date:20040311

Hi,

I like that my * talks German to the callers.
Google does not give me any reference about the availability of german 
announcement files.

Could somebody on this list help me out and make it available to me.

Thanks, best regards

Jakob

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

Re: RES: [Asterisk-Users] 403 Forbidden

2004-03-11 Thread Mireia Munoz de jesus

Hi, thanks a lot for your answer. When I call from SIP phone to analogic found I
get this log file:

(I only show, when there's the disconnection)

46:01.165 H245:816f650 H245Received capability set, is
accepted
 46:01.165 H245:816f650 H245TerminalCapabilitySet already in
progress: outSeq=1
 46:01.165 H245:816f650 H245Sending PDU: response
terminalCapabilitySetAck
 46:01.166 H245:816f650 H323   
InternalEstablishedConnectionCheck: connectionState=Await
ingSignalConnect fastStartState=FastStartDisabled
 46:01.167  H225 Caller:8141218 H225Set protocol version to 4
 46:01.167  H225 Caller:8141218 H323Clearing connection
ip$localhost/7705 reason=EndedByQ931C
ause
 46:01.167  H225 Caller:8141218 H323Call end reason for
ip$localhost/7705 set to EndedByQ931C
ause
 46:01.167  H225 Caller:8141218 H225Sending release complete PDU:
callRef=7705
 46:01.170  H225 Caller:8141218 H245Sending PDU: command
endSessionCommand
 46:01.170  H225 Caller:8141218 H225Sending PDU: releaseComplete
 46:01.171 H323 Cleaner H323Cleaning up connections

I suppose, from what you have told me in your mail, that the problem is in my
gateway so, have you any idea what can be the exact problem and how to
solve it?

Thanks a lot for you answer.

Best Regards,

Mireia

Quoting Vinicius Viana [EMAIL PROTECTED]:

 I believe your gatekeeper or your gateway is refusing the call. This can be
 a authorization problem in the gatekeeper or codec problem in the gateway.
 
 You need to see where your call is failing. Try to do the following:
 
 1 - Turn on the oh323 trace in the oh323.conf file adding these lines to
 your configuration:
 wrapLibTraceLevel=3
 libTraceLevel=3
 libTraceFile=/var/log/asterisk/oh323.log
 
 2 - Make a call from your SIP Phone to your PBX
 
 3 - Look into the /var/log/asterisk/oh323.log and verify if the call is
 failing in the Admission Request or in the Setup message.
 
 4 - If it fails in the Admission Request (you will see a Admission Reject
 into the log) the problem is in the configuration of your gatekeeper.
 5 - If it fails in the Setup message (you will see a Release Complete into
 the log) the problem is in the configuration of your gateway
 
 Other thing you can see is if your asterisk box is registered with your
 gatekeeper.
 
 With the information you supplied this is what I remember you can check to
 see what is wrong.
 
 Regards,
 
 Vinicius
 
 -Mensagem original-
 De: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] nome de Mireia Munoz de
 jesus
 Enviada em: quarta-feira, 10 de março de 2004 16:46
 Para: [EMAIL PROTECTED]; Martin Mielke
 Cc: [EMAIL PROTECTED]
 Assunto: Re: [Asterisk-Users] 403 Forbidden
 
 
 Hi,
 
 Thanks for your answer, but my asterisk is working as a H.323 - SIP gateway
 and
 calls between SIP clients (phone and soft clients) are working all right.
 The
 only problem I have, is like I have said in my mail is between sip phones
 and
 PBX.
 
 Best Regards,
 
 Mireia
 
 PS: Someone have other ideas?
 
 
 Quoting Martin Mielke [EMAIL PROTECTED]:
 
  Hi Mieria,
 
  Mireia Munoz de jesus wrote:
 
  Hi!
  
  When I try to call from a SIP phone to a PBX phone I get this error:
  
  chan_oh323.c [1004] Couldn`t call 483377839
  
  and if I get the messages from SIP debug, I have a 403 message. The
  configuration of my system is:
  
  SIP Phone  ASterisk  Gatekeeper - Gateway - PBX -
 Phone
  
  Have someone any idea of what is going on?. It will be very nice if
 someone
  helps... it`s been more than a week that I can`t solve this problem.
  
  Best Regards,
  
  Mireia
  
 
  Could it be that  you are using a *SIP* phone? Although you can add
  H.323 to Asteriskm, SIP and H.323 are different protocols...
 
 
  HTH,
 
  Martin
 
 
  ___
  Asterisk-Users mailing list
  [EMAIL PROTECTED]
  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
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
 
 ---
 
 Checked by AVG anti-virus system (http://www.grisoft.com).
 Version: 6.0.602 / Virus Database: 383 - Release Date: 01/03/2004
 
 ---
 
 Checked by AVG anti-virus system (http://www.grisoft.com).
 Version: 6.0.602 / Virus Database: 383 - Release Date: 01/03/2004
 
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

AW: [Asterisk-Users] who has German voice files ?

2004-03-11 Thread Thomas Haeger

Wait a week and you can have german files from one of our customers, who
wants to donate such files.


Regards,

Thomas.

-Ursprungliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Jakob
Strebel
Gesendet: Donnerstag, 11. Marz 2004 14:31
An: [EMAIL PROTECTED]
Betreff: [Asterisk-Users] who has German voice files ?


Hi,

I like that my * talks German to the callers.
Google does not give me any reference about the availability of german
announcement files.

Could somebody on this list help me out and make it available to me.

Thanks, best regards

Jakob

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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
[EMAIL PROTECTED]
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] who has German voice files ?

2004-03-11 Thread Stadlbauer Stephan

I'm also interested in german voice files...

in the meantime use http://www.rhetorical.com/cgi-bin/demo.cgi for
creating your own voice-files. I use them in my test enviroment.

regards,

stephan

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Jakob Strebel
 Sent: Thursday, March 11, 2004 2:31 PM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] who has German voice files ?
 
 Hi,
 
 I like that my * talks German to the callers.
 Google does not give me any reference about the availability 
 of german announcement files.
 
 Could somebody on this list help me out and make it available to me.
 
 Thanks, best regards
 
 Jakob
 
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 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
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] OpenBSD patches

2004-03-11 Thread Tor Houghton

Hi,

I've applied the OpenBSD patches as noted on

http://www.voip-info.org/tiki-index.php?page=Asterisk%20OpenBSD%20patch 

but there are a few files that still need changing with the current CVS.

I've collected them all here (including the ones from the wiki):

http://www.bogus.net/~torh/files/asterisk-20040311.patch

Of course, I hope these make it into the tree so that OpenBSD users don't
have to manually patch + search in future.. :-

Tor

--
Asterisk CVS-03/11/04-13:23:06 built by [EMAIL PROTECTED] on a i386 running OpenBSD

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

[Asterisk-Users] Have Voice Mail tell the extension?

2004-03-11 Thread Zot O'Connor

Is there an easy way to make the voicemail system say the extension
number after the directory find (via name)?

People want to know the extension once they have found the person to
speed up the process.

Thanks!

-- 
Zot O'Connor [EMAIL PROTECTED]
White Knight Hackers, Inc.

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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] 3com NBX phones

2004-03-11 Thread Clif Jones

I took apart an old broken 3com SIP phone so I could repair it last 
night and examined
the main board.  It is labeled as a NBX motherboard and was manufactured 
by NBX Inc.
I attached it to my Asterisk system and everything worked except for 
MWI.  The 3com
uses a simple text protocol and Asterisk attempts to use XML.  I took 
pictures of the
main board but forgot to bring them in to work.  If anyone wants any 
detailed info on the
unit, let me know in the next couple of days before I re-assemble the 
device.

Clif Jones wrote:

The IR device is a 3rd-party piece of hardware from Extended System 
(now owned by
iFoundry).  The SIP phone looks like all of the other 3com IP phones 
that I have seen
and turning it over with the front of the phone facing up the 
connectors go from left to
right as follows:
1. Handset connector
2. IRDA (serial) RJ-45 connector
3. PC Ethernet RJ-45 connector
4. Wall Ethernet RJ-45 connector
5. Power adapter

Maybe this will help in comparing the units.  I have posted my last 
SIP firmware
(with appropriate disclaimers) to the list but it is held up in 
moderator no man's land.

[EMAIL PROTECTED] wrote:

-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Nicolas 
Bougues
Sent: Friday, March 05, 2004 3:17 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] 3com NBX phones

  
[...]
 

Note that the hardware is probably not the same as the standard NBX 
phones : my SIP phones did feature an IR sensor to be used by a Palm 
for automated dialing.
  


That's actually an option on the better NBX phones...your is probably a
2102-IR or similar, and has been since at least when I did my last NBX
rollout about a year and a half ago.  What seems different is that you
could flash it at all.  When connecting to an NBX, these phones grab
their firmware from the NBX they pin up to.  I suppose there is a
flashable area on the phone that is used as a boot loader in NBX mode,
and probably to store the whole image when flashed with SIP.
Can anyone confirm these are the same phones?  Because I still have
boxes of them somewhere too (that seems to be a common thread here).
Daryl
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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
[EMAIL PROTECTED]
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: Have Voice Mail tell the extension?

2004-03-11 Thread Stephen R. Besch

Zot O'Connor wrote:

Is there an easy way to make the voicemail system say the extension
number after the directory find (via name)?
People want to know the extension once they have found the person to
speed up the process.
Thanks!

I know it's somewhat lame, and requires more management when extensions 
change, but the simplest solution is to instruct users to include the 
extension number when they record their name for the directory.

Stephen R. Besch
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] How can I use the # key normally?

2004-03-11 Thread Matt Lawson

Is there a way to disable the transfer function of the # key?  When 
calling other services,  we often need it to access other menus, other 
voicemail, etc.

Does this have anything to do with the T and t options in the Dial string?

Thanks.

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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] Radius

2004-03-11 Thread Greg Boehnlein

On Wed, 10 Mar 2004, Anton Tinchev wrote:

 Just make a wrapper.
 100 lines in perl.

Do you have an example that you can share?
 
 Derek Samford wrote:
 
 I know this has been hashed, and rehashed, but I saw that a few people
 had said they were going to release their code soon. Is there a working
 implementation of RADIUS for Asterisk out there? Not looking to start a
 debate on how bad it is for billing purposes, that's a given, but I need
 it for legacy systems.
 
  
 
 Thanks,
 
 Derek
 
 ###
 
 This message has been scanned by F-Secure Anti-Virus
   
 
 
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
 

-- 
Vice President of N2Net, a New Age Consulting Service, Inc. Company
 http://www.n2net.net Where everything clicks into place!
 KP-216-121-ST



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

[Asterisk-Users] G.729 passthrough notes (wiki fodder?)

2004-03-11 Thread John Todd

I did some cursory searching on the list archives, and was not able 
to come up with this solution, so I'll summarize.  Someone else 
should put this on the Wiki, since I am terribly lazy when it comes 
to web-ifying things.

I had previously passed G.729 (and G.723) through Asterisk, using 
SIP, between various SIP phones and a Cisco PRI gateway to which I 
have access.  I had previously remembered just ensuring that the Dial 
statement on the outbound call did not have T, or t nor Monitor 
associated with the channel, so that Asterisk didn't have to listen 
to the RTP stream.  This was simply packets in/packets out, no 
transcoding or de-coding required.  I didn't want it to do a SIP 
re-invite, since I wanted to use the Asterisk server as an RTP proxy 
for various reasons that I won't go into here.

