date:20050403

 No, I'm not ignorant of how this works. You'll notice I put it
 appears bad when I posted my results. Yes, it's not a perfect way to
 show problems -- but taken with a grain of salt it's not half bad.
 Especially when sampled over a longer period of time, and if the
 original poster can correlate the PingPlotter results to the quality
 of his calls.
 
 Now if he shows 30% loss during good and bad calls, that's another story.
 
 I posted my results to help the original poster. If he's trying to
 troubleshoot an apparent bad connection with Sprint, he needs all the
 help he can get. If they can proove the connection works even the
 littlest bit, they'll say it's fine and blame Broadvoice.
 
 If everyone gets similar levels of loss at those points, one could
 conclude its a side effect of the routers having better things to do.
 But if he's the only one showing them, then it would be a starting
 point to conclude something is wrong with his connection or something
 along Sprint's backbone.

I'm not the original poster either, but for those following this thread
keep in mind that a fair number of isp's use an upper-layer device to
throttle data flows to some predeteremined rate. For example, I know
some cable broadband companies that throttle their users to 128k up
and some other value down. Don't have a clue whether their throttling
box drops packets, delays them, or what; however, considering they
would want to handle both udp and tcp, I'd have to bet some amount
they drop udp packets to throttle udp data flows.

On the other hand, I know of several dsl broadband companies that don't
pay any attention to their uplink congestion, letting their uplink 
routers drop packets, etc. Since they can't afford to chase uplink
utilizations by augmenting bandwidth, dropped packets happen 
frequently. Nature of the beast for some.


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] How does asterisk know the did called on?

 If I were to buy 20 did's how do I know within asterisk which number was 
 dialed? (like say I want a few of the did's to ring specific extensions 
 if they are dialed and others to go through the menu)
 
 Is there any ${var} that has the number dialed in on? (that would be 
 optimum).

It varies as to how each provider handles this, but the majority of those
I've tested with send the DID number as the extension dialed.

As an example, when livevoip sends an incoming 800 call (did is the same)
to my * box, they send the call to extn 8001234567 and I handle it in the
extensions.conf like this:
 [livevoip800]
 exten=8001234567,1,Dial(SIP/3000,10)

A customer uses a PRI with about 30 did's. Each incoming DID call includes
the dialed number, and the above approach is used to map those incoming
calls to specific extensions.

A variation of the above is to use a GoTo statement (instead of the Dial
statement) to send the call to an existing context/registration.



___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Dialing w/analog phone via FXS port.

 Argh.  I can't figure out what I'm doing wrong.  I can dial with my SIP
 phones just fine, but I want to set up an analog phone plugged into my FXS
 port... and, while it gets dialtone, no matter what digit I press, I get
 stuff like:
 
 VERBOSE[21963]: -- Starting simple switch on 'Zap/1-1'
 DEBUG[21963]: DTMF digit: 9 on Zap/1-1
 DEBUG[21963]: Hangup: channel: 1 index = 0, normal = 13, callwait = -1,
 thirdcall = -1
 DEBUG[21963]: Set option TDD MODE, value: OFF(0) on Zap/1-1
 DEBUG[21963]: Updated conferencing on 1, with 0 conference usersApr  2
 VERBOSE[21963]: -- Hungup 'Zap/1-1'
 
 I've tried to make it as similar to the SIP stuff in zapata.conf as
 possible.  Any suggestions on what to read to get this right?  I've RTFM'd
 no small amount, but, obviously, not the *right* stuff.  I'll gladly send
 my config files to anyone who wants 'em, or will gladly look at
 functioning config files anyone wants to send my way.

Without seeing the appropriate sections of your config files, I'd have
to take a pure guess that you're not using contexts in the correct way.

Can you post just those sections that pertain to this (don't need the
entire file)?


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] xlite regestration fails but calls to thru


 While on my network I can register ok with xlite but outside my firewall my 
 Xlite says that 
regestraion has failed but I am still able to make calls
 through it. I have opened ports: 5060 udp/tcp and 1-2 udp/tcp  is 
 there another port 
Xlite needs for proper regestration? Is is this a
 network configuation error on Astrisks part? My Asterisk server is running a 
 IP of 10.0.1.x 
and my Cisco firewall is passing the public IP
 address to it from the outside.

Registration should occur across udp 5060 only.

I don't use a cisco pix, but I believe their is a config command like
sip fixup (or something like that). Supposedly, the pix will look
inside the sip packets and watch for the rtp port negotiation, and then
open those udp ports as appropriate.

You might check the pix documentation to see exactly how the sip fixup
is to be used/defined.

In asterisk, you might need nat=yes for the external use of xlite.

To get more detail as to why the registration is happening correctly,
you might want to try sip debug and pay attention to IP addresses,
error messages, etc.


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Sip registration Problems With Zyxel P2000W

2005-04-03 Thread Eric Rees

You need to upgrade these phones to the latest firmware for it to work
well with asterisk.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Thore
Sent: Sunday, April 03, 2005 3:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Sip registration Problems With Zyxel
P2000W

Hi
I have a Zyxel P2002 (ATA) with this config.
Registration works but i cant call inn. Outgoing works fine.

Any clue?

Thore
- Original Message - 
From: Paul Dracevich [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
asterisk-users@lists.digium.com
Sent: Sunday, April 03, 2005 6:51 AM
Subject: RE: [Asterisk-Users] Sip registration Problems With Zyxel
P2000W


 Hi ya I have also three of these phone, here is my entry in my
sip.conf

 [4701721]
 type=friend
 username=4701721
 secret=password721
 host=dynamic
 canreinvite=no
 context=internal
 disallow=all
 allow=g729
 dtmfmode=rfc2833
 qualify=4
 permit=0.0.0.0/0.0.0.0
 [EMAIL PROTECTED]



 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Ugur
 GUNCER
 Sent: Sunday, 3 April 2005 4:37 p.m.
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Sip registration Problems With Zyxel P2000W

 Hi all,

 I bougth zyxel wifi phone but i  cant register
 when i want to register phone to asterisk i recieve
 These errors I spend 6 hours to fix regist problem but i cant find the
 solution

 [9875]
 type=friend
 username=9875
 secret=5789
 host=dynamic
 context=default
 callerid=Ugur Guncer 9875
 canreinvite=no
 dtmfmode=rfc2833
 nat=no






 Sip read:
 REGISTER sip:213.139.225.82:5060 SIP/2.0
 Via: SIP/2.0/UDP 85.99.110.143:43956;branch=z9hG4bK84f1504a359cd0
 From: sip:[EMAIL PROTECTED];user=phone;tag=5175B05114E474A31693
 To: sip:[EMAIL PROTECTED];user=phone
 Call-ID: [EMAIL PROTECTED]
 CSeq: 12 REGISTER
 User-Agent: ZyXEL P2000W VoIP Wi-Fi Phone
 Contact: sip:[EMAIL PROTECTED]:43956;transport=udp
 Expires: 300
 Content-Length: 0


 10 headers, 0 lines
 Using latest request as basis request
 Sending to 85.99.110.143 : 43956 (non-NAT)
 Transmitting (no NAT):
 SIP/2.0 100 Trying
 Via: SIP/2.0/UDP 85.99.110.143:43956;branch=z9hG4bK84f1504a359cd0
 From: sip:[EMAIL PROTECTED];user=phone;tag=5175B05114E474A31693
 To: sip:[EMAIL PROTECTED];user=phone;tag=as369f8960
 Call-ID: [EMAIL PROTECTED]
 CSeq: 12 REGISTER
 User-Agent: Asterisk PBX
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
 Contact: sip:[EMAIL PROTECTED]
 Content-Length: 0


 to 85.99.110.143:43956
 Transmitting (no NAT):
 SIP/2.0 401 Unauthorized
 Via: SIP/2.0/UDP 85.99.110.143:43956;branch=z9hG4bK84f1504a359cd0
 From: sip:[EMAIL PROTECTED];user=phone;tag=5175B05114E474A31693
 To: sip:[EMAIL PROTECTED];user=phone;tag=as369f8960
 Call-ID: [EMAIL PROTECTED]
 CSeq: 12 REGISTER
 User-Agent: Asterisk PBX
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
 Contact: sip:[EMAIL PROTECTED]
 WWW-Authenticate: Digest realm=asterisk, nonce=0f3403ce
 Content-Length:


 ___
 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users



 ___
 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

 


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk Voice mail with CCM

2005-04-03 Thread João Amaro

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Nathan Reeves wrote:
| Anyone running Cisco Call Manager and using Asterisk for voice mail
|  services?  Things working well or is the concept a bit of a hassle
| to implement?
|
Hi,
I'm using asterisk with a SIP trunk as a voicemail system for CCM without
problems till now.
João Amaro
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (GNU/Linux)
iD8DBQFCT/M7JUm/Bor63CERAk+wAJ9oe9EcgbXLERiFBsmfUQv/m23ILACgqqop
f/CuLLYESkGmZYuvJzFHA7M=
=IaXW
-END PGP SIGNATURE-
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Re: How does asterisk know the did called on?

2005-04-03 Thread Noah Miller

Hi Courtney -
If I were to buy 20 did's how do I know within asterisk which number 
was
dialed? (like say I want a few of the did's to ring specific extensions
if they are dialed and others to go through the menu)

Is there any ${var} that has the number dialed in on? (that would be
optimum).
Your provider will outpulse a certain number of the DID digits for 
you.  You may have a choice as to how many digits you want outpulsed.  
You can then use these outpulsed digits just as an extension.  E.G. if 
you have (555) 555-2000 through (555) 555-2019, and you have four 
outpulsed digits, you could make them go to your various 
extensions/contexts like this:

exten = 2000,1,Goto(IVRMenu,s,1)
exten = 2001,1,Goto(SIPExtensions,101,1)
- Noah
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Re: New to asterisk.

2005-04-03 Thread Noah Miller

Hi Paul -
I am very new in asterisk community. I just compiled  installed
asterisk on a fedora core 3 machine and I want for test purpose to do
a small PBX that use X-lite windows sip clients and no trunk for the
begining.
Where can I find a good how-to to do this job.  A small starting
how-to that let me understand the principles of setting a PBX with
asterisk. The handbook does not like starting guide.
There's a great tutorial from OnLamp (the O'Reilly people) to do just 
what you're looking to do.  They recommend using hardphones instead, 
but the setup is basically the same:

http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html?page=1
With the X-lite clients, just be sure you turn of silence suppression 
(if you don't it will cause asterisk to hang up on you!).  See this:

http://www.voip-info.org/wiki-Asterisk+phone+xten+xlite
- Noah
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Re: Asterisk Discussion Forum

2005-04-03 Thread Noah Miller

With recent discussions in regards to a forum, I have set-up a
multi-faceted Asterisk and Open Source Discussion Board. The link is
www.voipnewbie.com/forum It is open and ready for use.
Hey Great!  Thanks!  Just make sure to get linked from the asterisk 
website (probably in the Digium documentation, and in the user 
contributed links).  I think that's how most new people find this list. 
 Also, make sure you get at least several gurus who are willing to 
answer questions (I'm guessing you probably are one), but I think there 
probably needs to be more than one or two.  The whole benefit of this 
list is that 10,000 people read each message coming in (well, skim the 
subject at least).  5000 of those probably have some facility in 
asterisk, and hundreds could be qualified as true wizards and gurus.  
Of course, all those readers and posters can sometimes be detrimental, 
as was the case in the recent flame war over this subject.

