Busy = DND).  With that being said,
I don't think it works very well even with all Polycom phones.  I can
change my status to Busy and look at the other Polycom Phones and they
still show me as Online.  (Yes, I have bw set to 1.)




Does that mean that DND doesn't show up as a hint either?

-Brian
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[asterisk-users] Asterisk Community lost a valuable contributor today

[Kevin P. Fleming]

I want to inform the community that last night one of our valuable
contributors and long-time community members, Rich Adamson, passed away
after a long battle with cancer.

Rich spent many, many hours sharing his valuable telecom and
transmission systems experience with everyone who could benefit from it;
he helped countless people solve audio quality and echo problems with
their analog lines when others had given up trying to help. He also
spent many hours helping those of us who work on Asterisk, Zaptel and
Digium hardware improve our products by providing valuable feedback and
testing our changes.

We want to express our deepest sympathies to Rich's family and thank
them for allowing Rich to spend much of his limited time with us. He
will be missed by everyone who had the pleasure of knowing him.
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Re: [asterisk-users] send fax by Iaxmodem ?

[Lee Howard]

Marco Mouta wrote:


Dec 13 11:28:07.51: [ 9242]: DIAL 2079^M
Dec 13 11:28:07.51: [ 9242]: <-- [9:ATDT2079\r]
Dec 13 11:28:16.70: [ 9242]: --> [4:BUSY]
Dec 13 11:28:46.70: [ 9242]: MODEM TIMEOUT: reading line from modem
Dec 13 11:28:46.71: [ 9242]: MODEM 
Dec 13 11:28:46.71: [ 9242]: SEND FAILED: JOB 1 DEST 2079^M ERR 
Unknown problem



In your case BUSY means exactly that, and you should take a look at the 
Asterisk CLI to get more information as to what "busy" really means.


However, your dialstring terminated by a carriage return ( or ^M) is 
problematic, too, because it essentially instructs HylaFAX to ignore all 
responses after ATDT2079 except for "OK" and then proceed from there.  
Basically you just need to get rid of that terminating carriage return.


Lee.
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Re: [asterisk-users] Measuring VoIP latency and packet loss

[Chris Mason (Lists)]

I like smokeping, as it gives a good sense of the quality of the route 
over time.


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(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 



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Re: [asterisk-users] Realtime +Mysql +Failover

[Dovid B]

You can have a MySQL Cluster.

- Original Message - 
From: "David Thomas" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Wednesday, December 13, 2006 8:36 PM
Subject: Re: [asterisk-users] Realtime +Mysql +Failover



On 12/13/06, Rob Schall <[EMAIL PROTECTED]> wrote:

Hoping someone out there has run into this or has some ideas for us.

We currently have asterisk set up with Realtime (using mysql) for its
extensions,sip and voicemail files.

The problem we are trying to solve, is one of a failover mechanism. What
if our mysql server went down.

Can Realtime be set up with a secondary mysql server to get its data
from. We can set up mysql to sync with its fellow server, and maybe when
it goes down, it couldn't make any changes (write), but either way, you
could still get the extension info, etc, so your phones would still
survive a mysql outage.

Any ideas?
Thanks,
Rob


I don't think Realtime can be setup with a secondary server (someone
please correct me if I'm wrong).

Two possibilities come to mind...

1. You can run MySQL in an HA arangement with on box as the hot standby.
2. If you can allow for ocassional asterisk reloads, you could use
Realtime Static

Regards,
David
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RE: [asterisk-users] Re: Core Dump: create_transaction (p=0x0) atpbx_dundi.c:2787

[Douglas Garstang]

> -Original Message-
> From: Tony Mountifield [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, December 13, 2006 1:19 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Re: Core Dump: create_transaction (p=0x0)
> atpbx_dundi.c:2787
> 
> 
> In article 
> <[EMAIL PROTECTED]>,
> Douglas Garstang <[EMAIL PROTECTED]> wrote:
> > Anyone seen this...? Is it a known issue?
> > 
> > I'd file a bug, but we're on 1.2.9.13, and every time I 
> file a bug and it isn't against the
> > latest code I get given crap for it. Given that most of the 
> time you don't know HOW to
> > reproduce a problem on the latest code anyway, not 
> accepting bugs from older versions does
> > the community no service, because potential bugs are never 
> accepted for submission.
> > 
> > (gdb) bt full
> > #0  0xb7da8d3c in mallopt () from /lib/libc.so.6
> > No symbol table info available.
> > #1  0xb7da7e3a in malloc () from /lib/libc.so.6
> > No symbol table info available.
> > #2  0xb7b30aa1 in create_transaction (p=0x0) at pbx_dundi.c:2787
> > trans = (struct dundi_transaction *) 0x0
> 
> Hmmm, that will be a tricky one to track down. There's no 
> reason to get
> a core dump from within malloc() unless something else has previously
> stomped outside of its own malloced area, smashing the free list.
> 
> So the problem is likely not within create_transaction(), but caused
> sometime before, possibly in some completely unrelated code.
> 
> Is it repeatable, or just happens at random (or even just once)?

Only the one time so far.
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Re: [asterisk-users] send fax by Iaxmodem ?

[Lee Howard]

Noc Phibee wrote:


déc 13 13:47:21.12: [13725]: <-- [15:ATDT0426690268\r]
déc 13 13:47:21.12: [13725]: --> [11:NO DIALTONE] 



Getting NO DIALTONE from iaxmodem means that the call was rejected.

Double-check your setup.  Make sure that iaxmodem can register.  Then 
make sure that the context that you're calling into (per the iax.conf 
iaxmodem settings) supports extens for the numbers you're dialing.  
Watch the Asterisk CLI.


Lee.
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Re: [asterisk-users] ssh access using zaptel channel to dial in.

[Noah Miller]

Hi Jordan -


Has anyone done this, or have a thought on how to do it.

I forsee it working like this...

Dial in to a main greeting, dial an extension using a modem string like
782-,,,##409*. The extension would some kind of modem emulator. I know
this compromises security. I was hoping to use an authenticate app in there
as well. My main question is using the zap hardware and some kind of
dialplan app to accomplish this


This sounds like a whole lot of unnecessary complication.  Why not
just use a regular old modem connected to a serial interface on the
computer you want to get CLI access to?  No need to involve asterisk
at all.  For security, let the OS handle authentication.

Old school? Yes, but it works.

- Noah
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[asterisk-users] ZAP multiline handset questions

[Noah Miller]

Hi All -

I haven't worked much with ZAP handsets before, but I've got a client
who is insistent on using a particular phone.  My questions:

1. With multiline analog phones, if I've got multiple phones, each
connected to a different FXS interface, is there a way to make the
line status lights on the other phones show that a particular FXO is
in use (like a key system, or like SIP hinting)?

2. Does anyone know of a good analog cordless phone (independent of
any base desk phone) that can handle multiple lines?

Thanks!
Noah
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[asterisk-users] ssh access using zaptel channel to dial in.

[Jordan Novak]

Has anyone done this, or have a thought on how to do it.
 
I forsee it working like this...
 
Dial in to a main greeting, dial an extension using a modem string like
782-,,,##409*. The extension would some kind of modem emulator. I
know this compromises security. I was hoping to use an authenticate app
in there as well. My main question is using the zap hardware and some
kind of dialplan app to accomplish this
 
Jordan Novak
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Re: [asterisk-users] Diva Server V-BRI-2 and internal numbers

[Armin Schindler]

On Wed, 13 Dec 2006, Gregory Duchatelet wrote:
> Hi,
> 
> I have an asterisk with a DIVA Server V-BRI-2 card, connected to a Siemens
> PABX. From a SIP phone, I can call other internal SIP phones, external
> numbers (to PSTN), but I can't call internal phones connected to the
> internal phone network.
> 
> When I call 107, which is an internal phone, heres the logs from asterisk:
> 
>  
> 
> -- Executing Dial("SIP/Greg-081f5a10", "CAPI/ISDN1/b:107||rtT") in new
> stack
> 
> -- Called ISDN1/b:107
> 
> -- CAPI/ISDN1/107-1a is proceeding passing it to SIP/Greg-081f5a10
> 
> -- CAPI/ISDN1/107-1a is busy
> 
>   == ISDN1: CAPI Hangingup
> 
>   == Everyone is busy/congested at this time (1:1/0/0)
> 
> -- Executing Hangup("SIP/Greg-081f5a10", "") in new stack
> 
>   == Spawn extension (macro-appel_sortant, s, 2) exited non-zero on
> 'SIP/Greg-081f5a10' in macro 'appel_sortant'
> 
>   == Spawn extension (macro-appel_sortant, s, 2) exited non-zero on
> 'SIP/Greg-081f5a10'
> 
>  
> 
> BUT! If I call an internal isdn number like 122 which is a fax, the call is
> answered.
>  
> 
> How can I call 107 ?

It looks like 107 is busy ;-)
Please increase verbosity, like
  set verbose 5
  capi debug
to see what is happening.

Armin

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Re: [asterisk-users] Programming soft buttons on the IP601?

[Noah Miller]

Hi Warren -


When the IP601 is sitting unused, it uses the first 2 of the 4 soft
buttons under the screen.  The third one is empty, which is good because
it is used for "Exit".

I would like to be able to use that 4th button for group pickup (*8#)
and have it read "Pickup".  Is this possible?  If so, how?


In short, no, the soft-buttons cannot be user-programmed.

Many on this list have tried to reprogram these buttons, but I'm not
aware of anyone who's actually gotten anything to work.  I'd love for
someone to prove me wrong, of course.

