date:20050624

oui 
--- Dave Cotton [EMAIL PROTECTED] a écrit :

 On Thu, 2005-06-23 at 23:46 +0400, Jean-Michel Hiver
 wrote:
  Asterisk wrote:
  
  Hello - sorry for my bad english.
  I will make a list of all sound files on asterisk
  and i'll record then on professional studio.
  the french prompts from sineapps sounds bad...
 sorry for her...
  tell me if their is many peoples want it !

  
  french french asterisk sound files would be
 lovely! I want 'em!
 
 She comes from Nice. Parisiens :)
 
 -- 
 Dave Cotton [EMAIL PROTECTED]
 PACA
 
 ___
 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
 







___ 
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger 
Téléchargez cette version sur http://fr.messenger.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] French Audio Files

2005-06-24 Thread Thierry Wehr

 -Message d'origine-
 De : [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] De la part 
 de Asterisk
 Envoyé : jeudi 23 juin 2005 22:47
 À : asterisk-users@lists.digium.com
 Objet : [Asterisk-Users] French Audio Files
 
 Hello - sorry for my bad english.
 I will make a list of all sound files on asterisk and i'll 
 record then on professional studio.
 the french prompts from sineapps sounds bad... sorry for her...
 tell me if their is many peoples want it !
 
 thank's.
 
 en francais:
 
 dites moi si ca vaut le coup que j'investissexr dans 
 l'enregistrement des messages en francais. La voix sera la 
 voix off d'M6...
 merci !

Hello

This is a great idea

Ceci est une tres bonne idee

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] French Audio Files

2005-06-24 Thread Jean-Michel Hiver


Dave Cotton wrote:


On Thu, 2005-06-23 at 23:46 +0400, Jean-Michel Hiver wrote:
 


Asterisk wrote:

   


Hello - sorry for my bad english.
I will make a list of all sound files on asterisk
and i'll record then on professional studio.
the french prompts from sineapps sounds bad... sorry for her...
tell me if their is many peoples want it !


 


french french asterisk sound files would be lovely! I want 'em!
   



She comes from Nice. Parisiens :)
 



The sound set I had was Canadian French. Appuyez sur la touche carré 
type messages.


If you know of other French sound sets, please let me know!

___
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] SpanDSP - Squished Faxes

2005-06-24 Thread El Flynn


Richard Cook wrote:

Hello,
 
Has anyone had issues with faxes showing up squished in the TIFF  file?
 
Any ideas what could be causing it?
 


We had some issues while getting fax-email and email-fax working. As far as I 
can tell, it ended up being a wonky version of libtiff that was causing it. On 
our box we've got libtiff 3.6.1.


Flynn

___
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] French Audio Files

How can you offer Fench audio files for free ?

Regards
Harry from France








___ 
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger 
Téléchargez cette version sur http://fr.messenger.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

[Asterisk-Users] Re: TDM400P DevKit Problem

2005-06-24 Thread Anatoliy Kounitskiy

You can look at this tutorial: 
http://www.asteriskguru.com/tutorials/wildcard_tdm11b.html  Yes, it is 
for TDM11B (TDM400P family), but TDM 400p is moduler card.


Best Regards,
Anatoliy Kounitskiy
___
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] chan_capi-cm-0.5.1 and CLIR

2005-06-24 Thread asterisk_on_oelf


Hi,

I have updated from chan_capi-0.3.5 to 0.5.1, but can't get CLIP/CLIR to work.

with 0.3.5 CLIR works fine:
exten = _1X.,1,Dial(CAPI/@12345:${EXTEN:1},180)


with 0.5.1 I tried as described in the README:
exten = _1X.,1,CallingPres(32)
exten = _1X.,2,Dial(CAPI/contr1/${EXTEN:1},180)

but it doesn't work. My first MSN is always shown.

Where is my mistake?

regards
Jens


___
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] Management: Reload performace Realtimeperformance ?

2005-06-24 Thread Rene Ott

This is interesting. But reloading the big conf files itself doesn't cause
any performance problems (with verbosity 0)? Do you have experience what
happens with incoming calls during reload?


 If you have configs that big you'll need to set verbose 0 before  
 doing that.. you'll notice the calls start to jitter and all hell  
 breaks loose if you don't do so.

/b
---
Anakin: Youre either with me, or youre my enemy.
Obi-Wan: Only a Sith could be an absolutist.

On Jun 23, 2005, at 3:10 PM, Rene Ott wrote:


 It is hard to describe a usage pattern cause I don't have a  
 concrete one.
 The questions are meant in a more general way which management  
 technique is
 better scalable.

 An example for a GUI using the

 DBwrite --- read db with perl and write new conf files --- reload

 process is the Asterisk Management Portal. It uses MySQL. Lets say  
 it is
 used in a company and the 1000 clients consist mostly of SIP  
 clients and
 some FAXes, and some ISDN phones.
 And I am asking myself if it is able to manage the lots of clients  
 in a good
 way (means one doesn't have to invest in lots of hardware to handle  
 the
 load). Or maybe it is better to use the realtime technique ?

 René

 Von: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Im Auftrag von  
 snacktime
 Gesendet: Donnerstag, 23. Juni 2005 20:46
 An: Asterisk Users Mailing List - Non-Commercial Discussion
 Betreff: Re: [Asterisk-Users] Management: Reload performace 
 Realtimeperformance ?



 I was reading around in the mailing lists and people say reloading is
 stable. Now this tool has to manage 1000 clients so the conf files  
 are
 quite big and reloading needs some time. What happens if a call  
 comes in
 during that reload time ?
 How is the performance in general of the process described above
 (assumed the used hardware is not under- and not overdimensioned),  
 can
 such a tool easily handle 1000 clients ?
 Does somebody use a similar tool with many clients ?


 This really depends on your usage patterns and a lot of other things
 like what database you are using and how it is configured, etc..   You
 probably need to list a lot more details about what it is you are
 trying to do before you will get an answer.

  I haven't had the time to test what happens when you reload a large
 static database, but I'm guessing it would load everything from the
 database first, then when it replaces what's in memory it only takes a
 second or so.



 Somewhere in the mailing lists someone said that the realtime uses  
 many
 database queries. If there are also 1000 clients to manage, this  
 should
 lead to lots of database queries.


 That's only for the realtime extensions.

 Chris
 ___
 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

[Asterisk-Users] BRIstuff/QuadBRI problem: Ring requested on unconfigured channel 255/255 span 5

2005-06-24 Thread Bruno . Voigt

Hi all,

I'm running a stable Asterisk on a HP DL380G2 1.4Ghz 0,5GB RAM
equipped with 1x TE410P and 2xJunghanns QuadBRI running in NT-mode.

Connected to the BRI-Ports are 12 Fax-Modems (Elsa MicroLink ISDN/TL V.34)
which are only operating in dial out analog mode to deliver fax messages.

After a while of running fine (50-200 dial out connections)
on some S0 spans the following message occurs over and over again:

chan_zap.c:8009 pri_dchannel: Ring requested on unconfigured channel 
255/255 span 5

The Modems connected to this span get NO DIALTONE for every ATD.
Modems on other spans continue to operate.

This error seems to appear mostly on  span 5 and 6.

After restarting asterisk everything is okay again for a while.

Any hints about what is going on here are greatly appreciated ;-)

TIA, Bruno - bruno @ ic3s.de

Connected to Asterisk 1.0.7-BRIstuffed-0.2.0-RC8g currently running on pbx 
(pid = 20305)
Verbosity is at least 5
pbx*CLI show channels
Channel  (ContextExtensionPri )   State Appl. Data
   Zap/25-1  (pri1   s1   )  Up Bridged Call 
Zap/128-1
  Zap/128-1  (from-s0-faxmodems 00711xxx 5   )  Up Dial 
Zap/r1/0711xxx
   Zap/24-1  (pri1   s1   )  Up Bridged Call 
Zap/132-1
  Zap/132-1  (from-s0-faxmodems 00242xxx  5   )  Up Dial 
Zap/r1/0242xxx
4 active channel(s)
Jun 24 11:49:19 WARNING[20305]: chan_zap.c:8009 pri_dchannel: Ring 
requested on unconfigured channel 255/255 span 5
  == Primary D-Channel on span 6 down for TEI 64
  == Primary D-Channel on span 6 up for TEI 64
-- Accepting overlap voice call from '' to '00394' on channel 0/2, 
span 6
-- Starting simple switch on 'Zap/129-1'
-- Channel 0/2, span 7 got hangup
-- Hungup 'Zap/24-1'
  == Spawn extension (from-s0-faxmodems, 00242xxx, 5) exited non-zero on 
'Zap/132-1'
-- Hungup 'Zap/132-1'
-- Executing SetCallerPres(Zap/129-1, prohib) in new stack
-- Executing NoOp(Zap/129-1, ) in new stack
-- Executing SetTransferCapability(Zap/129-1, 3K1AUDIO) in new 
stack
-- Setting transfer capability to: 0x10 - 3K1AUDIO.
-- Executing SetCIDNum(Zap/129-1, 0410612345) in new stack
-- Executing Dial(Zap/129-1, Zap/r1/039) in new stack
-- Requested transfer capability: 0x10 - 3K1AUDIO
-- Called r1/039
-- Zap/26-1 is ringing
-- Zap/26-1 answered Zap/129-1
-- Attempting native bridge of Zap/129-1 and Zap/26-1
Jun 24 11:49:35 WARNING[20305]: chan_zap.c:8009 pri_dchannel: Ring 
requested on unconfigured channel 255/255 span 5
  == Primary D-Channel on span 7 down for TEI 65
  == Primary D-Channel on span 7 up for TEI 65
___
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] polycom soundpoint ip 300

Hello,

if somebody is interested in Europe for 2 polycom
soundpoint ip 300 for testing with Asterisk contact me
out of list .

Regards
Harry






___ 
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger 
Téléchargez cette version sur http://fr.messenger.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] BRIstuff/QuadBRI problem: Ring requested on unconfigured channel 255/255 span 5

2005-06-24 Thread Klaus-Peter Junghanns

Hi,

can you please post the output of zap show channel for all channels
of an affected span? It seems that asterisk thinks that all B channels
are still in use. So i suspect some problem with call clearing.

best regards

Klaus
--
Klaus-Peter Junghanns

Am Freitag, den 24.06.2005, 12:07 +0200 schrieb [EMAIL PROTECTED]:
 Hi all,
 
 I'm running a stable Asterisk on a HP DL380G2 1.4Ghz 0,5GB RAM
 equipped with 1x TE410P and 2xJunghanns QuadBRI running in NT-mode.
 