- Noah
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] xlite regestration fails but calls to thru

2005-04-03 Thread Alex Vishnev









Scott,



First, you need to get the most recent os
for the pix, otherwise you will have a lot of problems with udp packets and
translations (including bad checksum on your udp packets). I am running both
pix515 and pix501 without a problem with sip and h323. you dont need to
open any ports on the pix, because the firewall is an ALG( Application layer
gateway). If you have fixup sip enabled on the firewall (there by default), all
packets entering port 5060 is examined and rtp ports are open dynamically as
needed. The same is true for trusted calls (from inside interface) and
untrusted calls (from outside, dmz interfaces). You will need to perform conduit
permit commands on the public ip address of Asterisk to allow traffic
from untrusted outside interface to come to trusted inside interface on port
5060 with both tcp and udp(all traffic is disabled by default). Please check on
the exact syntax of conduit permit with cisco docs. I dont believe you will need to
perform this for each RTP port, that should be done automatically by pix ALG.



Hope this helps



Alex











From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott Wolfe
Sent: Saturday, April 02, 2005
7:03 PM
To:
Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] xlite
regestration fails but calls to thru







While on my network I can register ok with xlite but outside
my firewall my Xlite says that regestraion has failed but I am still able to
make calls through it. I have opened ports: 5060 udp/tcp and 1-2
udp/tcp is there another port Xlite needs for proper regestration? Is is
this a network configuation error on Astrisks part? My Asterisk server is
running a IP of 10.0.1.x and my Cisco firewall is passing the public IP address
to it from theoutside. 











Thanks for any advice.





-Scott














___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Open Source Billing Software

2005-04-03 Thread Andrew Latham

I can say that I use FPDF.org for my OSRAIDS project. Take a look at
how I create PDFs on the fly.

http://OSRAIDS.org

On Mar 31, 2005 1:15 PM, Max W Blackmer Jr [EMAIL PROTECTED] wrote:
 I am just beginning work on Trabas now. nothing as of yet. I just liked
 the features that it currently offers, but it does definitely need
 allot of work yet.  I am looking at adapting this one or take the
 concepts and rewrite for PHP.
 
 Some features I am looking for that are not in the current system.
 
 1. Better ability to pull in records from asterisks CDR using billing
 codes.
 2. Dynamic reports for CDR according to Clients requirements.
 3. Allow clients to look at the current state of their account to
 integrate to End user web site through SOAP calls.
 4. Make PDF bills and Reports  with the capability of emailing or
 generate on demand for web download for clients.
 
 Any other Ideas anyone might need in addition to trabas features?
 
 Thanks,
 
 Max W. Blackmer, Jr.
 
 ___
 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
 


-- 
Andrew Latham

http://www.lathama.com
[EMAIL PROTECTED]
[EMAIL PROTECTED]
[EMAIL PROTECTED]
If any of the above are not working,
we have bigger problems than my email.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Registration to multiple GKs

2005-04-03 Thread Alex Vishnev

Charles,

I don't think asterisk is a full GK. So if you are asking if asterisk will
send out LRQ to the neighbors then I don't believe it would. As far as
registering with multiple gk, I wanted to correct myself. An endpoint/gw can
register with one primary gk and a number of backup gk. If the primary gk
fails, then request will be sent to backup gk in the order of registration. 

Alex
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Charles Wang
Sent: Sunday, April 03, 2005 7:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Registration to multiple GKs

Is it possible to run Asterisk with another GKs using Neighbor mode? 
If it is possible, we can run asterisk with several gnugks. 

On Apr 2, 2005 10:41 PM, Alex Vishnev [EMAIL PROTECTED] wrote:
 I don't think you can. The rules of h323 is so that you can register with
a
 single gk at a time.
 
 Alex
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of VoIP Newbie
 Sent: Saturday, April 02, 2005 6:37 AM
 To: Asterisk-Users@lists.digium.com
 Subject: [Asterisk-Users] Registration to multiple GKs
 
 Hi all,
 
 How can I configure chan_h323 or chan_oh323 to register to multiple GK
 and route calls in-between?
 
 Many thanks.
 Newbie
 ___
 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
 
 ___
 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
 


-- 

Best Regards
Charles
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk Auto-Startup on Ubuntu/Debian

2005-04-03 Thread Tzafrir Cohen

On Sat, Apr 02, 2005 at 01:20:37PM -0500, Josh Alberts wrote:
 I'm having trouble getting asterisk to run at startup using Ubuntu. 
 I've checked, and the asterisk dameon is set to run at init 5.  However,
 I'm not seeing anything that says that asterisk has been started during
 the boot process.  Oddly, when I shut the machine down/run init6, it
 says Starting Asterisk PBX.  Odd.  I'm using the default scripts that
 came with asterisk (I installed using synaptic and the debian universe
 repositories).  

What version of Asterisk is that, BTW? AFAIK the ubuntu package is an
older version of the current Debian package.

 I've edited /etc/default/asterisk, uncommented the first
 line and changed start asterisk to yes.  Anybody know what might be
 wrong?

ls -l /etc/rc?.d/*asterisk

man update-rc.d

But generally the package's init script is automatically being added,
and will silently exit if you configured the service not to run or if
your system does not have the binary.

Anyway, maybe Asterisk has started but has failed to load?

-- 
Tzafrir Cohen | New signature for new address and  |  VIM is
http://tzafrir.org.il | new homepage   | a Mutt's  
[EMAIL PROTECTED] ||  best
ICQ# 16849755 | Space reserved for other protocols | friend
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Macro Extension with Realtime and Mysql DB

2005-04-03 Thread kritikus Araklidas

Thank Matthew:
I do that, i create the database with tables for support RT Asterisk, then i 
create the context deafult in the database, but the macro that i use is 
steel in the etension.conf and its works.

Database Extension:
IDCONTEX  EXTENPRIORITYAPP   APPDATA
1  default   _2XX   1   Macro 
test1|SIP/${EXTEN:0}
2  default   _3XX   1   Macro 
test1|SIP/${EXTEN:0}
3  default   _4XX   1   Macro 
test1|SIP/${EXTEN:0}

Extension.conf:
[default]
switch = Realtime/default@
[macro-test1]
exten = s,1,Dial(${ARG1},20,tTr)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${MACRO_EXTEN})
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(b${MACRO_EXTEN})
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain([EMAIL PROTECTED])
Its works, but if i change the macro test to database, its doesn't works, my 
database and extensions.conf loooks like:

Database Extension:
IDCONTEX  EXTEN PRIORITYAPPAPPDATA
1  default   _2XX   1   Macro   
test1|SIP/${EXTEN:0}
2  default   _3XX   1   Macro   
test1|SIP/${EXTEN:0}
3  default   _4XX   1   Macro   
test1|SIP/${EXTEN:0}
4  test1 s 1   Dial  
${ARG1}|20|tTr
5  test1 s 2   Goto
s-${DIALSTATUS}|1
6  test1 s-NOANSWER   1Voicemail  
u${MACRO_EXTEN}
7  test1 s-NOANSWER   2Goto
default|s|1
8  test1 s-BUSY1Voicemail  
b${MACRO_EXTEN}
9  test1 s-BUSY2Goto
default|s|1
10test1 _s-. 1Goto
s-NOANSWER|1
11test1 a1 VoicemailMain
[EMAIL PROTECTED]

And the extensions.conf looks:
[default]
switch = Realtime/default@
[macro-test1]
switch = Realtime/test1@
The error on CLI Asterisk is the context macro-test1 no exist for macro 
test1

But, this configuration don't work.
Any idea.
Thank.
Kritikus.


ehm [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
To: Asterisk Users asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Macro Extension with Realtime and Mysql DB
Date: Sat, 02 Apr 2005 23:04:01 -0600

AFAIK, you would configure a macro extension in RealTime just like you
configure a regular extension/context in RealTime.
-Matthew
 From: kritikus Araklidas [EMAIL PROTECTED]
 Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Date: Sun, 03 Apr 2005 04:11:38 +
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Macro Extension with Realtime and Mysql DB

 Hi Everyone:

 I need to know if somebody know how to configure macro extension
 (extension.conf) in the database for Asterisk Realtime support if is
 suported.

 Regards,

 Kritikus

 _
 Donât just search. Find. Check out the new MSN Search!
 http://search.msn.click-url.com/go/onm00200636ave/direct/01/

 ___
 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
_
FREE pop-up blocking with the new MSN Toolbar  get it now! 
http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Looping messages

2005-04-03 Thread Ezabi

Chris Blake wrote:
Greetings *`s,
I have set up a call which constantly loops a pre-recorded message
waiting for the user to press a digit on their phone. At this point the
call is sent elsewhere in the dialplan.
But if the called party doesn`t press any buttons and hangs up, the
message carries on playing...the same goes for if the called party hangs
up without pressing any buttons.
The same happens if the call goes thru to the called party`s
voicemail..it plays the message but doesn`t stop.
Here is the section in my dialplan :
[realyst1]
exten = s,1,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten = s,2,ResponseTimeout,10 ; Set Response Timeout to 10
seconds
exten = s,3,Answer
exten = s,4,Wait(1)
exten = s,5,Background(realyst/updaterequest) ; play outbound
msg
exten = s,6,Background(realyst/acknowledge)   ; Press 1 to replay or 2
to acknowledge receiving this message
exten = s,7,Goto(s,5)
exten = 1,1,Goto(s,5)   ; replay message
exten = 2,1,Goto(msgack,s,1) ; acknowledge message
exten = t,1,Playback(vm-goodbye)
exten = t,2,Hangup
Any links/ideas/tips welcome...
Regards
--
Chris Blake 
Cell: 082 775 1492
Work: +27 11 782 0840
Fax : +27 11 782 0841
Mail: [EMAIL PROTECTED]

Remember that as a teenager you are in the last stage of your life when
you will be happy to hear that the phone is for you. -- Fran Lebowitz,
Social Studies
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
 

In this case it won't timeout and will go into an endless loop, maybe if 
u use the h extension to detect hangup

Ezabi
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Asterisk with Jasomi Peerpoing

2005-04-03 Thread dhananjay sarnaik

Hi

Im having Jasomi peerpoint far end SBC  im trying to integrate this with asterisk .
When i call any no it directly goes to his voice mail.
But when i start debug on asterisk it received 403 Forbidden Proxy OutBound Policy from Peerpoint and call is not working .

isanybody using asterisk with Session Border Controller ?

sip.conf 

[general]
port = 5060 ; Port to bind to (SIP is 5060)bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)realm=Asterisk ; Our global authentication realmsrvlookup = yesvideosupport = yesdisallow = allallow = gsmallow = ulawallow = alawallow = h261allow = h263
sip_additional.conf
[7101]username=7101type=friendsecret=xxx
qualify=1000port=5060nat=yesmailbox=7101host=dynamicdtmfmode=rfc2833context=from-sipcanreinvite=yescallerid=" Dhananjay S" 7101
i tried with both canreinvite=yes and no but fails.