I tried getting this to work the other day, and for some reason it 
was not functioning as I had recalled.  No matter how simple my 
dialplan, Asterisk insisted on transcoding the audio channel, even 
though it shouldn't have.  As I didn't have any G.729 channel 
licenses on that machine, that was obviously not optimal.  In any 
case, I wanted to push a lot of channels through the system and not 
be hamstrung by the processing power of the Asterisk server being the 
bottleneck.

Here is what I had configured for the general section, the Cisco PRI 
gateway, and the UA, respectively:

[general]
port = 5060
bindaddr = 0.0.0.0
context = default
[cisco1]
type=friend
host=10.10.22.18
context=from-cisco1
canreinvite=yes
disallow=all
allow=g729
[3013534299]
type=friend
host=dynamic
nat=1
secret=somesecretpassword
canreinvite=no
context=from-clients
disallow=all
allow=g729
I had added/subtracted various allow/disallow parameters in each SIP 
peer to no avail.  After some halfhearted searching through the 
archives, I didn't come up with anything that seemed to solve the 
problem, though I did find some people asking the same question. 
Searching on passthrough gave no useful results, and searching on 
G.729 or g729 led to too many results, so I was forced to ask for 
help.  :-)  The folks at Digium suggested the following:

Add to the [general] section in sip.conf the following:

disallow=all
allow=g729
allow=ulaw
allow=alaw
Without this block of permissions, apparently Asterisk will not pass 
audio through itself without trying to transcode.  Why this is not 
implicitly understood by the configuration options under each SIP 
peer, I don't know, but when I added those lines to the [general] 
section of sip.conf, the system started to pass through the G.729 
media streams without trying to perform codec translation.  Now it 
works!

Next up: testing the number of RTP streams an Asterisk box can handle 
without transcoding...

JT



___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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] Have Voice Mail tell the extension?

2004-03-11 Thread Ben Miller

Have the person record their name and extension when they record their
name.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zot O'Connor
Sent: Thursday, March 11, 2004 8:52 AM
To: asterisk list
Subject: [Asterisk-Users] Have Voice Mail tell the extension?

Is there an easy way to make the voicemail system say the extension
number after the directory find (via name)?

People want to know the extension once they have found the person to
speed up the process.

Thanks!

-- 
Zot O'Connor [EMAIL PROTECTED]
White Knight Hackers, Inc.

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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
[EMAIL PROTECTED]
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 gui client

2004-03-11 Thread mattf

Right now it really helps if you are a programmer or someone who is familiar
with the configuration of an Asterisk system to setup the astguiclient
suite. I will be adding more documentation in a few weeks and maybe even a
simple how-to or a how I installed a new Asterisk T1-internal-VOIP system
with astguiclient from scratch page.

Right now we're concentrating on filling some feature gaps that we have like
adding a callerID popup(which we will release a beta for by Friday night)
and a voicemail indicator. Also, we will be tweeking the features of the
VICIDIAL dialer app and creating a new receptionist module.

I'll post on the list when we have additions or changes to the project or
you can just look on the project website:

http://astguiclient.sf.net/

MATT---


-Original Message-
From: dkwok [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 11, 2004 5:10 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] asterisk gui client


I have looked at matt's asterisk gui client at sourceforge. I am not a 
programmer by trade. The documentation there seems to be a bit lacking. 
Has anyone have the experience in installing the gui client and may 
perhaps have a how-to document available for sharing.

-- 
David Kwok
Tel: 612 99292086 ext 1002
Iaxtel/FWD # 17001813482 ext 1002
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Doubt about IP address setting for Asterisk

2004-03-11 Thread Francisco Perez-Landaeta

Hi, I have a doubt with the installation of asterisk and redhat 9

when i tried setting up the redhat, and said something about the HOST FILE.
I had to modify it and put my address. XXX.XXX.XXX.XXX.

is this correct or will this affect the configuration of asterisk in another
way.

Then, since i dont have a domain like pulver.com or iax. dot something. what
will i have my domain or server be ?

thanks,


- Original Message - 
From: Greg Boehnlein [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, March 11, 2004 10:51 AM
Subject: Re: [Asterisk-Users] Radius


 On Wed, 10 Mar 2004, Anton Tinchev wrote:

  Just make a wrapper.
  100 lines in perl.

 Do you have an example that you can share?

  Derek Samford wrote:
 
  I know this has been hashed, and rehashed, but I saw that a few people
  had said they were going to release their code soon. Is there a working
  implementation of RADIUS for Asterisk out there? Not looking to start a
  debate on how bad it is for billing purposes, that's a given, but I
need
  it for legacy systems.
  
  
  
  Thanks,
  
  Derek
  
  ###
  
  This message has been scanned by F-Secure Anti-Virus
  
  
 
  ___
  Asterisk-Users mailing list
  [EMAIL PROTECTED]
  http://lists.digium.com/mailman/listinfo/asterisk-users
  To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users
 

 -- 
 Vice President of N2Net, a New Age Consulting Service, Inc. Company
  http://www.n2net.net Where everything clicks into place!
  KP-216-121-ST



 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 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
[EMAIL PROTECTED]
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] Outbound Transfer and the # key

2004-03-11 Thread mattf

works great, thanks for posting it.

This illustrates my point perfectly that to have this functionality you have
to modify the patch every time you want to upgrade your CVS. Is there any
way we can pursuade Mark to at least make it a compile-time option if not a
parking.conf option?

Thanks,

MATT---


-Original Message-
From: Iain Stevenson [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 11, 2004 2:31 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Outbound Transfer and the # key



Oh dear.  You can either manually enter in the missing line or apply the 
attached patch as before (you need a clean res_parking.c which you can get 
by deleting the file and then doing cvs co asterisk again).  This patch 
works on my system updated to the latest cvs.

  Iain


--On Wednesday, March 10, 2004 4:54 pm -0500 mattf 
[EMAIL PROTECTED] wrote:

 Here's my patch results:

 [EMAIL PROTECTED] asterisk]# patch -p0  ./Parking.patch
 patching file res/res_parking.c
 Hunk #1 FAILED at 25.
 Hunk #2 succeeded at 228 (offset 13 lines).
 Hunk #3 succeeded at 288 (offset 12 lines).
 Hunk #4 succeeded at 408 (offset 13 lines).
 1 out of 4 hunks FAILED -- saving rejects to file res/res_parking.c.rej

 [EMAIL PROTECTED] asterisk]# cat res/res_parking.c.rej
 ***
 *** 25,30 
   #include asterisk/musiconhold.h
   #include asterisk/config.h
   #include asterisk/cli.h
   #include stdlib.h
   #include errno.h
   #include unistd.h
 --- 25,31 
   #include asterisk/musiconhold.h
   #include asterisk/config.h
   #include asterisk/cli.h
 + #include asterisk/indications.h
   #include stdlib.h
   #include errno.h
   #include unistd.h


 is the first fail a bad thing?

 This is CVS from 15 minutes ago.

 Thanks,

 MATT---



 -Original Message-
 From: Iain Stevenson [mailto:[EMAIL PROTECTED]
 Sent: Wednesday, March 10, 2004 4:33 PM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] Outbound Transfer and the # key



 Try the attached patch.  Go to your asterisk root directory and type:

   patch -p0  path_to_patch/Parking.patch

 .. then rebuild asterisk.

   Iain



 --On Wednesday, March 10, 2004 7:43 am -0500 John Congdon [EMAIL PROTECTED]
 wrote:

 I have applied the patch and restarted Asterisk.

 But it still only requires a single # to transfer.
 Did I possibly miss something?

 This is just to show that it was applied.

 [EMAIL PROTECTED] asterisk]# pwd
 /usr/src/asterisk
 [EMAIL PROTECTED] asterisk]# patch -p0  ../old_asterisk/doublehash.patch
 patching file res/res_parking.c
 Reversed (or previously applied) patch detected!  Assume -R? [n]
 Apply anyway? [n]
 Skipping patch.
 3 out of 3 hunks ignored -- saving rejects to file res/res_parking.c.rej


 John



 On Mar 9, 2004, at 4:53 PM, mattf wrote:

 There is a better way to deal with this, it's the doublehash patch. This
 patch makes it so you have to press the hash key twice to transfer a call
 instead of once as is default in Asterisk.

 Sad thing is that every time the parking code changes the patch has to
 change(sometimes twice a week) and I don't have a patch for the most
 recent CVS. I've asked numerous times for some wonderful
 Asterisk-code-God(please Mark ;)) to make it a configurable variable in
 the parking.conf file but noone seems to think it's worthy of doing. It's
 actually a rather simple code change from what I can guess reading the
 patch code. I've been told that the core developers(Mark) don't want to
 mess with doublehash, but maybe if enough people say they want it we can
 get them to make this harmless addition to the parking code.

 Here's a bug where it's been talked about:
 http://bugs.digium.com/bug_view_page.php?bug_id=885

 MATT---


 -Original Message-
 From: John Congdon [mailto:[EMAIL PROTECTED]
 Sent: Tuesday, March 09, 2004 4:24 PM
 To: Asterisk Mailling List
 Subject: [Asterisk-Users] Outbound Transfer and the # key


 Has there been any resolution to this?

 Does anyone have a good way to allow
 someone to choose whether they want to
 be able to transfer a call vs send the # to
 the other end.

 Is there  a simple way to change the Transfer
 key for # to *?


 John

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




 ___
 Asterisk-Users

Re: [Asterisk-Users] Cisco 7960 and short delay before voice starts after ring.

2004-03-11 Thread James Sizemore




exten = 6500,1,Answer
exten = 6500,2,Wait,1
exten = 6500,3,VoicemailMain2
Or should I say, Me too!

Is this the bug for the case in question?
 CSCed48311: Media takes 0.4 sec to be set up
Thanks.

-Andrew

Yes the problem is that when making outgoing calls, there is enough of a 
delay in the call setup once the remote side picks up, that people that 
answer the phone hello will be heard saying o  or if they talk fast 
enough not heard at all therefor leaving a very awkward silence at the 
start of a call.

This is very annoying. A earlier  person  suggested  answering the  
calls before  dialing  and playing a ringing sound till the start of the 
voice.  That may be a work around of sorts for some,  you will hear a 
ring then a congestion tone on call that can't connect, or a ring before 
a operator messages (say to dial one before the number) that most users 
may not be used to.  I'll be playing with that ideal to see what odd 
effect a ring has before call setup causes. 

The work around may be less annoying then the problem. smile I'll see.



___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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 CAPI DECT problem

2004-03-11 Thread Jon Lawrence

On Thursday 11 March 2004 11:41, Ignace CARIA wrote:

 - Plug the DECT base into a X100P Digium Card.

Plug the DECT phone into a Handytone-286 which is in turn plugged into your 
network.
It works fine for me.

HTH
Jon

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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] who has German voice files ?

2004-03-11 Thread Fran Boon

Thomas Haeger wrote:
Wait a week and you can have german files from one of our customers, who
wants to donate such files.
Great :)

Please could you make them available from the following webpage?
http://voip-info.org/wiki-Asterisk+sound+files+international
If anyone has Spanish or Portuguese, then that would make me very happy!

Best Wishes,
Fran.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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_sip.c:495 retrans_pkt: Maximum retries exceeded on call

2004-03-11 Thread James Sizemore

You do have :
nat_enable: 1
nat_received_processing: 1
On the Ciscos?

AstGrp wrote:

I am having a similar problem... I get the same message, but inbound
calls can go through This is only Cisco phones that are behind NAT.
I have tried your recommendations from below, but still no luck.. User
can make outbound calls, just can't receive any.  Any ideas would be
greatly appreciated.. I even tried to change the timeout value in
chan_sip, but it just waits longer to fail.. Just dosen't seem to want
to communicate...
Thanks,

gcc

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Bittner
Posted At: Tuesday, March 02, 2004 11:46 PM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum
retries exceeded on call
Subject: RE: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum
retries exceeded on call
Are you using Cisco phones. ? 