 Connected to the BRI-Ports are 12 Fax-Modems (Elsa MicroLink ISDN/TL V.34)
 which are only operating in dial out analog mode to deliver fax messages.
 
 After a while of running fine (50-200 dial out connections)
 on some S0 spans the following message occurs over and over again:
 
 chan_zap.c:8009 pri_dchannel: Ring requested on unconfigured channel 
 255/255 span 5
 
 The Modems connected to this span get NO DIALTONE for every ATD.
 Modems on other spans continue to operate.
 
 This error seems to appear mostly on  span 5 and 6.
 
 After restarting asterisk everything is okay again for a while.
 
 Any hints about what is going on here are greatly appreciated ;-)
 
 TIA, Bruno - bruno @ ic3s.de
 
 Connected to Asterisk 1.0.7-BRIstuffed-0.2.0-RC8g currently running on pbx 
 (pid = 20305)
 Verbosity is at least 5
 pbx*CLI show channels
 Channel  (ContextExtensionPri )   State Appl. Data
Zap/25-1  (pri1   s1   )  Up Bridged Call 
 Zap/128-1
   Zap/128-1  (from-s0-faxmodems 00711xxx 5   )  Up Dial 
 Zap/r1/0711xxx
Zap/24-1  (pri1   s1   )  Up Bridged Call 
 Zap/132-1
   Zap/132-1  (from-s0-faxmodems 00242xxx  5   )  Up Dial 
 Zap/r1/0242xxx
 4 active channel(s)
 Jun 24 11:49:19 WARNING[20305]: chan_zap.c:8009 pri_dchannel: Ring 
 requested on unconfigured channel 255/255 span 5
   == Primary D-Channel on span 6 down for TEI 64
   == Primary D-Channel on span 6 up for TEI 64
 -- Accepting overlap voice call from '' to '00394' on channel 0/2, 
 span 6
 -- Starting simple switch on 'Zap/129-1'
 -- Channel 0/2, span 7 got hangup
 -- Hungup 'Zap/24-1'
   == Spawn extension (from-s0-faxmodems, 00242xxx, 5) exited non-zero on 
 'Zap/132-1'
 -- Hungup 'Zap/132-1'
 -- Executing SetCallerPres(Zap/129-1, prohib) in new stack
 -- Executing NoOp(Zap/129-1, ) in new stack
 -- Executing SetTransferCapability(Zap/129-1, 3K1AUDIO) in new 
 stack
 -- Setting transfer capability to: 0x10 - 3K1AUDIO.
 -- Executing SetCIDNum(Zap/129-1, 0410612345) in new stack
 -- Executing Dial(Zap/129-1, Zap/r1/039) in new stack
 -- Requested transfer capability: 0x10 - 3K1AUDIO
 -- Called r1/039
 -- Zap/26-1 is ringing
 -- Zap/26-1 answered Zap/129-1
 -- Attempting native bridge of Zap/129-1 and Zap/26-1
 Jun 24 11:49:35 WARNING[20305]: chan_zap.c:8009 pri_dchannel: Ring 
 requested on unconfigured channel 255/255 span 5
   == Primary D-Channel on span 7 down for TEI 65
   == Primary D-Channel on span 7 up for TEI 65
 ___
 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] chan_capi-cm-0.5.1 and CLIR

2005-06-24 Thread Armin Schindler

On Fri, 24 Jun 2005, asterisk_on_oelf wrote:
 Hi,
 
 I have updated from chan_capi-0.3.5 to 0.5.1, but can't get CLIP/CLIR to work.
 
 with 0.3.5 CLIR works fine:
 exten = _1X.,1,Dial(CAPI/@12345:${EXTEN:1},180)
 
 
 with 0.5.1 I tried as described in the README:
 exten = _1X.,1,CallingPres(32)
 exten = _1X.,2,Dial(CAPI/contr1/${EXTEN:1},180)
 
 but it doesn't work. My first MSN is always shown.
 
 Where is my mistake?

This is the correct way. I took this change from 0.4.0pre1 and I have to 
admit I did not test it yet. Maybe there is a bug, but the code looks good 
so far.
Can you please provide a verbose log of level 5 with 'capi debug' 
(at least the CONNECT_REQ messages) to see if the correct presentation value 
is set.

Armin
___
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] Whole configuration for SMS

2005-06-24 Thread Gireesh Hariharasubramani









Hi,



 Could I get the whole configuration details
regarding the SMS application?



Thanks,

H.Gireesh






___
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] Voicemail

2005-06-24 Thread MDM



I am trying to setup voicemail for my iaxy device, however,
i cannot get it to work voicemail never picks up. Below is my config.
Am i doing anything wrong here

From my Extensions.conf file
exten = 7403,1,Dial(IAX2/u7403/s)
exten = 7403,2,Voicemail(u7403)
exten = 7403,102,Voicemail(b7403)
exten = 7403,103,Hangup

From my voicemail.conf
[telx.com]
7001 = 7001
7002 = 7002
7402 = 7403
___
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] Voicemail

2005-06-24 Thread jurczak

What is your default context in the voicemail.conf
Maybe you should try Voicemail([EMAIL PROTECTED])


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of MDM
Sent: Thursday, June 23, 2005 11:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Voicemail


I am trying to setup voicemail for my iaxy device, however,
i cannot get it to work voicemail never picks up. Below is my config.
Am i doing anything wrong here

 From my Extensions.conf file
exten = 7403,1,Dial(IAX2/u7403/s)
exten = 7403,2,Voicemail(u7403)
exten = 7403,102,Voicemail(b7403)
exten = 7403,103,Hangup

 From my voicemail.conf
[telx.com]
7001 = 7001
7002 = 7002
7402 = 7403
___
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] Voicemail

2005-06-24 Thread Peter Bowyer

On 23/06/05, MDM [EMAIL PROTECTED] wrote:
 
 I am trying to setup voicemail for my iaxy device, however,
 i cannot get it to work voicemail never picks up. Below is my config.
 Am i doing anything wrong here
 
  From my Extensions.conf file
 exten = 7403,1,Dial(IAX2/u7403/s)
 exten = 7403,2,Voicemail(u7403)
 exten = 7403,102,Voicemail(b7403)
 exten = 7403,103,Hangup
 

Please check the documentation for the 'dial' command at

http://www.voip-info.org/wiki-Asterisk+cmd+Dial

and note that unless you specify a timeout value, the channel will
ring indefinitely. Which is the same answer as you got last time you
asked this question in the past few hours.

Peter

-- 
Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[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] Using 2 x DSL

2005-06-24 Thread Remco Barende


How about using SER to balance the load for outgoing calls?



On Thu, 23 Jun 2005, jltaylor wrote:


You can't really do true bonding unless you control both ends of the link.
I had a customer who tried this.

It's easy to do with ATM and IMA interfaces on T1/T3 type stuff.

The $300-$1000 dual wan routers will not work off the shelf.

Policy based routing helped but it's tough to make it work.

Now, what you can do is put the Asterisk on ONE network and use policy based
routing to share other stuff like surfing, smtp, telnet, etc. You can
prioritize the traffic so that the packets to and from the Asterisk are
mangled to have the higher priority.

If both DSL's are for Asterisk ONLY then you might try round-robin DNS or
manually setup traffic.

Asteris will work on multiple LAN's - I have both a PUBLIC and PRIVATE ip in
the same box on different NIC's. Just set your routing.

James Taylor
MetroTel
3505 Summerhill Road
Suite 11
Texarkana, Tx  75503
903-793-1956


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of VoIP-PBX
Sent: Thursday, June 23, 2005 1:46 PM
To: Jorge Carrasquillo; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: [Asterisk-Users] Using 2 x DSL


Hi all, my client wants to double his bandwidth by using 2 x DSL lines
into one Asterisk network
How can I do this ?

Thanks

Henry
___
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 to setup two Asterisk boxes - keeping theregistration

2005-06-24 Thread Haydn.Kemmery

You'll need to create a trunk between the two systems 
Then configure your out bound routing to use that trunk

Eg if you use 1 as the prefix for the trunk

BOX A Ext200 will call box B EXT201 by dialing 1201 



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ronald
Wiplinger
Sent: Sunday, 19 June 2005 12:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] How to setup two Asterisk boxes - keeping
theregistration

I have now two asterisk boxes running, one on the IP *18 and one on IP
*20

Both are working, use the same dialplan and realtime.

I did not find out how to set it up that if a phone is registered in *18

can reach a phone registered in *20.

For the wake up call I also see some troubles, since both machines point

to the same NFS space. It could be that both machines start the wakeup
call.

Has anybody solved that?


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

This email and any files transmitted with it are confidential and
intended solely for the use of the individual or entity to whom
they are addressed.
If you have received this email in error please notify the
originator of the message. This footer also confirms that this
email message has been scanned for the presence of computer viruses.

Any views expressed in this message are those of the individual
sender, except where the sender specifies and with authority,
states them to be the views of LMC.



___
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] Re: Cisco 7960 firmware upgrade promblems

2005-06-24 Thread Tom Rymes

For what it is worth, this is not what I did. Since you have already
upgraded two phones to 7.4, I will assume that you know how to do that
properly. Actually, a quick run-down:

1.) install SIP v5.x by specifying the image name in the OS79XX.txt file
and the SIPMAC.cnf file. The file name should be the same in both
files.
2.) Upgrade to SIP v7.0 (This step might be unneccessary, but I'm not
sure). You need to put P003-07-0-0 in the OS79XX.txt file, and
P0S3-07-0-0 in the SIPMAC.cnf file. Make sure that you have the
P0S3-07-0-00.loads file in the tftpboot directory, too!
3.) Now load the v7.4 firmware by specifying P003-07-4-0 in the
OS79XX.txt file, and P0S3-07-4-0 in the SIPMAC.cnf file. Make sure
that you have the P0S3-07-4-00.loads file in the tftpboot directory,
too!