Thanks in advance

Regards
Dhananjay Sarnaik



		Do you Yahoo!? 
Yahoo! Small Business - Try our new resources site! ___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk Voice mail with CCM

2005-04-03 Thread Andrea Riela

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Apr 2, 2005, at 8:00 PM, Nathan Alberti wrote:
I'm currently in the process of getting it to work for a CCME install, 
I have it all working except for one thing.. I think it was calling a 
phone from the asterisk server the call transfer back to asterisk 
would fail with an authentication issue and die. I'm pretty sure this 
issue can be resolved I just have not had the time recently wo work on 
it, I can provide more info when I'm back in the office next week.

If you have any question, please ask :)
I've a solution like that in production, with a SIP trunk between ccme 
and *, without problems. The MWI works too.

Regards
Andrea
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (Darwin)
iD8DBQFCUBuTMakHrsrHP9wRAvk9AKC1YqetsRZXw1wrOKXrqemSwFxDOACdFjS1
vpGDh7BzIUwDAQBnwMwzVq8=
=C1ka
-END PGP SIGNATURE-
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Are there online forums instead of this email forum??

2005-04-03 Thread Tzafrir Cohen

Hi

I haven't read all of the messages in this lengthy thread, so I hope 
I'm not repeating something from it.

Just a couple of questions:

1. What about mail-archive.com for archiving the list?

2. The archive need not be related to the list. It just needs to be
subsribed to it. Anybody want to set up searchable archives for the
list?

3. What about web-forum - mailing list gateways? E.g:
http://www.phorum.org (never used it, just heard about it).

4. I'll just mention again that this list has a very high load. Can
anybody suggest separate topics that could be moved to sublists?

-- 
Tzafrir Cohen | New signature for new address and  |  VIM is
http://tzafrir.org.il | new homepage   | a Mutt's  
[EMAIL PROTECTED] ||  best
ICQ# 16849755 | Space reserved for other protocols | friend
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] SIP dialing in two extensions

2005-04-03 Thread Jozeph Brasil

Hi guys,

Is it possible to make Dial to call two extensions at the same time?
I want when the user pressed extension it call to two SIP phones at the same
time... Who wakeup first get the call...
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

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

2005-04-03 Thread Olle E. Johansson

Welcome to the Asterisk users community!

Asterisk is the leading Open Source Telephony platform,
with support both for classical telephony and IP telephony.
Asterisk.org is a fast moving project. New code is added every
day.
Our community is also growing fast and we're having a lot
of interaction, on the IRC and on the mailing lists.
It's great to have you participating in this Open Source project
- building an Open Source PBX. Here are a few things to know and
remember while working with the project.
Again, welcome to the Asterisk.org Open Source PBX Project!
Astricon Europe registration is new open!
http://www.astricon.net
Meet you on the IRC channel :-), the bug tracker or
on the mailing list!
/oej
** Asterisk version information
At this moment we have two current versions of Asterisk, the
developer version and the stable version. The stable version
is distributed as .tar.gz archives on several servers. The
current stable version of Asterisk is 1.0.7. The stable version
contains no new functions and only changes when bugs are fixed.
The development version is to be used by people that can test
new functions and live with bugs and unexpected shortcomings.
The development version is branded 1.1 and will be the basis
for the next stable version, version 1.2. We will hopefully
soon reach a code freeze and start testing the stability
of version 1.1, so we will need your help.
** The mailing list is growing
Today, we propably have over 10,000 readers on the -users list. This
means that everything anyone write to this mailing list, is sent to
thousands of mailboxes that are already flowing over with messages.
That's why we all need to follow some simple rules on how to use
the mailing list and the other tools that are available.
** Think before sending a message, think twice
I would like to stress the fact that you have to think before you send a
message to such a big list. Do *not* send out personal replies on the list.
If you offer services to someone, do *not* CC: or reply to the list, it
will annoy more potential customers than get you new customers. If you
send out a message by mistake, you don't have to apologize to all of us,
we understand you're embarassed. We will get more annoyed by your
apology than over your first message.
And please do not send out test messages to the list.
** Try finding the answer first, then ask the list
The Asterisk Wiki at http://www.voip-info.org is an important
knowledge base for the project.
Go there to find your answer first, then search the mailing list
archives (Google or http://search.voip-forum.com) and then
go to the IRC channel. The IRC channel is populated with Asterisk gurus
around the clock (literally) and they'll help you move forward.
* IRC info: http://www.asterisk.org/index.php?menu=support#irc
* There's many links to Asterisk web pages on the documentation
  page at http://www.asterisk.org
* The Asterisk FAQ is found on the wiki
  http://www.voip-info.org/wiki-Asterisk+FAQ
* The Asterisk documentation project (which needs your help)
  is at http://www.asteriskdocs.org
  Their handbook The hitchhiker's guide to Asterisk is already
  well worth reading.
* Asterisk Daily news is at
  http://www.sineapps.com/news.php
* VoIP-search (Asterisk mailing list etc)
  http://search.voip-forum.com
Finally, if you don't find the answer elsewhere, try the list.
** Mailing lists
For developers, there is a developer's list, asterisk-dev.
Do not use this list as a secondary support line if you do
not get an answer on the -users list. It is meant for developer
discussions, not advanced support. If you need answers, there
is a better chance that you will get help on the irc channel.
For BSD users (FreeBSD, NetBSD, OpenBSD and OS/X) there's a
list called asterisk-bsd. There is also a business list
for those that want to ask for commercial services and
inform their community about new services (asterisk-biz).
You'll find all lists on http://lists.digium.com, which is the
site where you manage your subscription to this list as well.
Please, do not crosspost the same message to multiple mailing
lists. It will not help you, it will only add to the mail flow
and get people that read both lists irritated. If you are
unsure which list to use, send only to the -users list.
Make sure that you remove unnecessary text when you reply,
to make it easy to browse the mailing list quickly. And please
do not send HTML mail to a mailing list.
** Reporting bugs
If you think you have found a bug, report it. We need bug reports.
Read this document http://www.digium.com/bugtracker.html and then
go to the bugtracker http://bugs.digium.com to file a report.
If you are unsure, find a bug marshal on the IRC channel to help
you. They're appointed to support you with how to handle bugs.
Please check the bugtracker thoroughly before posting a new bug;
often, your bug or feature already exists but is simply slowly
making it's way through the system.  Duplicate

Re: [Asterisk-Users] SIP dialing in two extensions

2005-04-03 Thread administrator tootai

Jozeph Brasil a écrit :
Hi guys,
Is it possible to make Dial to call two extensions at the same time?
I want when the user pressed extension it call to two SIP phones at the same
time... Who wakeup first get the call...
 

Dial(SIP/extensionIAX2/otherextensionOH323/...)
--
Daniel
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] SET CHECK group

2005-04-03 Thread Mark Halverson

I attempted to use the incominglimit and outgoinglimit in iax.conf and it
doesnt seem to work anylonger, running CVS-HEAD 3/16/05

So I tried using the SetGroup but, in the dialplan I am already using Get
and Check Group.  I tried it with different variables and it still doesn't
workany ideas?

Basically I want each IAX Client coming in to be limited to a single call -
I then only want 1 call on each outbound SIP channel/account.

I believe the problem to be with (${CALLERIDNUM}) as it is setting the group
to: CALLERID/something  -  that something is always changing with each call
so the group using calleridnum never exceeds 1 and all calls go through.

Example:

exten = _1NXXNXX,1,SetGroup(${CALLERIDNUM})
exten = _1NXXNXX,2,Checkgroup(1)
exten = _1NXXNXX,3,SetGroup(CH1)
exten = _1NXXNXX,4,CheckGroup(1)
exten = _1NXXNXX,5,Dial(SIP/[EMAIL PROTECTED])
exten = _1NXXNXX,103,background(busy) 
exten = _1NXXNXX,104,hangup 
exten = _1NXXNXX,105,SetGroup(CH2) 
exten = _1NXXNXX,106,CheckGroup(1) 
exten = _1NXXNXX,107,Dial(SIP/[EMAIL PROTECTED])
exten = _1NXXNXX,207,Dial(IAX2/[EMAIL PROTECTED])


-- 
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.9.1 - Release Date: 4/1/2005
 

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Packetization

IAX is not an option as Sipura devices do not support AIX.Yes, the sipura will handle the different packet sizes...

Is it possible to reprogram asteris to do this?
On Apr 3, 2005 1:55 AM, Steven Critchfield [EMAIL PROTECTED] wrote:On Sat, 2005-04-02 at 21:16 -0500, Matt wrote: I'm aware that asterisk only supports 20ms packetization rates.Due to the fact that I will be using some voip devices on a wireless network which is highly sensative to framerate.. is there any way I can hard code the packetization rate at say 30 or 40ms and then compile astrisk?If so, can anyone in the know tell me what variables I need to look at to change?Are you sure your other devices support different packet sizes? Are yousure the added delay in audio delivery can be handled decently and notcause added echo?Have you considered what IAX trunking can do for you? It will reduceframe rate as you add channels since each packet will then hold theframes for each of the consecutive calls.--Steven Critchfield [EMAIL PROTECTED]___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Router with QoS recommendations

2005-04-03 Thread Philipp von Klitzing

Hi!

 As I have a Cisco PIX 515, with NO QoS functionality, and Im looking for
 a router that does outgoing QoS to put in front of my PIX. Problem is
 that Im using my 768/8096Kbit ADSL for both data and VoIP, and as soon
 as data is being sent to the internet the sound quality drops to
 something that is of NO use.

 Any suggestions or recommendations is appreciated.

Checkout m0n0wall on a Soekris or WRAP device.

Cheers, Philipp


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Passing varibles out of macros

2005-04-03 Thread Joe Presto

An option, but what about multiple inbound calls?  I'd be worried that they
trip over each other.

But - given the odds of this happening (variable is set and then read
instantly) - it may be the route to go.

Thanks - Joe

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Wilson Pickett
 Sent: Sunday, April 03, 2005 5:35 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Passing varibles *out* of macros
 
 How about setGlobalVar()
 ___
 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Router with QoS recommendations

2005-04-03 Thread NVC List Manager

On Sunday 03 April 2005 06:33, [EMAIL PROTECTED] wrote:
 Hi List



 As I have a Cisco PIX 515, with NO QoS functionality, and I'm looking for a
 router that does outgoing QoS to put in front of my PIX. Problem is that
 I'm using my 768/8096Kbit ADSL for both data and VoIP, and as soon as data
 is being sent to the internet the sound quality drops to something that is
 of NO use.



 Any suggestions or recommendations is appreciated.