The hard buttons are programmable.  With the 2.0.x firmwares you can
program a string of digits (*8#) by re-programming one of the hard
keys as a speed dial.  The services button isn't normally used, so you
could use it for this purpose.  I generally reprogram the
"Directories" and "Call Lists" keys, too, as they're redundant.


- Noah
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Re: [asterisk-users] Playing a sound file on handset pickup

[Matt]

How about put it in the dial plan?  So anytime you try to make an
outbound call it would play a reminder saying that the alternate
greeting is enabled.   You could just use a DB variable.

On 12/13/06, Mailinglisten <[EMAIL PROTECTED]> wrote:

John French wrote:
> I've added the ability for a user to record a custom message associated
> with a special IVR menu for occasions when business will be closed for
> some non-standard amount of time (Maybe 4 days at Christmas...)   They
> just dial 800, record the message then hang up and dial 801 to enable
> it.  Presumably, when they return after the holiday, they should dial
> 802 to disable it and return to the normally scheduled menus.  But they
> will most likely forget so I'd like to set up some type of reminder
> functionality; perhaps playing a message back to them stating that the
> custom message is still enabled before giving them dialtone or something
> to the same effect.  Is this possible and can anyone offer
> recommendations?
>
> Thanks.
>
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>
Why not just add that functionality to the s extension? If no extension
is given, they will end up there, won't they? So if that "I'm not here"
message is set up, and the client picks up the phone, we assume that
he/she is back and thus delete the notification without notice.

- Fabian Foerster
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Re: [asterisk-users] Polycom IP4000 and vsftpd 2.0.1

[Andrew Joakimsen]

Do you have the latest firmware files from polycom and sample
configurations? Can you get the phone to accept those? Any reason why you
are using FTP? Http has worked without a hitch. What does your logs say?

On 12/13/06, Anthony Rodgers <[EMAIL PROTECTED]> wrote:


Is anyone else having trouble getting a Polycom IP4000 (running SIP
1.6.7 and BootROM 3.1.3) to download its configuration files from a
vsftpd 2.0.1 server? We have 100+ IP501s that manage this without
problems, but the IP4000 keeps timing out.

We have opened a case with Polycom, but they are insisting that it is
our configuration files that are at fault, even though the phone times
out on bootrom.ld, long before it attempts to load the configuration
files.

I did turn up some postings about IP501s, BootROM 3.1.3 and vsftpd
2.0.3, and wonder if this might be a similar issue.

CP

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Re: [asterisk-users] Measuring VoIP latency and packet loss

[J. Oquendo]

Carlos Rojas wrote:

iftop

On 12/12/06, *Mochamad Susantok* <[EMAIL PROTECTED] 
> wrote:


Dear all,
Are there anyone have ben to use some tool or method to measure
latency
and packet loss for VoIP packet ?




Commercial or Open Source?

For Open Source, try IPTraf, PKStat, Netperf, Softflowd, MRTG, make your 
own is pretty much what I try to do... Commercial? Opera from Opticom, 
or Fluke Networks' NetTools

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Re: [asterisk-users] webvoicemail

[Brian Roy]

On 12/13/06, Ed Nuñez <[EMAIL PROTECTED]> wrote:


I've been trying to find where to download the Web Vmail application and
instructions on how to install it for Asterisk BE.  Any ideas?




Is this any different than the vmail.cgi that comes with the open version?
Otherwise, you will just need to grab a compiled copy off of another box.
Only needs vmail.cgi and a couple of supporting graphics.

-Brian
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[asterisk-users] Re: Core Dump: create_transaction (p=0x0) at pbx_dundi.c:2787

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
Douglas Garstang <[EMAIL PROTECTED]> wrote:
> Anyone seen this...? Is it a known issue?
> 
> I'd file a bug, but we're on 1.2.9.13, and every time I file a bug and it 
> isn't against the
> latest code I get given crap for it. Given that most of the time you don't 
> know HOW to
> reproduce a problem on the latest code anyway, not accepting bugs from older 
> versions does
> the community no service, because potential bugs are never accepted for 
> submission.
> 
> (gdb) bt full
> #0  0xb7da8d3c in mallopt () from /lib/libc.so.6
> No symbol table info available.
> #1  0xb7da7e3a in malloc () from /lib/libc.so.6
> No symbol table info available.
> #2  0xb7b30aa1 in create_transaction (p=0x0) at pbx_dundi.c:2787
> trans = (struct dundi_transaction *) 0x0

Hmmm, that will be a tricky one to track down. There's no reason to get
a core dump from within malloc() unless something else has previously
stomped outside of its own malloced area, smashing the free list.

So the problem is likely not within create_transaction(), but caused
sometime before, possibly in some completely unrelated code.

Is it repeatable, or just happens at random (or even just once)?

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[asterisk-users] Polycom IP4000 and vsftpd 2.0.1

[Anthony Rodgers]

Is anyone else having trouble getting a Polycom IP4000 (running SIP 
1.6.7 and BootROM 3.1.3) to download its configuration files from a 
vsftpd 2.0.1 server? We have 100+ IP501s that manage this without 
problems, but the IP4000 keeps timing out.


We have opened a case with Polycom, but they are insisting that it is 
our configuration files that are at fault, even though the phone times 
out on bootrom.ld, long before it attempts to load the configuration 
files.


I did turn up some postings about IP501s, BootROM 3.1.3 and vsftpd 
2.0.3, and wonder if this might be a similar issue.


CP

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[asterisk-users] Programming soft buttons on the IP601?

[Warren (mailing lists)]

When the IP601 is sitting unused, it uses the first 2 of the 4 soft 
buttons under the screen.  The third one is empty, which is good because 
it is used for "Exit".


I would like to be able to use that 4th button for group pickup (*8#) 
and have it read "Pickup".  Is this possible?  If so, how?


Thanks,
Warren
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Re: [asterisk-users] CallerID Issue (asterisk newbie)

[Sven Beisiegel]

Hi everybody...

I have a similar problem... I don't get the ID of the person that i
called on my phone... Does anyone know something about this problem?

greets,
Sven

2006/12/13, Bruce Ferrell <[EMAIL PROTECTED]>:

Anselm Martin Hoffmeister wrote:
> Am Mittwoch, den 13.12.2006, 15:03 + schrieb
> [EMAIL PROTECTED]:
>
>>Hi guys. This is my 1st post here (after much reading). I have a test
>>asterisk system setup using X-Lite Soft Phones, and the issue I am
>>running into is that caller id shows up as "asterisk" on all incoming
>>calls and on all local to local calls (internal). I have showcallerid,
>>etc. configured in zapata.conf, but I'm drawing a blank.  When I check
>>my voicemails it tells me that the message is from an unknown caller.
>>I would appreciate any info.
>
>
> Zapata.conf is not usually related to sip device callerid, if you have
> no Zap interface.
>
> Try setting the callerid= stuff in sip.conf appropriately.
> Mine looks like this, for my desktop phone:
>
> [sip504]
> mailbox=01
> callerid=504
> type=friend
> username=sip504
> secret=YouDontKnowThis
> context=sipclient
> host=dynamic
> nat=yes
> disallow=all
> allow=alaw
> allow=gsm
>
> so the callerid= line is what you should adapt.
>
> BR
> Anselm

I've seen this recently when the caller ID comes in
NPANXXy somehow the very long callerid isn't handled
well.   I ended up peeling off the first 10 digits and re-stuffing the
callerid with that
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Re: [asterisk-users] ParkAndAnnounce + Paging

[Noah Miller]

It is possible to announce the parking position through a paging to a group
of extensions?

I would like that when someone parks a call, some phones will announce with
the speaker the position.

Something like:

exten => s,1,ParkAndAnnounce(call-parked-at:PARKED|30|PAGE(LOCAL/[EMAIL 
PROTECTED]&LOCAL/[EMAIL PROTECTED]|)

Is there a way, maybe with a different approach?


I think your method should work.  Have you tried it yet?  It's a very
good idea, BTW.  Talk about an auto-attendant!


- Noah
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Re: [asterisk-users] Asterisk 1.4 realtime with mysql 5.0 and unixODBC.

[Noah Miller]

Hi Lan -


I am trying to upgrade my testing asterisk realtime 1.2.13 with MySQL 5.0
and unixODBC to the beta asterisk 1.4.
I run the make and make install for the asterisk-addon just fine, It created
the modules res_config_mysql.so and  cdr_addon_mysql.so without any problem
or error.  However, when I run the asterisk, it comes up with the error :

  == Parsing '/etc/asterisk/res_mysql.conf': Found
asterisk: symbol lookup error:
/usr/lib/asterisk/modules/res_config_mysql.so: undefined
symbol: mysql_init


One question: Did you remove the old 1.2.x addon modules from your
modules directory (/usr/lib/asterisk/modules) before you installed the
new 1.4.x addons?


- Noah
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Re: [asterisk-users] Playing a sound file on handset pickup

[Mailinglisten]

John French wrote:
I've added the ability for a user to record a custom message associated 
with a special IVR menu for occasions when business will be closed for 
some non-standard amount of time (Maybe 4 days at Christmas...)   They 
just dial 800, record the message then hang up and dial 801 to enable 
it.  Presumably, when they return after the holiday, they should dial 
802 to disable it and return to the normally scheduled menus.  But they 
will most likely forget so I'd like to set up some type of reminder 
functionality; perhaps playing a message back to them stating that the 
custom message is still enabled before giving them dialtone or something 
to the same effect.  Is this possible and can anyone offer 
recommendations?
 
Thanks.