One note to add, make sure that you copy the .zip file to the asterisk
box first, thenunzip it. Do not unzip it on your PC and copy the files
individually. Also double check permissions on the files (though again,
you have had no problems with the othe rtwo phones.

If you can't get to the settings portion of the phone to manually
specify the TFTP server and the Network address, I would recommend that
you try downgrading the firmware. It's possible that the phone has the
correct settings, but the upgrade is failing.

Failing all else, you could install a DHCP server that specifies the
TFTP server to the phone so that it gets the right address from the DHCP
server.

Tom

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of
 Tarpo, Louie
 Sent: Thursday, June 23, 2005 7:29 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Re: Cisco 7960 firmware upgrade
 promblems


 If you've followed the advice of previous posters, delete
 your 7-4 firmware and re-extract it.  You should NOT rename it.

 Your OS79XX.txt should contain
 P003-07-3-00

 and your SIPMAC.cnf should contain
 # SIP Configuration Generic File (start)
 image_version: P0S3-07-4-00

 The P003-07-3-00 is the loader file
 the loader file loads the actual image which is the P0S3-07-4-00

 Louie


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of
 Patrick Lidstone (Personal E-mail)
 Sent: Thursday, June 23, 2005 3:30 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Re: Cisco 7960 firmware upgrade promblems



  Make sure that you have done the following:
 
  1.) Set up the phone to use DHCP to get an address *or* manually
  configured an e-mail address using the settings on the phone.
 
  2.) Set the DHCP server to give out the correct TFTP server address,
  *or* configure Alternate TFTP Server = yes and manually
 specify the
  server address.
 
  I assume that you have already done that, but you never know!
 
  Tom

 Hi Tom, the problem is that the Universal Application Loader
 has never successfully loaded a firmware image, so there is
 no way to set these options manually. The phone is definitely
 doing something with DHCP, but never generates a TFTP request
 - apparently?

 Patrick

 ___
 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] Dial peer preference

2005-06-24 Thread kurt x

Does Asterisk support preference for the dial peers.  

For example:

I have two outbound peers in *.  The first is a SIP dial peer and the
second peer is to
the PSTN via a T1.

The SIP dial peer is the main outbound peer for all calls. However, if
the my SIP providers network goes down, I need to be able to
automatically route the call out the T1 card.  Is this
possible in Asterisk.  I have not seen any preference commands for Asterisk.

If not, is there a work around for this type of set up.

Kurt
___
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] FOP related questions

2005-06-24 Thread Daniel ANDRE


Seth Remington a écrit :


On Wed, 2005-06-22 at 15:58 +0200, Daniel ANDRE wrote:
 


Hello,

I have downloaded and installed Flash Operator Panel. version 0.21. It 
works pretty well and I have some questions about it.


1. The text label of the buttons are partially hidden by their icons. Is 
there a way to adjust right margin for the buttons?
   




Take a look at the op_style.cfg file. You'll probably find something to
adjust in there that will help your issue.
 


This was the fist p lace I looked but found no clue on how to do that

2. I would like to have the fop brought in the front of screen whenever 
and extension rings. Sort of crm feature but with fop and not another 
url. Is there a way to do that?
   




Not out of the box that I'm aware of. Would probably require a change to
the flash code.
 


I d on't know how to change it. Do you have any idea?

3. This question is notre directly related to fop but you may have the 
answer. I would like to have fop panel in tis own windows (no toolbar, 
menu, title, ...) either with FireFox and Internet Explorer. Any Idea?
   




With firefox you can turn off the navigation and bookmarks tool bar and
set it to run full screen. Make sure you have the Hide the tab bar when
only one web site is open option selected in the preferences. Maybe not
exactly what you were looking for.
 

This is not really what I am looking for. I'd like to have some link or 
shortcut that opens my browser with no menu or toolbar.


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] Errors on SuSE 9.3 default install.

2005-06-24 Thread Zoltan Szecsei


Hi,
Total newbie - I spent most of yesterday looking through various docs 
and could not come up with anything useful.


I have 2 different SuSE 9.3 boxes - a P3 with 2 * ISDN BRI pci cards and 
a P4 with no asterisk relevant HW at all.


I am using the default asterisk install that was loaded when I did a 
(full) fresh install of SuSE 9.3.


I get the *same* error on *both* boxes, even though one of them does not 
have any ISDN HW at all.


The error is:

[app_capiCD.so]Jun 24 14:33:42 WARNING[1921]: loader.c:258 
ast_load_resource: /usr/lib/asterisk/modules/app_capiCD.so: undefined 
symbol: capidebug
Jun 24 14:33:42 WARNING[1921]: loader.c:440 load_modules: Loading module 
app_capiCD.so failed!



I've searched for docs relating to how asterisk loads (and therefore 
what makes it load certain modules) and couldn't find anything on the 
voip-info.org/wiki-Asterisk.


Any help would be welcome.

TIA,
Zoltan

___
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] wcte11xp hardlock problem

2005-06-24 Thread John Mylchreest

Hi,

Can anyone help at all. I am running an Asus Pundit-R with a TE100P (1
port PRI) card, and experience hardlock problems. I believe the problem
is with interrupt/apic although I cannot get very far with the true
diagnosis :)

Is anyone here running the same setup? Perhaps similar, and have the
same kind of problems?

Regards,
John
___
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 server with remote monitoring capabilities

2005-06-24 Thread Rich Adamson

 I'm tired of having to drive out to the colocation
 facility each time my dedicated asterisk server craps
 out, just to press the button to do a hard reboot.
 (I'm running 1.05 stable at present, no telephony
 hardware, as this is mainly a system that receives
 calls, no dial-out ability is needed.) 

Then fix the root-cause. Rebooting a box is not a fix. There
are plenty of uptime examples in the months/years timeframes.


___
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] French Audio Files

2005-06-24 Thread Dave Cotton

On Fri, 2005-06-24 at 12:56 +0400, Jean-Michel Hiver wrote:

 The sound set I had was Canadian French. Appuyez sur la touche carré 
 type messages.

I know, I thought that, I commented on the list about a year ago and was
assured that she came from Nice, I can't say too much because my accent
and sometimes my construction also comes out like Garou. Perhaps it's
living in New Zealand that's changed her accent and that the sound set
was aimed at Canada with a direct translation of the American.  

Pound in US English, Hash in English.


-- 
Dave Cotton [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] FOP related questions

2005-06-24 Thread Tony Hoyle


Daniel ANDRE wrote:

This is not really what I am looking for. I'd like to have some link or 
shortcut that opens my browser with no menu or toolbar.


The old netscape builds used to have a '-kiosk' flag that did that.

With some versions of IE you can run 'iexplore -k'.

There's no way I can see in firefox without modifying the browser 
itself, unfortunately (once inside the browser you can hit F11 but that 
doesn't remove the address bar).


Tony
___
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] Errors on SuSE 9.3 default install.

2005-06-24 Thread Doug Lytle


Zoltan Szecsei wrote:


Hi,
Total newbie - I spent most of yesterday looking through various docs 
and could not come up with anything useful.


I've searched for docs relating to how asterisk loads (and therefore 
what makes it load certain modules) and couldn't find anything on the 
voip-info.org/wiki-Asterisk.



Look at the modules.conf in the Asterisk directory.

Doug

___
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] Dial peer preference

2005-06-24 Thread Damon Estep

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of kurt x
 Sent: Friday, June 24, 2005 6:24 AM
 To: Asterisk
 Subject: [Asterisk-Users] Dial peer preference

 Does Asterisk support preference for the dial peers.

 For example:

 I have two outbound peers in *.  The first is a SIP dial peer and the
 second peer is to
 the PSTN via a T1.

 The SIP dial peer is the main outbound peer for all calls. However, if
 the my SIP providers network goes down, I need to be able to
 automatically route the call out the T1 card.  Is this
 possible in Asterisk.  I have not seen any preference commands for
 Asterisk.

 If not, is there a work around for this type of set up.

 Kurt

Have you tried putting in something like this?

Etxen s,1,Dial(sip/[EMAIL PROTECTED],duration)
Exten s,2,Dial(zap/chan/number,duration)
Exten s,3,Congestion(5)
Exten s,4,(hangup)
___
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 Zoom x5v 5565

2005-06-24 Thread Geoff Manning

 I trying to obtain some information relation to implement
 Zoom x5v 5565 and Asterisk

What exactly are you trying to do? Are you trying to use Asterisk with the
Global Village service? I assume you have the X5v up and running and
providing internet access.
___
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 server with remote monitoringcapabilities

2005-06-24 Thread Matt Schulte

I would have to agree, an IAD locking up is bad either way you look at
it. Even if you're there to reboot it on demand, it takes nearly 5
minutes to come back up. What kind of servers are they? What kind of
phones? In all honesty, none of our IAD's ever lock up. And ones that
did were defective and replaced.

-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED] 
Sent: Friday, June 24, 2005 8:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk server with remote
monitoringcapabilities

 I'm tired of having to drive out to the colocation
 facility each time my dedicated asterisk server craps
 out, just to press the button to do a hard reboot.
 (I'm running 1.05 stable at present, no telephony
 hardware, as this is mainly a system that receives
 calls, no dial-out ability is needed.)

Then fix the root-cause. Rebooting a box is not a fix. There are plenty
of uptime examples in the months/years timeframes.

___
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] realtime sip confusion

2005-06-24 Thread Matthew Boehm


snacktime wrote:

Trying to use realtime sip for the first time, and it's not working as
expected.
I have one user entry in the sip database.  Everthing else is still in
sip.conf.  When I get an incoming call, this is the database query:

SELECT * FROM ast_home_sip_realtime WHERE name = '+18003859169'

The 800# is the caller id of the caller, which doesnt' make any sense to me.

Is there any documentation about how realtime sip/iax actually work
beyond just the schema's that are on the wiki?

Chris


Seems to me that your UA is sending that number as its SIP Username.

You can look in /var/log/asterisk/debug for lots of RealTime info if 
using res_config_mysql.


-Matthew

___
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] SendText

2005-06-24 Thread Christian Hiller


Hello,

i dont get this feature, how can i send a text to a certain SIP-phone 
that support this kind of messaging. The WIKI shows an example, but it 
shows how the receiving phone got to make a call to receive a message.