I had this issue with my cisco phones. I didn't had any issues with
dropped calls. All I did to fix this was set a prefered_codex and set
proxy_register to 0. 

I hope this helps.

John Bittner
Simlab.net
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of dkwok
Sent: Wednesday, March 03, 2004 7:04 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum 
retries exceeded on call

*CLI Mar  3 12:55:05 WARNING[1150495040]: chan_sip.c:495
retrans_pkt: 
Maximum retries exceeded on call 
[EMAIL PROTECTED] for seqno 102 (Request)

This has been brought up in the previous post but it does not seem to
have an answer for it so far.
I cvs the stable v1.0 this morning after compiling and
installing I have 
calls drop 1 minutes into the connection with the above message.

If anyone has any idea of this occurrence.

I have set up sip.conf:

canreinvite=no

--
David Kwok
Tel: 612 99292086 ext 1002
Iaxtel/FWD # 17001813482 ext 1002
   

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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
[EMAIL PROTECTED]
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
[EMAIL PROTECTED]
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_sip.c:495 retrans_pkt: Maximum retries exceeded on call

2004-03-11 Thread AstGrp

Here's a copy of the cisco config

-- Current *FLASH* Configuration --

Platform : Cisco IP Phone 7940
Elasped Time: 00:01:37

dhcp_server : 10.100.0.2
my_ip_addr : 10.100.0.150
subnet_mask : 255.255.255.0
defaultgw : 10.100.0.2
dyn_dns_addr_1 : 0.0.0.0
dyn_dns_addr_2 : 0.0.0.0
dns_addr : 10.100.254.7
dns_backup_1: 24.93.68.65
tftp_addr : 66.64.246.36
dyn_tftp_addr : 0.0.0.0
my_mac_addr : 000f:23ac:4559
domain_name : tnessentials.com
my_name : SIP000F23AC4559
Status Flags : 1230

image_version : P0S3-06-2-00
FirmLoadID : PC030301
DSPLoadID : PS03AT38
network_media_type : Auto
network_port2_type : Hub/Switch
tos_media : 5
phone_label : TNE PBX VOIP
tftp_cfg_dir : 
phone_password : **
phone_prompt : SIP Phone
language : english
sntp_mode : DirectedBroadcast
sntp_server : 
time_zone : EST
dst_offset : 1
dst_start_month : April
dst_start_day : 0
dst_start_day_of_week : Sun
dst_start_week_of_month : 1
dst_start_time : 02
dst_stop_month : Oct
dst_stop_day : 0
dst_stop_day_of_week : Sunday
dst_stop_week_of_month : 8
dst_stop_time : 2
dst_auto_adjust : 1
time_format_24hr : 1
date_format : M/D/Y
nat_enable : 1
nat_address : 
voip_control_port : 5060
start_media_port : 16456
end_media_port : 17456
sync : 1
xml_card_dir : 
xml_card_file : CARD.XML
telnet_level : 2
services_url : 
directory_url : 
logo_url : 
http_proxy_addr : 
http_proxy_port : 80
enable_vad : 0
dial_template : dialplan
callerid_blocking : 0
anonymous_call_block : 0
autocomplete : 1
messages_uri : 55
dnd_control : 0
preferred_codec : g711ulaw
dtmf_outofband : avt
dtmf_avt_payload : 101
dtmf_db_level : 3
dtmf_inband : 1
line1_name : khome
line2_name : UNPROVISIONED
line1_authname : khome
line2_authname : UNPROVISIONED
line1_password : **
line2_password : **
line1_shortname : UNPROVISIONED
line2_shortname : UNPROVISIONED
line1_displayname : Kyle Elworthy
line2_displayname : 
proxy1_address : 66.64.246.36
proxy2_address : 
proxy1_port : 5060
proxy2_port : 5060
sip_retx : 10
sip_invite_retx : 6
timer_t1 : 500
timer_t2 : 4000
timer_invite_expires : 180
timer_register_expires : 3600
proxy_register : 1
proxy_backup : 
proxy_emergency : 
proxy_backup_port : 5060
proxy_emergency_port : 5060
outbound_proxy : 
outbound_proxy_port : 5060
nat_received_processing : 1
mwi_status : 0
call_waiting : 1
user_info : none
cnf_join_enable : 1
remote_party_id : 0
semi_attended_transfer : 1
call_hold_ringback : 0
stutter_msg_waiting : 0
cfwd_url : 
call_stats : 1
auto_answer : 0
local_cfwd_enable : 1
timer_register_delta : 5

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James
Sizemore
Posted At: Thursday, March 11, 2004 10:47 AM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum
retries exceeded on call
Subject: Re: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum
retries exceeded on call


You do have :
nat_enable: 1
nat_received_processing: 1

On the Ciscos?

AstGrp wrote:

I am having a similar problem... I get the same message, but inbound 
calls can go through This is only Cisco phones that are behind NAT.

I have tried your recommendations from below, but still no luck.. User 
can make outbound calls, just can't receive any.  Any ideas would be 
greatly appreciated.. I even tried to change the timeout value in 
chan_sip, but it just waits longer to fail.. Just dosen't seem to want 
to communicate...

Thanks,

gcc

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John 
Bittner Posted At: Tuesday, March 02, 2004 11:46 PM Posted To: Asterisk

User Group
Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum 
retries exceeded on call
Subject: RE: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum 
retries exceeded on call


Are you using Cisco phones. ?

I had this issue with my cisco phones. I didn't had any issues with 
dropped calls. All I did to fix this was set a prefered_codex and set 
proxy_register to 0.

I hope this helps.

John Bittner
Simlab.net


  

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of dkwok
Sent: Wednesday, March 03, 2004 7:04 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum
retries exceeded on call

*CLI Mar  3 12:55:05 WARNING[1150495040]: chan_sip.c:495
retrans_pkt:
Maximum retries exceeded on call 
[EMAIL PROTECTED] for seqno 102 (Request)

This has been brought up in the previous post but it does not seem to 
have an answer for it so far.

I cvs the stable v1.0 this morning after compiling and installing I 
have calls drop 1 minutes into the connection with the above message.

If anyone has any idea of this occurrence.

I have set up sip.conf:

canreinvite=no

--
David Kwok
Tel: 612 99292086 ext 1002
Iaxtel/FWD # 17001813482 ext 1002




___
Asterisk-Users mailing list
[EMAIL PROTECTED]

[Asterisk-Users] PRI errors blocking Asterisk

Hi Asterisk community,

Every once in a while (can be several times a day, or every few days),
I get that kind of error (with a TE405P) :

 PRI: Short write: -1/66 (Unknown error 500)

After that, the E1 links on the server get jammed : all the current
channels, or any new zap channel is simply unkillable.

Restarting Asterisk (after kill -9) solves the problem.

It seems to me that the Q921 layer in libpri has an unrecoverable
error (such as the fd being wrong/closed).

Anybody know where it could come from, and/or what should be done to
avoid it ?

-- 
Nicolas Bougues
Axialys Interactive
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: AW: [Asterisk-Users] who has German voice files ?

2004-03-11 Thread Mark Messmore, Technical Support, University Telcom Inc.

Thomas,

At 14:45 11.03.2004 +0100, you wrote:
Wait a week and you can have german files from one of our customers, who
wants to donate such files.
Please let us know when they are available.

Jakob 

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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] G.729 passthrough notes (wiki fodder?)

2004-03-11 Thread Fran Boon

John Todd wrote:
I did some cursory searching on the list archives, and was not able to 
come up with this solution, so I'll summarize.  Someone else should put 
this on the Wiki, since I am terribly lazy when it comes to web-ifying 
things.
http://voip-info.org/tiki-index.php?page=Asterisk+G.729+pass-thru

Thanks John for the legwork :)

F
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] SIP native bridge vs. SIP reinvite

2004-03-11 Thread Jeremy Jones

Hi,

I'm trying to get rtp media streams to run between endpoints rather than
through my * server, and I think I'm getting something wrong.  I have an
AS5300 speaking both h323 (for a different voip system I run) and sip
for *.  Dial-peers on the as5300 differentiate inbound from pstn to
different chunks of DID numbers between h323 and sip.  I'm testing with
xlite on a PC.

So here's what I have:

Outbound trunks are defined in my extensions.conf that send _9whatever
to SIP/pstn_gw/${EXTEN}.

In sip.conf I have two friends, one for my xlite softphone, one for
pstn_gw:

[2085551212]
type=friend
username=2085551212
secret=1234
host=dynamic
canreinvite=yes
disallow=all
allow=ulaw
context=testme
mailbox=5551212
callerid=Jeremy Jones 2085551212

[pstn_gw]
type=friend
username=pstn_gw
disallow=all
allow=ulaw
context=default
canreinvite=yes
host=10.0.0.201

I can place a call from the PSTN to 5551212 successfully, and I can
place calls from xlite to the PSTN successfully.  But in either case I
always see two sip channels active on *, and the endpoints (as5300 
xlite) are sending their rtp via *.  Here's what I see when I place a
call from xlite to:


*CLI -- Executing Prefix(SIP/2085551212-f04d, 9) in new stack
-- Prepended prefix, new extension is 93532533
-- Executing Dial(SIP/20825551212-f04d, SIP/pstn_gw/93532533) in
new stack
-- Called pstn_gw/93532533
-- SIP/pstn_gw-85a0 is making progress passing it to
SIP/2085551212-f04d
-- SIP/pstn_gw-85a0 answered SIP/2085551212-f04d
-- Attempting native bridge of SIP/2085551212-f04d and
SIP/pstn_gw-85a0

*CLI
*CLI sip show channels
Peer User/ANRCall ID  Seq (Tx/Rx)  Lag  Jitter
Format
10.0.0.201   9353253302e2e09e167  00103/00651  0ms  ms
ULAW
10.0.0.100 2082874602E0541F6D-81  00102/03763  0ms  ms
ULAW
2 active SIP channel(s)
*CLI

(I have a Prefix rule for outbound 'cuz this is a system for residential
users, and the as5300 has dial-peers that need a 9 prefix...)

The output in * is similar for inbound from PSTN to xlite.

I can send output from sip debug if that'd help.

Thanks,
Jeremy Jones
Network Nerd
WestCom, LLC




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

[Asterisk-Users] Cannot use # key to transfer calls

2004-03-11 Thread Rana Dutt

I cannot use the # key to transfer a call. I have two kinds of SIP phones,
Grandstream and IpDialog, and the # key cannot be used to transfer on either
one. If I press the # key during a call, I hear the touchtone for it, but
Asterisk does nothing.
The documentation for parking a call says that I must first transfer the
call using #, so that's why I need this feature to work.  Thanks for any
pointers.

-Ron Dutt


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

[Asterisk-Users] CVS Update Frequency

Just as a matter of curiosity...how often do most of you update your *
installation from the CVS?  

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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] Cisco 7960 and short delay before voice starts after ring.

2004-03-11 Thread Barton Hodges

[EMAIL PROTECTED] wrote:
 exten = 6500,1,Answer
 exten = 6500,2,Wait,1
 exten = 6500,3,VoicemailMain2
 
 Or should I say, Me too!
 
 Is this the bug for the case in question?
  CSCed48311: Media takes 0.4 sec to be set up
 
 Thanks.
 