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Why not just add that functionality to the s extension? If no extension 
is given, they will end up there, won't they? So if that "I'm not here" 
message is set up, and the client picks up the phone, we assume that 
he/she is back and thus delete the notification without notice.


- Fabian Foerster
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Re: [asterisk-users] how to define a secure trunk

[Pavel Jezek]

http://www.voip-info.org/wiki/view/IAX+encryption



Joao Pereira wrote:

Hello
I would like to define a trunk from my Asterisk to a VoIP provider, 
but I want to make it secure, because its through the Internet.
I want to be sure no one makes calls as being me, and that my calls 
aren't intercepted.
Is it possible to define encrypted trunks? And should I define the 
trunk in SIP, IAX or something else?


Thanks
Joao Pereira


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[asterisk-users] Playing a sound file on handset pickup

[John French]

I've added the ability for a user to record a custom message associated 
with a special IVR menu for occasions when business will be closed for 
some non-standard amount of time (Maybe 4 days at Christmas...)   They 
just dial 800, record the message then hang up and dial 801 to enable 
it.  Presumably, when they return after the holiday, they should dial 
802 to disable it and return to the normally scheduled menus.  But they 
will most likely forget so I'd like to set up some type of reminder 
functionality; perhaps playing a message back to them stating that the 
custom message is still enabled before giving them dialtone or something 
to the same effect.  Is this possible and can anyone offer 
recommendations?
 
Thanks.

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Re: [asterisk-users] CallerID Issue (asterisk newbie)

[Bruce Ferrell]

Anselm Martin Hoffmeister wrote:

Am Mittwoch, den 13.12.2006, 15:03 + schrieb
[EMAIL PROTECTED]:

Hi guys. This is my 1st post here (after much reading). I have a test  
asterisk system setup using X-Lite Soft Phones, and the issue I am  
running into is that caller id shows up as "asterisk" on all incoming  
calls and on all local to local calls (internal). I have showcallerid,  
etc. configured in zapata.conf, but I'm drawing a blank.  When I check  
my voicemails it tells me that the message is from an unknown caller.   
I would appreciate any info.



Zapata.conf is not usually related to sip device callerid, if you have
no Zap interface.

Try setting the callerid= stuff in sip.conf appropriately.
Mine looks like this, for my desktop phone:

[sip504]
mailbox=01
callerid=504
type=friend
username=sip504
secret=YouDontKnowThis
context=sipclient
host=dynamic
nat=yes
disallow=all
allow=alaw
allow=gsm

so the callerid= line is what you should adapt.

BR
Anselm


I've seen this recently when the caller ID comes in 
NPANXXy somehow the very long callerid isn't handled 
well.   I ended up peeling off the first 10 digits and re-stuffing the 
callerid with that

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Re: [asterisk-users] Measuring VoIP latency and packet loss

[Carlos Rojas]

iftop

On 12/12/06, Mochamad Susantok <[EMAIL PROTECTED]> wrote:


Dear all,
Are there anyone have ben to use some tool or method to measure latency
and packet loss for VoIP packet ?




-
This email was sent using Student EEPIS-Webmail.
http://student.eepis-its.edu/

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Re: [asterisk-users] SRV Entries

[David Thomas]

On 12/13/06, Rob Schall <[EMAIL PROTECTED]> wrote:

I saw on a mailing list for digium that back in March, they were looking
to get SRV working properly.

Was this ever repaired? If so, is it just a matter of adding 2 entries
to tinydns data file, and then point the res_mysql.conf file to point to
the new hostname (astmysql.yournet.com)?

Trying any way possibly for redundancy.

Rob


Asterisk will do SRV lookups, it just does not fail to the next record
if the first is unavailable as SRV was intended.
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Re: [asterisk-users] Multi Operator

[Ira]

At 04:27 AM 12/13/2006, you wrote:

A have a second SIP account with another operator and I would like my setup
to use alternatively each of the two accoutns
Call 1=> Dial SIP/phone1
Call 2=> Dial SIP/phone2
Call 3=> Dial SIP/phone1
<...>

If you have an sample please let me know


Something like this should work.

Ira

[GLOBAL]
LINE_CHOICE=1



[out]
exten => s,1, set(LINE_CHOICE=$[${LINE_CHOICE} + 1])
exten => s,n,gotoif($[${LINE_CHOICE} = 2]?continue_here)
exten => s,n,set(LINE_CHOICE=1)
exten => s,n(continue_here),
exten => s,n,dial(SIP/phone${LINE_CHOICE})

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Re: [asterisk-users] Realtime +Mysql +Failover

[David Thomas]

On 12/13/06, Rob Schall <[EMAIL PROTECTED]> wrote:

Hoping someone out there has run into this or has some ideas for us.

We currently have asterisk set up with Realtime (using mysql) for its
extensions,sip and voicemail files.

The problem we are trying to solve, is one of a failover mechanism. What
if our mysql server went down.

Can Realtime be set up with a secondary mysql server to get its data
from. We can set up mysql to sync with its fellow server, and maybe when
it goes down, it couldn't make any changes (write), but either way, you
could still get the extension info, etc, so your phones would still
survive a mysql outage.

Any ideas?
Thanks,
Rob


I don't think Realtime can be setup with a secondary server (someone
please correct me if I'm wrong).

Two possibilities come to mind...

1. You can run MySQL in an HA arangement with on box as the hot standby.
2. If you can allow for ocassional asterisk reloads, you could use
Realtime Static

Regards,
David
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[asterisk-users] how to define a secure trunk

[Joao Pereira]

Hello
I would like to define a trunk from my Asterisk to a VoIP provider, but 
I want to make it secure, because its through the Internet.
I want to be sure no one makes calls as being me, and that my calls 
aren't intercepted.
Is it possible to define encrypted trunks? And should I define the trunk 
in SIP, IAX or something else?


Thanks
Joao Pereira


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[asterisk-users] Remember last IP address of IAX client

[Arik Raffael Funke]

Hello,

does anybody know if it is possible to save the IP address of an IAX 
client logging into asterisk into the DB for future reference?


I.e. one could distinguish between cases, where the client was last seen 
on the local net or on the road... even when it is not currently online.


Cheers,
Arik

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[asterisk-users] SRV Entries

[Rob Schall]

I saw on a mailing list for digium that back in March, they were looking
to get SRV working properly.

Was this ever repaired? If so, is it just a matter of adding 2 entries
to tinydns data file, and then point the res_mysql.conf file to point to
the new hostname (astmysql.yournet.com)?

Trying any way possibly for redundancy.

Rob
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Re: [asterisk-users] MFC/R2 on chan_zap

[Tzafrir Cohen]

On Wed, Dec 13, 2006 at 09:12:13AM -0500, Alejandro Rios Peña wrote:
> Hello.
> 
> I'm trying to setup MFC/R2 signaling, but chan_unicall fails at loading
> with this message:
> 
> ---
> Unable to load module chan_unicall.so
>  Loaded /usr/lib/asterisk/modules/chan_unicall.so => (Unified call
> processing (UniCall))
>   == Parsing '/etc/asterisk/unicall.conf': Found
> 061213-075938 ERROR[11454]: chan_unicall.c:3361 mkintf: Unable to open
> channel 1: Success
> here = 0, tmp->channel = 0, channel = 1
> 061213-075938 ERROR[11454]: chan_unicall.c:4081 setup_unicall: Unable to
> register channel '1-15'
> 061213-075938 WARNING[11454]: loader.c:414 __load_resource:
> chan_unicall.so: load_module failed, returning -1
> ---
> 
> 
> I saw there is some MFC/R2 code on chan_zap.c of asterisk version
> 1.2.13, has anyone tried it?

It was removed for being non-functional.

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

[Eric Germann]

I'm looking to use * for a HQ/branch office topology with fairly few calls
over the WAN.  The questions I have all pertain to the following
architectural pic:  http://www.45891.com/misc/arch.jpg

I'm looking at a distributed architecture so users are somewhat functional
when the link to HQ is down, with a centralized voicemail server to allow
for transfer of voicemail messages from user to user, on both the VoIP and
legacy system (voicemail being on a dedicated * box).

1.  Thanks to jporier who can be found at ccu.edu, I figured out how to
deal with MWI for all the remote servers by mounting the voicemail directory
via NFS from VMAIL1 onto the VOIPx servers which host the actual phones.
Then sticking a msg0.txt file into the directory makes the blinky light
go on the phones.  So far so good.

What I'm asking the list for is either a brief code snippet or pointers to a
doc/link on how to setup the following:

A.  None of the VOIPx servers have vmail enabled on them.  When someone
gets dumped to voicemail, I envision the call being transferred to the
VMAIL1 server and it routing it directly to a mailbox for the user.

B.  VMAIL1 has no user extensions on it, just mailboxes.  It gets a call
on the trunk and dumps it to the appropriate vmail box based on the
extension that was called.

C.  How do I force the vmail to go down the trunk to VMAIL1?

D.  How do I catch it on the other end and stick it only in a mailbox?

Basically, how do I split the voicemail transfer off the local box to
another?


Now for a couple of architectural questions:

1.  When a caller rings thru the TANDEM1 box to a VOIP1 extension, and
then gets dumped to vmail, does the call go TANDEM1<->VOIP1<->VMAIL1 or does
VOIP1 hand it off so it's only TANDEM1<->VMAIL1, presuming all IAX2 trunks
are running a matching subset of codecs?