Thx for a hint! :)


___
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] voicemail

2005-06-24 Thread Michael Di Martino

Ok I have added the timeout value but it still does not pick. However
jus to test voicemail function
I comment out the first line and voice does pick up. What could be
wrong.

exten = 7403,1,Dial(IAX2/u7403/1/5)
exten = 7403,2,Voicemail(u7403)
exten = 7403,102,Voicemail(b7403)
exten = 7403,103,Hangup

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Seth
Remington
Sent: Friday, June 24, 2005 12:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] voicemail

On Thu, 2005-06-23 at 23:19 -0400, Michael Di Martino wrote:
 I am trying to setup voicemail for my iaxy device, however, i cannot 
 get it to work voicemail never picks up. Below is my config.
 Am i doing anything wrong here
 
 From my Extensions.conf file
 exten = 7403,1,Dial(IAX2/7403/10)

You did not specify a timeout in the dial command. Change it to:
exten = 7403,1,Dial(IAX2/7403/10,xx) --- where xx is the number of
seconds you want the Dial command to attempt to connect the call before
it returns and proceeds to the next priority (i.e. voicemail).

 exten = 7403,2,Voicemail(u7403) 
 exten = 7403,102,Voicemail(b7403) 
 exten = 7403,103,Hangup 
 
 From my voicemail.conf 
 [telx.com] 
 7403 = 7403 
 
 
 Thanks 
 Mike

Hope that helps.

-Seth



___
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] Errors on SuSE 9.3 default install.

2005-06-24 Thread Zoltan Szecsei


Doug Lytle wrote:


Zoltan Szecsei wrote:


Hi,
Total newbie - I spent most of yesterday looking through various docs 
and could not come up with anything useful.


I've searched for docs relating to how asterisk loads (and therefore 
what makes it load certain modules) and couldn't find anything on the 
voip-info.org/wiki-Asterisk.



Look at the modules.conf in the Asterisk directory.

Doug

___
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



Hi Doug,

Thanks for the thought - but I neglected to say that I have already 
edited that. The results were identical barring the timestamps.


I just took a flyer at changing:

;zls noload = chan_alsa.so

because I have on-board sound and, in the [global] section:

;zls chan_capi.so=yes


What does intrigue me is, at the top, under [modules] is says: autoload=yes

Maybe thats the part I need to understand better.

TIA,
Zoltan.


___
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] RTP session between two end users

2005-06-24 Thread Erdem HAKİ









Is it possible that a RTP session between two end users  (so i want to
use asterisk as a signaling proxy and bypass RTP sessions)?



I used canreinvite=yes but it didnt work. 







Description from asterisk conf. File;

(canreinvite=yes   
; allow RTP voice traffic to bypass Asterisk)







Thanks



Erdem HAKI  [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] French Audio Files

2005-06-24 Thread Jean-Michel Hiver



The sound set I had was Canadian French. Appuyez sur la touche carré 
type messages.
   



I know, I thought that, I commented on the list about a year ago and was
assured that she came from Nice, I can't say too much because my accent
and sometimes my construction also comes out like Garou.

Well it's great to have some free french sound files already, but I 
wouldn't mind an extra set. Maybe I should hire my girlfriend to produce 
a set of 'creole french' sounds, that'd be fun: si ou vé ékout lo 
messaj syvan, press sis :)




Perhaps it's
living in New Zealand that's changed her accent and that the sound set
was aimed at Canada with a direct translation of the American.  


Pound in US English, Hash in English.
 

Pound = carré = direct translation? Mind you, appuyez sur la touche 
livre would be worse :)


Cheers,
Jean-Michel.

___
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 server with remote monitoringcapabilities

2005-06-24 Thread Chris Mason (Lists)




I'm tired of having to drive out to the colocation
facility each time my dedicated asterisk server craps
out, just to press the button to do a hard reboot.
(I'm running 1.05 stable at present, no telephony
hardware, as this is mainly a system that receives
calls, no dial-out ability is needed.)
   

I have never had a pbx lock up. I suggest you change your hardware. I 
would think your problem is RAM, as Asterisk is not hard on the drives 
or other hardware.


--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [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] voicemail

2005-06-24 Thread Robert Webb



On Fri, 24 Jun 2005 09:27:45 -0400
 Michael Di Martino [EMAIL PROTECTED] wrote:
Ok I have added the timeout value but it still does not 
pick. However

jus to test voicemail function
I comment out the first line and voice does pick up. 
What could be

wrong.

exten = 7403,1,Dial(IAX2/u7403/1/5)
exten = 7403,2,Voicemail(u7403)
exten = 7403,102,Voicemail(b7403)
exten = 7403,103,Hangup




AS someone esle suggested, go to www.voip-info.org and 
read about the dial command. It is obvious you haven't as 
your syntax in for the commands above are incorrect. 
Between the extension and the timeout, you use a , not 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] INBAND DTMF G729 ASTERISK

On Thursday 23 June 2005 19:57, Brian West wrote:
 With inband its at least not sent in clear text.

It's trivial to pull DTMF out of an inband stream too.  Perhaps not AS trivial 
but just the same, you should be using SRTP if you're paranoid about this 
kind of thing.

-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] INBAND DTMF G729 ASTERISK

On Thursday 23 June 2005 15:19, [EMAIL PROTECTED] wrote:
 Why don't Asterisk support inband DTMF with G729? Is  there a way to do
 that!?

If you figure out how to do it, I'm sure you will have a very captive and 
attentive audience to explain how you're sending continuous tones over a 
lossy voice codec.

It's just not possible to reliably send continuous tones (which include DTMF) 
over a lossy voice codec such as g729, sorry.  

-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] PRI auto reset?

2005-06-24 Thread Wilson Pickett

 The P4 is the one that won't behave. Any way to force it via config to reset 
 the PRI channels every so often?

Can't help you there, Bro, but I think we'd sound great topgether on
stage, ya dig?
___
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] Extensions Puzzle: Contexts Confligting with each other.

2005-06-24 Thread Joan Bautista

I have a setup for 2 companies. Both ones should have separate
schedules and differents menu options. here is an example:
[default]
include= company1
include= company2

[company1]
include = optionscompany1

[company2]
include = optionscompany2

[optionscompany1]

; option 0 Operator
exten = 0,1,Playback,CallTransfer_SP
exten = 0,3,SetCIDName,Operator
exten = 0,4,Goto,CMP1|1100|1

; option 1 Spanish
;exten = 1,1,SetLanguage(sp)
;exten = 1,2,Goto(day1|${CENTRAL1}|7))

; option 2 Addresss
exten = 2,1,Directory,default
exten = 2,2,Goto(day1|${CENTRAL1}|7))

; option 3
exten = 3,1,Playback,CallTransfer_SP
exten = 3,2,Goto(day1|${CENTRAL1}|7))


[optionscompany2]
; option 0 Operator
exten = 0,1,Playback,CallTransfer_SP
exten = 0,3,SetCIDName,Operator
exten = 0,4,Goto,CMP2|1100|1

; option 1 Spanish
;exten = 1,1,SetLanguage(sp)
;exten = 1,2,Goto(day2|${CENTRAL2}|7))

; option 2 Addresss
exten = 2,1,Directory,default
exten = 2,2,Goto(day2|${CENTRAL2}|7))

; option 3
exten = 3,1,Playback,CallTransfer_SP
exten = 3,2,Goto(day2|${CENTRAL2}|7))


The problem I have is that if a valid comes in for Company2 and the
caller select any available options it goes to the context for options
on Company1. Is there any way to correct or prevent this from
happening?
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] RTP session between two end users

2005-06-24 Thread Steve Clark


Erdem HAKÝ wrote:
Is it possible that a RTP session between two end users  (so i want to 
use asterisk as a signaling proxy and bypass RTP sessions)?


 


I used “canreinvite=yes” but it didn’t work.

 

 

 


Description from asterisk conf. File;

(canreinvite=yes; allow RTP voice traffic to bypass 
Asterisk)


 

 

 


Thanks

 


Erdem HAKI – [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

You have to make sure that you are not using t or T in the dial command.

___
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] Whole configuration for SMS

2005-06-24 Thread Wilson Pickett

 Could I get the whole configuration details regarding the SMS
 application? 

See the wiki (sorry I don't have a link handy)
___
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] French Audio Files

2005-06-24 Thread Time Bandit

 The sound set I had was Canadian French. Appuyez sur la touche carré
 type messages.
 
 If you know of other French sound sets, please let me know!
I would be interested in the French-Canadian set, where can I get them ?

From a French-Canadian view, the french prompts from sineapps sounds VERY bad.

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] More IP address in bindaddr directive

2005-06-24 Thread Kevin P. Fleming


Coufal Bohuslav wrote:


is it possible to bind SIP protokol not to all but to more that one
interfaces. I did try use bindaddr, but i don't know right syntax.


Not at this time, but the development branch will have support for this 
soon.

___
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] SendText

2005-06-24 Thread Jerry Geis


I used the outgoing spool directory, added a variable like TEXT=Hello world
and going to context send_my_text. tehn send_my_text has exten = 
s,1,SendText($TEXT)

Works great.

Jerry


---

Hello,

i dont get this feature, how can i send a text to a certain SIP-phone 
that support this kind of messaging. The WIKI shows an example, but it 
shows how the receiving phone got to make a call to receive a message.


Thx for a hint! :)

___
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] INBAND DTMF G729 ASTERISK

 On Thursday 23 June 2005 15:19, [EMAIL PROTECTED] wrote:
 Why don't Asterisk support inband DTMF with G729? Is  there a way to do
 that!?

 If you figure out how to do it, I'm sure you will have a very captive and
 attentive audience to explain how you're sending continuous tones over a
 lossy voice codec.

But there are some products that supports DTMF inband on G729. Ok, it will
not work in most cases(like everyone told) but why Asterisk dont support
it? Is this hardcoded, or is possible to try it out?

 It's just not possible to reliably send continuous tones (which include
 DTMF) over a lossy voice codec such as g729, sorry.

 -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] Management: Reload performace Realtimeperformance ?

2005-06-24 Thread Brian West


You'll have either blocked calls.. or choppy calls.

/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”

On Jun 24, 2005, at 5:02 AM, Rene Ott wrote:

This is interesting. But reloading the big conf files itself  
doesn't cause
any performance problems (with verbosity 0)? Do you have experience  
what

happens with incoming calls during reload?