 -Andrew
 
 Yes the problem is that when making outgoing calls, there is
 enough of a
 delay in the call setup once the remote side picks up, that
 people that
 answer the phone hello will be heard saying o  or if they
 talk fast
 enough not heard at all therefor leaving a very awkward
 silence at the
 start of a call.
 
 This is very annoying. A earlier  person  suggested  answering the
 calls before  dialing  and playing a ringing sound till the
 start of the
 voice.  That may be a work around of sorts for some,  you will hear
a
 ring then a congestion tone on call that can't connect, or a
 ring before
 a operator messages (say to dial one before the number) that
 most users
 may not be used to.  I'll be playing with that ideal to see
 what odd
 effect a ring has before call setup causes.
 
 The work around may be less annoying then the problem. smile I'll
 see. 

I've seen the same thing, and it appears to be from attempting a
native bridge.  You can try the attached patch to disable native
bridging.  It cut out the annoying silence completely for me.  This
may be a bad thing (unnecessary CPU utilization due to same-codec
translation), but I have not experienced any problems.

Barton







channel.c.diff
Description: Binary data

[Asterisk-Users] MySQL VM config

2004-03-11 Thread Tim Sailer

In Monastery, I'm using the show voicemail users command to get a list
of defined users, and how many VM messages they have. It seems that this
doesn't work when MySQL is used for the VM config. I can get the mbox
info out of the correct table, but where can I find the number of unread
messages?

Tim

-- 

 Tim Sailer Coastal Internet, Inc.  
 Network and Systems Operations PO Box 726  
 http://www.buoy.comMoriches, NY 11955  
 [EMAIL PROTECTED]   (631) 399-2910  (888) 924-3728  

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

[Asterisk-Users] Music on Hold sound goes off if environment is silent

Hi,

Music on hold works if the environment is noisy.
But in case of silence the sound goes off.
If I scratch continuously on the mikrofone, then the replay works without 
any interruption.

Q: is there a parameter which influences this behaviour?

Thanks, best regards

Jakob 

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

[Asterisk-Users] Need help with MGCP.CONF and dual voice per host

2004-03-11 Thread Duane Cox

I setup this config, but I had to comment out the voice port 2
because it conflict with my voice port 1.

Is this the correct format?


[00060D0F4FBF]
host=dynamic
context=default
line = aaln/1
callerid=217378
;context=default
;line = aaln/2
;callerid=217379

Thanks
Duane Cox



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

[Asterisk-Users] Soundcard question

Hi all,

I am getting an error about the soundcard not responding when * is run. 
There is a Creative Labs card in the slot, but it doesn't come up as 
SoundBlaster when linux (Slackware 9.1) boots. It looks like it might be 
working though. Looking at the IRQ list, the card is deteced as an 
Ensoniq 1317 or something -(I am not near that box at the moment.)

Will * work with this card or what cards will it work with? I really 
want to be able to have dialup music. I have an old DAL CardD+ ISA 
soundcard but I'm assuming that won't ever work.

tia,

randulo
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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] Cannot use # key to transfer calls

2004-03-11 Thread Barton Hodges

[EMAIL PROTECTED] wrote:
 I cannot use the # key to transfer a call. I have two kinds of SIP
 phones, Grandstream and IpDialog, and the # key cannot be used to
 transfer on either one. If I press the # key during a call, I hear
 the touchtone for it, but Asterisk does nothing.
 The documentation for parking a call says that I must first transfer
 the call using #, so that's why I need this feature to work. Thanks
 for any pointers.
 
 -Ron Dutt

Make sure your Dial() line contains the 'T' and/or 't' options.

Also make sure that your DTMF entries in sip.conf match the phones.

I've found that with Grandstream HandyTones, the only reliable method
of using '#' to transfer is by using inband DTMF, which means using
ULAW/ALAW as well.


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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 errors blocking Asterisk

2004-03-11 Thread Scott Stingel

Hi Nichoas-

Are you are getting lots of frame re-transmission messages in
/var/log/asterisk/messages as well?

regards
Scott

Scott M. Stingel 
Emerging Voice Technology Inc.
Palo Alto, California and London, England

Email:  scott at evtmedia.com  
URL:www.evtmedia.com  

 

-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
Nicolas Bougues
Sent: Thursday, March 11, 2004 4:00 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] PRI errors blocking Asterisk

Hi Asterisk community,

Every once in a while (can be several times a day, or every few days),
I get that kind of error (with a TE405P) :

 PRI: Short write: -1/66 (Unknown error 500)

After that, the E1 links on the server get jammed : all the current
channels, or any new zap channel is simply unkillable.

Restarting Asterisk (after kill -9) solves the problem.

It seems to me that the Q921 layer in libpri has an unrecoverable
error (such as the fd being wrong/closed).

Anybody know where it could come from, and/or what should be done to
avoid it ?

-- 
Nicolas Bougues
Axialys Interactive
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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
[EMAIL PROTECTED]
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] Cannot use # key to transfer calls

2004-03-11 Thread Steven Sokol

Does the entry for your extension include the 't' option? 

Example: Dial(SIP/|20|t)

The 't' option allows you (the called party) to transfer.  The 'T' option
can also be added to allow the calling party to transfer.

See: http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Dial

Steven Sokol
Owner/Manager
Sokol  Associates, LLC

Phone:  816.822.1807
IaxTel: 700.613.9004
Web:http://www.sokol-associates.com

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Rana Dutt
 Sent: Thursday, March 11, 2004 10:15 AM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Cannot use # key to transfer calls
 
 I cannot use the # key to transfer a call. I have two kinds of SIP phones,
 Grandstream and IpDialog, and the # key cannot be used to transfer on
 either
 one. If I press the # key during a call, I hear the touchtone for it, but
 Asterisk does nothing.
 The documentation for parking a call says that I must first transfer the
 call using #, so that's why I need this feature to work.  Thanks for any
 pointers.
 
 -Ron Dutt
 
 
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 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
[EMAIL PROTECTED]
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] CVS Update Frequency

2004-03-11 Thread Scott Stingel

Never - unless I must have a new feature, or need a critical bug fix!

but, seriously, mine are production systems, and I don't use many of the
VoIP features of asterisk.  There is so much development going on in
asterisk, that you may want to update only an in-house, non-production
system, at first when you get a new CVS.  Then implement a rigorous test
protocol that you follow before you release new CVS's to the field.

Regards


Scott M. Stingel 
Emerging Voice Technology Inc.
Palo Alto, California and London, England

Email:  scott at evtmedia.com  
URL:www.evtmedia.com  

 

-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
Mark Messmore, Technical Support, University Telcom Inc.
Sent: Thursday, March 11, 2004 4:16 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] CVS Update Frequency

Just as a matter of curiosity...how often do most of you update your *
installation from the CVS?  

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

[Asterisk-Users] remote dtmf

2004-03-11 Thread Chris Clifton

Using a cisco 7960 + ulaw, calling a long distance 800 # via voicepulse,
when the remote ivr transfers the call using a couple dtmf tones, asterisk
disconnects with a fast busy. Anything I can do to prevent this behaviour ?

Thanks,
Chris

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

[Asterisk-Users] IAX Phone: Now Supports Windows 98/ME

2004-03-11 Thread Steven Sokol

Having received several requests from users of Windows 98 and ME, I have
changed the installer for IAX Phone to install on those versions of Windows.

Please note that I don't have any Win 9X or ME boxes about to test on, so I
cannot guarantee is proper operation on those platforms. 

grin (But then again, its beta code and I don't guarantee anything about
it anyway so what the heck?) /grin

Please let me know what issues you find.

Download at: http://www.sokol-associates.com/

Thanks

Steve

Steven Sokol
Owner/Manager
Sokol  Associates, LLC

Phone:  816.822.1807
IaxTel: 700.613.9004
Web:http://www.sokol-associates.com



___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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] CVS Update Frequency

On Thu, 2004-03-11 at 10:16, Mark Messmore, Technical Support,
University Telcom Inc. wrote:
 Just as a matter of curiosity...how often do most of you update your *
 installation from the CVS?  

What does it say about your technical knowledge if you don't do the
minor things such as properly start a new thread?

I tend to update production machines only when there is problems or
features that require them. That being said, I have our core phone
switch stuck on an older version of asterisk as it does nothing more
than route calls around and doesn't need special features. It takes all
our inbound calls from our PRI and drops them on our T1 channel bank,
redirects some back to the PRI, and the rest go via IAX2 to 2-3 other
asterisk machines.

phone:/home/critch# w  
 10:49:14 up 134 days, 21:17,  2 users,  load average: 0.06, 0.06, 0.01
USER TTY  FROM  LOGIN@   IDLE   JCPU   PCPU WHAT
critch   pts/1steven.basesys.c 10:490.00s  0.04s  0.04s sshd: critch [p
phone:/home/critch# asterisk -r
Asterisk CVS-10/22/03-06:38:52, Copyright (C) 1999-2001 Linux Support Services, Inc.
Written by Mark Spencer [EMAIL PROTECTED]
=
Connected to Asterisk CVS-10/22/03-06:38:52 currently running on phone (pid = 24942)
phone*CLI show uptime 
System uptime: 10 weeks, 4 days, 10 hours, 40 minutes, 39 seconds
Last reload: 3 days, 25 minutes, 41 seconds

I have not updated our office switch since the loss of the original
voicemail system due to not wanting to remake the voicemail configs. 
 10:55:23 up 97 days, 14:41,  1 user,  load average: 0.00, 0.03, 0.04
USER TTY  FROM  LOGIN@   IDLE   JCPU   PCPU WHAT
critch   pts/0steven.basesys.c 10:550.00s  0.04s  0.02s w
pbx:/home/critch# asterisk -r
Asterisk CVS-10/31/03-13:16:40, Copyright (C) 1999-2001 Linux Support Services, Inc.
Written by Mark Spencer [EMAIL PROTECTED]
=
Connected to Asterisk CVS-10/31/03-13:16:40 currently running on pbx (pid = 22802)
pbx*CLI show uptime 
System uptime: 13 weeks, 2 days, 21 hours, 24 minutes, 8 seconds
Last reload: 5 weeks, 5 days, 23 hours, 38 minutes, 14 seconds

Our other main asterisk machine just had a hard lock up over the
weekend, and we have replacement hardware enroute to us now that should
get here early next week. At that point I will be hard pressed to go
with current CVS when I haven't been testing more recent versions when I
know the versions I have have long uptime records. 

-- 
Steven Critchfield  [EMAIL PROTECTED]

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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] Music on Hold sound goes off if environment is silent

2004-03-11 Thread Ernest W. Lessenger

At 08:37 AM 3/11/2004, you wrote:
Music on hold works if the environment is noisy.
But in case of silence the sound goes off.
If I scratch continuously on the mikrofone, then the replay works without
any interruption.
Q: is there a parameter which influences this behaviour?
Whatever phone or softphone you are using, you need to disable silence 
suppression. Why? Dunno exactly. In the newest version of Xten, the feature 
is Advanced System Settings - Audio Settings - Silence Settings - 
Transmit Silence - Should be Yes.

--Ernest 

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

[Asterisk-Users] Nitsuko 124i interface, anyone?

A client has a Nitsuko 124i phone system with an accompanying voicemail
based on a single dialogic card with two ports.

Has anyone tried to replace the Nitsuko NVM-2000 with asterisk?

Right now there are two RJ-11's strung from the phonesystem to the
voicemail. All calls that come into the business are first dropped onto
the voicemail system to run through a menu, allow choosing of extension,
etc. After that, the voicemail transmits a signal back to the
phonesystem which causes it to grab the line back and send it on to the
called party. 