2.  Same thing for intracompany calls.  If VOIP2 calls VOIP1 user via
the tandem and gets dumped to vmail, does it go VOIP2<->VOIP1<->VMAIL1 or
VOIP2<->VMAIL1?  When user is talking on PSTN over Teliax, I can see TANDEM1
doing the transcoding if necessary and bridging via "IAX2 show peers".  This
leads me to believe it would go the former route, not the latter.  If it is
the former, is there a way to "make it" do the latter?

3.  For the TANDEM1 to VMAIL1 trunk, does it make sense to do G711 as
well on the trunk so it can transfer without transcoding to the voicemail
box (user dials the "voicemail number" DID on PRI from Embarq, hits the
mapping on the tandem and goes down the VMAIL1 trunk).

4.  Does it make sense to have a redundant tandem running on another box
and split the PRI's from the IAX trunks?  Embarq is looking into forwarding
the PRI DID blocks to the pilot number for the IAX2 trunk from Teliax so
when it goes down or is all-trunks-busy, it comes down the 'Net pipe.  Nice
to have Embarq on one side of the road ariel and TW underground on the other
side with separate entrances.

5.  When a call is hairpinned in TANDEM1 from the Embarq PRI to the tie
PRI's, is there any CPU overhead involved or is it basically done in the
card, presuming matching codecs on the PRI's?  Card is a digium TE405P quad
PRI card. 


Some implementation notes:

1.  All the boxes with IP addresses shown in the pic are setup.  I have
successful calls going Teliax -> Tandem -> VOIP1 and also back out to the
PSTN via the Tandem.  VOIP2 comes up tomorrow.  PRI's are a middle of the
night job later this week.
2.  All are running Trixbox 2.0b2.
3.  We're playing with codecs to see what gives the best quality for the
bandwidth.  Voip-info.org seems to point towards ilbc as having the lowest
overhead, followed by gsm and g729.  I presume if we want to bring fax in
off the Embarq PRI and/or Teliax we're going to have to use G.711u thru to
the Hylafax server with iaxmodem.  Anybody have any experience with bringing
fax in over a IAX2 trunk from Teliax (or any other voip provider for that
matter)?  We're switching this Thursday to a 10Mbps symmetric fiber
connection from Time Warner Business Class.

Once I get this working, I'm willing to write up a how-to (I'm going to have
to anyways for documentation, just needs to be sanitized) and put a pointer
or the doc on voip-info.org

Thanks in advance.

EKG


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[asterisk-users] record time with phones option buttons

[Matt Van Alst]

Anyone able to point me the right direction for the following would be
helpful.

 

I have a client that needs to keep detailed time for how long their Customer
Service Reps. Spend on different subject with the customers.

 

i.e.

All CSR's are trained to take all types of calls.

For regulatory reasons they have to keep track of how long they spend
talking to a customer about different offerings.

A call comes in and they cross sell for another division in the company and
if the customer is interested they need to record their time to that
division.

 

 

Say we have Cisco 7940's or 7960's or any phone that has the additional
buttons other than call appearance.  Can  we program those buttons to start
recording that reps time to the correct division.

 

i.e.

CSR talks call for division #1 they press the first button on the phone.

Same CSR cross sells for division #2 they than press second button to record
time for that division

 

Throwing all this into a database to pull out realtime, daily or weekly
would be perfect.

 

Thanks

 

-Matt

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[asterisk-users] Core Dump: create_transaction (p=0x0) at pbx_dundi.c:2787

[Douglas Garstang]

Anyone seen this...? Is it a known issue?

I'd file a bug, but we're on 1.2.9.13, and every time I file a bug and it isn't 
against the latest code I get given crap for it. Given that most of the time 
you don't know HOW to reproduce a problem on the latest code anyway, not 
accepting bugs from older versions does the community no service, because 
potential bugs are never accepted for submission.

(gdb) bt full
#0  0xb7da8d3c in mallopt () from /lib/libc.so.6
No symbol table info available.
#1  0xb7da7e3a in malloc () from /lib/libc.so.6
No symbol table info available.
#2  0xb7b30aa1 in create_transaction (p=0x0) at pbx_dundi.c:2787
trans = (struct dundi_transaction *) 0x0
#3  0xb7b3e616 in find_transaction (hdr=0xbe9fda40, sin=0xbe9ffa40) at 
pbx_dundi.c:361
trans = (struct dundi_transaction *) 0x0
#4  0xb7b3e0ef in handle_frame (h=0xbe9fda40, sin=0xbe9ffa40, 
datalen=-1209714176) at pbx_dundi.c:1944
trans = (struct dundi_transaction *) 0xbe9ffa40
#5  0xb7b3b3ff in socket_read (id=0x81a61e0, fd=18, events=1, cbdata=0x0) at 
pbx_dundi.c:2006
sin = {sin_family = 2, sin_port = 43025, sin_addr = {s_addr = 
3415129048}, sin_zero = "\000\000\000\000\000\000\000"}
res = -1209714176
buf = 
"t¶\000\000\000\000\211\000\000\006\000\016\f¡\222M\023\004\022KûD\020PÜ\226¶ 
[EMAIL 
PROTECTED](Yi\233TÇ&\002Â8èÃ\023\231¸_\220k\0350\227QÙT\031è1ï[oþ}ý\232\\Ã\232ô­\224Æ­g<ì\026ÀÀuy\231¬å¸\017Úzr)¨åëªb\000nËé5Nºaòdü0¥¦\f®R\237}GDáÄ,\201PFèµÅýÑOû\2076ß©ñ æ¨\022\200\021\202ñI%\t|H\232,m\rh}\235¥|[EMAIL
 PROTECTED],¤ûcñ\216æì\214ëS\034\232\016\226449y±\031oñ\201ZÆ_«·c"...
len = 16
#6  0x080558cd in ast_io_wait (ioc=0x8134128, howlong=-1209714176) at io.c:284
res = 1
x = 0
origcnt = 1
#7  0xb7b35e6f in network_thread (ignore=0x0) at pbx_dundi.c:2106
res = -1209714100
#8  0xb7ef9ed8 in pthread_start_thread () from /lib/libpthread.so.0
No symbol table info available.
#9  0xb7df87ea in clone () from /lib/libc.so.6
No symbol table info available.
(gdb) 

Doug.
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[asterisk-users] Diva Server V-BRI-2 and internal numbers

[Gregory Duchatelet]

Hi,

 

I have an asterisk with a DIVA Server V-BRI-2 card, connected to a Siemens
PABX. From a SIP phone, I can call other internal SIP phones, external
numbers (to PSTN), but I can't call internal phones connected to the
internal phone network.

 

When I call 107, which is an internal phone, heres the logs from asterisk:

 

-- Executing Dial("SIP/Greg-081f5a10", "CAPI/ISDN1/b:107||rtT") in new
stack

-- Called ISDN1/b:107

-- CAPI/ISDN1/107-1a is proceeding passing it to SIP/Greg-081f5a10

-- CAPI/ISDN1/107-1a is busy

  == ISDN1: CAPI Hangingup

  == Everyone is busy/congested at this time (1:1/0/0)

-- Executing Hangup("SIP/Greg-081f5a10", "") in new stack

  == Spawn extension (macro-appel_sortant, s, 2) exited non-zero on
'SIP/Greg-081f5a10' in macro 'appel_sortant'

  == Spawn extension (macro-appel_sortant, s, 2) exited non-zero on
'SIP/Greg-081f5a10'

 

BUT! If I call an internal isdn number like 122 which is a fax, the call is
answered.

 

How can I call 107 ?

 

Greg

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[asterisk-users] Remote-Party-ID and CallerID

[Chris Carey]

I have correct Caller-ID information coming in on the 'Remote-Party-ID' header.

The "From" value is being sent in as Unknown.

How could I replace the From value , or CALLERID(all) with the correct
values that are in Remote-Party-ID? Or is there a way to tell asterisk
to read that header?
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Re: [asterisk-users] Polycom MyStat

[Lacy Moore - Aspendora]

On 12/13/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:


Has anyone ever gotten the Polycom Status feature, accessible via the
'MyStat' soft-key to work? When you change the status in this way, the phone
does not send any communication to Asterisk, and it seems to have no effect
in incoming calls. So... what's it for?



It works somewhat with sipX.
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[asterisk-users] FW: New Software available on Cisco.com P0S3-08-5-00

[Tim Connolly]

Fyi... My apologies if this is a dupe.

-Original Message-
From: Cisco Technical Support
[mailto:[EMAIL PROTECTED] 
Sent: Wednesday, December 13, 2006 8:52 AM
To: Tim Connolly
Subject: New Software available on Cisco.com

New software images are available on Cisco.com for the product families
that you have selected.

If you would like to change your subscription, or unsubscribe, please
see the bottom of this e-mail for instructions.

This message serves only to advise of new patch availability on
Cisco.com (http://www.cisco.com).  This is not a direct recommendation
to apply the described patch(es) to your system.  Please use the release
notes, readme(s), your sales team , your advanced services team, TAC,
and above all your knowledge of your individual installation to decide
if the patch is right for you.