If you have configs that big you'll need to set verbose 0 before
doing that.. you'll notice the calls start to jitter and all hell
breaks loose if you don't do so.



/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”

On Jun 23, 2005, at 3:10 PM, Rene Ott wrote:




It is hard to describe a usage pattern cause I don't have a
concrete one.
The questions are meant in a more general way which management
technique is
better scalable.

An example for a GUI using the

DBwrite --- read db with perl and write new conf files --- reload

process is the Asterisk Management Portal. It uses MySQL. Lets say
it is
used in a company and the 1000 clients consist mostly of SIP
clients and
some FAXes, and some ISDN phones.
And I am asking myself if it is able to manage the lots of clients
in a good
way (means one doesn't have to invest in lots of hardware to handle
the
load). Or maybe it is better to use the realtime technique ?

René

Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von
snacktime
Gesendet: Donnerstag, 23. Juni 2005 20:46
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [Asterisk-Users] Management: Reload performace 
Realtimeperformance ?





I was reading around in the mailing lists and people say  
reloading is

stable. Now this tool has to manage 1000 clients so the conf files
are
quite big and reloading needs some time. What happens if a call
comes in
during that reload time ?
How is the performance in general of the process described above
(assumed the used hardware is not under- and not overdimensioned),
can
such a tool easily handle 1000 clients ?
Does somebody use a similar tool with many clients ?




This really depends on your usage patterns and a lot of other things
like what database you are using and how it is configured, etc..
You

probably need to list a lot more details about what it is you are
trying to do before you will get an answer.

 I haven't had the time to test what happens when you reload a large
static database, but I'm guessing it would load everything from the
database first, then when it replaces what's in memory it only  
takes a

second or so.





Somewhere in the mailing lists someone said that the realtime uses
many
database queries. If there are also 1000 clients to manage, this
should
lead to lots of database queries.




That's only for the realtime extensions.

Chris
___
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



___
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] More IP address in bindaddr directive

2005-06-24 Thread Coufal Bohuslav

Thanks for info.

Bob.

Kevin P. Fleming píše v Pá 24. 06. 2005 v 09:43 -0500:
 Coufal Bohuslav wrote:
 
  is it possible to bind SIP protokol not to all but to more that one
  interfaces. I did try use bindaddr, but i don't know right syntax.
 
 Not at this time, but the development branch will have support for this 
 soon.
 ___
 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] INBAND DTMF G729 ASTERISK

 On Thursday 23 June 2005 19:57, Brian West wrote:
 With inband its at least not sent in clear text.

 It's trivial to pull DTMF out of an inband stream too.  Perhaps not AS
 trivial
 but just the same, you should be using SRTP if you're paranoid about this
 kind of thing.

We are on a real world... Every cyber cafe has its own little
hacker/cracker that is sniffing out... A simple ethereal capture could
give me a bank pin number... It is REALLY trivial!

Denis.

___
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 server with remote monitoring capabilities

2005-06-24 Thread Max W Blackmer Jr

  Original Message 
 Subject: Re: [Asterisk-Users] Asterisk server with remote monitoring
 capabilities
 From: beonice [EMAIL PROTECTED]
 Date: Thu, June 23, 2005 7:52 pm

 --- Michael Welter [EMAIL PROTECTED] wrote:

  William Boehlke wrote:
   Dell sells a remote management card for under $400
  that enables remote
   reboots. I know there are others out there but
  have no experience with them.

   William Boehlke
   Signate

   -Original Message-
   From: [EMAIL PROTECTED]
   [mailto:[EMAIL PROTECTED]
  On Behalf Of beonice

   I'm tired of having to drive out to the colocation
  facility each time my
   dedicated asterisk server craps out, just to press
  the button to do a hard
   reboot.
   (I'm running 1.05 stable at present, no telephony
  hardware, as this is
   mainly a system that receives calls, no dial-out
  ability is needed.)

  APC makes a power strip with a web server.  Each
  socket has its own IP
  address.  All you have to do to power cycle is
  access the IP address
  from your web browser and give the power cycle
  command.  It is sooo cool.

 Thanks for your responses, folks.

 Okay, so what makes more sense:
   1) a remote management card that will let me
 actually log in to the machine to monitor it as well
 as to reboot it
 vs.
   2) a remote-accessible powerstrip that will allow me
 to remotely reboot the server?

A little note. make sure your server motherboard/bios supports power on
after power loss to use the remote control power strip. Secondly make
sure the power strip control uses SSH and NOT telnet to control it.
Telnet is too insecure because passwords are sent plain text.

Another possibility is to write a reboot script and set up a cron job to
automatically reboot every night until you solve the bigger problem of
why is the server having problems?

With Linux their is little need to reboot Linux. There is only one time
that you have to reboot Linux. When you upgrade the kernel or its
modules. Kernel modules do not always need a reboot. Kernel module that
do require a reboot are critical to operation of your system for example
RAID# .

The best way is to have a script that uses the init script to restart
the applications that are questionable on a cron job schedule for low
usage.  With a good script you could also check on the status of the
service and perform functional test of the service. Then the script
would perform the necessary tasks to recover from application failure. 
This wont help with a total system failure as the script will not work.
Some of the remote monitoring cards can detect a system lockup and
preform a system reboot automatically.  When all of these fail you can
use remote control power strips or a KVM (Keyboard Video Mouse) over IP
to remotely control the hardware as if you are there.  Cyclades
(www.cyclades.com) sells both KVM and Remote Power management solutions
that are secure. They even have RSA authentication tokens and a
Biometric/RSA token authentications for secure management of the remote
locations.

Cheers,

Max W. Blackmer, Jr.
Consultant, Knowledge Power IT

___
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] INBAND DTMF G729 ASTERISK

On Friday 24 June 2005 10:58, [EMAIL PROTECTED] wrote:
 But there are some products that supports DTMF inband on G729. Ok, it will
 not work in most cases(like everyone told) but why Asterisk dont support
 it? Is this hardcoded, or is possible to try it out?

Asterisk can do it too, it's just not reliable on any platform.  Set 
dtmfmode=inband and use the g729 codec; that's all there is to it.  You will 
be disappointed 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] Asterisk 1.0.8 Released

2005-06-24 Thread Iqbal


does ast_data have apatch for this release

iqbal

Russell Bryant wrote:


Greetings!

Version 1.0.8 has been released of Asterisk, Asterisk-addons, Zaptel, 
and Libpri.  This release contains a significant amount of bug fixes 
(possibly the most of the 1.0.X releases).  Tarballs are available on 
the asterisk web site as well as the asterisk ftp server.


A complete list of all changes made to the v1-0 branch is available 
through the archives of the cvs mailing list.  See 
http://lists.digium.com for more information.


ChangeLogs that represent an overview of the larger changes are 
available in the source, as well as the following web site for 
convenience - http://dev.asteriskdocs.org/.


Thanks for your support,

Russell Bryant
___
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] SendText

2005-06-24 Thread Christian Hiller

Hmm, thx for answering... i understand the idea with the spool 
directory..When somebody wants to send a message, the variable in spool 
directorry gets set to message-text.


But how can i use the new context, e.g. i want to sent Hello to 
sip:[EMAIL PROTECTED] ?


Christian


Jerry Geis schrieb:

I used the outgoing spool directory, added a variable like TEXT=Hello 
world
and going to context send_my_text. tehn send_my_text has exten = 
s,1,SendText($TEXT)


Works great.

Jerry


---

Hello,

i dont get this feature, how can i send a text to a certain SIP-phone 
that support this kind of messaging. The WIKI shows an example, but it 
shows how the receiving phone got to make a call to receive a message.


Thx for a hint! :)

___
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] INBAND DTMF G729 ASTERISK

On Friday 24 June 2005 11:02, [EMAIL PROTECTED] wrote:
 We are on a real world... Every cyber cafe has its own little
 hacker/cracker that is sniffing out... A simple ethereal capture could
 give me a bank pin number... It is REALLY trivial!

And this is different exactly HOW with inband DTMF??  They can do the EXACT 
same thing!  If you want security don't use VOIP unless it's encrypted and/or 
over a VPN.  It's really that simple.

Please don't lecture me on the real world because while it's obvious that 
you are intelligent, you are also a little naive as to the application and 
difficulty level of sniffing VOIP traffic.

-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] Loosing hair on connecting Panasonic PBX- * - Euroisdn Italy

2005-06-24 Thread Emanuele Pucciarelli

Robert Rozman wrote:

 I'm pulling my hair down and getting bold :-) . I have Asterisk between
 Panasonic KXTD816 and Euroisdn in Italy (beronet octobri and bristuff
 Asterisk)

(hint: spend the extra $$ and support who's written the software!)

 I'm trying to do just plain transfer of call from pbx to ISDN through
 Asterisk...

I'm doing that without any problems via normal HFC-S PCI A cards with
Samsung PBX's.

 It seems like PBX hangsup, when call is progressing with no apparent
 reason.
 I'd kindly ask for any advice or some working example for this

Would you mind checking if Layer 1 is UP (cat /proc/zaptel/*) and
reporting bri debug span ... traces?

 On isdn side I also have a problem. Asterisk quite often says that it
 cannot create ZAP channel, although partticular span is reported up and
 active. I've also tried to connect loop between NT and TE port and
 call doesn't get through

So it looks like it does not depend on the Panasonic gear!

-- 
Emanuele
___
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] INBAND DTMF G729 ASTERISK

2005-06-24 Thread Julio Arruda


[EMAIL PROTECTED] wrote:

On Thursday 23 June 2005 19:57, Brian West wrote:


With inband its at least not sent in clear text.


It's trivial to pull DTMF out of an inband stream too.  Perhaps not AS
trivial
but just the same, you should be using SRTP if you're paranoid about this
kind of thing.



We are on a real world... Every cyber cafe has its own little
hacker/cracker that is sniffing out... A simple ethereal capture could
give me a bank pin number... It is REALLY trivial!


I think the point(s) the others are trying to make:

1- It is not feasible to use inband in G.729 (or, as far as I know, any 
other compressed codec), and that is final. Other than that.