I believe we can replace the NVM-2000 with asterisk, but I'm not sure
what kind of signalling or cards would be needed to allow asterisk to
handle and process the calls.

Comments from anyone who has worked with this hardware and knows more
about it than myself are appreciated, even if you've not actually tried
to swap it out with *.

-
Andrew Thompson
http://aktzero.com/ 


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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] Cannot use # key to transfer calls

2004-03-11 Thread Dave Cotton

On Thu, 2004-03-11 at 17:14, Rana Dutt wrote:
 I cannot use the # key to transfer a call. I have two kinds of SIP phones,
 Grandstream and IpDialog, and the # key cannot be used to transfer on either
 one. If I press the # key during a call, I hear the touchtone for it, but
 Asterisk does nothing.
 The documentation for parking a call says that I must first transfer the
 call using #, so that's why I need this feature to work.  Thanks for any
 pointers.

I can't speak for the IpDialog but the Grandstream can handle this using
either the transfer button or #. If you post your configs perhaps we can
see what's wrong.
-- 
Dave Cotton [EMAIL PROTECTED]

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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] Soundcard question

randulo wrote:
 Will * work with this card or what cards will it work with? I really
 want to be able to have dialup music. I have an old DAL CardD+ ISA
 soundcard but I'm assuming that won't ever work.
 

If by dialup music you mean music-on-hold, a soundcard is not required
for that, go to the wiki and read.

http://www.voip-info.org/wiki-Asterisk

-
Andrew Thompson
http://aktzero.com/ 


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

RES: RES: [Asterisk-Users] 403 Forbidden

2004-03-11 Thread Vinicius Viana

The call end reason EndedByQ931Cause is used by the OpenH323 stack when it
doesn't know the real cause.
Try to see if the codecs in the gateway are compatible with the codecs in
asterisk.
What are the codecs you are using in SIP Phones, in Asterisk and in the
gateway?

Regards,

Vinicius



-Mensagem original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nome de Mireia Munoz de
jesus
Enviada em: quinta-feira, 11 de março de 2004 11:37
Para: [EMAIL PROTECTED]; Vinicius Viana
Assunto: Re: RES: [Asterisk-Users] 403 Forbidden


Hi, thanks a lot for your answer. When I call from SIP phone to analogic
found I
get this log file:

(I only show, when there's the disconnection)

46:01.165 H245:816f650 H245Received capability set, is
accepted
 46:01.165 H245:816f650 H245TerminalCapabilitySet
already in
progress: outSeq=1
 46:01.165 H245:816f650 H245Sending PDU: response
terminalCapabilitySetAck
 46:01.166 H245:816f650 H323
InternalEstablishedConnectionCheck: connectionState=Await
ingSignalConnect fastStartState=FastStartDisabled
 46:01.167  H225 Caller:8141218 H225Set protocol version to 4
 46:01.167  H225 Caller:8141218 H323Clearing connection
ip$localhost/7705 reason=EndedByQ931C
ause
 46:01.167  H225 Caller:8141218 H323Call end reason for
ip$localhost/7705 set to EndedByQ931C
ause
 46:01.167  H225 Caller:8141218 H225Sending release complete
PDU:
callRef=7705
 46:01.170  H225 Caller:8141218 H245Sending PDU: command
endSessionCommand
 46:01.170  H225 Caller:8141218 H225Sending PDU: releaseComplete
 46:01.171 H323 Cleaner H323Cleaning up connections

I suppose, from what you have told me in your mail, that the problem is in
my
gateway so, have you any idea what can be the exact problem and how to
solve it?

Thanks a lot for you answer.

Best Regards,

Mireia

Quoting Vinicius Viana [EMAIL PROTECTED]:

 I believe your gatekeeper or your gateway is refusing the call. This can
be
 a authorization problem in the gatekeeper or codec problem in the gateway.

 You need to see where your call is failing. Try to do the following:

 1 - Turn on the oh323 trace in the oh323.conf file adding these lines to
 your configuration:
 wrapLibTraceLevel=3
 libTraceLevel=3
 libTraceFile=/var/log/asterisk/oh323.log

 2 - Make a call from your SIP Phone to your PBX

 3 - Look into the /var/log/asterisk/oh323.log and verify if the call is
 failing in the Admission Request or in the Setup message.

 4 - If it fails in the Admission Request (you will see a Admission Reject
 into the log) the problem is in the configuration of your gatekeeper.
 5 - If it fails in the Setup message (you will see a Release Complete into
 the log) the problem is in the configuration of your gateway

 Other thing you can see is if your asterisk box is registered with your
 gatekeeper.

 With the information you supplied this is what I remember you can check to
 see what is wrong.

 Regards,

 Vinicius

 -Mensagem original-
 De: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] nome de Mireia Munoz de
 jesus
 Enviada em: quarta-feira, 10 de março de 2004 16:46
 Para: [EMAIL PROTECTED]; Martin Mielke
 Cc: [EMAIL PROTECTED]
 Assunto: Re: [Asterisk-Users] 403 Forbidden


 Hi,

 Thanks for your answer, but my asterisk is working as a H.323 - SIP
gateway
 and
 calls between SIP clients (phone and soft clients) are working all right.
 The
 only problem I have, is like I have said in my mail is between sip phones
 and
 PBX.

 Best Regards,

 Mireia

 PS: Someone have other ideas?


 Quoting Martin Mielke [EMAIL PROTECTED]:

  Hi Mieria,
 
  Mireia Munoz de jesus wrote:
 
  Hi!
  
  When I try to call from a SIP phone to a PBX phone I get this error:
  
  chan_oh323.c [1004] Couldn`t call 483377839
  
  and if I get the messages from SIP debug, I have a 403 message. The
  configuration of my system is:
  
  SIP Phone  ASterisk  Gatekeeper - Gateway - PBX -
 Phone
  
  Have someone any idea of what is going on?. It will be very nice if
 someone
  helps... it`s been more than a week that I can`t solve this problem.
  
  Best Regards,
  
  Mireia
  
 
  Could it be that  you are using a *SIP* phone? Although you can add
  H.323 to Asteriskm, SIP and H.323 are different protocols...
 
 
  HTH,
 
  Martin
 
 
  ___
  Asterisk-Users mailing list
  [EMAIL PROTECTED]
  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
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

 ---

 Checked by AVG anti-virus system

[Asterisk-Users] GSM Bandwidth - Test x Measures

2004-03-11 Thread Joel Barbosa Moraes




Hi all!

 Everybody was talking about 
bandwidth consumption of the codecs but I have one doubt about it.
 I prepared an Asterisk box with 
a TDM20B and a X100P (two FXS and one FXO). Last weekI traveled 
andcarried my notebook so I could dial to a local ISP and then connect to 
my asterisk box to make local calls.
 Onthe hotel, the quality 
of the phone line was terrible and the maxium connection speed I made was 
26.4k.Even with thebad connection I connected with my asterisk box 
and then madea call and the quality was good. The client I used was the 
IAX Phone that uses the GSM codec (as said at the web page).
 Any explanations? This is 
bothering me because I have to decide how many clients a DSL (256/128) 
connection can support and if I take the measures I can put a maximum of 3 
(since its 35k with the GSM codec) and if I use my tests I could use almost 5. 
What should I do? Help me. :-)

Thanks a lot in advance,

Joel Moraes

[Asterisk-Users] Agents and delay before and after they handle a call

2004-03-11 Thread Jeff Crews

Is there a way for Agents logging in with AgentLogin to have the the agent 
hear the beep and then have the option to press # or some button to 
indicate they are ready to take the next call?Sometimes an agent is 
taking a drink of water or coughing...and logging off and logging back seem 
lengthy to do.

I have tried to use AgentCallbackLogin but it seems to require that each 
Agent has their own DID phone number so that that the application can call 
them back at that specific number.  We do not have DID to each agent 
implemented yet...as we are using Asterisk with our old phone system.

Thanks.

---
Jeff Crews
Eastern Oregon Net, Inc.
La Grande Oregon
Email [EMAIL PROTECTED]
Voice 541-963-2625 or 800-785-7873,  extension 11
personal efax 503-907-6704
standard company fax 541-962-7818
web http://www.eoni.com 

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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] OpenBSD patches

2004-03-11 Thread Tilghman Lesher

On Thursday 11 March 2004 07:45, Tor Houghton wrote:
 Of course, I hope these make it into the tree so that OpenBSD users
 don't have to manually patch + search in future.. :-

Anything you hope makes it into the tree should be posted to
http://bugs.digium.com/

-Tilghman

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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] Soundcard question

Hi,

Andrew Thompson wrote:
If by dialup music you mean music-on-hold, a soundcard is not required
for that, go to the wiki and read.
I do mean music on hold, or in this case music on demand.
http://www.voip-info.org/wiki-Asterisk
You mean the part that says Asterisk needs no additional hardware for 
Voice over IP. ? I see nothing about music on hold here. Out of 
curiousity, why does * complain about the sound card when it starts? 
Maybe it doesn't matter...

I tried to make an extension 6000 that plays music but it hangs up 
immediately. mpg123 is in the right place.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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] Music on Hold sound goes off if environment is silent

2004-03-11 Thread hank

can you play music on hold using the line in feature of your sound card to
the phone?
thanks
- Original Message -
From: Jakob Strebel [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, March 11, 2004 8:37 AM
Subject: [Asterisk-Users] Music on Hold sound goes off if environment is
silent


 Hi,

 Music on hold works if the environment is noisy.
 But in case of silence the sound goes off.
 If I scratch continuously on the mikrofone, then the replay works without
 any interruption.

 Q: is there a parameter which influences this behaviour?


 Thanks, best regards

 Jakob

 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 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
[EMAIL PROTECTED]
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] Soundcard question

2004-03-11 Thread hank

are you using alsa drivers?
- Original Message - 
From: randulo [EMAIL PROTECTED]
To: Asterisk List [EMAIL PROTECTED]
Sent: Thursday, March 11, 2004 8:40 AM
Subject: [Asterisk-Users] Soundcard question

 Hi all,

 I am getting an error about the soundcard not responding when * is run. 
 There is a Creative Labs card in the slot, but it doesn't come up as 
 SoundBlaster when linux (Slackware 9.1) boots. It looks like it might be 
 working though. Looking at the IRQ list, the card is deteced as an 
 Ensoniq 1317 or something -(I am not near that box at the moment.)

 Will * work with this card or what cards will it work with? I really 
 want to be able to have dialup music. I have an old DAL CardD+ ISA 
 soundcard but I'm assuming that won't ever work.

 tia,

 randulo
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 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
[EMAIL PROTECTED]
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] Music on Hold sound goes off if environment is silent

2004-03-11 Thread Bob Knight

Ernest W. Lessenger wrote:

At 08:37 AM 3/11/2004, you wrote:

Music on hold works if the environment is noisy.
But in case of silence the sound goes off.
If I scratch continuously on the mikrofone, then the replay works 
without
any interruption.

Q: is there a parameter which influences this behaviour?


Whatever phone or softphone you are using, you need to disable silence 
suppression. Why? Dunno exactly. In the newest version of Xten, the 
feature is Advanced System Settings - Audio Settings - Silence 
Settings - Transmit Silence - Should be Yes. 
Why?
Because the * community is just a little on the lazy side.
* can not self clock RTP packets.  Instead of clocking itself and just
locking on to received packets, it totally relies on received packets
for it's timing.
No packets coming in for timing, no packets going out.