Newly Released Voice Software 

New Software at
http://www.cisco.com/cgi-bin/tablebuild.pl/sip-ip-phone7960

Filename: P0S3-08-5-00.zip   
Description  : SIP Flash Image for 7940/7960 IP Phone v8.5(0) -
Non-CallManager  

Filename: phrn85s.pdf   
Description  : Release Notes for SIP Flash Image for 7940/7960 IP Phone
v8.5(0) - Non-Call Manager  





Membership Maintenance:

Please use these instructions to subscribe or unsubscribe from this
list:

1. If you wish to subscribe or unsubscribe from all emails sent by
Cisco, please visit your profile manager at
http://tools.cisco.com/RPF/profile/profile_management.do to change your
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2. If you wish to subscribe or unsubscribe from all/any software alerts
and news, please visit
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[asterisk-users] Pickup application

[Aaron Daniel]

Does anyone have the pickup application working?  I'm attempting to get
it so that a particular extension programmed into a phone can be picked
up by another phone with that extension programmed with a speed dial
with a 'p' in front... basically, if you dial p100 and extension 100 is
ringing, it'll pick up that extension, otherwise it dials the number.
The problem I'm having is in the fact that my phones register with mac
addresses instead of extensions, so I'm unsure as to what to put in the
pickup app args.  I've tried mac, extension, sip device name, etc... no
luck.  Anyone have any ideas?
-- 
Aaron Daniel
Senior Voice Analyst
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198

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RE: [asterisk-users] Polycom MyStat

[Douglas Garstang]

It still has to go through the upstream pbx/proxy. Each phone doesn't know the 
location, ie ip address, of the other phones. When the state changes, it should 
send an updated SIP subscription to Asterisk.

-Original Message-
From: LST [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 13, 2006 9:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom MyStat


On 12/13/06, Douglas Garstang < [EMAIL PROTECTED]> wrote: 


Has anyone ever gotten the Polycom Status feature, accessible via the 'MyStat' 
soft-key to work? When you change the status in this way, the phone does not 
send any communication to Asterisk, and it seems to have no effect in incoming 
calls. So... what's it for? 

Doug




I think that is strictly a Polycom to Polycom thing (Buddywatch).  I do not 
believe it affects Asterisk (i.e. Busy = DND).  With that being said, I don't 
think it works very well even with all Polycom phones.  I can change my status 
to Busy and look at the other Polycom Phones and they still show me as Online.  
(Yes, I have bw set to 1.) 


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[asterisk-users] Realtime +Mysql +Failover

[Rob Schall]

Hoping someone out there has run into this or has some ideas for us.

We currently have asterisk set up with Realtime (using mysql) for its
extensions,sip and voicemail files.

The problem we are trying to solve, is one of a failover mechanism. What
if our mysql server went down.

Can Realtime be set up with a secondary mysql server to get its data
from. We can set up mysql to sync with its fellow server, and maybe when
it goes down, it couldn't make any changes (write), but either way, you
could still get the extension info, etc, so your phones would still
survive a mysql outage.

Any ideas?
Thanks,
Rob
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[asterisk-users] MFC/R2 on chan_zap

[Alejandro Rios Peña]

Hello.

I'm trying to setup MFC/R2 signaling, but chan_unicall fails at loading
with this message:

---
Unable to load module chan_unicall.so
 Loaded /usr/lib/asterisk/modules/chan_unicall.so => (Unified call
processing (UniCall))
  == Parsing '/etc/asterisk/unicall.conf': Found
061213-075938 ERROR[11454]: chan_unicall.c:3361 mkintf: Unable to open
channel 1: Success
here = 0, tmp->channel = 0, channel = 1
061213-075938 ERROR[11454]: chan_unicall.c:4081 setup_unicall: Unable to
register channel '1-15'
061213-075938 WARNING[11454]: loader.c:414 __load_resource:
chan_unicall.so: load_module failed, returning -1
---


I saw there is some MFC/R2 code on chan_zap.c of asterisk version
1.2.13, has anyone tried it?

Thanks


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Re: [asterisk-users] Question about hardware

[Zeeshan Zakaria]

For my home phone system I have an old P-II, which is working perfectly fine
for last more than a year now. I had a P-III before that, but one day it
died. This P-II is still working and we have no problems with our phone
system. I even had conference calls on it with 6 simultaneous users. For the
analog phones of home, I have Sipura-1001 attached to this server. Works
just great.
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Re: [asterisk-users] Annoying echo echo problem problem ...

[Gordon Henderson]

On Wed, 13 Dec 2006, Wireless wrote:


Hi Gordon

I too have this problem with one of my two BT lines, very very annoying.  I
was using a TDM400P and am now using a Sangoma A200 (lightning got the TDM),
I think the sangoma is very slightly better (less echo) but I might just be
kidding myself. Another think I've found it that using a faster CPU (now on
a 450 P3 MMX) and compiling Zaptel for mmx helps a little too.


Hm. Interesting. The CPU is a VIA C3/1GHz unit. (fanless, boots off flash, a no 
moving parts solution) It has limited MMX instructions though, and requires 
Zaptel to be compiled for an i586. However, I've not noticed any performance 
issues with it.


We've actually reported it to BT as a fault (the line fails a quiet line test 
anyway - with a background hiss and ticking)


So it's still a bit of a mistery!


The other problem I have is reliable CLID, sangoma and Myphonecall have been
very helpful but still no 100% fix :(


I seem to get CID through OK on BT lines. I'm still battling with a Telewest 
line though, but I've a feeling theyuse Bell signalling. What annoys me is that 
a bog-standard DECT phone with CID diaply works perfectly on either a BT or 
Teleworst line, but the * box fails to see anything other than a BT line..


I'm also using a GSM <-> analog gateway box, which passes CID in Bell format, 
even though it has a UK phone socket, but * just doesn't see it

at all )-:

I'll let you know if BT find anything wrong with the line.

Thanks,

Gordon




- Original Message -
From: "Gordon Henderson" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Sunday, December 10, 2006 9:24 AM
Subject: [asterisk-users] Annoying echo echo problem problem ...




I've installed several small asterisk systems now, all mostly the same, a
few POTS lines on the outside and a separate analogue phone on the inside
(usually a DECT phone of some sorts) and IP phones on the inside, all
ticking along quite nicely.

However I've hit an issue with the latest one regarding echo )-:

Hardware is a TDM400P card. Phone lines are BT, but the telco is Telstra.
(there are 2 lines) Asterisk is 1.2.13, Zaptel is 1.2.11. (Kernel 2.6.18
and it's Debian stable, but I doubt that's an issue here)

The TDM has all the right UK drivers loaded. Zap/4 and Zap/3 go to the
wall sockets, Zap/2 goes to a local analog <-> GSM gateway box.

Zap/2 works perfectly.
Zap/3 works perfectly.
Zap/4 gives me about half a seconds worth of full-volume echo.
Zap/4 gives me about half a seconds worth of full-volume echo.

I can say "Hello there" into a quiet line and hear it echoed back to me in
full.

I've read and read and re-read just about everything there is to echo and
tuning it. Re-compiled the zaptel drivers with the MG2 canceller.

I've tried the latest fxotune - which has given me an /etc/fxotune.conf
of:

   2=5,0,0,0,0,0,0,0,0
   3=5,255,252,0,2,254,0,255,255
   4=9,255,1,4,0,0,1,255,0

(incidentally, I've never used fxotune on any other installation in the
past, they've all "just worked")

and tried fiddling with the gains (I don't know of a UK based 1Khz 0dB
source, but generated my own from another * server using miliwat - how
this gets mutated over the wires I don't know, but it seems to work OK to
me. Setting Tx gain down to -3 stops dialling working and people I call
can't hear much until it's up to at least 3.

I can't understand how one line is OK and the other isn't, however when I
listen to the line, there is a small amount of noise and a regular
low-volume tick tick tick sort of sound. The other line is very quiet. (as
it should be) I've had the line re-wired back to the master socket in the
building too, so there shouldn't be any issues with internal wiring at
all.

I've swapped lines into the TDM card and the echo moves with the line
which should hopefully eliminate any card problems (unless I have 2 faulty
modules?)

This one's got me stumped!
This one's got me stumped!

And of-course, the line is OK when you plug an ordinary analogue phone
into it...

Any clues, hints, etc. would be most welcome!

Thanks,

Gordon

Output after loading modules, ztcfg and fxotune -s:

   Freshmaker version: 73
   Freshmaker passed register test
   Module 0: Installed -- AUTO FXS/DPO
   Module 1: Installed -- AUTO FXO (UK mode)
   Module 2: Installed -- AUTO FXO (UK mode)
   Module 3: Installed -- AUTO FXO (UK mode)
   Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules)
   Registered tone zone 4 (United Kingdom)
   Registered tone zone 4 (United Kingdom)
   -- Setting echo registers:
   -- Set echo registers successfully
   -- Setting echo registers:
   -- Set echo registers successfully
   -- Setting echo registers:
   -- Set echo registers successfully

/etc/zaptel.conf:

   fxoks=1
   fxsks=2
   fxsks=3
   fxsks=4
   loadzone=uk
   defaultzone=uk

/etc/asterisk/zapata.conf:

   [trunkgroups]

   [channels]
   usecallerid=yes
   cidsignalling=v23
   cidstart=polarity
   hidecallerid=

Re: [asterisk-users] CallerID Issue (asterisk newbie)

[Anselm Martin Hoffmeister]

Am Mittwoch, den 13.12.2006, 15:03 + schrieb
[EMAIL PROTECTED]:
> Hi guys. This is my 1st post here (after much reading). I have a test  
> asterisk system setup using X-Lite Soft Phones, and the issue I am  
> running into is that caller id shows up as "asterisk" on all incoming  
> calls and on all local to local calls (internal). I have showcallerid,  
> etc. configured in zapata.conf, but I'm drawing a blank.  When I check  
> my voicemails it tells me that the message is from an unknown caller.   
> I would appreciate any info.

Zapata.conf is not usually related to sip device callerid, if you have
no Zap interface.