As usual there's nothing that will beat OpenBSD. Takes 15 minutes to build 
following the instructions on the CD cover.

-- 

NVC List Manager
(For external lists)
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Router with QoS recommendations

2005-04-03 Thread Tim Pushor

NVC List Manager wrote:

As usual there's nothing that will beat OpenBSD. Takes 15 minutes to build 
following the instructions on the CD cover.

 

To someone who has never installed OpenBSD (or FreeBSD + pf for that 
matter) the learning curve is going to be much much higher than 15 
minutes, although one you learn PF you will never go back!

Tim
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Router with QoS recommendations

2005-04-03 Thread Andrew Kohlsmith

On April 3, 2005 08:13 am, Tim Pushor wrote:
 To someone who has never installed OpenBSD (or FreeBSD + pf for that
 matter) the learning curve is going to be much much higher than 15
 minutes, although one you learn PF you will never go back!

I've never seen the great advantage to pf over ip and tc.  Perhaps I'm just 
not that learned though.  :-)

-A.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Passing varibles out of macros

2005-04-03 Thread Gary Reuter

Have you tried putting in some NoOp lines to verify the values of
${screenresult}?

Also, wouldn't you get the desired result by removing the 'g' option
from your Dial()?
You might want to add an 'h' extension for further processing on the
dead channel.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Authenticating username

2005-04-03 Thread Martijn van Oosterhout

Hi,

From what I can see in the documentation the title of the section in
sip.conf is the username that the user logs in as. Is there a way of
seperating the names so that you can login with a normal username, but
call them with SIP/extension. Like so:

[904]
authuser=john
secret=password
etc...

Dial(SIP/904)calls whoever logged on as john.

Any ideas?
-- 
Martijn van Oosterhout
Ecomtel Pty Ltd
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Snom and Multiple calls

2005-04-03 Thread Philipp von Klitzing

Hi!

 On the snom (I've tested this on the 220 and 360), the phone will 
 immediately reject any new INVITE that arrives with 486 BUSY HERE if 
 there's already a call on the phone opening

That is very interesting - can you present a review of the Snom 360 
hardware, even if it is a short one? Possibly compare it to the 220 and 
illustrate the differences? 

Some questions:
- quality of the handset and the speaker phone?
- how do the buttons feel?
- are the line LEDs multi-colour ones?

As the 360 softphone is available to everyone the software part isn't 
_that_ interesting as I can play with it myself... :-)

Cheers, Philipp


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Sipura SPA 2000 - Miltiple Ring Tones

2005-04-03 Thread Trevor Peirce

Rod Bacon wrote:
I'm glad I'm not the only one
Now... for a solution?
Well at least this rules out a misconfiguration on the telco's end 
(unless both our telco's made the same mistake). Does /anyone/ at all 
have any suggestions, or is there some debug information we can send to 
the list to figure this out?

It is quite annoying :)
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Buying some Polycom IP300s

2005-04-03 Thread Jim Van Meggelen

Dan Morin wrote:
 Sorry for the double post, I tried to paste and accidently sent the
 email 
 
 I've been playing with Asterisk for a few weeks now, and I've gotten
 everything to work well with softphones, so I'm ready to move on to
 normal VoIP phones.  I've been looking around and reading comments
 that people have had, and I was convinced that the Polycom IP300 was
 a great phone for a good price.  But, then I ran into this page,
 which has been update in the last few days: 
 
 http://www.voip-info.org/wiki-Polycom+SoundPoint+IP+500
 
 The page in the wiki used to say that the person would not recomed
 Polycom phones to anyone.  So anyway, I just want to make sure that
 the IP300 is a good choice.  I don't want to get cheap phones that
 aren't business quality, since I do play on using them for my
 business after testing.  Also, is the IP500 worth the extra money? 
 What can it do that the IP300 can't.  And finally, will the IP300 do
 ulaw encoding?  

The IP300 is a nice entry-level business phone. It does not have a
speakerphone, and cannot handle PoE, but other than that it is
excellent.

It is more expensive than some of the fully-featured generic phones, but
it also is built to a much higher standard, including a properly
weighted handset and high impact plastic.

If price is the main thing, then this phone might be a bit too expensive
($130-$150), but if quality (or even just the *feeling* of quality) is
important, this phone will serve well.

The IP500 is a similar phone with more line appearances, a higher
resolution display, full handsfree (Polycom-quality) and PoE. The IP500
has been favorably compared to the Cisco 7940.

Cheers,


--
Jim Van Meggelen
[EMAIL PROTECTED]

-- 
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.9.1 - Release Date: 01/04/2005
 

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Detecting when a called mobile is not reachable?

2005-04-03 Thread Ian Hailey

Hello all,
I was hoping to be able to call a mobile and if it is un-reachable for 
whatever reason (e.g. switched off) then I was expecting an unobtainable 
response that would be detected in Asterisk. It seems that the operator 
(Virgin in UK) imedately completes the call and plays an automated 
message before clearing the call. Does anyone know if there a way of 
avoiding the call completion for mobiles? I have noticed that Sipgate 
charge for a calls to an unavailable mobile regardless.

Thanks.
Ian.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Packetization

2005-04-03 Thread Bruce Komito

The packet size is a function of the number of milliseconds of sound sent
in the RTP packet. I don't know how to force * to change this, but you
*can* unilaterally change the RTP packet size on the Sipura. By doing
this, RTP packets sent by the Sipura will be larger or smaller than the
default (.03 ms is the default), and I know * will swallow whatever the
Sipura sends it. So, I know it's possible to change this in at least one
direction if you are using a Sipura.

Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815

On Sun, 3 Apr 2005, Matt wrote:

IAX is not an option as Sipura devices do not support AIX.
Yes, the sipura will handle the different packet sizes...

Is it possible to reprogram asteris to do this?

On Apr 3, 2005 1:55 AM, Steven Critchfield [EMAIL PROTECTED] wrote:

On Sat, 2005-04-02 at 21:16 -0500, Matt wrote:
I'm aware that asterisk only supports 20ms packetization rates. Due
to the fact that I will be using some voip devices on a wireless
network which is highly sensative to framerate.. is there any way I
can hard code the packetization rate at say 30 or 40ms and then
compile astrisk? If so, can anyone in the know tell me what variables
I need to look at to change?

Are you sure your other devices support different packet sizes? Are you
sure the added delay in audio delivery can be handled decently and not
cause added echo?

Have you considered what IAX trunking can do for you? It will reduce
frame rate as you add channels since each packet will then hold the
frames for each of the consecutive calls.
--
Steven Critchfield [EMAIL PROTECTED]

This message has been categorized as Indeterminate by Bayesian Analyzer.
Please click on this link if this message is a Spam
http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-04-03%5C140ee012d55b40a08232629f70c89189C=2

Or on this link if this message is a legitimate mail
http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-04-03%5C140ee012d55b40a08232629f70c89189C=1

--
---
This message has been inspected by DynaComm i:mail
---

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Looking for res_config_pgsql

2005-04-03 Thread Martijn van Oosterhout

A google search shows exactly one reference, so it appears to exist
somewhere. It's in somebodies CVS, any ideas?
-- 
Martijn van Oosterhout
Ecomtel Pty Ltd
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Detecting when a called mobile is not reachable?

2005-04-03 Thread David John Walsh

This is traditional accross the mobile / cell providers, and there is
no real way around it.

Background : The only way to ensure that a mobile is truly there is to
page the mobile, normally based on the Mobile Switching Centre (MSC)
coverage area, and thats after looking up on the subscirbers HLR, its
a lot of signalling for a call not to connect, and a cost to the
operator.

With the rate that mobile operators charge the A party for the call,
they get a percentage of the call from the originating operator, so
they get cash as soon as it connects, and therefor its in their
interest to connect that call, even if its to an announcement shelf.

Its one of the reasons they invented voicemail

If there is a way around it, don't shout it too loudly

David

On Apr 3, 2005 8:56 PM, Ian Hailey [EMAIL PROTECTED] wrote:
 Hello all,
 
 I was hoping to be able to call a mobile and if it is un-reachable for
 whatever reason (e.g. switched off) then I was expecting an unobtainable
 response that would be detected in Asterisk. It seems that the operator
 (Virgin in UK) imedately completes the call and plays an automated
 message before clearing the call. Does anyone know if there a way of
 avoiding the call completion for mobiles? I have noticed that Sipgate
 charge for a calls to an unavailable mobile regardless.
 
 Thanks.
 
 Ian.
 ___
 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Buying some Polycom IP300s

2005-04-03 Thread Courtney Couch

We have a majority of IP300's, and a few IP500's.  The IP300's are great 
phones if you need to simply drop in a bunch of VoIP phones quickly and 
cheaply.  The IP300's simply lack certain features like speakerphone 
that you may want.  Aside from that, its a great phone.

-Courtney
Dan Morin wrote:
Sorry for the double post, I tried to paste and accidently sent the email
 
I've been playing with Asterisk for a few weeks now, and I've gotten 
everything to work well with softphones, so I'm ready to move on to 
normal VoIP phones.  I've been looking around and reading comments 
that people have had, and I was convinced that the Polycom IP300 was a 
great phone for a good price.  But, then I ran into this page, which 
has been update in the last few days:
 
http://www.voip-info.org/wiki-Polycom+SoundPoint+IP+500
 
The page in the wiki used to say that the person would not recomed 
Polycom phones to anyone.  So anyway, I just want to make sure that 
the IP300 is a good choice.  I don't want to get cheap phones that 
aren't business quality, since I do play on using them for my business 
after testing.  Also, is the IP500 worth the extra money?  What can it 
do that the IP300 can't.  And finally, will the IP300 do ulaw encoding?
 
Thanks in advance.


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Asterisk on Suse minimal installation based on Suse Rescue - what to add to be bootable on HD partition ?

2005-04-03 Thread Robert Rozman

Hi,
I'm trying to go route some of Asterisk users already proposed for Asterisk 
minimal system. I've started from Suse Rescue system image - I've put it 
into HD partition. But since rescue is spawned from working system it has 
empty /boot directories and is not directly bootable if put on HD. I've 
tried to transfer or install kernel and grub to this partition, but no 
success (I first access to partition with chroot to make additions...).

I get errors on kernel rpm -ivh installation (I guess there are no 
directories and dependencies found in chroot) and also get error on 
grub-install /dev/hda7:
Could not find device for /boot: Not found or not a block device

Is there anyone more experienced with some advice, howto or example what 
need to be added to partition to be bootable on HD ?

