RE: [Asterisk-Users] Dial Command M(x) Option

[Brian West]

Its in CVS-HEAD no need to patch it..   The bug notes have some examples on
how to use it


bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Shaun Tierney
> Sent: Friday, December 03, 2004 11:50 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Dial Command M(x) Option
> 
> What version of Asterisk should I be applying this patch to?  The patch
> command doesn't seem to be working.  I think because the dates on the
> files
> in Asterisk 1.0.2 don't match the dates in the diff file.  Any ideas?  The
> patch seems to work partially.  When I run patch -p4 < app_dial_rev5.diff
> from the asterisk-1.0.2 directory, this is what I get.
> 
> missing header for unified diff at line 8 of patch
> can't find file to patch at input line 8
> Perhaps you used the wrong -p or --strip option?
> The text leading up to this was:
> --
> |Index: pbx.c
> |===
> |RCS file: /usr/cvsroot/asterisk/pbx.c,v
> |retrieving revision 1.173
> |diff -u -r1.173 pbx.c
> |--- pbx.c  19 Nov 2004 05:18:10 -  1.173
> |+++ pbx.c  22 Nov 2004 22:19:48 -
> --
> File to patch: pbx.c
> patching file pbx.c
> Hunk #1 FAILED at 4935.
> Hunk #2 succeeded at 4986 with fuzz 2 (offset -428 lines).
> 1 out of 2 hunks FAILED -- saving rejects to file pbx.c.rej
> missing header for unified diff at line 174 of patch
> can't find file to patch at input line 174
> Perhaps you used the wrong -p or --strip option?
> The text leading up to this was:
> --
> |Index: include/asterisk/pbx.h
> |===
> |RCS file: /usr/cvsroot/asterisk/include/asterisk/pbx.h,v
> |retrieving revision 1.33
> |diff -u -r1.33 pbx.h
> |--- include/asterisk/pbx.h 13 Nov 2004 22:44:33 -  1.33
> |+++ include/asterisk/pbx.h 22 Nov 2004 22:19:48 -
> --
> File to patch: include/asterisk/pbx.h
> patching file include/asterisk/pbx.h
> Hunk #1 FAILED at 578.
> 1 out of 1 hunk FAILED -- saving rejects to file
> include/asterisk/pbx.h.rej
> missing header for unified diff at line 191 of patch
> can't find file to patch at input line 191
> Perhaps you used the wrong -p or --strip option?
> The text leading up to this was:
> --
> |Index: apps/app_dial.c
> |===
> |RCS file: /usr/cvsroot/asterisk/apps/app_dial.c,v
> |retrieving revision 1.107
> |diff -u -r1.107 app_dial.c
> |--- apps/app_dial.c21 Nov 2004 20:38:32 -  1.107
> |+++ apps/app_dial.c22 Nov 2004 22:19:49 -
> --
> File to patch: apps/app_dial.c
> patching file apps/app_dial.c
> Hunk #1 succeeded at 67 (offset -1 lines).
> Hunk #2 succeeded at 458 (offset -32 lines).
> Hunk #3 FAILED at 466.
> Hunk #4 succeeded at 660 (offset -8 lines).
> Hunk #5 succeeded at 951 (offset -57 lines).
> Hunk #6 succeeded at 1009 (offset -8 lines).
> Hunk #7 succeeded at 974 (offset -57 lines).
> 1 out of 7 hunks FAILED -- saving rejects to file apps/app_dial.c.rej
> 
> Am I just applying the patch incorrectly or something?
> 
> Thanks for the help,
> 
> Shaun
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Brian West
> Sent: Friday, December 03, 2004 10:09 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Dial Command M(x) Option
> 
> 
> http://bugs.digium.com/bug_view_page.php?bug_id=0002905
> 
> > This email and any attached files are confidential and copyright
> > protected.  If you are not the addressee, any dissemination,
> distribution
> > or copying of this communication is strictly prohibited.  Unless
> otherwise
> > expressly agreed in writing, nothing stated in this communication shall
> be
> > legally binding.
> 
> FYI these types of disclaimers ARE STUPID... And crack me up when I see
> them...
> 
> bkw
> 
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RE: [Asterisk-Users] SIP SECURITY WARNING: v1-0 (cvs today) sip contextin general section ignored goes to default instead - allowingunauthorized sip devices to place calls in default context