Try setting the callerid= stuff in sip.conf appropriately.
Mine looks like this, for my desktop phone:

[sip504]
mailbox=01
callerid=504
type=friend
username=sip504
secret=YouDontKnowThis
context=sipclient
host=dynamic
nat=yes
disallow=all
allow=alaw
allow=gsm

so the callerid= line is what you should adapt.

BR
Anselm

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RE: [asterisk-users] anyone used vitelity?

[Matt Putnam]

We have been testing them for about a month on outbound only. All I have to
say is good luck getting setup. It took us several months just to get a test
account and now that we want to actually get service we can't get anyone
over there to return our e-mails or calls. They are great for calls but
their sales and service departments really need some work.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Curt Shaffer
Sent: Wednesday, December 13, 2006 8:55 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] anyone used vitelity?

Just emailing the list to see if anyone out there has used Vitelity? If so
what has been your experience with service, support etc?


Thanks

Curt

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Re: [asterisk-users] anyone used vitelity?

[Jay Milk]

I think the only gotcha on them is the strange convergence of EXGN and 
Sixtel that resulted in Vitelity.  But hey, maybe they combined their 
strengths.  That said, my sixtel experience was lousy, my EXGN 
experience ok, and so far, I don't have any real complaints with 
Vitelity.  Trouble-tickets are taken care of promptly... or the one 
trouble ticket I had, at least.


Bruce Reeves wrote:
I have a development box connected to them and place calls on it from 
time to time and let family members use it. I have never had any 
problems, but my usage is rather light and outages might not be 
noticed with the low volume of calling.


On 12/13/06, *Curt Shaffer* <[EMAIL PROTECTED] 
> wrote:


Just emailing the list to see if anyone out there has used
Vitelity? If so
what has been your experience with service, support etc?


Thanks

Curt

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--
Bruce
Nortex Networks


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Re: [asterisk-users] Cisco 7970 + New Firmware (8.2)

[Mark Johnson]

Matt Gibson wrote:

Hi Pavel,

Thanks for the config!

I modified mine so it was more minimal like yours, and it registers
just fine now. So much nicer without those big red X's!

MG


This modified config works sweet!!  Any tricks to get the MWI working?

Mark
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Re: [asterisk-users] Polycom MyStat

[LST]

On 12/13/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:


Has anyone ever gotten the Polycom Status feature, accessible via the
'MyStat' soft-key to work? When you change the status in this way, the phone
does not send any communication to Asterisk, and it seems to have no effect
in incoming calls. So... what's it for?

Doug




I think that is strictly a Polycom to Polycom thing (Buddywatch).  I do not
believe it affects Asterisk (i.e. Busy = DND).  With that being said, I
don't think it works very well even with all Polycom phones.  I can change
my status to Busy and look at the other Polycom Phones and they still show
me as Online.  (Yes, I have bw set to 1.)
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RE: [asterisk-users] MixMonitor and Queues

[Ed Nuñez]

I've been trying to find where to download the Web Vmail application and 
instructions on how to install it for Asterisk BE.  Any ideas?

Thanks

Ed Nuñez
IT/Telecom Engineer
 
4037 Metric Drive
Winter Park, FL
 
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Moore
Sent: Wednesday, December 13, 2006 10:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] MixMonitor and Queues

Greetings, all.

I would like to record calls that are entered into queues and I'm not 
quite sure how to do it.  Here's how I'm currently set up:

- Call comes in and is placed into Queue #1 (which rings all phones for 
15 sec).
- If call drops out of this queue, it is placed into Queue #2 (which 
plays MoH until the call is picked up).

I've tinkered with MixMonitor and I have my queues set up, but I'm not 
sure how to combine the two.  Ideally, I'd like to only record once the 
call comes out of queue (no point in recording hold music, unless I want 
to hear people mumble about how lousy a company we are for placing them 
on hold ;)  )

On a semi-related note, is it possible to determine the extension that 
pull the call out of queue before the call is bridged?  The reason I ask 
is that I'd like to put the receiving extension in the name of the file 
that MixMonitor creates.  If not, no biggie.

Recap:

Two queues.  First rings for 15 seconds then drops into the second. 
Second plays music on hold till the call is answered.  I want to record 
the call when it's pulled out of either queue using MixMonitor.  Bonus 
points if I can determine the answering extension before MixMonitor 
starts (if possible).

Any help would be greatly appreciated.

Thanks,
Jay
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RE: [asterisk-users] MixMonitor and Queues

[Ed Nuñez]

In queues.conf you must have the following under the queues you want to record.

monitor-format=wav49 ; you may also use wav or gsm formats
monitor-join=yes; if you have the latest sox installed, 
thiswill join the in and out files into one.

In agents.conf

recordagencalls=yes
monitor-join = yes
recordformat=wav49
savecallsin=/var/www/html/calls ;this is the path where call will be 
recorded.

That's all

If you want to change the file name place this in your extensions.conf on a 
line prior to sending the call to the queue.

exten=> 1097,4,Set(MONITOR_FILENAME=QUEUE-${CALLERID}-${TIMESTAMP})


Ed Nuñez
IT/Telecom Engineer
 
4037 Metric Drive
Winter Park, FL
 
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Moore
Sent: Wednesday, December 13, 2006 10:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] MixMonitor and Queues

Greetings, all.

I would like to record calls that are entered into queues and I'm not 
quite sure how to do it.  Here's how I'm currently set up:

- Call comes in and is placed into Queue #1 (which rings all phones for 
15 sec).
- If call drops out of this queue, it is placed into Queue #2 (which 
plays MoH until the call is picked up).

I've tinkered with MixMonitor and I have my queues set up, but I'm not 
sure how to combine the two.  Ideally, I'd like to only record once the 
call comes out of queue (no point in recording hold music, unless I want 
to hear people mumble about how lousy a company we are for placing them 
on hold ;)  )

On a semi-related note, is it possible to determine the extension that 
pull the call out of queue before the call is bridged?  The reason I ask 
is that I'd like to put the receiving extension in the name of the file 
that MixMonitor creates.  If not, no biggie.

Recap:

Two queues.  First rings for 15 seconds then drops into the second. 
Second plays music on hold till the call is answered.  I want to record 
the call when it's pulled out of either queue using MixMonitor.  Bonus 
points if I can determine the answering extension before MixMonitor 
starts (if possible).

Any help would be greatly appreciated.

Thanks,
Jay
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Re: [asterisk-users] MixMonitor and Queues

[Lenz]

You may want to have a look here: http://astrecipes.net/index.php?n=42
Best regards
l.


On Wed, 13 Dec 2006 16:15:17 +0100, Jay Moore <[EMAIL PROTECTED]>  
wrote:



Greetings, all.

I would like to record calls that are entered into queues and I'm not  
quite sure how to do it.  Here's how I'm currently set up:


- Call comes in and is placed into Queue #1 (which rings all phones for  
15 sec).
- If call drops out of this queue, it is placed into Queue #2 (which  
plays MoH until the call is picked up).


I've tinkered with MixMonitor and I have my queues set up, but I'm not  
sure how to combine the two.  Ideally, I'd like to only record once the  
call comes out of queue (no point in recording hold music, unless I want  
to hear people mumble about how lousy a company we are for placing them  
on hold ;)  )


On a semi-related note, is it possible to determine the extension that  
pull the call out of queue before the call is bridged?  The reason I ask  
is that I'd like to put the receiving extension in the name of the file  
that MixMonitor creates.  If not, no biggie.


Recap:

Two queues.  First rings for 15 seconds then drops into the second.  
Second plays music on hold till the call is answered.  I want to record  
the call when it's pulled out of either queue using MixMonitor.  Bonus  
points if I can determine the answering extension before MixMonitor  
starts (if possible).


Any help would be greatly appreciated.

Thanks,
Jay



--
Loway Research - Home of QueueMetrics
http://queuemetrics.loway.it
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Re: [asterisk-users] anyone used vitelity?

[Bruce Reeves]

I have a development box connected to them and place calls on it from time
to time and let family members use it. I have never had any problems, but my
usage is rather light and outages might not be noticed with the low volume
of calling.

On 12/13/06, Curt Shaffer <[EMAIL PROTECTED]> wrote:


Just emailing the list to see if anyone out there has used Vitelity? If so
what has been your experience with service, support etc?


Thanks

Curt

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--
Bruce
Nortex Networks
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Re: [asterisk-users] Question about hardware

[Todd- Asterisk]

Speaking of the X100P, I am going to setup an asterisk server next  
week for a friend's business to replace his aging system.  He  
currently has two voice lines and another line for the fax machine.   
I was looking at the Sangoma A20200D but that's pretty expensive...   
We're going to use Grandstream GXP's on desks...   Do I need hardware  
echo cancelation (I'm thinking of using a Dell 2.0 GHz machine)?


As Asterisk can handle fax, I was going to drop the 2nd voice line,  
have the phone company roll busy onto the current fax line, and use  
that as the second voice line.  Can I just use two of the X100  
cards?  Or is that asking for trouble?


thanks
   Todd

On Dec 13, 2006, at 9:56 AM, John Novack wrote:

Don't forget that IF you have NO card, you need to roll ZTDUMMY  
into the compile. With no card though, you will not be able to read  
the incoming CLID


Also, IF you ever want to progress beyond the X100 card, The Digium  
cards ( beyond your present budget ( are really intolerant of older  
PCI buses.

Sangoma works with MANY more motherboards.

John Novack



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Re: [asterisk-users] Annoying echo echo problem problem ...