Thanks in advance,
regards,
Rob.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Packetization

2005-04-03 Thread Leif Madsen - Certified Asterisk Consultant

Ok.. wow .03ms (if that's the default) is DEFINATELY part of the issue
(audio going from the phone over wireless is slightly choppy).. while
audio coming in (20ms) is ok... where do you change it on the sipura?On Apr 3, 2005 4:07 PM, Bruce Komito [EMAIL PROTECTED] wrote:The packet size is a function of the number of milliseconds of sound sentin the RTP packet.I don't know how to force * to change this, but you*can* unilaterally change the RTP packet size on the Sipura.By doingthis, RTP packets sent by the Sipura will be larger or smaller than thedefault (.03 ms is the default), and I know * will swallow whatever theSipura sends it.So, I know it's possible to change this in at least onedirection if you are using a Sipura.Bruce KomitoHigh Sierra Networks, Inc.www.servers-r-us.com(775) 236-5815On Sun, 3 Apr 2005, Matt wrote: IAX is not an option as Sipura devices do not support AIX. Yes, the sipura will handle the different packet sizes... Is it possible to reprogram asteris to do this? On Apr 3, 2005 1:55 AM, Steven Critchfield [EMAIL PROTECTED] wrote: On Sat, 2005-04-02 at 21:16 -0500, Matt wrote: I'm aware that asterisk only supports 20ms packetization rates. Due to the fact that I will be using some voip devices on a wireless network which is highly sensative to framerate.. is there any way I can hard code the packetization rate at say 30 or 40ms and then compile astrisk? If so, can anyone in the know tell me what variables I need to look at to change? Are you sure your other devices support different packet sizes? Are you sure the added delay in audio delivery can be handled decently and not cause added echo? Have you considered what IAX trunking can do for you? It will reduce frame rate as you add channels since each packet will then hold the frames for each of the consecutive calls. -- Steven Critchfield [EMAIL PROTECTED] This message has been categorized as Indeterminate by Bayesian Analyzer. Please click on this link if this message is a Spam http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-04-03%5C140ee012d55b40a08232629f70c89189C=2 Or on this link if this message is a legitimate mail http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-04-03%5C140ee012d55b40a08232629f70c89189C=1 -- --- This message has been inspected by DynaComm i:mail ---___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Are there online forums instead of this email forum??

On Mar 31, 2005 11:26 AM, Chuck Bunn [EMAIL PROTECTED] wrote:
 I am new to Asterisk and the first thing I have noticed about Asterisk
 and Pingtels open PBX's is that they are using this dinosaur method of
 running forums. It is a real pain getting every message in the forum and
 essentially keeping my own database of issues. With that said are there
 any forums that are well used or that might even convert this email in a
 true forum that is searchable and that doesn't require me downloading
 every email. Before you go and rant on me go see how Mambo Server does
 it at  http://forum.mamboserver.com. The forums are easy to use and thus
 are easy to participate in. I use mozilla Thunderbird and I have setup
 filters and all but it still is a pain to use this outdated email forum.

If you are looking for a web interface to this mailing list, get
yourself a Gmail account and subscribe from it. It does fantastic
threading. If you need an account, I'll be happy to help you out, I
have 50 invites.

-- 
Leif Madsen
http://www.leifmadsen.com
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Packetization

Never mind... blah spoke before I thought :P

Found the setting
On Apr 3, 2005 5:23 PM, Matt [EMAIL PROTECTED] wrote:Ok.. wow .03ms (if that's the default) is DEFINATELY part of the issue
(audio going from the phone over wireless is slightly choppy).. while
audio coming in (20ms) is ok... where do you change it on the sipura?On Apr 3, 2005 4:07 PM, Bruce Komito [EMAIL PROTECTED] wrote:The packet size is a function of the number of milliseconds of sound sentin the RTP packet.I don't know how to force * to change this, but you*can* unilaterally change the RTP packet size on the Sipura.By doingthis, RTP packets sent by the Sipura will be larger or smaller than thedefault (.03 ms is the default), and I know * will swallow whatever theSipura sends it.So, I know it's possible to change this in at least onedirection if you are using a Sipura.Bruce KomitoHigh Sierra Networks, Inc.www.servers-r-us.com(775) 236-5815On Sun, 3 Apr 2005, Matt wrote: IAX is not an option as Sipura devices do not support AIX. Yes, the sipura will handle the different packet sizes... Is it possible to reprogram asteris to do this? On Apr 3, 2005 1:55 AM, Steven Critchfield [EMAIL PROTECTED] wrote: On Sat, 2005-04-02 at 21:16 -0500, Matt wrote: I'm aware that asterisk only supports 20ms packetization rates. Due to the fact that I will be using some voip devices on a wireless network which is highly sensative to framerate.. is there any way I can hard code the packetization rate at say 30 or 40ms and then compile astrisk? If so, can anyone in the know tell me what variables I need to look at to change? Are you sure your other devices support different packet sizes? Are you sure the added delay in audio delivery can be handled decently and not cause added echo? Have you considered what IAX trunking can do for you? It will reduce frame rate as you add channels since each packet will then hold the frames for each of the consecutive calls. -- Steven Critchfield [EMAIL PROTECTED] This message has been categorized as Indeterminate by Bayesian Analyzer. Please click on this link if this message is a Spam http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-04-03%5C140ee012d55b40a08232629f70c89189C=2 Or on this link if this message is a legitimate mail http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-04-03%5C140ee012d55b40a08232629f70c89189C=1 -- --- This message has been inspected by DynaComm i:mail ---
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Packetization

I have to admit this still doesn't make sence.. if sipura's default is
.03ms and asterisk is 20ms.. why is the sipura dumping out around 60
frames/sec while the sipura is dumping out around 30 frames/sec??

Shouldn't the frames / packets per second go UP as the packetization gets smaller?On Apr 3, 2005 5:25 PM, Matt [EMAIL PROTECTED] wrote:Never mind... blah spoke before I thought :P

Re: [Asterisk-Users] Asterisk Discussion Form

2005-04-03 Thread John Novack





One would hope so, but one
of the fist posts I see is someone ranting on about how if you haven't
read x or y you don't deserve an answer.

It is that kind of a social misfit that should not be welcome anywhere,
but seems to have too loud a voice here.

JN


Ty Carter wrote:
Thank
you for your contribution Now maybe this is a good place where
people can ask a question without getting slammed because they don't
understand their own ignorance. :-)
  
  
  
  
[EMAIL PROTECTED] wrote:
  
  
  List:


With recent discussions in regards to a forum, I have set-up a

multi-faceted Asterisk and Open Source Discussion Board. The link is

www.voipnewbie.com/forum It is open and ready for use.


Enjoy!


VoIPNewbie

___

Asterisk-Users mailing list

Asterisk-Users@lists.digium.com

http://lists.digium.com/mailman/listinfo/asterisk-users

To UNSUBSCRIBE or update options visit:

 http://lists.digium.com/mailman/listinfo/asterisk-users




  
  
  
  
  
___
  
Asterisk-Users mailing list
  
Asterisk-Users@lists.digium.com
  
http://lists.digium.com/mailman/listinfo/asterisk-users
  
To UNSUBSCRIBE or update options visit:
  
 http://lists.digium.com/mailman/listinfo/asterisk-users
  
  
  
  
  



___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: Fwd: NDN: Re: [Asterisk-Users] Delaying answer of incoming calls

2005-04-03 Thread John Novack





Well, you COULD use your
delete key.
You DO have one, don't you?

And you complain of others posting stupidity

JN


C F wrote:

  What can be done to this shmuck?
Everytime I post anything to the list I get one of these. I'm sure
I'll get one for posting this one as well.

-- Forwarded message --
From: [EMAIL PROTECTED] [EMAIL PROTECTED]
Date: Apr 3, 2005 12:24 AM
Subject: NDN: Re: [Asterisk-Users] Delaying answer of incoming calls
To: C F [EMAIL PROTECTED]


Sorry. Your message could not be delivered to:

Joshua Chessman (Mailbox or Conference is full.)
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users




  



___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] really small box

2005-04-03 Thread Irakli Natsvlishvili

Hello, Matt!
MR fine. If you have to do any sort of transcoding a soekris is not the
MR way to go but for a small installation it works great.
Well.. Cisco's 17xx series router is a device which you can take, plug, 
configure and have office PBX. But price tag is $2K.

Why the same can't be done for a fraction of this price using * and not 
involving active cooling and graphics cards? 20-30 office users + 3-4 
transcoding sessions + voicemail. What kind of horsepower do you need for 
this?

MR I run an entire asterisk installation off of a 512 MB CF card (have
MR ~250 MB to spare for voicemails and logs)
Do you have install/configuration/HOWTO document? If yes, could you post it 
here or just send it to mail email?

I.N. 

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Router with QoS recommendations

2005-04-03 Thread Irakli Natsvlishvili

From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Sunday, April 03, 2005 3:33 AM
Subject: [Asterisk-Users] Router with QoS recommendations

As I have a Cisco PIX 515, with NO QoS functionality,
and I'm looking for a router that does outgoing QoS to put in front of my 
PIX.
PixOS 7.0.1 supports QoS. Yesterday it was on TAC's download page. No, I 
have not installed yet.

I.N. 

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Router with QoS recommendations

2005-04-03 Thread Tim Pushor

iptables looks very powerful, thats for sure.
I prefer PF's approach to security first, convenience second, and I 
*really* like the fact that PF has a real parser. As the requements get 
more complex, having everything in one file, and very readable and 
structured is a huge plus. Also, the integration with ALTQ is nice, 
especially for these types of applications.

Andrew Kohlsmith wrote:
On April 3, 2005 08:13 am, Tim Pushor wrote:
 

To someone who has never installed OpenBSD (or FreeBSD + pf for that
matter) the learning curve is going to be much much higher than 15
minutes, although one you learn PF you will never go back!
   

I've never seen the great advantage to pf over ip and tc.  Perhaps I'm just 
not that learned though.  :-)

-A.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
 

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] CAPI/Dialing out

2005-04-03 Thread Philip Hofstetter

Hi,
Philip Hofstetter wrote:
Now may next step has been to enable dialing out with the softphones.
This does not work as expected.
I was able to fix this problems by downgrading from kernel 2.6.11 to 
2.6.10. There must be a CAPI-Problem hidden somewhere.

Last saturday was so much fun for me, trying out all the stuff that can 
be done with asterisk.

Thanks to all for this wonderful program!
Philip
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Zaptel Anti-MMX Optimizations

2005-04-03 Thread I put the Who? in Mishehu

Did you try issuing show translation recalc #  where # is any given 
number of seconds to recalculate for?  For example, speex tends to show 
weird numbers for me on my dual proc xeon 2.8ghz, until I do a show 
translation recalc 1, then I get more sane numbers.

Just my thoughts.
-mishehu
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: Fwd: NDN: Re: [Asterisk-Users] Delaying answer of incoming calls

2005-04-03 Thread C F

On Apr 3, 2005 5:45 PM, John Novack [EMAIL PROTECTED] wrote:
  Well, you COULD use your delete key.

Actually nope, I can't because I'm using gmails web client to read my email.

  You DO have one, don't you?

Yep I do, how did you know?

  And you complain of others posting stupidity

Please read what I answered about NOT being able to use the deleted
key, then read the line about complaing.. then look in the mirror.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] IAX messages

2005-04-03 Thread pabut

I'm trying to get IAXTEL inbound working  in my log I'm seeing all this
noise (below).
I understand I'm in DEBUG mode but I'm not doing anything yet ... what do
all these messages mean???