[Brian West]

It's known that YOU DO this:

sip.conf you do 
[general]
context=from-sip

extensions.conf:
[from-sip]
exten => s,1,Congestion

This is a config issue.  Not really a security issue.

bkw


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Andy Reinke
> Sent: Friday, December 03, 2004 6:48 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Cc: [EMAIL PROTECTED]; [EMAIL PROTECTED]
> Subject: [Asterisk-Users] SIP SECURITY WARNING: v1-0 (cvs today) sip
> contextin general section ignored goes to default instead -
> allowingunauthorized sip devices to place calls in default context
> 
> SIP SECURITY WARNING
> 
> 
> 
> Version: v1-0 (cvs today)
> 
> 
> 
> Problem:  sip context in general section ignored - goes to default -
> allowing unauthorized sip devices to place calls in default context
> 
> 
> 
> Fix [workaround]:
> 
> 
> 
> Remove or rename "default" context in extensions.conf
> 
> 
> 
> Notes:
> 
> 
> 
> I am not sure what other asterisk functionality may be affected by this -
> review your other config files for references to the "default" context.
> Test your configurations to ensure calls are landing in the correct
> context.  I suggest removing "default" and creating others like sip-
> default which include demo and then testing from a sip channel to make
> sure you still hit the demo from a registered device but, not from
> unregistered devices.  Repeat for other channels as necessary.
> 
> 
> 
> Detail:
> 
> 
> 
> I have been working with asterisk for a while now but, had never
> tested/noticed this scenario - I had always created device entries in
> sip.conf for any devices I tested so I never ran into this.  Today on a
> new config the phone came up before I had put anything in sip.conf and I
> thought - let's see what happens if we try to call someone - and it WORKED
> which was the least expected behavior.
> 
> 
> 
> I am using a cisco 7960 with SIP firmware v6.3 (dosen't really matter any
> sip phone will do this) With a bare asterisk build and setup of v1-0
> (pulled from cvs today) on FC3 minimal + asterisk requirements + up2date
> and the configs (sip, extensions) below.
> 
> 
> 
> Without placing any peer,friend,user entries in sip.conf for the phone
> device/extension, I am able to make calls through the "default" context.
> In the below example dialing "500" from a sip phone will execute the inter
> asterisk connection test (IAX) to digium even though the context defined
> in the general section of sip.conf is "sip-unauthorized" which should play
> congestion and hang up (as was suggested in "Getting started with
> asterisk").
> 
> 
> 
> Removing or renaming the "default" context in extensions.conf appears to
> resolve this issue - congestion is played.  However, adding a real
> extension such as 900 and mapping it to something like voicemail shows
> that the context sip-unauthorized is not being used - also the following
> error is logged on the console (verbose = 7) which hints to this as well -
> and explains why congestion was played.  Instead of looking for sip-
> unauthorized as expected it looked for the missing default and then played
> congestion when default was not found.
> 
> 
> 
> Dec  3 20:26:42 NOTICE[15447]: pbx.c:1318 pbx_extension_helper:  Cannot
> find extension context 'default'
> 
> 
> 
> 
> 
> 
> 
> Sip.conf
> 
> [general]
> 
> contex=sip-unauthorized
> 
> port=5060
> 
> bindaddr=0.0.0.0
> 
> localnet=172.16.0.0/255.255.255.0
> 
> 
> 
> 
> 
> 
> 
> Extensions.conf
> 
> [general]
> 
> static=yes
> 
> writeprotect=no
> 
> 
> 
> [globals]
> 
> ;CONSOLE=Console/dsp ; Console interface for demo
> 
> IAXINFO=guest; IAXtel username/password
> 
> ;TRUNK=Zap/g2; Trunk interface
> 
> ;TRUNKMSD=1  ; MSD digits to strip (usually 1
> or 0)
> 
> 
> 
> [macro-stdexten];
> 
> ;
> 
> ; Standard extension macro:
> 
> ;   ${ARG1} - Extension  (we could have used ${MACRO_EXTEN} here as well
> 
> ;   ${ARG2} - Device(s) to ring
> 
> ;
> 
> exten => s,1,Dial(${ARG2},20) ; Ring the
> interface, 20 seconds maximum
> 
> exten => s,2,Goto(s-${DIALSTATUS},1)  ; Jump based
> on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
> 
> 
> 
> exten => s-NOANSWER,1,Voicemail(u${ARG1})  ; If unavailable,
> send to voicemail w/ unavail announce
> 
> exten => s-NOANSWER,2,Goto(default,s,1); If they press #,
> return to start
> 
> 
> 
> exten => s-BUSY,1,Voicemail(b${ARG1})  ; If busy, send to
> voicemail w/ busy announce
> 
> exten => s-BUSY,2,Goto(default,s,1)   ; If they
> press #, return to start
> 
> 
> 
> exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything
> else as no answer
> 
> 
> 
> exten => a,1,VoicemailMain(${ARG1})   ; If they
> press *, send t