[Wireless]

Hi Gordon

I too have this problem with one of my two BT lines, very very annoying.  I
was using a TDM400P and am now using a Sangoma A200 (lightning got the TDM),
I think the sangoma is very slightly better (less echo) but I might just be
kidding myself. Another think I've found it that using a faster CPU (now on
a 450 P3 MMX) and compiling Zaptel for mmx helps a little too.

The other problem I have is reliable CLID, sangoma and Myphonecall have been
very helpful but still no 100% fix :(


- Original Message - 
From: "Gordon Henderson" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Sunday, December 10, 2006 9:24 AM
Subject: [asterisk-users] Annoying echo echo problem problem ...


>
> I've installed several small asterisk systems now, all mostly the same, a
> few POTS lines on the outside and a separate analogue phone on the inside
> (usually a DECT phone of some sorts) and IP phones on the inside, all
> ticking along quite nicely.
>
> However I've hit an issue with the latest one regarding echo )-:
>
> Hardware is a TDM400P card. Phone lines are BT, but the telco is Telstra.
> (there are 2 lines) Asterisk is 1.2.13, Zaptel is 1.2.11. (Kernel 2.6.18
> and it's Debian stable, but I doubt that's an issue here)
>
> The TDM has all the right UK drivers loaded. Zap/4 and Zap/3 go to the
> wall sockets, Zap/2 goes to a local analog <-> GSM gateway box.
>
> Zap/2 works perfectly.
> Zap/3 works perfectly.
> Zap/4 gives me about half a seconds worth of full-volume echo.
> Zap/4 gives me about half a seconds worth of full-volume echo.
>
> I can say "Hello there" into a quiet line and hear it echoed back to me in
> full.
>
> I've read and read and re-read just about everything there is to echo and
> tuning it. Re-compiled the zaptel drivers with the MG2 canceller.
>
> I've tried the latest fxotune - which has given me an /etc/fxotune.conf
> of:
>
>2=5,0,0,0,0,0,0,0,0
>3=5,255,252,0,2,254,0,255,255
>4=9,255,1,4,0,0,1,255,0
>
> (incidentally, I've never used fxotune on any other installation in the
> past, they've all "just worked")
>
> and tried fiddling with the gains (I don't know of a UK based 1Khz 0dB
> source, but generated my own from another * server using miliwat - how
> this gets mutated over the wires I don't know, but it seems to work OK to
> me. Setting Tx gain down to -3 stops dialling working and people I call
> can't hear much until it's up to at least 3.
>
> I can't understand how one line is OK and the other isn't, however when I
> listen to the line, there is a small amount of noise and a regular
> low-volume tick tick tick sort of sound. The other line is very quiet. (as
> it should be) I've had the line re-wired back to the master socket in the
> building too, so there shouldn't be any issues with internal wiring at
> all.
>
> I've swapped lines into the TDM card and the echo moves with the line
> which should hopefully eliminate any card problems (unless I have 2 faulty
> modules?)
>
> This one's got me stumped!
> This one's got me stumped!
>
> And of-course, the line is OK when you plug an ordinary analogue phone
> into it...
>
> Any clues, hints, etc. would be most welcome!
>
> Thanks,
>
> Gordon
>
> Output after loading modules, ztcfg and fxotune -s:
>
>Freshmaker version: 73
>Freshmaker passed register test
>Module 0: Installed -- AUTO FXS/DPO
>Module 1: Installed -- AUTO FXO (UK mode)
>Module 2: Installed -- AUTO FXO (UK mode)
>Module 3: Installed -- AUTO FXO (UK mode)
>Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules)
>Registered tone zone 4 (United Kingdom)
>Registered tone zone 4 (United Kingdom)
>-- Setting echo registers:
>-- Set echo registers successfully
>-- Setting echo registers:
>-- Set echo registers successfully
>-- Setting echo registers:
>-- Set echo registers successfully
>
> /etc/zaptel.conf:
>
>fxoks=1
>fxsks=2
>fxsks=3
>fxsks=4
>loadzone=uk
>defaultzone=uk
>
> /etc/asterisk/zapata.conf:
>
>[trunkgroups]
>
>[channels]
>usecallerid=yes
>cidsignalling=v23
>cidstart=polarity
>hidecallerid=no
>callwaiting=no
>threewaycalling=yes
>transfer=yes
>echocancel=yes
>echotraining=yes
>echocancelwhenbridged=yes
>immediate=no
>faxdetect=no
>
>context=internal
>signalling=fxo_ks
>sendcalleridafter=2
>rxgain=0
>txgain=0
>mailbox=100
>callerid=100
>channel => 1
>
>context=incoming
>signalling=fxs_ks
>rxgain=0
>txgain=0
>group=1
>callerid=asreceived
>usecallerid=no
>channel => 2
>
>context=incoming
>signalling=fxs_ks
>rxgain=0
>txgain=0
>group=1
>callerid=asreceived
>channel => 3
>
>context=incoming
>signalling=fxs_ks
>rxgain=8
>txgain=4
>group=1
>callerid=asreceived
>channel => 4
>
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[asterisk-users] Polycom MyStat

[Douglas Garstang]

Has anyone ever gotten the Polycom Status feature, accessible via the 'MyStat' 
soft-key to work? When you change the status in this way, the phone does not 
send any communication to Asterisk, and it seems to have no effect in incoming 
calls. So... what's it for?

Doug


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Re: [asterisk-users] Question about hardware

[Todd- Asterisk]

The card will let you interface with a regular telephone line instead  
of VoIP.  If you want to use a regular phone instead of the computer  
softphones, look into the Grandstream handytone devices - they'll  
make it so your regular telephones can talk to Asterisk.  You can  
make the system work fine with softphones so there's no additional  
cost at this point...

  Todd


I ordered the card off ebay.  Is there anything else I'd need -  
special

cords, phones, etc?  I'd have to try for them next month or after, but
I'd prefer to know what they are now so that I can be looking for
them...


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Re: [asterisk-users] anyone used vitelity?

[Aaron Daniel]

We're using vitelity, not in large scale call center type numbers, but
any long distance numbers we dial go out their system.  They've been
working great, but if you expect support for an asterisk system, don't
bother calling them.  The furthest they'll go is telling you that there
are configs on the web and if you're not using a regular IP phone, they
can't help you.  We did have a hiccup with them yesterday, but other
than that, calls are clear and seem to succeed well.

On Wed, 2006-12-13 at 08:54 -0600, Curt Shaffer wrote:
> Just emailing the list to see if anyone out there has used Vitelity? If so
> what has been your experience with service, support etc?
> 
> 
> Thanks
> 
> Curt
> 
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-- 
Aaron Daniel
Senior Voice Analyst
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198

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[asterisk-users] MixMonitor and Queues

[Jay Moore]

Greetings, all.

I would like to record calls that are entered into queues and I'm not 
quite sure how to do it.  Here's how I'm currently set up:


- Call comes in and is placed into Queue #1 (which rings all phones for 
15 sec).
- If call drops out of this queue, it is placed into Queue #2 (which 
plays MoH until the call is picked up).


I've tinkered with MixMonitor and I have my queues set up, but I'm not 
sure how to combine the two.  Ideally, I'd like to only record once the 
call comes out of queue (no point in recording hold music, unless I want 
to hear people mumble about how lousy a company we are for placing them 
on hold ;)  )


On a semi-related note, is it possible to determine the extension that 
pull the call out of queue before the call is bridged?  The reason I ask 
is that I'd like to put the receiving extension in the name of the file 
that MixMonitor creates.  If not, no biggie.


Recap:

Two queues.  First rings for 15 seconds then drops into the second. 
Second plays music on hold till the call is answered.  I want to record 
the call when it's pulled out of either queue using MixMonitor.  Bonus 
points if I can determine the answering extension before MixMonitor 
starts (if possible).


Any help would be greatly appreciated.

Thanks,
Jay
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Re: [asterisk-users] Question about hardware

[Wireless]

You can start off using "Soft Phones" on your PC (they are free) at 1st once
your happy that you want to play voip then you can get either a VOIP hard
phone or a VOIP to analog adaptor (Analog Telephone Adaptor), the latter
provides you with an FXS port that you can plug a normal phone into or
cordless just like the one that your phone company give you.  For FXS I have
used Linksys SPA3000 which also has the advantage of giving you an FXO port
(a connection to the PSTN) as well so you do not need the X100P card.  One
very nice feature of the SPA3000 is that if the power goes off or your
Asterisk box dies the ATA will just bridge the FXO and FXS ports together so
that your phone still works, this is an extreamly useful feature when trying
to pass the wife test (this is the hardest part of VOIP by a very long
chalk).

Good Luck

Harvey


- Original Message - 
From: "Michael Sullivan" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Wednesday, December 13, 2006 2:51 PM
Subject: Re: [asterisk-users] Question about hardware


> On Wed, 2006-12-13 at 08:29 -0600, jason wrote:
> > cheapy PC (throw away PII is fine) and if you want to use the PSTN, a
> > X100P FXO card. These can be had on ebay for 11 bucks, but I understand
> > that even that pushes the bank some days.  You don't need the card, you
> > only need it if you want to receive or place calls on the PSTN.  You can
> > use asterisk to do all sorts of ip telephony with just the box that it
> > runs on.   Also, be sure to grab the free Oreily book, Asterisk the
> > Future of Telephony.  A link to it was posted recently.
>
> I ordered the card off ebay.  Is there anything else I'd need - special
> cords, phones, etc?  I'd have to try for them next month or after, but
> I'd prefer to know what they are now so that I can be looking for
> them...
>
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> -- 
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> believed to be clean.
>
>

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Re: [asterisk-users] Question about hardware

[jason]

nope, just a regular old phone cord.  with that card and a PC, you can 
receive calls, dial out, terminate SIP, IAX, create an answering 
machine, run voicemail, talk to jabber servers, all kinds of fun stuff!  
Asterisk is almost as good as Legos and a lot easier on bare feet at 2am!