Apr  2 02:47:44 DEBUG[28339]: Immediately destroying 2, having received
INVAL
Apr  2 02:47:44 DEBUG[28339]: Raw Hangup 69.73.19.178:4569, src=2, dst=941
Apr  2 02:48:44 DEBUG[28339]: Immediately destroying 1, having received
INVAL
Apr  2 02:49:32 DEBUG[28339]: Sending VNAK
Apr  2 02:49:32 DEBUG[28339]: Sending VNAK
Apr  2 02:49:32 DEBUG[28339]: Sending VNAK
Apr  2 02:49:34 DEBUG[28339]: Sending VNAK
Apr  2 02:49:34 VERBOSE[28339]: -- Registered to '69.73.19.178', who
sees us as 68.52.23.171:19548
Apr  2 02:49:34 DEBUG[28339]: Raw Hangup 69.73.19.178:4569, src=2, dst=1330
Apr  2 02:49:34 DEBUG[28339]: Raw Hangup 69.73.19.178:4569, src=2, dst=1330
Apr  2 02:49:34 DEBUG[28339]: Raw Hangup 69.73.19.178:4569, src=2, dst=1330
Apr  2 02:49:34 DEBUG[28339]: Raw Hangup 69.73.19.178:4569, src=2, dst=1330
Apr  2 02:50:23 DEBUG[28339]: Immediately destroying 1, having received
INVAL
Apr  2 02:50:23 DEBUG[28339]: Raw Hangup 69.73.19.178:4569, src=1, dst=1047
Apr  2 02:50:26 DEBUG[28339]: Raw Hangup 69.73.19.178:4569, src=1, dst=1047
Apr  2 02:50:33 DEBUG[28339]: Raw Hangup 69.73.19.178:4569, src=1, dst=1047
Apr  2 02:50:35 DEBUG[28339]: Raw Hangup 69.73.19.178:4569, src=1, dst=1047
Apr  2 02:50:36 DEBUG[28339]: Raw Hangup 69.73.19.178:4569, src=1, dst=1047
Apr  2 02:51:22 DEBUG[28339]: Immediately destroying 2, having received
INVAL
Apr  2 02:52:04 DEBUG[28339]: Immediately destroying 1, having received
INVAL
Apr  2 02:52:04 DEBUG[28339]: Raw Hangup 69.73.19.178:4569, src=1, dst=68
Apr  2 02:52:38 DEBUG[28339]: Raw Hangup 69.73.19.178:4569, src=2, dst=336
Apr  2 02:52:49 DEBUG[28339]: Raw Hangup 69.73.19.178:4569, src=2, dst=336
Apr  2 02:52:59 DEBUG[28339]: Raw Hangup 69.73.19.178:4569, src=2, dst=336
Apr  2 02:53:44 DEBUG[28339]: Immediately destroying 1, having received
INVAL
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] SET CHECK group

2005-04-03 Thread C F

Look at:
http://www.voip-info.org/wiki-Asterisk+cmd+setgroup
read example 2 revised.

On Apr 3, 2005 1:20 PM, Mark Halverson [EMAIL PROTECTED] wrote:
 I attempted to use the incominglimit and outgoinglimit in iax.conf and it
 doesn't seem to work anylonger, running CVS-HEAD 3/16/05
 
 So I tried using the SetGroup but, in the dialplan I am already using Get
 and Check Group.  I tried it with different variables and it still doesn't
 workany ideas?
 
 Basically I want each IAX Client coming in to be limited to a single call -
 I then only want 1 call on each outbound SIP channel/account.
 
 I believe the problem to be with (${CALLERIDNUM}) as it is setting the group
 to: CALLERID/something  -  that something is always changing with each call
 so the group using calleridnum never exceeds 1 and all calls go through.
 
 Example:
 
 exten = _1NXXNXX,1,SetGroup(${CALLERIDNUM})
 exten = _1NXXNXX,2,Checkgroup(1)
 exten = _1NXXNXX,3,SetGroup(CH1)
 exten = _1NXXNXX,4,CheckGroup(1)
 exten = _1NXXNXX,5,Dial(SIP/[EMAIL PROTECTED])
 exten = _1NXXNXX,103,background(busy)
 exten = _1NXXNXX,104,hangup
 exten = _1NXXNXX,105,SetGroup(CH2)
 exten = _1NXXNXX,106,CheckGroup(1)
 exten = _1NXXNXX,107,Dial(SIP/[EMAIL PROTECTED])
 exten = _1NXXNXX,207,Dial(IAX2/[EMAIL PROTECTED])
 
 --
 No virus found in this outgoing message.
 Checked by AVG Anti-Virus.
 Version: 7.0.308 / Virus Database: 266.9.1 - Release Date: 4/1/2005
 
 ___
 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] How to reset IAXy?

2005-04-03 Thread I put the Who? in Mishehu

I'd put the device and another machine on a separate physical network 
where you can make whatever IP configurations you need in order to be 
able to send data to the IAXy.  Then you can load new configuration to 
it there.

There might be a better way to do i, but I don't know for sure.
-mishehu
Lam H. Nguyen wrote:
Can anyone tell me how to reset the IAXy? I used I put
it the wrong ip config in the IAXy and it conflicts
with my network whenever I plug it in. Currently the
DHCP is disable. I need to re-enable it to change the
settings.
The hard reset button on the IAXy doesn't seem to work 

		
__ 
Do you Yahoo!? 
Yahoo! Personals - Better first dates. More second dates. 
http://personals.yahoo.com

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
!DSPAM:424f6cce180855966097315!
 

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] VG248 and Asterisk

2005-04-03 Thread Steve Blair

 Has anyone been successful getting a Cisco VG248 gateway
to speak MGCP with Asterisk? If so can you share either
your mgcp.conf or at least tips on getting the two devices
working together.
Thanks
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Authenticating username

2005-04-03 Thread Nabeel Jafferali

 Dial(SIP/904)calls whoever logged on as john.

You could define a variable in extensions.conf.

Nabeel
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] problems with call-forward from ccme to * on sip trunk

2005-04-03 Thread Andrea Riela

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi folks
I've a strange problem, probably a mistake but I don't see it :(
Description:
My ephone-dn number on ccme, that is a simple connection plar for all 
ISDN calls, is 601
The voicemailmain on asterisk is 5900.
CCME: 192.168.17.1
*: 192.168.17.10

My sip.conf: http://www.pastebin.com/266718
My extension.conf: http://www.pastebin.com/266720
My voicemail.conf: http://www.pastebin.com/266722
when I call the asterisk server from SIP free accounts, I receive the 
call on 601 (my 7960 phone) and then the call will be forwarded to 
voicemail without any problem.
But when I receive a call from ISDN cloud, the 601 rings, the call is 
forwarded (see debug) on voicemail (number 5601), but the line goes 
down.

This is the debug, that is I suppose the problem is on my Asterisk 
config (the 'ext-number' is the caller ID): 
http://www.pastebin.com/266724

I hope you could help me :)
Thanks for all
Regards
Andrea
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (Darwin)
iD8DBQFCUIYAMakHrsrHP9wRAlR1AKDKNzARotrmFMPphvjwqjp8da4SwACfQ6lo
hxesZUu9t220j8zfQHW2DX0=
=zJCw
-END PGP SIGNATURE-
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Zaptel Anti-MMX Optimizations

2005-04-03 Thread Trevor Peirce

I put the Who? in Mishehu wrote:
Did you try issuing show translation recalc #  where # is any given 
number of seconds to recalculate for?  For example, speex tends to 
show weird numbers for me on my dual proc xeon 2.8ghz, until I do a 
show translation recalc 1, then I get more sane numbers.

I actually get really weird sounding calls when MXX_OPTIMIZATIONS are 
enabled as well as random crashes... so I don't think I need to worry 
how that table is calculated.  It's just the best way I've found to 
qualify the problem.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] AS5300+SIP+ASTERISK or AS5300+MGCP

2005-04-03 Thread jafar mohammed

hi's

i have been trying to configure my AS5300 to work with
my asterisk box. i have tried SIP, calls come,
answered and AS5300 sends BYE message after not more
than 5 secs. I have also tried MGCP, but i believe i
am not configuring that right. here is the output of
the sip debug. please help me out or lead me to the
direction of sorting this problem out.

thank you

INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0

Via: SIP/2.0/UDP  66.178.100.66:5060

From: sip:66.178.100.66;tag=8CB7504-1904

To: sip:[EMAIL PROTECTED]

Date: Mon, 04 Apr 2005 00:16:50 GMT

Call-ID:
[EMAIL PROTECTED]

Supported: timer

Min-SE:  600

Cisco-Guid:
2899651584-2748649945-2861211020-3122285050

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK,
COMET, REFER, SUBSCRIBE, NOTIFY, INFO

CSeq: 101 INVITE

Max-Forwards: 6

Remote-Party-ID:
sip:66.178.100.66;party=calling;screen=no;privacy=off

Timestamp: 1112573810

Contact: sip:66.178.100.66:5060

Expires: 180

Allow-Events: telephone-event

MIME-Version: 1.0

Content-Type: multipart/mixed;boundary=uniqueBoundary

Content-Length: 431



--uniqueBoundary

Content-Type: application/sdp



v=0

o=CiscoSystemsSIP-GW-UserAgent 5042 571 IN IP4
66.178.100.66

s=SIP Call

c=IN IP4 66.178.100.66

t=0 0

m=audio 18992 RTP/AVP 3 19

c=IN IP4 66.178.100.66

a=rtpmap:3 GSM/8000

a=rtpmap:19 CN/8000

a=ptime:10

--uniqueBoundary

Content-Type: application/gtd

Content-Disposition: signal;handling=optional



IAM,

GCI,acd52c00a3d511d9aa8a9d8cba1a49fa



--uniqueBoundary--

-*-

   - 21 headers, 21 lines

* Using latest SIP request as basis request

* Sending to 66.178.100.66 : 5060 (NAT)
Apr  4 00:16:40 NOTICE[25296]: chan_sip2.c:5872
check_user_full: User name from URI: 66.178.100.66,
Digest auth user: (null)

  == Authentication turned off, no secret for user
66.178.100.66

* No RDNIS header in SIP packet

-- - SIPFromURI:
sip:66.178.100.66;tag=8CB7504-1904

--  Transmitting (no NAT) response to
66.178.100.66:5060 

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 66.178.100.66:5060

From: sip:66.178.100.66;tag=8CB7504-1904

To: sip:[EMAIL PROTECTED];tag=as08ade073

Call-ID:
[EMAIL PROTECTED]

CSeq: 101 INVITE

User-Agent: Asterisk PBX

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

Contact: sip:[EMAIL PROTECTED]

Content-Length: 0



-*-

-- Executing Answer(SIP/66.178.100.66-bf34, )
in new stack

* SDP preparation: We're at 62.56.250.198 port 17962

* Answering with preferred capability 0x2 (gsm)

* Answering with preferred capability 0x4 (ulaw)