This would be something fun to work on, but who has time when
there are work arounds.  I am unemployed and I do not have the time.
--
Bob Knight
[-w] the work option
[EMAIL PROTECTED]
925-449-9163
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] IPC5000 - Wireless Sip phone

2004-03-11 Thread Craig Waddington









I am looking to buy a wireless sip phone, probably the
IPC5000, I have looked at Wisip phone and read tons of posts regarding that
phone.



Do any * admins have any feedback on this phone?



Is there any major differences between the phones, besides
looks?



The site has very limited information regarding prices etc.



Ta.

Re: [Asterisk-Users] Soundcard question

hank wrote:
are you using alsa drivers?
Forgive me, I just installed Slackware two days ago, I'm not up to speed 
yet, but I see ALSA mixer app is there. I also saw somewhere that the 
soundcard is muted at boot time and needs to be manually unbooted using 
the alsamixer app. I ran that and it looked like it worked. There are a 
bunch of snd- drivers showing with lsmod.

I was also seeing a complaint about Warning, flexibel rate... so I'm 
reconverting some mp3 files here at home.

Any soundcard tips gratefully accepted like where to look to see if all 
is well. I need to hook up phones and try to play an mp3 into them to 
see if that is working.

I'll look tomorrow at the office to see if any of these things work :)

thx
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Using MultiTech MVP-210 as FXO/FXS gateway

2004-03-11 Thread Stephen Foster









Hi all,

 Im
trying to use my 2-port multi-tech VoIP gateway to
talk to asterisk. Ideally I want to put it in a remote location with a POTS
line one port1 and an analog phone on port2 to call that location. Both the MultiTech and Asterisk have non-natted
static IPs.



I have tried every different type of configuration possible
for the sip.conf file. I can call from the analog
phone on the multitech to a local asterisk extension
and it rings, but when I
pickup I get a busy signal at both ends.



When I try and call from asterisk to the phone on the multitech, I dont even get that far. I receive this
from the CLI:



 --
Starting simple switch on 'Zap/10-1'

 --
Executing Dial(Zap/10-1, SIP/multitech) in new stack

 --
Called multitech

 --
Got SIP response 486 Busy Here back from 122.33.44.55

 --
SIP/multitech-964c is busy

 == Everyone is busy at this time

n
Hungup 'Zap/10-1'



The MultiTech seems pretty simple
to configure, just the IP of asterisk, username and pass. The only field I
havent tried its SIP URL. I was recently at a MultiTech
show and I saw them use x-lite to call to the MultiTech. Since neither is a sip proxy, I cant
figure out why that worked for them but I cant get this working with
asterisk.



Here is the current version of my sip.conf



[multitech]

context=local

;disallow=all

allow=all

;disallow=all

allow=gsm

allow=ulaw

allow=alaw

type=friend

username=multitech

secret=pass

nat=no

;mailbox=200

host=dynamic

reinvite=no

;canreinvite=yes

qualify=1000

dtmfmode=info

canreinvite=no

callerid=Multi
Tech

;defualtip=1.2.3.4



Thanks everyone,

 Steve

RE: [Asterisk-Users] Nitsuko 124i interface, anyone?

2004-03-11 Thread Micke Andersson

Andrew Thompson  wrote on the Thursday, March 11, 2004 6:06 PM 

 Comments from anyone who has worked with this hardware and knows more
 about it than myself are appreciated, even if you've not actually
 tried to swap it out with *.  
 


I have a Nitsuka system here at home.. somewhere in a box.

I'm not sure wich model.  I dont have any voicemail system to that though,
so I'm interessted in your idea to use a *

/Mike

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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] OpenBSD patches

2004-03-11 Thread Tor Houghton

On Thu, Mar 11, 2004 at 11:42:05AM -0600, Tilghman Lesher wrote:
 On Thursday 11 March 2004 07:45, Tor Houghton wrote:
  Of course, I hope these make it into the tree so that OpenBSD users
  don't have to manually patch + search in future.. :-
 
 Anything you hope makes it into the tree should be posted to
 http://bugs.digium.com/
 

Yeah, John gave me a heads up on that earlier, so I did.

Cheers,

Tor
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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] Soundcard question

On Thu, 2004-03-11 at 12:26, randulo wrote:
 hank wrote:
  are you using alsa drivers?
 
 Forgive me, I just installed Slackware two days ago, I'm not up to speed 
 yet, but I see ALSA mixer app is there. I also saw somewhere that the 
 soundcard is muted at boot time and needs to be manually unbooted using 
 the alsamixer app. I ran that and it looked like it worked. There are a 
 bunch of snd- drivers showing with lsmod.
 
 I was also seeing a complaint about Warning, flexibel rate... so I'm 
 reconverting some mp3 files here at home.
 
 Any soundcard tips gratefully accepted like where to look to see if all 
 is well. I need to hook up phones and try to play an mp3 into them to 
 see if that is working.
 
 I'll look tomorrow at the office to see if any of these things work :)

All those snd- modules sounds exactly like alsa. The error message is
probably related to the chan_oss module trying to get access, but not
having a OSS driver to talk to. This isn't a problem, but if you don't
want to see it, put a noload = chan_oss in modules.conf for asterisk. 
-- 
Steven Critchfield  [EMAIL PROTECTED]

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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] Soundcard question

Steven Critchfield wrote:

All those snd- modules sounds exactly like alsa. The error message is
probably related to the chan_oss module trying to get access, but not
having a OSS driver to talk to. This isn't a problem, but if you don't
want to see it, put a noload = chan_oss in modules.conf for asterisk. 
Thanks for that suggestion. I guess my problem of not hearing the music 
is yet another .config ignorance that hopefully will be cured soon. Only 
been around for a few days. ;)
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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] Soundcard question

On Thu, 2004-03-11 at 13:03, randulo wrote:
 Steven Critchfield wrote:
 
  All those snd- modules sounds exactly like alsa. The error message is
  probably related to the chan_oss module trying to get access, but not
  having a OSS driver to talk to. This isn't a problem, but if you don't
  want to see it, put a noload = chan_oss in modules.conf for asterisk. 
 
 Thanks for that suggestion. I guess my problem of not hearing the music 
 is yet another .config ignorance that hopefully will be cured soon. Only 
 been around for a few days. ;)

MoH seems to bite many people. I haven't been interested in that so I
haven't learned it. If you are just wanting to dial an extension and
listen to music, try the mp3player app.
-- 
Steven Critchfield  [EMAIL PROTECTED]

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

[Asterisk-Users] stealth asterisk (XP100-PBX Handset)

2004-03-11 Thread Zot O'Connor

Since no one answered my other question.

Is anyone stealth using asterisk?

I have a nec handset.  I would love to pipe it to an xp100 and then VoIP
to the asterisk box (even if on the same box).

The two issue I see are

  Intercom (it blasts to the speak and is used as a PA)
  Digital signaling vs pots.

Any ideas?

-- 
Zot O'Connor [EMAIL PROTECTED]
White Knight Hackers, Inc.

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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 MultiTech MVP-210 as FXO/FXS gateway

2004-03-11 Thread Jorge Mendoza

I tested Multitech with the same scenario and it works.

Stephen Foster wrote:

The MultiTech seems pretty simple to configure, just the IP of asterisk, 
username and pass. The only field I havent tried its SIP URL. I was 
recently at a MultiTech show and I saw them use x-lite to call to the 
MultiTech. Since neither is a sip proxy, I cant figure out why that 
worked for them but I cant get this working with asterisk.

No so simple. At least you must to elaborate the following windows:
IP, Voice/Fax, Interface, Phone Book configuration,Outbound Phone Book, 
Inbound Phone Book.


Here is the current version of my sip.conf

 

[multitech]

context=local

My sip.conf:

[multitech]
context=default
type=friend
host=192.168.YY.XX ; multitech IP
dtmfmode=inband; we use alaw
Hope this help.

Jorge
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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] Music on Hold sound goes off if environment is silent

Hank,


can you play music on hold using the line in feature of your sound card to
the phone?
I have a Logitech USB Headset, which has integrated Sound Card. I cant find 
the line feature, can you give me a hint where to find it?

Jakob

BTW: the silence suppression as a workaround is working. But how to tell 
every user that he has to enable it? 

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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] Music on Hold sound goes off if environment is silent

2004-03-11 Thread hank

I don't know where it is
I use a creative labs sound blaster  audigy
fwd number
91013
us phone number phone to fwd
3602070445
uk phone number phone to fwd
0870 - 3403466
email
[EMAIL PROTECTED]

- Original Message -
From: Jakob Strebel [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, March 11, 2004 11:35 AM
Subject: Re: [Asterisk-Users] Music on Hold sound goes off if environment
is silent


 Hank,


 can you play music on hold using the line in feature of your sound card
to
 the phone?

 I have a Logitech USB Headset, which has integrated Sound Card. I cant
find
 the line feature, can you give me a hint where to find it?

 Jakob

 BTW: the silence suppression as a workaround is working. But how to tell
 every user that he has to enable it?

 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 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
[EMAIL PROTECTED]
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] IPC5000 - Wireless Sip phone

2004-03-11 Thread Craig Waddington









Thanks for the info. Sounds good.



Does that mean I can contact them for a test
unit also, to try before I buy?















From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Michael Devenijn
Sent: 11 March 2004 18:25
To:
[EMAIL PROTECTED]
Subject: RE: [Asterisk-Users]
IPC5000 - Wireless Sip phone







I ordered a test unit and will recieve it this week (already shipped
from sweden),
i will post some comments on this list when it is tested .. I hope it will do
his job !! ...











the mail they sent to : 













Hello Michael,











Hope you are well.











Your sample is on the way and pls find attached delivery
note for your reference.











Ps. frieght charge was USD10 lower, so we own you USD10 that
we will pretty reduced it with your next order or we transfer it to your bank
account.











I'll the coming days send you updated information about the
handset and its new design i.e. it has L2 roaming feature now. The handoff time
is 200 ~ 300ms between the AP. We aim to short it to 100 ~ 200ms. 









The implementation of Web Authentication(web-login)
what we call HTTPS(SSL)is ongoing and should be releasedon June. It
can be software upgrade.















Best Regards,
Mohammed Fahd























-Oorspronkelijk
bericht- 
Van:
[EMAIL PROTECTED]namensCraig Waddington 
Verzonden: do 11/03/2004 19:15 
Aan: [EMAIL PROTECTED]

CC: 
Onderwerp: [Asterisk-Users]
IPC5000 - Wireless Sip phone



I am looking to buy a wireless sip phone, probably the
IPC5000, I have looked at Wisip phone and read tons of posts regarding that
phone.



Do any * admins have any feedback on this phone?



Is there any major differences between the phones, besides
looks?



The site has very limited information regarding prices etc.



Ta.

Re: [Asterisk-Users] PRI errors blocking Asterisk

On Thu, Mar 11, 2004 at 05:12:24PM +0100, Klaus-Peter Junghanns wrote:
 Nicolas,
 
 does your TE405P share the irq?

No, it's alone on IRQ 17 (with IO-APIC).

-- 
Nicolas Bougues
Axialys Interactive
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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 errors blocking Asterisk

On Thu, Mar 11, 2004 at 04:48:42PM -, Scott Stingel wrote:
 Hi Nichoas-
 
 Are you are getting lots of frame re-transmission messages in
 /var/log/asterisk/messages as well?
 

No.

I get a few of these messages, though :

Mar 11 16:11:11 WARNING[81926]: PRI: Read on 131 failed: Unknown error 500
Mar 11 16:11:11 NOTICE[81926]: PRI got event: 8 on span 2 
Mar 11 16:11:11 WARNING[81926]: PRI: Read on 131 failed: Unknown error 500
Mar 11 16:11:11 NOTICE[81926]: PRI got event: 8 on span 2

Note that I'm not sure about the timing settings on my board. The
board has (currently) 3 E1 spans connected, from 3 different
operators, all of them providing a clock (no guarantee they are
synchronised).