2- Out-of-band is as safe/unsafe as having the conversation recorded, 
including pin, by the hacker, if no encrypted voice path is being used. 
as others mentioned, DTMF tones would be very obvious in a trace 
(maybe someone may want to post an example). Remember, if the other end 
need to be able to regenerate the DTMF info, it MUST be present in the 
stream, so is as easy/hard as the other endpoint 'decoding' it.


PS: I seem to recall some Voice over data products that would upspeed to 
G.711, upon detecting of DTMF tones, this may have given someone the 
wrong impression, that the DTMF was being sent as G.729, when it was not 
in fact.

[], O-O




___
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] INBAND DTMF G729 ASTERISK

2005-06-24 Thread Julio Arruda


Andrew Kohlsmith wrote:

On Friday 24 June 2005 10:58, [EMAIL PROTECTED] wrote:


But there are some products that supports DTMF inband on G729. Ok, it will
not work in most cases(like everyone told) but why Asterisk dont support
it? Is this hardcoded, or is possible to try it out?



Asterisk can do it too, it's just not reliable on any platform.  Set 
dtmfmode=inband and use the g729 codec; that's all there is to it.  You will 
be disappointed though.


I think that IAX, as one example, won't allow this ? Have a faint memory 
of some error message when trying it (maybe was ILBC+iax ?)

___
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] format_base64.c released on pbxfreeware.org

2005-06-24 Thread Brian West

This module allows you to read and write base64 files with record and  
playback.  The resulting files are RFC-822 and able to drop/cat into  
a mailspool.


l8tr,
Brian

___
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] Exposing Zap Channels on Server A to be Used By Server B

2005-06-24 Thread Wiley Siler









Hello All,



I remember there is a way to use two Asterisk servers and
allow one to see a virtual trunk that makes it so server B can use the ZAP
channels on server A.

Does anyone know where I can find this? I am racking my
brain trying to remember the terminology.

It was like creating a 24 channel virtual T1 connection from
server B to Server A that allowed server B to not have any ZAP hardware.

Anyone know what I am talking about? I am searching the
Wiki now but not hitting



Setup: 

Server A has TMD lines and Voip.providers

Server B has only some extensions, needs to connect to
Server A and use its ZAP channels



Thanks,

Wiley








___
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] realtime sip confusion

2005-06-24 Thread snacktime

 
  Chris
 
 Seems to me that your UA is sending that number as its SIP Username.
 
 You can look in /var/log/asterisk/debug for lots of RealTime info if
 using res_config_mysql.


This was an incoming call via a DID.  I can call from any phone and
the query is always on the callerid.  Part of my problem is I'm
completely guessing on how sip realtime works, there is absolutely
nothing I can find that say's 'this is what sip realtime does in a
user/peer/friend context'.

Also there is a bug where if a context has a dash, realtime splits the
string on the dash and does two queries.  I don't know why it's
picking up the context's in the first place since I don't understand
the logic.  I do know that I have a couple of unique context names
such as 'from-teliax' or 'voicepulse-out', and in mysql I see realtime
making queries like the following:

SELECT * from sip where name = 'from'
SELECt * from sip where name = 'teliax'

Chris
___
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] Exposing Zap Channels on Server A to be Used By Server B

2005-06-24 Thread Colin Anderson




Iremember there is a way to use two 
Asterisk servers and allow one to see a virtual trunk thatmakes 
it so server B can use the ZAP channels on server A.

  
  You are looking for this:
  
  http://www.voip-info.org/tiki-index.php?page=Asterisk%20TDMoE
  
  It's 
  Layer 2 I think so you can only run it in the same subnet or with linux 
  bridging
  
  hth
___
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] INBAND DTMF G729 ASTERISK

On Friday 24 June 2005 11:59, Julio Arruda wrote:
 I think that IAX, as one example, won't allow this ? Have a faint memory
 of some error message when trying it (maybe was ILBC+iax ?)

IAX2 sends all DTMF tones out of band.  It's designed that way.

-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] INBAND DTMF G729 ASTERISK

2005-06-24 Thread Kevin P. Fleming


Julio Arruda wrote:

I think that IAX, as one example, won't allow this ? Have a faint memory 
of some error message when trying it (maybe was ILBC+iax ?)


IAX does not support inband DTMF for any codec.
___
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] flash panel only works with IP address

2005-06-24 Thread Carlos Chavez

On Fri, 2005-06-24 at 08:50 +0200, Daniel ANDRE wrote:
 Carlos Chavez a écrit :
 
 On Thu, 2005-06-23 at 08:54 +0200, [EMAIL PROTECTED] wrote:
   
 
  ...
 
  There is a specific list for FOP you should directo your questions to.
 
   
 
 What is this list and how to subscribe to?
 
Go to http://www.asternic.org

-- 
--
Telecomunicaciones Abiertas de Mexico
Carlos Chavez
Director de Tecnologia
+52-55-91169161 Ext. 2001

___
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] Distinctive Ring for Agents (Was: Re: Asterisk 1.0.8)

2005-06-24 Thread alan

Russell Bryant wrote:

 Greetings!

 Version 1.0.8 has been released of Asterisk, Asterisk-addons, Zaptel,
 and Libpri.  This release contains a significant amount of bug fixes
 (possibly the most of the 1.0.X releases).  Tarballs are available on
 the asterisk web site as well as the asterisk ftp server.

Thanks!  I appreciate the effort you put into these releases.
I've upgraded from 1.0.7 to 1.0.8 and have had no problems so far.

I was happy to see that the distinctive rings for queues bug was
patched in this version.


However, I have a related question about distinctive rings for agents,
and I'm not sure whether it's a bug or working as intended.

Currently, if I setvar(ALERT_INFO...) for the purposes of setting up a
SIP distinctive ring, and then dial an agent extension, the ALERT_INFO
variable does not make it to the SIP channel which the agent is logged
in on. When I added debugging to the extension chan_agent is dialing,
the ALERT_INFO didn't even make it that far.

Is this the way it's supposed to work? Is there any known way around
this problem?

My goal is to have an agent who is a member of two queues, hear a
different ring depending on what queue the call is coming from. (Yes,
I'm aware of the queue announcement, and I intend to use it. But this is
another requested feature which I'd like to implement if it's feasible.)

Thanks,

Alan Ferrency
pair Networks, Inc.
[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] SpanDSP - Squished Faxes

2005-06-24 Thread Carlos Chavez

On Fri, 2005-06-24 at 10:03 +0200, Marco Parmeggiani wrote:
 Richard Cook ha scritto:
  Hello,
   
  Has anyone had issues with faxes showing up squished in the TIFF  file?
   
  Any ideas what could be causing it?
   
 
The best solution that I have found so far is to convert the image to
PDF before delivery.  I tried several image viewers and many of them
would show the squished image so in order to prevent this I just use
tiff2pdf before sending the email to the user.

-- 
--
Telecomunicaciones Abiertas de Mexico
Carlos Chavez
Director de Tecnologia
+52-55-91169161 Ext. 2001

___
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] SendText

2005-06-24 Thread Christian Hiller

Ok, i did some research and its working fine - is there a way to change 
the change the callerID to something like: MessageCenter or something 
like this? I always get this realm asterisk. is it the realm, right ?


Christian


Jerry Geis schrieb:

I used the outgoing spool directory, added a variable like TEXT=Hello 
world
and going to context send_my_text. tehn send_my_text has exten = 
s,1,SendText($TEXT)


Works great.

Jerry


---

Hello,

i dont get this feature, how can i send a text to a certain SIP-phone 
that support this kind of messaging. The WIKI shows an example, but it 
shows how the receiving phone got to make a call to receive a message.


Thx for a hint! :)

___
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] SpanDSP - Squished Faxes

2005-06-24 Thread Richard Cook

Hello Carlos,

Thank you for the reply.

It does appear to be the actual viewer that squishes the image, not SpanDSP.
I tried a different viewer on the workstation and the fax appears correct.

I like your suggestion to convert it to PDF, thank you. :)

--
Richard Cook
[EMAIL PROTECTED]
T: 705-497-9320  ext 2010
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Carlos Chavez
Sent: Friday, June 24, 2005 12:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SpanDSP - Squished Faxes

On Fri, 2005-06-24 at 10:03 +0200, Marco Parmeggiani wrote:
 Richard Cook ha scritto:
  Hello,
   
  Has anyone had issues with faxes showing up squished in the TIFF  file?
   
  Any ideas what could be causing it?
   
 
The best solution that I have found so far is to convert the image
to PDF before delivery.  I tried several image viewers and many of them
would show the squished image so in order to prevent this I just use
tiff2pdf before sending the email to the user.

--
--
Telecomunicaciones Abiertas de Mexico
Carlos Chavez
Director de Tecnologia
+52-55-91169161 Ext. 2001

___
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] INBAND DTMF G729 ASTERISK

2005-06-24 Thread Matt Fredrickson

On Fri, Jun 24, 2005 at 11:59:51AM -0400, Julio Arruda wrote:
 Andrew Kohlsmith wrote:
 On Friday 24 June 2005 10:58, [EMAIL PROTECTED] wrote:
 
 But there are some products that supports DTMF inband on G729. Ok, it will
 not work in most cases(like everyone told) but why Asterisk dont support
 it? Is this hardcoded, or is possible to try it out?
 
 
 Asterisk can do it too, it's just not reliable on any platform.  Set 
 dtmfmode=inband and use the g729 codec; that's all there is to it.  You 
 will be disappointed though.
 
 I think that IAX, as one example, won't allow this ? Have a faint memory 
 of some error message when trying it (maybe was ILBC+iax ?)

IAX always sends DTMF out of band.  That's why SIP sucks.  Too many options,
too many ways to mess something up.