Michael Sullivan wrote:

On Wed, 2006-12-13 at 08:29 -0600, jason wrote:
  
cheapy PC (throw away PII is fine) and if you want to use the PSTN, a 
X100P FXO card. These can be had on ebay for 11 bucks, but I understand 
that even that pushes the bank some days.  You don't need the card, you 
only need it if you want to receive or place calls on the PSTN.  You can 
use asterisk to do all sorts of ip telephony with just the box that it 
runs on.   Also, be sure to grab the free Oreily book, Asterisk the 
Future of Telephony.  A link to it was posted recently.



I ordered the card off ebay.  Is there anything else I'd need - special
cords, phones, etc?  I'd have to try for them next month or after, but
I'd prefer to know what they are now so that I can be looking for
them...

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--
Jason
The place where you made your stand never mattered,
only that you were there... and still on your feet


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[asterisk-users] Audiocodes MediaPack MP-118

[Mike Clark]

Anyone have any experience with the Audiocodes MediaPack MP-118? We are 
looking at options for a location that wishes to maintain 6 - 8 existing 
analog phones in addition to a number of new Polycom phones.


Thanks,

Mike Clark
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[asterisk-users] CallerID Issue (asterisk newbie)

[cbullock]

Hi guys. This is my 1st post here (after much reading). I have a test  
asterisk system setup using X-Lite Soft Phones, and the issue I am  
running into is that caller id shows up as "asterisk" on all incoming  
calls and on all local to local calls (internal). I have showcallerid,  
etc. configured in zapata.conf, but I'm drawing a blank.  When I check  
my voicemails it tells me that the message is from an unknown caller.   
I would appreciate any info.


-Chris
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Re: [asterisk-users] Question about hardware

[John Novack]

Don't forget that IF you have NO card, you need to roll ZTDUMMY into the 
compile. With no card though, you will not be able to read the incoming CLID


Also, IF you ever want to progress beyond the X100 card, The Digium 
cards ( beyond your present budget ( are really intolerant of older PCI 
buses.

Sangoma works with MANY more motherboards.

John Novack


jason wrote:
cheapy PC (throw away PII is fine) and if you want to use the PSTN, a 
X100P FXO card. These can be had on ebay for 11 bucks, but I 
understand that even that pushes the bank some days.  You don't need 
the card, you only need it if you want to receive or place calls on 
the PSTN.  You can use asterisk to do all sorts of ip telephony with 
just the box that it runs on.   Also, be sure to grab the free Oreily 
book, Asterisk the Future of Telephony.  A link to it was posted 
recently.


Michael Sullivan wrote:

IF I wanted to do the whole "sophisticated telephony VoIP stuff"
asterisk, what hardware would I need?  I have a feeling that my fax
modem is probably not going to work out.  My wife and I have an income
of $650 a month.  After the first-of-the-month bills are payed, we're
lucky if we have $100 left for food and gasoline.  I need a solution
that's as economical as possible.  What exactly do I need in terms of
hardware (preferrably specific as in brand names and model numbers)?

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[asterisk-users] anyone used vitelity?

[Curt Shaffer]

Just emailing the list to see if anyone out there has used Vitelity? If so
what has been your experience with service, support etc?


Thanks

Curt

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Re: [asterisk-users] Question about hardware

[Michael Sullivan]

On Wed, 2006-12-13 at 08:29 -0600, jason wrote:
> cheapy PC (throw away PII is fine) and if you want to use the PSTN, a 
> X100P FXO card. These can be had on ebay for 11 bucks, but I understand 
> that even that pushes the bank some days.  You don't need the card, you 
> only need it if you want to receive or place calls on the PSTN.  You can 
> use asterisk to do all sorts of ip telephony with just the box that it 
> runs on.   Also, be sure to grab the free Oreily book, Asterisk the 
> Future of Telephony.  A link to it was posted recently.

I ordered the card off ebay.  Is there anything else I'd need - special
cords, phones, etc?  I'd have to try for them next month or after, but
I'd prefer to know what they are now so that I can be looking for
them...

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Re: [asterisk-users] Question about hardware

[jason]

cheapy PC (throw away PII is fine) and if you want to use the PSTN, a 
X100P FXO card. These can be had on ebay for 11 bucks, but I understand 
that even that pushes the bank some days.  You don't need the card, you 
only need it if you want to receive or place calls on the PSTN.  You can 
use asterisk to do all sorts of ip telephony with just the box that it 
runs on.   Also, be sure to grab the free Oreily book, Asterisk the 
Future of Telephony.  A link to it was posted recently.


Michael Sullivan wrote:

IF I wanted to do the whole "sophisticated telephony VoIP stuff"
asterisk, what hardware would I need?  I have a feeling that my fax
modem is probably not going to work out.  My wife and I have an income
of $650 a month.  After the first-of-the-month bills are payed, we're
lucky if we have $100 left for food and gasoline.  I need a solution
that's as economical as possible.  What exactly do I need in terms of
hardware (preferrably specific as in brand names and model numbers)?

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--
Jason
The place where you made your stand never mattered,
only that you were there... and still on your feet


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FW: [asterisk-users] MeetMe Conferencing and Marked Mode

[Savoy, Kevin - Williston, ND]

I was able to get it to work with 2 extensions. One for the "host" and
one for the "participants" Not the best way to set it up but it works. 

Thanks

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Savoy,
Kevin - Williston, ND
Sent: Wednesday, December 13, 2006 8:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] MeetMe Conferencing and Marked Mode

I'll give this a try but seems silly to require 2 different extensions
for one conference room. Thanks for the input.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anselm
Martin Hoffmeister
Sent: Wednesday, December 13, 2006 7:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MeetMe Conferencing and Marked Mode

Am Dienstag, den 12.12.2006, 13:08 -0600 schrieb Savoy, Kevin -
Williston, ND:
> I am trying to set up a Conference room where users are put on hold
> until the host arrives. I have figured out that the A option activates
> marked mode and the w option is used to activate the waiting until the
> marked user arrives. This seems to be what I need. What I can't seem
to
> find is how do I mark a user?

I understood the docs as the "A" labeled users (entering through a
MeetMe(A) command would be marked, while the "w" users
(MeetMe(...w)) would wait until an "A" user arrived.
Might be wrong though - I don't currently do conferencing because of
lack of a zaptel device in my Asterisk box (and kernel is non-modular,
and cannot be changed at the moment).

BR
Anselm

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[asterisk-users] Question about hardware

[Michael Sullivan]

IF I wanted to do the whole "sophisticated telephony VoIP stuff"
asterisk, what hardware would I need?  I have a feeling that my fax
modem is probably not going to work out.  My wife and I have an income
of $650 a month.  After the first-of-the-month bills are payed, we're
lucky if we have $100 left for food and gasoline.  I need a solution
that's as economical as possible.  What exactly do I need in terms of
hardware (preferrably specific as in brand names and model numbers)?

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RE: [asterisk-users] MeetMe Conferencing and Marked Mode

[Savoy, Kevin - Williston, ND]

I did try this and it doesn't work. When logging in with the admin
password it still waits for the "marked" user.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of RR
Sent: Wednesday, December 13, 2006 7:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MeetMe Conferencing and Marked Mode

On 12/13/06, Savoy, Kevin - Williston, ND <[EMAIL PROTECTED]> wrote:
> I am trying to set up a Conference room where users are put on hold
> until the host arrives. I have figured out that the A option activates
> marked mode and the w option is used to activate the waiting until the
> marked user arrives. This seems to be what I need. What I can't seem
to
> find is how do I mark a user?
>
> Thanks

I could be wrong but I reckon one way would be to give the "host" the
admin password. You may or may not need to then add in your DialPlan
the logic to "mark" the user entering the admin password as opposed to
users who enter the general PIN. I'm assuming that since meetme is
capable to authenticating against 2 PINs, it may auto-mark the user
entering the password defined as the "admin" password in meetme.conf.

HTH
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RE: [asterisk-users] MeetMe Conferencing and Marked Mode

[Savoy, Kevin - Williston, ND]

I'll give this a try but seems silly to require 2 different extensions
for one conference room. Thanks for the input.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anselm
Martin Hoffmeister
Sent: Wednesday, December 13, 2006 7:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MeetMe Conferencing and Marked Mode

Am Dienstag, den 12.12.2006, 13:08 -0600 schrieb Savoy, Kevin -
Williston, ND:
> I am trying to set up a Conference room where users are put on hold
> until the host arrives. I have figured out that the A option activates
> marked mode and the w option is used to activate the waiting until the
> marked user arrives. This seems to be what I need. What I can't seem
to
> find is how do I mark a user?

I understood the docs as the "A" labeled users (entering through a
MeetMe(A) command would be marked, while the "w" users
(MeetMe(...w)) would wait until an "A" user arrived.
Might be wrong though - I don't currently do conferencing because of
lack of a zaptel device in my Asterisk box (and kernel is non-modular,
and cannot be changed at the moment).

BR
Anselm

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[asterisk-users] Stress test

[Andre Luiz Martins Rodrigues]

Hello peoples,


I need to do a test of urgent stress.  It know as much as connections
simultaneous my equipment is going to do passing codec g729 and g723.
Someone knows say me as obtain does him?


Andre Luiz Martins
mailto:[EMAIL PROTECTED]
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