Upon module loading, the driver says 

TE410P: Timing from source 0

I chose one (quite random) E1 span as a primary sync source in
zaptel.conf

SPAN 3: Primary Sync Source

I'm not sure how this setting is used. Do I really have to set one ? 

--
Nicolas Bougues
Axialys Interactive
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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 errors blocking Asterisk

2004-03-11 Thread Scott Stingel

could you please post your zaptel.conf?

You're right, maybe this has something to do with your clock source or
timing

Thanks

Scott M. Stingel 
Emerging Voice Technology Inc.
Palo Alto, California and London, England

Email:  scott at evtmedia.com  
URL:www.evtmedia.com  

 

-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
Nicolas Bougues
Sent: Thursday, March 11, 2004 8:06 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] PRI errors blocking Asterisk

On Thu, Mar 11, 2004 at 04:48:42PM -, Scott Stingel wrote:
 Hi Nichoas-
 
 Are you are getting lots of frame re-transmission messages in
 /var/log/asterisk/messages as well?
 

No.

I get a few of these messages, though :

Mar 11 16:11:11 WARNING[81926]: PRI: Read on 131 failed: 
Unknown error 500
Mar 11 16:11:11 NOTICE[81926]: PRI got event: 8 on span 2 
Mar 11 16:11:11 WARNING[81926]: PRI: Read on 131 failed: 
Unknown error 500
Mar 11 16:11:11 NOTICE[81926]: PRI got event: 8 on span 2

Note that I'm not sure about the timing settings on my board. The
board has (currently) 3 E1 spans connected, from 3 different
operators, all of them providing a clock (no guarantee they are
synchronised).

Upon module loading, the driver says 

TE410P: Timing from source 0

I chose one (quite random) E1 span as a primary sync source in
zaptel.conf

SPAN 3: Primary Sync Source

I'm not sure how this setting is used. Do I really have to set one ? 

--
Nicolas Bougues
Axialys Interactive
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] German ringtone

2004-03-11 Thread Norbert Wegener

Hello,
I have setup my first asterisk using an isdn card and i4l.
I can make calls from the fixed network to a sip phone via asterisk and 
vice versa.
Unfortunately I do not get any ring (or busy) tone at my Grandstream, 
when making a call via the isdn card and i4l.
The problem of no ring  tone only occurs in this one direction. The 
other way is okay.
Is this, because in indications.conf German signals are not defined or 
is this a missing feature of i4l?
I googled and found 
http://lists.digium.com/pipermail/asterisk-users/2002-October/005355.html
which let me believe, the problem has been addressed long time ago.  The 
patch there seems not to be integrated in asterisk, or am I wrong?

Thanks for any answer.
Norbert
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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] IPC5000 - Wireless Sip phone

2004-03-11 Thread Miguel Cavazos

you buy the unit thats what its call a test unit ipc5000 looks great and
its 28 USD more than wisip i think the lcd is worth

Miguel
On Thu, 2004-03-11 at 19:58, Craig Waddington wrote:
 Thanks for the info. Sounds good.
 
  
 
 Does that mean I can contact them for a test unit also, to try before
 I buy?
 
  
 
  
 
  
 

 __
 
 From:[EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Michael
 Devenijn
 Sent: 11 March 2004 18:25
 To: [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] IPC5000 - Wireless Sip phone
 
 
  
 
 I ordered a test unit and will recieve it this week (already shipped
 from sweden), i will post some comments on this list when it is tested
 .. I hope it will do his job !! ...
 
 
  
 
 
 the mail they sent to : 
 
 
  
 
 
 Hello Michael,
 
 
  
 
 
 Hope you are well.
 
 
  
 
 
 Your sample is on the way and pls find attached delivery note for your
 reference.
 
 
  
 
 
 Ps. frieght charge was USD10 lower, so we own you USD10 that we will
 pretty reduced it with your next order or we transfer it to your bank
 account.
 
 
  
 
 
 I'll the coming days send you updated information about the handset
 and its new design i.e. it has L2 roaming feature now. The handoff
 time is 200 ~ 300ms between the AP. We aim to short it to 100 ~ 200ms.
 
  
 
 
 The implementation of Web Authentication (web-login) what we call
 HTTPS(SSL) is ongoing and should be released on June. It can be
 software upgrade.
 
 
  
 
 
 Best Regards,
 Mohammed Fahd
 
 
  
 
 
  
 
 
 -Oorspronkelijk bericht- 
 Van: [EMAIL PROTECTED] namens Craig
 Waddington 
 Verzonden: do 11/03/2004 19:15 
 Aan: [EMAIL PROTECTED] 
 CC:
 Onderwerp: [Asterisk-Users] IPC5000 - Wireless Sip phone
 
 
 I am looking to buy a wireless sip phone, probably the
 IPC5000, I have looked at Wisip phone and read tons of posts
 regarding that phone.
 
  
 
 Do any * admins have any feedback on this phone?
 
  
 
 Is there any major differences between the phones, besides
 looks?
 
  
 
 The site has very limited information regarding prices etc.
 
  
 
 Ta.
 
  
 
  
 
 
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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] Cisco 7960 and short delay before voice startsafter ring.

2004-03-11 Thread Steve Dolloff

We have the same complaint here.  The caller doesn't hear the receiver
say hello and so no-one knows what's going on.

Stephen

 -Original Message-
 From: James Sizemore [mailto:[EMAIL PROTECTED]
 Sent: Thursday, March 11, 2004 9:38 AM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] Cisco 7960 and short delay before voice
 startsafter ring.

  exten = 6500,1,Answer
  exten = 6500,2,Wait,1
  exten = 6500,3,VoicemailMain2

  Or should I say, Me too!

  Is this the bug for the case in question?
   CSCed48311: Media takes 0.4 sec to be set up

  Thanks.

  -Andrew

 Yes the problem is that when making outgoing calls, there is enough of
a
 delay in the call setup once the remote side picks up, that people
that
 answer the phone hello will be heard saying o  or if they talk
fast
 enough not heard at all therefor leaving a very awkward silence at the
 start of a call.

 This is very annoying. A earlier  person  suggested  answering the
 calls before  dialing  and playing a ringing sound till the start of
the
 voice.  That may be a work around of sorts for some,  you will hear a
 ring then a congestion tone on call that can't connect, or a ring
before
 a operator messages (say to dial one before the number) that most
users
 may not be used to.  I'll be playing with that ideal to see what odd
 effect a ring has before call setup causes.

 The work around may be less annoying then the problem. smile I'll
see.

 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 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
[EMAIL PROTECTED]
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 errors blocking Asterisk

On Thu, Mar 11, 2004 at 08:13:48PM -, Scott Stingel wrote:
 could you please post your zaptel.conf?
 

Here it is :

span=1,1,0,ccs,hdb3
span=2,0,0,ccs,hdb3
span=3,1,0,ccs,hdb3 # Colt est source de timing
span=4,0,0,ccs,hdb3

defaultzone=fr

bchan=1-15,17-31
dchan=16

bchan=32-46,48-62
dchan=47

bchan=63-77,79-93
dchan=78

bchan=94-108,110-124
dchan=109

-- 
Nicolas Bougues
Axialys Interactive
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Night menu not working

2004-03-11 Thread Justin Carlson

Hi all,

I am trying to get day and nighttime menus to work in * and no matter what
time I specify the first include entry that matches the number dialed is
used.  I have included my extentions.conf and my sip phones have a default
context of default.


[general]
static=yes
writeprotect=no
[globals]
MARYKAY = 21
RECEPTIONIST = 20
KATHY = 22

[daytime]
include = parkedcalls
exten = t,1,Playback(vm-goodbye)
exten = t,2,Hangup
exten = i,1,Playback(invalid)

switch = IAX2/[EMAIL PROTECTED]/dialout

;sip extentions


exten = ${MARYKAY},1,Dial,SIP/21|20
exten = ${MARYKAY},2,Voicemail,u21

exten = ${RECEPTIONIST},1,Dial,SIP/20|20
exten = ${RECEPTIONIST},2,Dial,SIP/20SIP/21SIP/22|20
exten = ${RECEPTIONIST},3,Voicemail,u20

exten = ${KATHY},1,Dial,SIP/22|20
exten = ${KATHY},2,Voicemail,u22

; for Local Voicemail access
exten = *98,1,VoicemailMain
exten = asterisk,1,VoicemailMain

exten = 25,1,Dial,SIP/fax

; voicemail extentions
exten = 621,1,Voicemail,u21
exten = 620,1,Voicemail,u20
exten = 622,1,Voicemail,u22
exten = 679,1,VoicemailMain
; direct extentions
exten = 201,1,Dial,IAX/[EMAIL PROTECTED]/6515526201
exten = 307,1,Dial,IAX/[EMAIL PROTECTED]/6515522307
exten = 309,1,Dial,IAX/[EMAIL PROTECTED]/6515522309
exten = 313,1,Dial,IAX/[EMAIL PROTECTED]/6515522313
exten = 317,1,Dial,IAX/[EMAIL PROTECTED]/6515522317
exten = 601,1,Dial,IAX/[EMAIL PROTECTED]/6515523601
exten = 603,1,Dial,IAX/[EMAIL PROTECTED]/6515523603
exten = 609,1,Dial,IAX/[EMAIL PROTECTED]/6515523609
exten = 664,1,Dial,IAX/[EMAIL PROTECTED]/6515523664
exten = 694,1,Dial,IAX/[EMAIL PROTECTED]/6515523694
exten = 816,1,Dial,IAX/[EMAIL PROTECTED]/6515526816
exten = 817,1,Dial,IAX/[EMAIL PROTECTED]/6515526817
exten = 821,1,Dial,IAX/[EMAIL PROTECTED]/6515526821
exten = 842,1,Dial,IAX/[EMAIL PROTECTED]/6515526842

[faxmachine]
switch = IAX2/[EMAIL PROTECTED]/faxmachine

[nighttime]
exten = 21,1,Playback(tt-monkeys)

[default]
include = daytime|8:00-14:48|mon-fri
include = nighttime

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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 MultiTech MVP-210 as FXO/FXS gateway

2004-03-11 Thread John Chester

I am using an MVP-210 as FXS -- I haven't tried FXO.

Here's my sip.conf entry:

[mvp-x303]
type=friend
host=192.168.1.93
username=303
dtmfmode=rfc2833
context=fs1
disallow=all
allow=ulaw
(Not sure if dtmfmode is correct.)

Username must be an extension number that appears in the MVP210's inbound 
phone book.

Here's an MVP210 outbound phone book entry to call x352 on the Asterisk server:

Destination Pattern: 352
Total Digits: 3
IP Address: 192.168.1.94(the Asterisk server)
Protocol Type: SIP
Transport Protocol: UDP (MVP210 defaults to TCP)
SIP Port Number: 5060
At 01:23 PM 3/11/2004 -0500, Stephen Foster wrote:

Hi all,

Im trying to use my 2-port multi-tech VoIP gateway to talk to 
asterisk. Ideally I want to put it in a remote location with a POTS line 
one port1 and an analog phone on port2 to call that location. Both the 
MultiTech and Asterisk have non-natted static IPs.



I have tried every different type of configuration possible for the 
sip.conf file. I can call from the analog phone on the multitech to a 
local asterisk extension and it rings, but when I  pickup I get a busy 
signal at both ends.