Matthew Fredrickson
___
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] Exposing Zap Channels on Server A to be Used By Server B

2005-06-24 Thread Robert Goodyear


On Jun 24, 2005, at 9:07 AM, Wiley Siler wrote:

x-tad-biggerHello All,/x-tad-bigger
x-tad-bigger /x-tad-bigger
x-tad-biggerI remember there is a way to use two Asterisk servers and allow one to see a virtual trunk that makes it so server B can use the ZAP channels on server A./x-tad-bigger
x-tad-biggerDoes anyone know where I can find this?  I am racking my brain trying to remember the terminology./x-tad-bigger
x-tad-biggerIt was like creating a 24 channel virtual T1 connection from server B to Server A that allowed server B to not have any ZAP hardware./x-tad-bigger
x-tad-biggerAnyone know what I am talking about?  I am searching the Wiki now but not hitting…/x-tad-bigger
x-tad-bigger /x-tad-bigger
x-tad-biggerSetup: /x-tad-bigger
x-tad-biggerServer A has TMD lines and Voip.providers/x-tad-bigger
x-tad-biggerServer B has only some extensions, needs to connect to Server A and use its ZAP channels
/x-tad-bigger
FWIW I've just been IAX2 trunking over to my other server with the TE110P in it; works very reliably and I can do a failover to VoIP if, say, all channels are busy or something else bad happens. It can also give me two-way (inbound AND outbound) failover with timeout forwarding from my VoIP provider, where if after xx seconds my external inbound IAX2 trunk does not pick up (either by design or ISP failure) the call is routed to my Cox DID, then internally IAX2 trunked across to my PBX.

I'm sure all this can be done with the TDMoE method, but I was just throwing this at you so can make an informed decision.

Robert Goodyear
Brand Up LLC
http://www.brand-up.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] INBAND DTMF G729 ASTERISK

 On Friday 24 June 2005 11:02, [EMAIL PROTECTED] wrote:
 We are on a real world... Every cyber cafe has its own little
 hacker/cracker that is sniffing out... A simple ethereal capture could
 give me a bank pin number... It is REALLY trivial!

 And this is different exactly HOW with inband DTMF??  They can do the
 EXACT
 same thing!  If you want security don't use VOIP unless it's encrypted
 and/or
 over a VPN.  It's really that simple.

Ok, point me on HOW may I get DTMF inband with ethereal.

Andrew, I'm just looking for the most quality/security solution to use
Asterisk with G729, ok?! I think this is good for all of us.

Thanks.

Denis.

___
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] INBAND DTMF G729 ASTERISK

 On Friday 24 June 2005 11:02, [EMAIL PROTECTED] wrote:
 We are on a real world... Every cyber cafe has its own little
 hacker/cracker that is sniffing out... A simple ethereal capture could
 give me a bank pin number... It is REALLY trivial!

 And this is different exactly HOW with inband DTMF??  They can do the
 EXACT
 same thing!  If you want security don't use VOIP unless it's encrypted
 and/or
 over a VPN.  It's really that simple.

Ok, point me on HOW may I get DTMF inband with ethereal.

Andrew, I'm just looking for the most quality/security solution to use
Asterisk with G729, ok?! I think this is good for all of us.

Thanks.

Denis.

___
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] INBAND DTMF G729 ASTERISK

 On Friday 24 June 2005 11:02, [EMAIL PROTECTED] wrote:
 We are on a real world... Every cyber cafe has its own little
 hacker/cracker that is sniffing out... A simple ethereal capture could
 give me a bank pin number... It is REALLY trivial!

 And this is different exactly HOW with inband DTMF??  They can do the
 EXACT
 same thing!  If you want security don't use VOIP unless it's encrypted
 and/or
 over a VPN.  It's really that simple.

Ok, point me on HOW may I get DTMF inband with ethereal.

Andrew, I'm just looking for the most quality/security solution to use
Asterisk with G729, ok?! I think this is good for all of us.

Thanks.

Denis.

___
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] SpanDSP - Squished Faxes

2005-06-24 Thread Brian West

Check the clocking on your T1's if you're using a TDM board GIVE UP NOW those don't do faxing well due to frame slips.Squished faxes are the number one sign of clocking issues on your boards./bOn Jun 23, 2005, at 2:44 PM, Richard Cook wrote: Hello,   Has anyone had issues with faxes showing up squished in the TIFF  file?   Any ideas what could be causing it?   -- Richard Cook [EMAIL PROTECTED] T: 705-497-9320  ext 2010   Blank Bkgrd.gif___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo 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] INBAND DTMF G729 ASTERISK

2005-06-24 Thread Robert Webb



On Fri, 24 Jun 2005 13:10:13 -0400 (EDT)
 [EMAIL PROTECTED] wrote:

On Friday 24 June 2005 11:02, [EMAIL PROTECTED] wrote:
We are on a real world... Every cyber cafe has its own 
little
hacker/cracker that is sniffing out... A simple ethereal 
capture could

give me a bank pin number... It is REALLY trivial!


And this is different exactly HOW with inband DTMF?? 
They can do the

EXACT
same thing!  If you want security don't use VOIP unless 
it's encrypted

and/or
over a VPN.  It's really that simple.


Ok, point me on HOW may I get DTMF inband with ethereal.

Andrew, I'm just looking for the most quality/security 
solution to use
Asterisk with G729, ok?! I think this is good for all of 
us.


Thanks.

Denis.



People, could you PLEASE check first as to who your 
respons is going to. This double posting that has started 
recently is getting VERY annoying.



To: Asterisk Users Mailing List - Non-Commercial 
Discussion asterisk-users@lists.digium.com
Cc: Asterisk Users Mailing List - Non-Commercial 
Discussion asterisk-users@lists.digium.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] meetme mute status

2005-06-24 Thread bdz

yeah, but if you track your executed commands that only
tracks what you have done but if someone mutes/unmutes a
conferencee from a telnet manager session you will never
know.
you can add a manager event to report mute/unmute but then
you have to have a monitor session running all the time.
and what if your monitor session dies? you restart it and
you don't know what statuses are real.
so the only thing is good if the meetme list command reports
the mute status...

On Wed, Jun 22, 2005 at 09:24:03AM -0500, Moises Silva wrote:
 In my little experience with Meetme, i have not found how to know if
 certain user is muted or not, so im keeping track of the commands i
 execute from the web interface, so i know if its muted or not. Its not
 so hard to add a manager event, check manager.c to know how to add
 events.
 
 On 6/22/05, bdz [EMAIL PROTECTED] wrote:
  hi,
  
  is there any way to figure out what the mute status
  is of the meetme conference participants?
  
  i personally can no see any difference on the output:
  kamikaze*CLI meetme
  Conf Num   PartiesMarked Activity  Creation
  5000   0002   N/A00:00:40  Static
  * Total number of MeetMe users: 2
  kamikaze*CLI
  kamikaze*CLI meetme list 5000
  User #: 1  Channel: SIP/fizik-c4eb
  User #: 2  Channel: H323/ip$192.168.42.10:10659/14231
  kamikaze*CLI
  kamikaze*CLI meetme mute 5000 1
  kamikaze*CLI meetme list 5000
  User #: 1  Channel: SIP/fizik-c4eb
  User #: 2  Channel: H323/ip$192.168.42.10:10659/14231
  kamikaze*CLI
  
  i also can not see any mute/unmute event on the manager
  interface only the join/leave events come.
  
  any idea?
  ___
  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
  
 
 
 -- 
 Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org;
 ___
 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 1.0.8 Released

you're kidding Spencer don't think it's the right way
don't waste your time SEMS need devoloppers

harry 
--- Iqbal [EMAIL PROTECTED] a écrit :

 does ast_data have apatch for this release
 
 iqbal
 
 Russell Bryant wrote:
 
  Greetings!
 
  Version 1.0.8 has been released of Asterisk,
 Asterisk-addons, Zaptel, 
  and Libpri.  This release contains a significant
 amount of bug fixes 
  (possibly the most of the 1.0.X releases). 
 Tarballs are available on 
  the asterisk web site as well as the asterisk ftp
 server.
 
  A complete list of all changes made to the v1-0
 branch is available 
  through the archives of the cvs mailing list.  See
 
  http://lists.digium.com for more information.
 
  ChangeLogs that represent an overview of the
 larger changes are 
  available in the source, as well as the following
 web site for 
  convenience - http://dev.asteriskdocs.org/.
 
  Thanks for your support,
 
  Russell Bryant
  ___
  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
 







___ 
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger 
Téléchargez cette version sur http://fr.messenger.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

[Asterisk-Users] Unable to open pseudo channel for timing... Sound may be choppy.

2005-06-24 Thread Joseph

1.)
I'm getting messages in my log:
Unable to open pseudo channel for timing...  Sound may be choppy.
Unable to open IAX timing interface: No such file or directory

I'm using kernel 2.6, I don't think I timing, do I?

2.)
I'm losing IAX registration with provider nor IAX protocol will go
through.
Though, I can ping both providers just fine.  
When I reboot the firewall, the registration stays up for several hours
and gets lost again.

What could be the problem?

-- 
#Joseph
___
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] SendText

2005-06-24 Thread Michiel van Baak

On 18:39, Fri 24 Jun 05, Christian Hiller wrote:
 Ok, i did some research and its working fine - is there a way to change 
 the change the callerID to something like: MessageCenter or something 
 like this? I always get this realm asterisk. is it the realm, right ?
 
 Christian
 

If it's a callfile you can use the keyword callerid
Else do it in the context in extensions.conf:
s,1,SetCallerID(your name)
s,2,SendText($TEXT)

 
 Jerry Geis schrieb:
 
 I used the outgoing spool directory, added a variable like TEXT=Hello 
 world
 and going to context send_my_text. tehn send_my_text has exten = 
 s,1,SendText($TEXT)
 
 Works great.
 
 Jerry
 
 
 ---
 
 Hello,
 
 i dont get this feature, how can i send a text to a certain SIP-phone 
 that support this kind of messaging. The WIKI shows an example, but it 
 shows how the receiving phone got to make a call to receive a message.
 
 Thx for a hint! :)
 
 ___
 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

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

Why is it drug addicts and computer afficionados are both called users?
___
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] SpanDSP - Squished Faxes

On Friday 24 June 2005 13:23, Brian West wrote:
 Check the clocking on your T1's if you're using a TDM board GIVE UP
 NOW those don't do faxing well due to frame slips.

EASY there, Brian, easy...  There are some very smart cookies (and me, but I'm 
not a smart cookie) working on this.  It is rumoured (I have not yet tested) 
that 1.0.7 does NOT have this problem, and I KNOW I had it working with HEAD 
but that was probably back March or even Februaryish timeframe.

 Squished faxes are the number one sign of clocking issues on your
 boards.

??  I find it highly doubtful that missing an entire line (and not more or 
less) is indicative of a clocking issue...  usually you get data corruption 
with clock slips, not complete and neat and tidy missing lines, but I suppose 
anything's possible.