* Answering with preferred capability 0x8 (alaw)

Answering with non-codec capability 0x1(g723)

-- Reliably  Transmitting (no NAT) response to
66.178.100.66:5060 

SIP/2.0 200 OK

Via: SIP/2.0/UDP 66.178.100.66:5060

From: sip:66.178.100.66;tag=8CB7504-1904

To: sip:[EMAIL PROTECTED];tag=as08ade073

Call-ID:
[EMAIL PROTECTED]

CSeq: 101 INVITE

User-Agent: Asterisk PBX

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

Contact: sip:[EMAIL PROTECTED]

Content-Type: application/sdp

Content-Length: 265



v=0

o=root 25296 25296 IN IP4 62.56.250.198

s=session

c=IN IP4 62.56.250.198

t=0 0

m=audio 17962 RTP/AVP 3 0 8 101

a=rtpmap:3 GSM/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

-*-

-- Executing Wait(SIP/66.178.100.66-bf34, 2)
in new stack

--- Sip read from 66.178.100.66:50341 
ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0

Via: SIP/2.0/UDP  66.178.100.66:5060

From: sip:66.178.100.66;tag=8CB7504-1904

To: sip:[EMAIL PROTECTED];tag=as08ade073

Date: Mon, 04 Apr 2005 00:16:50 GMT

Call-ID:
[EMAIL PROTECTED]

Max-Forwards: 6

Content-Length: 0

CSeq: 101 ACK



-*-

   - 9 headers, 0 lines

--- Sip read from 66.178.100.66:53065 
BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0

Via: SIP/2.0/UDP  66.178.100.66:5060

From: sip:66.178.100.66;tag=8CB7504-1904

To: sip:[EMAIL PROTECTED];tag=as08ade073

Date: Mon, 04 Apr 2005 00:16:50 GMT

Call-ID:
[EMAIL PROTECTED]

User-Agent: Cisco-SIPGateway/IOS-12.x

Max-Forwards: 6

Timestamp: 1112573810

CSeq: 102 BYE

Content-Length: 0



-*-

   - 11 headers, 0 lines

* Sending to 66.178.100.66 : 5060 (non-NAT)

--  Transmitting (no NAT) response to
66.178.100.66:5060 

SIP/2.0 200 OK

Via: SIP/2.0/UDP 66.178.100.66:5060

From: sip:66.178.100.66;tag=8CB7504-1904

To: sip:[EMAIL PROTECTED];tag=as08ade073

Call-ID:
[EMAIL PROTECTED]

CSeq: 102 BYE

User-Agent: Asterisk PBX

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

Contact: sip:[EMAIL PROTECTED]

Content-Length: 0



-*-

  == Spawn extension (AS5300, 9001, 2) exited non-zero
on 'SIP/66.178.100.66-bf34'

-- Executing Hangup(SIP/66.178.100.66-bf34, )
in new stack

  == Spawn extension (AS5300, h, 1) exited non-zero on
'SIP/66.178.100.66-bf34'

Destroying SIP dialogue '[EMAIL PROTECTED]'



__ 
Do you Yahoo!? 
Yahoo! Small Business - Try our new resources site!
http://smallbusiness.yahoo.com/resources/

[Asterisk-Users] AS5300+SIP+ASTERISK or AS5300+MGCP

2005-04-03 Thread jafar mohammed

hi's

i have been trying to configure my AS5300 to work with
my asterisk box. i have tried SIP, calls come,
answered and AS5300 sends BYE message after not more
than 5 secs. I have also tried MGCP, but i believe i
am not configuring that right. here is the output of
the sip debug. please help me out or lead me to the
direction of sorting this problem out.

thank you

INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0

Via: SIP/2.0/UDP  66.178.100.66:5060

From: sip:66.178.100.66;tag=8CB7504-1904

To: sip:[EMAIL PROTECTED]

Date: Mon, 04 Apr 2005 00:16:50 GMT

Call-ID:
[EMAIL PROTECTED]

Supported: timer

Min-SE:  600

Cisco-Guid:
2899651584-2748649945-2861211020-3122285050

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK,
COMET, REFER, SUBSCRIBE, NOTIFY, INFO

CSeq: 101 INVITE

Max-Forwards: 6

Remote-Party-ID:
sip:66.178.100.66;party=calling;screen=no;privacy=off

Timestamp: 1112573810

Contact: sip:66.178.100.66:5060

Expires: 180

Allow-Events: telephone-event

MIME-Version: 1.0

Content-Type: multipart/mixed;boundary=uniqueBoundary

Content-Length: 431



--uniqueBoundary

Content-Type: application/sdp



v=0

o=CiscoSystemsSIP-GW-UserAgent 5042 571 IN IP4
66.178.100.66

s=SIP Call

c=IN IP4 66.178.100.66

t=0 0

m=audio 18992 RTP/AVP 3 19

c=IN IP4 66.178.100.66

a=rtpmap:3 GSM/8000

a=rtpmap:19 CN/8000

a=ptime:10

--uniqueBoundary

Content-Type: application/gtd

Content-Disposition: signal;handling=optional



IAM,

GCI,acd52c00a3d511d9aa8a9d8cba1a49fa



--uniqueBoundary--

-*-

   - 21 headers, 21 lines

* Using latest SIP request as basis request

* Sending to 66.178.100.66 : 5060 (NAT)
Apr  4 00:16:40 NOTICE[25296]: chan_sip2.c:5872
check_user_full: User name from URI: 66.178.100.66,
Digest auth user: (null)

  == Authentication turned off, no secret for user
66.178.100.66

* No RDNIS header in SIP packet

-- - SIPFromURI:
sip:66.178.100.66;tag=8CB7504-1904

--  Transmitting (no NAT) response to
66.178.100.66:5060 

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 66.178.100.66:5060

From: sip:66.178.100.66;tag=8CB7504-1904

To: sip:[EMAIL PROTECTED];tag=as08ade073

Call-ID:
[EMAIL PROTECTED]

CSeq: 101 INVITE

User-Agent: Asterisk PBX

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

Contact: sip:[EMAIL PROTECTED]

Content-Length: 0



-*-

-- Executing Answer(SIP/66.178.100.66-bf34, )
in new stack

* SDP preparation: We're at 62.56.250.198 port 17962

* Answering with preferred capability 0x2 (gsm)

* Answering with preferred capability 0x4 (ulaw)

* Answering with preferred capability 0x8 (alaw)

Answering with non-codec capability 0x1(g723)

-- Reliably  Transmitting (no NAT) response to
66.178.100.66:5060 

SIP/2.0 200 OK

Via: SIP/2.0/UDP 66.178.100.66:5060

From: sip:66.178.100.66;tag=8CB7504-1904

To: sip:[EMAIL PROTECTED];tag=as08ade073

Call-ID:
[EMAIL PROTECTED]

CSeq: 101 INVITE

User-Agent: Asterisk PBX

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

Contact: sip:[EMAIL PROTECTED]

Content-Type: application/sdp

Content-Length: 265



v=0

o=root 25296 25296 IN IP4 62.56.250.198

s=session

c=IN IP4 62.56.250.198

t=0 0

m=audio 17962 RTP/AVP 3 0 8 101

a=rtpmap:3 GSM/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

-*-

-- Executing Wait(SIP/66.178.100.66-bf34, 2)
in new stack

--- Sip read from 66.178.100.66:50341 
ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0

Via: SIP/2.0/UDP  66.178.100.66:5060

From: sip:66.178.100.66;tag=8CB7504-1904

To: sip:[EMAIL PROTECTED];tag=as08ade073

Date: Mon, 04 Apr 2005 00:16:50 GMT

Call-ID:
[EMAIL PROTECTED]

Max-Forwards: 6

Content-Length: 0

CSeq: 101 ACK



-*-

   - 9 headers, 0 lines

--- Sip read from 66.178.100.66:53065 
BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0

Via: SIP/2.0/UDP  66.178.100.66:5060

From: sip:66.178.100.66;tag=8CB7504-1904

To: sip:[EMAIL PROTECTED];tag=as08ade073

Date: Mon, 04 Apr 2005 00:16:50 GMT

Call-ID:
[EMAIL PROTECTED]

User-Agent: Cisco-SIPGateway/IOS-12.x

Max-Forwards: 6

Timestamp: 1112573810

CSeq: 102 BYE

Content-Length: 0



-*-

   - 11 headers, 0 lines

* Sending to 66.178.100.66 : 5060 (non-NAT)

--  Transmitting (no NAT) response to
66.178.100.66:5060 

SIP/2.0 200 OK

Via: SIP/2.0/UDP 66.178.100.66:5060

From: sip:66.178.100.66;tag=8CB7504-1904

To: sip:[EMAIL PROTECTED];tag=as08ade073

Call-ID:
[EMAIL PROTECTED]

CSeq: 102 BYE

User-Agent: Asterisk PBX

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

Contact: sip:[EMAIL PROTECTED]

Content-Length: 0



-*-

  == Spawn extension (AS5300, 9001, 2) exited non-zero
on 'SIP/66.178.100.66-bf34'

-- Executing Hangup(SIP/66.178.100.66-bf34, )
in new stack

  == Spawn extension (AS5300, h, 1) exited non-zero on
'SIP/66.178.100.66-bf34'

Destroying SIP dialogue '[EMAIL PROTECTED]'

__
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com

Re: [Asterisk-Users] *** Asterisk 2.0 Stable release out now

2005-04-03 Thread Matt Riddell

John Novack wrote:
An even BETTER question is: When will what is already out and more or 
less working have enough accurate documentation to make it acceptable to 
a wider audience?
Once more people start contributing.
As one small example: the recent postings regarding wctdm. If all the 
options are at the end of the driver source, how long does it take to 
put into a more accessible form?
Probably not long: just head over to http://asteriskdocs.org and see how 
you can help out, or pop on to the wiki ( http://www.voip-info.org ) and 
add an entry.

--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Asterisk Realtime Capabilities

2005-04-03 Thread Rod Bacon

Hello all. I am trying to architect a large-scale solution and need to 
know some of the capabilities of * using realtime configuration (I have 
read some docuemntation on the WIKI, but have not yet played with Realtime).

As the supporting docco is a little light-on at the moment, I'm hoping 
to get some meaningful feedback from those who've used it successfully.

Essentially, I'm trying to build a distributed network of * boxes which 
will terminate SIP/IAX/PSTN calls. I would potentially like to use a 
central farm of redundant * boxes in place of a single SER box (if 
possible) to act as the SIP (and IAX) registrar/proxy. I know that SER 
is a much more efficient SIP proxy, but I'm hoping that the addition of 
multiple * boxes will largely negate this fact, and provide redundancy.

Is it indeed conceivable to build a national network entirely of * 
servers, have users register via SIP/IAX to a load-balanced (not sure 
which mechanism I'll empoy here yet) farm of * servers (ie. an unknown 
server), and then receive calls from anywhere in the network (and 
outside)? In other words, can the registering server update a USRLOC 
type database on the fly, so all other servers know where to route calls 
for a given (dynamic?) client?