When I try and call from asterisk to the phone on the multitech, I dont 
even get that far. I receive this from the CLI:



-- Starting simple switch on 'Zap/10-1'

-- Executing Dial(Zap/10-1, SIP/multitech) in new stack

-- Called multitech

-- Got SIP response 486 Busy Here back from 122.33.44.55

-- SIP/multitech-964c is busy

  == Everyone is busy at this time

n   Hungup 'Zap/10-1'



The MultiTech seems pretty simple to configure, just the IP of asterisk, 
username and pass. The only field I havent tried its SIP URL. I was 
recently at a MultiTech show and I saw them use x-lite to call to the 
MultiTech. Since neither is a sip proxy, I cant figure out why that worked 
for them but I cant get this working with asterisk.



Here is the current version of my sip.conf



[multitech]

context=local

;disallow=all

allow=all

;disallow=all

allow=gsm

allow=ulaw

allow=alaw

type=friend

username=multitech

secret=pass

nat=no

;mailbox=200

host=dynamic

reinvite=no

;canreinvite=yes

qualify=1000

dtmfmode=info

canreinvite=no

callerid=Multi Tech

;defualtip=1.2.3.4



Thanks everyone,

Steve
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] * and PrePaid

2004-03-11 Thread Barry Fawthrop





Greetings

What would it take (all hardware etc..) to setup * 
on a 
prepaid card server.

I have an * server a T1 and TDM10B card, thus 
allowing 24 
simultaneous calls I 
guessing I need a VoIP Termination 
Provider (eg: NuFone, etc..)

How do I print and create the cards, and what are 
the 
requirements, and how do I setup * to handle the 
billing, of 
minutes used, etc.. or does this not exist yet ?

Thanks In Advance
B

RE: [Asterisk-Users] IPC5000 - Wireless Sip phone

2004-03-11 Thread Michael Devenijn

no i
bought this one

-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Craig
WaddingtonSent: Thursday, March 11, 2004 8:58 PMTo:
[EMAIL PROTECTED]Subject: RE: [Asterisk-Users]
IPC5000 - Wireless Sip phone

Thanks for the info.
Sounds good.

Does that mean I can
contact them for a test unit also, to try before I
buy?

From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael DevenijnSent: 11 March 2004 18:25To:
[EMAIL PROTECTED]Subject: RE: [Asterisk-Users] IPC5000 -
Wireless Sip phone

I ordered a test unit and will recieve it this week
(already shipped from sweden), i will post some comments
on this list when it is tested .. I hope it will do his job !!
...

the mail they sent to :

Hello
Michael,

Hope you are
well.

Your sample is on the way and pls
find attached delivery note for your
reference.

Ps. frieght charge was USD10
lower, so we own you USD10 that we will pretty reduced it with your next order
or we transfer it to your bank account.

I'll the coming days send you
updated information about the handset and its new design i.e. it has L2
roaming feature now. The handoff time is 200 ~ 300ms between the AP. We aim to
short it to 100 ~ 200ms.

The implementation of Web
Authentication(web-login) what we call HTTPS(SSL)is ongoing and
should be releasedon June. It can be software
upgrade.

Best Regards,Mohammed
Fahd

-Oorspronkelijk bericht-
Van:
[EMAIL PROTECTED]namensCraig Waddington
Verzonden: do 11/03/2004
19:15 Aan:
[EMAIL PROTECTED] CC: Onderwerp: [Asterisk-Users] IPC5000 -
Wireless Sip phone
I am looking to buy a wireless
sip phone, probably the IPC5000, I have looked at Wisip phone and read tons
of posts regarding that phone.

Do any * admins have any
feedback on this phone?

Is there any major differences
between the phones, besides looks?

The site has very limited
information regarding prices etc.

Ta.

DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer.

Re: [Asterisk-Users] Night menu not working

2004-03-11 Thread Tilghman Lesher

On Thursday 11 March 2004 14:53, Justin Carlson wrote:
 I am trying to get day and nighttime menus to work in * and no
 matter what time I specify the first include entry that matches the
 number dialed is used.  I have included my extentions.conf and my
 sip phones have a default context of default.

Many people find that using the application GotoIfTime in the dialplan
logic is more intuitive.  Try that.

-Tilghman

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

[Asterisk-Users] MGCP RELOAD function

2004-03-11 Thread Duane Cox

Hello

I was just wondering if anyone was working on the MGCP RELOAD
functionality.

Thanks,
Duane Cox



___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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-oh323, new version 0.5.10

2004-03-11 Thread T. Chan


Dear Michael

Do you foresee implementing these in the near future, one or the other or
both? Thanks

Tc


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michael
Manousos
Sent: Thursday, March 11, 2004 4:49 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] asterisk-oh323, new version 0.5.10



Hi TC,
T.38 FAX and native bridging are not supported by asterisk-oh323.

Michael.


T. Chan wrote:
 Dear Michael,

 Does your H323 driver run T38 Fax? Also, does your H323 driver have the
 capability of just proxying signal, and NOT proxying signal and media,
just
 like the canrevite=yes in the sip scenario? Thanks

 TC

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Michael
 Manousos
 Sent: Wednesday, March 10, 2004 7:11 AM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] asterisk-oh323, new version 0.5.10



 Hello all,

 asterisk-oh323 has been updated. The new version 0.5.10 fixes
 the incorrect answering of H.323 channels (thanks to the people
 of the list who helped to trace the problem). Also, I have added
 support for Gnomemeeting text messages (just for fun).
 Additionally, the new version contains stability improvements.

 This will be the last version using the OpenH323/Pwlib v1.12.2/1.5.2.
 The next version will move on to the latest versions of these
 libraries.

 Regards,
 Michael.


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

 ---
 Incoming mail is certified Virus Free.
 Checked by AVG anti-virus system (http://www.grisoft.com).
 Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004

 ---
 Outgoing mail is certified Virus Free.
 Checked by AVG anti-virus system (http://www.grisoft.com).
 Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004

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

--
./M

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

---
Incoming mail is certified Virus Free.
Checked by AVG anti-virus system (http://www.grisoft.com).
Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004

---
Outgoing mail is certified Virus Free.
Checked by AVG anti-virus system (http://www.grisoft.com).
Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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] Night menu not working

These suggestions may not help get your daytime stuff working, but it
should make life easier later.

On Thu, 2004-03-11 at 14:53, Justin Carlson wrote:
 [general]
 static=yes
 writeprotect=no
 [globals]
 MARYKAY = 21
 RECEPTIONIST = 20
 KATHY = 22
 
 [daytime]
 include = parkedcalls
 exten = t,1,Playback(vm-goodbye)
 exten = t,2,Hangup
 exten = i,1,Playback(invalid)
 
 switch = IAX2/[EMAIL PROTECTED]/dialout
 
 ;sip extentions
 

Make the following section it's own context and have it included in the
above section.

 exten = ${MARYKAY},1,Dial,SIP/21|20
 exten = ${MARYKAY},2,Voicemail,u21
 
 exten = ${RECEPTIONIST},1,Dial,SIP/20|20
 exten = ${RECEPTIONIST},2,Dial,SIP/20SIP/21SIP/22|20
 exten = ${RECEPTIONIST},3,Voicemail,u20
 
 exten = ${KATHY},1,Dial,SIP/22|20
 exten = ${KATHY},2,Voicemail,u22

The below section probably needs to be defined in a section that can be
included in both daytime and nighttime. You may want to call afterhours
to access your voicemail.

 ; for Local Voicemail access
 exten = *98,1,VoicemailMain
 exten = asterisk,1,VoicemailMain
 
 exten = 25,1,Dial,SIP/fax

Maybe this should be included in the above mentioned newly needed
section for your extensions.

 ; voicemail extentions
 exten = 621,1,Voicemail,u21
 exten = 620,1,Voicemail,u20
 exten = 622,1,Voicemail,u22
 exten = 679,1,VoicemailMain
 ; direct extentions
 exten = 201,1,Dial,IAX/[EMAIL PROTECTED]/6515526201
 exten = 307,1,Dial,IAX/[EMAIL PROTECTED]/6515522307
 exten = 309,1,Dial,IAX/[EMAIL PROTECTED]/6515522309
 exten = 313,1,Dial,IAX/[EMAIL PROTECTED]/6515522313
 exten = 317,1,Dial,IAX/[EMAIL PROTECTED]/6515522317
 exten = 601,1,Dial,IAX/[EMAIL PROTECTED]/6515523601
 exten = 603,1,Dial,IAX/[EMAIL PROTECTED]/6515523603
 exten = 609,1,Dial,IAX/[EMAIL PROTECTED]/6515523609
 exten = 664,1,Dial,IAX/[EMAIL PROTECTED]/6515523664
 exten = 694,1,Dial,IAX/[EMAIL PROTECTED]/6515523694
 exten = 816,1,Dial,IAX/[EMAIL PROTECTED]/6515526816
 exten = 817,1,Dial,IAX/[EMAIL PROTECTED]/6515526817
 exten = 821,1,Dial,IAX/[EMAIL PROTECTED]/6515526821
 exten = 842,1,Dial,IAX/[EMAIL PROTECTED]/6515526842
 
 [faxmachine]
 switch = IAX2/[EMAIL PROTECTED]/faxmachine
 
 [nighttime]
 exten = 21,1,Playback(tt-monkeys)
 
 [default]
 include = daytime|8:00-14:48|mon-fri
 include = nighttime

Seems you are missing the days of month and months arguments there.
Also, you would probably want to conditionally include nighttime also.
-- 
Steven Critchfield  [EMAIL PROTECTED]

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

RE[3]: [Asterisk-Users] Crossconnect VoIP and PSTN in India. Is it allowed? {Scanned}

2004-03-11 Thread Vasyl Rublyov

Hi Art,

I am actually interesting more in legal site for this, and most of for 
India.

Vasyl

What PBX systems do you have in the US and Ukrain?
There are a couple of ways I believe you could do this.
a) set the PRI port on the Definity as an EM Tie Line, then have * just
perform the VOIP Gateway functions. (as if it were point to point tie lines)
b) set the PRI port on the Definity as an EM Tie Line, which passes those
calls to * which then handles the Gateway and routing functions for the
calls to the other locations. (point to multipoint)
c) set the PRI port on the Definity as a PRI Line, and use * as a
softswitch/gateway.
you may want to check out Multitech's website. They have a line of products
certified to work with Avaya equipment. Check out their diagrams to get
some ideas on how to set up your network for Toll bypass
Not sure about the legal issues, I know in the US as long as it remains a
private network (not available to the public and only for internal office
use) it's ok.
It's once you open up all that nice low cost routing of voice traffic to the
public, the Public Utilities Commissions start to take notice.
-Art

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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] Cisco 7960 and short delay before voice startsafter ring.

Steve Dolloff wrote:
 We have the same complaint here.  The caller doesn't hear the
 receiver say hello and so no-one knows what's going on. 
 
 Stephen

I get this also, on my Sipura SPA-2000.

-
Andrew Thompson
http://aktzero.com/ 


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

[Asterisk-Users] XML Phone book software.

2004-03-11 Thread Brian R. Swan

Hi gang,

I'm looking into writing a some phone book XML/PHP software for my Cisco 
phones.  Specifically, I'd like to be able to use a web interface (on the 
computer) to maintain a contact list, and then dial from it on the phone.  
Maybe using MySql on the back end or something (to be determined).  Before I 
start, and duplicate something else that exists, I wanted to see if anyone 
has heard of software like that?  Searches of Sourceforge, Freshmeat, and 
Google didn't turn up much or anything.

Thanks!
Brian
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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] Nitsuko 124i interface, anyone?