-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] INBAND DTMF G729 ASTERISK

On Friday 24 June 2005 13:10, [EMAIL PROTECTED] wrote:
 Ok, point me on HOW may I get DTMF inband with ethereal.

You capture the data stream, then pull the audio frames out and reassemble to 
a nice slinear audio file (say wav?) -- put that through code cobbled 
together with asterisk's dsp.c or spandsp or even some already-available 
audio software and you can get data like this:

Detected tone 5
Detected tone 8
Detected tone 9
Detected tone 3
...

Hell there was a slashdot article not too far back that gave the exact 
scenario.  

Honestly it is *not* difficult, there are tools out there that do it now and 
can be put together for free. 

Yes, it's a LITTLE more difficult than just running strings over a packet 
trace, but it's not much more difficult and besides... when did script 
kiddies ever build their own tools?  They wait for someone like me to write 
it and then just download it.

 Andrew, I'm just looking for the most quality/security solution to use
 Asterisk with G729, ok?! I think this is good for all of us.

I agree.  However:

1. Inband DTMF with any compressed voice codec is flakey.
2. Inband DTMF is only slightly harder to see than out of band DTMF.
3. If you want voice quality you'll be using ulaw anyway.

I know what you're trying to accomplish but I'm telling you that you're 
chasing ghosts here...  all you'll end up doing is giving yourself a false 
sense of security and when someone drains your bank account you'll be 
flabbergasted because you were so certain that your DTMF was unhackable since 
it was inband, and it's simply not a valid security measure.

We're all on the same team here, I'm just trying to prevent some headaches for 
you.

-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] SpanDSP - Squished Faxes

2005-06-24 Thread Barton Fisher

Which T1 card?

Had same problem with TE410P. Things I
did:

1. Move card to higher priority IRQ fixed problem
(IRQ10).
2. Make sure IRQ is not shared.
3. Disable everything in CMOS that's not needed or
using - COM, LPT, USB, Hyper-Threading, and the likes.
4. Use the latestZAPTEL Drivers.
5. Use Telco for timing source in zaptel.conf. Only
set Telco as source.4 ports cards only need one
source

Bart

- Original Message -
From:
Brian West

To: [EMAIL PROTECTED] ; Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Friday, June 24, 2005 10:23
AM
Subject: Re: [Asterisk-Users] SpanDSP -
Squished Faxes
Check the clocking on your T1's if you're using a TDM board
GIVE UP NOW those don't do faxing well due to frame slips.

Squished faxes are the number one sign of clocking issues on your
boards.

On Jun 23, 2005, at 2:44 PM, Richard Cook wrote:

Hello,

Has anyone had issues with faxes showing
up squished in theTIFF file?

Any ideas what could be causing
it?

--
Richard Cook
[EMAIL PROTECTED]
T: 705-497-9320 ext 2010

Blank
Bkgrd.gif___
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
This message was check with eTrust Antivirus [undefined] and found virus
free.

___Asterisk-Users
mailing
listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo
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] MusicOnHold

2005-06-24 Thread Giordano Grandis




Hi, I installed 
mpg123 v0.59r without error and defined as defaut folder 
/var/lib/asterisk/mohmp3. When i set a call on hold everythinghs seem 
ok

*CLI -- Executing Dial("SIP/2339-4da6", 
"SIP/2391|60|Thtr") in new stack -- Called 
2391 -- SIP/2391-79a0 is ringing -- 
Saved useragent "PA168S" for peer 2319 -- SIP/2391-79a0 
answered SIP/2339-4da6 -- Attempting native bridge of 
SIP/2339-4da6 and SIP/2391-79a0 -- Started music on hold, 
class 'default', on SIP/2339-4da6 -- Stopped music on hold 
on SIP/2339-4da6 == Spawn extension (local, 2391, 1) exited non-zero 
on 'SIP/2339-4da6'
Anyone can help 
me

Thanks

Giordano 
 


___
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] INBAND DTMF G729 ASTERISK

2005-06-24 Thread Michael D Schelin

Hello, I'm not sure about Asterisk and in band DTMF without careful 
reading, but i do know that most ATA's and soft phones all have in band 
capabilities if set. G729 may not pass in band DTMF correctly all the 
time,in fact it's very poor and this is the reason for out of band. I 
think from reading the rest of the comments on the this post that you 
may have to look closer at encryption to keep all eyes from sniffing out 
the pins. I understand why you wouldn't want to slow down VoIP any more 
than you have to but customer security is more important.



Robert Webb wrote:



On Fri, 24 Jun 2005 13:10:13 -0400 (EDT)
 [EMAIL PROTECTED] wrote:


On Friday 24 June 2005 11:02, [EMAIL PROTECTED] wrote:


We are on a real world... Every cyber cafe has its own little
hacker/cracker that is sniffing out... A simple ethereal capture could
give me a bank pin number... It is REALLY trivial!



And this is different exactly HOW with inband DTMF?? They can do the
EXACT
same thing!  If you want security don't use VOIP unless it's encrypted
and/or
over a VPN.  It's really that simple.



Ok, point me on HOW may I get DTMF inband with ethereal.

Andrew, I'm just looking for the most quality/security solution to use
Asterisk with G729, ok?! I think this is good for all of us.

Thanks.

Denis.



People, could you PLEASE check first as to who your respons is going to. 
This double posting that has started recently is getting VERY annoying.



To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Cc: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.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




___
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 global variables without extension..

2005-06-24 Thread chouck




Is it at all possible to set a Global Variable 
freely whenever a context gets used without having to enter an extension 
priority to use SetGlobalVar? This is really limiting the dialplan for 
me. Heres an example of what I would like to be able to do.


[globals]
AREACODE=

[local]
exten=_NXX,1,Dial(SIP/${AREACODE}${EXTEN}/blah)

[anyoldcontext1]
AREACODE=313
include=local


[anyoldcontext2]
AREACODE=810
include=local


so then sip accounts would point towards either 
anyoldcontext1 or anyoldcontext2 and depending on what is set in sip.conf for a 
context depends on what their area code the sip account would use. Anyone 
have any ideas? I suppose my last resort will be to modify the source but 
an simple alternative would be nice. Thanks a lot

-Chad
___
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 global variables without extension..

2005-06-24 Thread chouck





Is it at all possible to set a Global Variable 
freely whenever a context gets used without having to enter an extension 
priority to use SetGlobalVar? This is really limiting the dialplan for 
me. Heres an example of what I would like to be able to do.


[globals]
AREACODE=

[local]
exten=_NXX,1,Dial(SIP/${AREACODE}${EXTEN}/blah)

[anyoldcontext1]
AREACODE=313
include=local


[anyoldcontext2]
AREACODE=810
include=local


so then sip accounts would point towards either 
anyoldcontext1 or anyoldcontext2 and depending on what is set in sip.conf for a 
context depends on what their area code the sip account would use. Anyone 
have any ideas? I suppose my last resort will be to modify the source but 
an simple alternative would be nice. Thanks a lot

-Chad
___
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 server with remote monitoring capabilities

2005-06-24 Thread Mark Musone

Warning - Shameless plug..

Why not just use a managed service provider (like www.shatterit.com)
that is _really_ there 24/7 and can not only reboot your box for you
at any time, but can also monitor it so that it doesnt go down in the
first place.

I apologize for the commerical nature, but this is a real solution for
this real problem...all those expensive hardware solutions is no
replacement for a human..

-Mark


On 6/24/05, Max W Blackmer Jr [EMAIL PROTECTED] wrote:
   Original Message 
  Subject: Re: [Asterisk-Users] Asterisk server with remote monitoring
  capabilities
  From: beonice [EMAIL PROTECTED]
  Date: Thu, June 23, 2005 7:52 pm
 
  --- Michael Welter [EMAIL PROTECTED] wrote:
 
   William Boehlke wrote:
Dell sells a remote management card for under $400
   that enables remote
reboots. I know there are others out there but
   have no experience with them.
   
   
William Boehlke
Signate
   
   
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
   On Behalf Of beonice
   
I'm tired of having to drive out to the colocation
   facility each time my
dedicated asterisk server craps out, just to press
   the button to do a hard
reboot.
(I'm running 1.05 stable at present, no telephony
   hardware, as this is
mainly a system that receives calls, no dial-out
   ability is needed.)
   
   APC makes a power strip with a web server.  Each
   socket has its own IP
   address.  All you have to do to power cycle is
   access the IP address
   from your web browser and give the power cycle
   command.  It is sooo cool.
 
  Thanks for your responses, folks.
 
  Okay, so what makes more sense:
1) a remote management card that will let me
  actually log in to the machine to monitor it as well
  as to reboot it
  vs.
2) a remote-accessible powerstrip that will allow me
  to remotely reboot the server?
 
 
 A little note. make sure your server motherboard/bios supports power on
 after power loss to use the remote control power strip. Secondly make
 sure the power strip control uses SSH and NOT telnet to control it.
 Telnet is too insecure because passwords are sent plain text.
 
 Another possibility is to write a reboot script and set up a cron job to
 automatically reboot every night until you solve the bigger problem of
 why is the server having problems?
 
 With Linux their is little need to reboot Linux. There is only one time
 that you have to reboot Linux. When you upgrade the kernel or its
 modules. Kernel modules do not always need a reboot. Kernel module that
 do require a reboot are critical to operation of your system for example
 RAID# .
 
 The best way is to have a script that uses the init script to restart
 the applications that are questionable on a cron job schedule for low
 usage.  With a good script you could also check on the status of the
 service and perform functional test of the service. Then the script
 would perform the necessary tasks to recover from application failure.
 This wont help with a total system failure as the script will not work.
 Some of the remote monitoring cards can detect a system lockup and
 preform a system reboot automatically.  When all of these fail you can
 use remote control power strips or a KVM (Keyboard Video Mouse) over IP
 to remotely control the hardware as if you are there.  Cyclades
 (www.cyclades.com) sells both KVM and Remote Power management solutions
 that are secure. They even have RSA authentication tokens and a
 Biometric/RSA token authentications for secure management of the remote
 locations.
 
 
 Cheers,
 
 Max W. Blackmer, Jr.
 Consultant, Knowledge Power IT
 
 ___
 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] INBAND DTMF G729 ASTERISK