Also, I will be using multiple * boxes as media gateways. Is there an 
existing mechanism whereby a given server can record the number of 
busy/available ZAP channels to a central database for the purpose of 
call routing?



___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Buying some Polycom IP300s

2005-04-03 Thread Kong

erm, how much u willing to sell ip500?, i would like to get 1 or 2 for my 
developments testing purposes. BTW if u do sell me, I'm in Malaysia, is it 
a problem for u to send it over? :D

thanz.
At 04:39 AM 4/4/2005, you wrote:
We have a majority of IP300's, and a few IP500's.  The IP300's are great 
phones if you need to simply drop in a bunch of VoIP phones quickly and 
cheaply.  The IP300's simply lack certain features like speakerphone that 
you may want.  Aside from that, its a great phone.

-Courtney
Dan Morin wrote:
Sorry for the double post, I tried to paste and accidently sent the email
I've been playing with Asterisk for a few weeks now, and I've gotten 
everything to work well with softphones, so I'm ready to move on to 
normal VoIP phones.  I've been looking around and reading comments that 
people have had, and I was convinced that the Polycom IP300 was a great 
phone for a good price.  But, then I ran into this page, which has been 
update in the last few days:

http://www.voip-info.org/wiki-Polycom+SoundPoint+IP+500
The page in the wiki used to say that the person would not recomed 
Polycom phones to anyone.  So anyway, I just want to make sure that the 
IP300 is a good choice.  I don't want to get cheap phones that aren't 
business quality, since I do play on using them for my business after 
testing.  Also, is the IP500 worth the extra money?  What can it do that 
the IP300 can't.  And finally, will the IP300 do ulaw encoding?

Thanks in advance.

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] How does asterisk know the did called on?

Courtney Couch wrote:
If I were to buy 20 did's how do I know within asterisk which number was 
dialed? (like say I want a few of the did's to ring specific extensions 
if they are dialed and others to go through the menu)

Is there any ${var} that has the number dialed in on? (that would be 
optimum).
Your carrier can tell you how many digits they will send to you. 
Asterisk sees these digits and will match exten = 1234,1,Blah if the 
carrier sends you 4 digits.

Remember Asterisk does not really support DID on analog ports, only 
T-1/E-1 (including PRI) ports , BRI ports, and VoIP ports.

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] problems with call-forward from ccme to * on sip trunk

2005-04-03 Thread Nathan Alberti

I think this problem is exactly the one I am having.
The issue is:
http://www.pastebin.com/266724
042 Found no matching peer or user for '192.168.17.1:56730'
to which asterisk generates a SIP/2.0 404 Not Found (line 057)
yet you have it configured here:
[operator]
type=peer
canreinvite=no
host=192.168.17.1
context=cme-pbx
Hopefully someone with a working configuration can provide feedack.
Regards,
Nathan.
Andrea Riela wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi folks
I've a strange problem, probably a mistake but I don't see it :(
Description:
My ephone-dn number on ccme, that is a simple connection plar for all 
ISDN calls, is 601
The voicemailmain on asterisk is 5900.
CCME: 192.168.17.1
*: 192.168.17.10

My sip.conf: http://www.pastebin.com/266718
My extension.conf: http://www.pastebin.com/266720
My voicemail.conf: http://www.pastebin.com/266722
when I call the asterisk server from SIP free accounts, I receive the 
call on 601 (my 7960 phone) and then the call will be forwarded to 
voicemail without any problem.
But when I receive a call from ISDN cloud, the 601 rings, the call is 
forwarded (see debug) on voicemail (number 5601), but the line goes down.

This is the debug, that is I suppose the problem is on my Asterisk 
config (the 'ext-number' is the caller ID): http://www.pastebin.com/266724

I hope you could help me :)
Thanks for all
Regards
Andrea
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (Darwin)
iD8DBQFCUIYAMakHrsrHP9wRAlR1AKDKNzARotrmFMPphvjwqjp8da4SwACfQ6lo
hxesZUu9t220j8zfQHW2DX0=
=zJCw
-END PGP SIGNATURE-
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Re: using unixODBC

2005-04-03 Thread Kamran Ahmad

hello

i dont know why unixodbc is not working. i am trying
to make odbc connection. yesterday my odbc connection
was working with mysql on my one mechine but now it is
not working. is there any problem in code.

/etc/odbc.ini
[test]
Description = My test dsn
Trace = Off
TraceFile = stderr
Driver = mytestdriver
SERVER = 127.0.0.1
USER = asteriskuser
PASSWORD = asteriskpassword
PORT = 3306
DATABASE = asteriskcdrdb

/etc/odbcinst.ini
[mytestdriver]
Description = MySql driver for linux
Driver = /usr/lib/libodbc-2.50.39.so



__ 
Yahoo! Messenger 
Show us what our next emoticon should look like. Join the fun. 
http://www.advision.webevents.yahoo.com/emoticontest
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Detecting when a called mobile is not reachable?


On Apr 3, 2005 8:56 PM, Ian Hailey [EMAIL PROTECTED] wrote:
Hello all,
I was hoping to be able to call a mobile and if it is un-reachable for
whatever reason (e.g. switched off) then I was expecting an unobtainable
response that would be detected in Asterisk. It seems that the operator
(Virgin in UK) imedately completes the call and plays an automated
message before clearing the call. Does anyone know if there a way of
avoiding the call completion for mobiles? I have noticed that Sipgate
charge for a calls to an unavailable mobile regardless.
Bellsouth at least WILL play an automated message, but NOT answer the 
line.  I work around this by adding the r option to the Dial command. 
 The r option of course provides a fake ringing sound to the caller, 
even if it REALLY should be doing something else like playing telco 
audio before answer, or a busy tone.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] How to reset IAXy?

2005-04-03 Thread Ronald Wiplinger

I put the Who? in Mishehu wrote:
I'd put the device and another machine on a separate physical network 
where you can make whatever IP configurations you need in order to be 
able to send data to the IAXy.  Then you can load new configuration to 
it there.

There might be a better way to do i, but I don't know for sure.
-mishehu
Lam H. Nguyen wrote:
Can anyone tell me how to reset the IAXy? I used I put
it the wrong ip config in the IAXy and it conflicts
with my network whenever I plug it in. Currently the
DHCP is disable. I need to re-enable it to change the
settings.
The hard reset button on the IAXy doesn't seem to work

The reset button works! You just need to keep pressing it while you turn 
on the power, and keep holding it for at least 4

hours
(I know I am three days too late for that joke, ... )
bye
Ronald
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] SET CHECK group

Mark Halverson wrote:
exten = _1NXXNXX,1,SetGroup(${CALLERIDNUM})
Try using ${ACCOUNTCODE} and make sure the account code is unique to 
each phone.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Detecting when a called mobile is not reachable?

2005-04-03 Thread Rod Bacon

This is quite interesting.
I tested calls to 2 mobiles that I knew were off, and not diverted to 
voicemail. 1 with Telstra, the other with vodafone (I'm in Australia). Via 
ISDN, both calls were shown as unanswered by asterisk. When the calls went 
to voicemail, the call was deemed to be answered.

Via analogue circuits, the call is shown as answered, no matter what.

- Original Message - 
From: Ian Hailey [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Monday, April 04, 2005 5:56 AM
Subject: [Asterisk-Users] Detecting when a called mobile is not reachable?


Hello all,
I was hoping to be able to call a mobile and if it is un-reachable for 
whatever reason (e.g. switched off) then I was expecting an unobtainable 
response that would be detected in Asterisk. It seems that the operator 
(Virgin in UK) imedately completes the call and plays an automated message 
before clearing the call. Does anyone know if there a way of avoiding the 
call completion for mobiles? I have noticed that Sipgate charge for a 
calls to an unavailable mobile regardless.

Thanks.
Ian.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Joshua Chessman

2005-04-03 Thread Rod Bacon

Empty yer bloody mailbox...
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 9, Issue 21

2005-04-03 Thread Kamran Ahmad

hello

can any one tell me what is the problem in my odbc
connection.
here is my sql.log connection with mysql is working
and with freetds is giving me error jawad is one
windows server having MS Sql server

#isql kdsn
src/tds/login.c: tds_connect: jawad:1433: Connection
refused
[ISQL]ERROR: Could not SQLConnect


[ODBC][3085][__handles.c][444]
Exit:[SQL_SUCCESS]
Environment = 0x823cb58
[ODBC][3085][SQLAllocHandle.c][345]
Entry:
Handle Type = 2
Input Handle = 0x823cb58
[ODBC][3085][SQLAllocHandle.c][463]
Exit:[SQL_SUCCESS]
Output Handle = 0x823d130
[ODBC][3085][SQLConnect.c][3526]
Entry:
Connection = 0x823d130
Server Name = [kdsn][length = 7 (SQL_NTS)] 
  
User Name = [kami][length = 11 (SQL_NTS)]  
 
Authentication = [***][length = 11
(SQL_NTS)]
UNICODE Using encoding ASCII 'ISO8859-1' and UNICODE
'UCS-2LE'

DIAG [08S01] [FreeTDS][SQL Server]Server is
unavailable or does not exist.

DIAG [S1000] [FreeTDS][SQL Server]Unable to connect
to data source

[ODBC][3085][SQLConnect.c][3894]
Exit:[SQL_ERROR]
[ODBC][3085][SQLFreeHandle.c][268]
Entry:
Handle Type = 2
Input Handle = 0x823d130
[ODBC][3085][SQLFreeHandle.c][317]
Exit:[SQL_SUCCESS]
[ODBC][3085][SQLFreeHandle.c][203]
Entry:
Handle Type = 1
Input Handle = 0x823cb58

/usr/local/etc/odbc.ini
[mydsn]
Description = test
Driver  = test
Server  = localhost
Database= asterisk
Port= 3306
Socket  =
Option  =
Stmt=
  
  
[kdsn]
Description = tds dsn
Driver  = tdsdriver
Servername  =
Server  = jawad
Address = jawad
Port= 1433
Database= kdb
TDS_Version = 4.2
Language= us_english
TextSize=
Domain  =
PacketSize  =
  
  
/usr/local/etc/odbcinst.ini
[ODBC]
Trace   = Yes
TraceFile   = /tmp/sql.log
ForceTrace  = Yes
Pooling = No
  
  
[test]
Description = test
Driver  = /usr/lib/libmyodbc.so
Driver64=
Setup   = /usr/local/lib/libodbcmyS.so.1
Setup64 =
UsageCount  = 1
CPTimeout   =
CPReuse =
  
  
[tdsdriver]
Description = driver for tds
Driver  = /usr/local/lib/libtdsodbc.so
Driver64= /usr/local/lib
Setup   = /usr/local/lib/libtdsodbc.so.0
Setup64 = /usr/local/lib
UsageCount  = 1
CPTimeout   =
CPReuse =
  
  



__ 
Do you Yahoo!? 
Yahoo! Personals - Better first dates. More second dates. 
http://personals.yahoo.com

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Detecting when a called mobile is not reachable?