RE: [Asterisk-Users] Alpha Paging

[Brian West]

Hylafax can do TAP/IXO via sendpage.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Paul Crick
> Sent: Friday, December 03, 2004 7:52 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Alpha Paging
> 
> > I have a need to be able to send faxes and alpha pages from
> > the same box as asterisk is running on.   Since asterisk
> > handles all of the calls would I need to get a separate fax
> > modem for this or can I run it through one of the FXO ports?
> You're probably gonna need a modem - are you asking if you can plug that
> in
> to your asterisk box? or if you can do what you want directly with a
> Zaptel
> FXO port connected to a phone line?
> 
> Either way.. Hylafax will take care of your faxing requirement, and qpage
> will let you interface with dial up access to a TAP/IXO compliant paging
> gateway for sending alphanumeric pages. I've set up and played with both,
> they're pretty cool. Not sure about them sharing a modem between them,
> have
> to have a poke around..
> 
> Cheers
> Paul
> 
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RE: [Asterisk-Users] Asterisk and Cisco IP Phones

[Brian West]

Pfft ya right if you want half assed supported channel drivers.  Use SIP.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Keith O'Brien
> Sent: Saturday, December 04, 2004 12:57 PM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] Asterisk and Cisco IP Phones
> 
> No you don't have to use SIP.   You can also use the SCCP channel on *
> with Cisco phones.
> 
> 
> 
> 
> 
> Message: 16
> 
> Date: Sat, 4 Dec 2004 12:45:53 +0200
> 
> From: "Walid Azab" <[EMAIL PROTECTED]>
> 
> Subject: [Asterisk-Users] Asterisk and Cisco IP Phones
> 
> To: <[EMAIL PROTECTED]>
> 
> Message-ID: <[EMAIL PROTECTED]>
> 
> Content-Type: text/plain; charset="us-ascii"
> 
> 
> 
> Hello Everyone,
> 
> 
> 
> I want to start using Asterisk with Cisco IP Phones 7960 / 7940/ and 7905.
> 
> Any info or help is appreciated.
> 
> 
> 
> I know I'll have to use SIP but if I want to use the phones off site
> meaning
> 
> from my home for example, how can this be done?
> 
> Also, regarding external lines what are the options for Asterisk?
> 
> 
> 
> Thanks
> 
> Walid
> 
> 


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RE: [Asterisk-Users] Asterisk and Cisco IP Phones

[Brian West]

Let me CLARIFY for those that might not get what I ment.. DO NOT RECOMMEND
SCCP unless you have actually installed and used it.  Its crap... 

SIP is what you want if you use a cisco phone with asterisk.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Brian West
> Sent: Saturday, December 04, 2004 1:33 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Asterisk and Cisco IP Phones
> 
> Pfft ya right if you want half assed supported channel drivers.  Use SIP.
> 
> bkw
> 
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Keith O'Brien
> > Sent: Saturday, December 04, 2004 12:57 PM
> > To: [EMAIL PROTECTED]
> > Subject: RE: [Asterisk-Users] Asterisk and Cisco IP Phones
> >
> > No you don't have to use SIP.   You can also use the SCCP channel on *
> > with Cisco phones.
> >
> >
> >
> >
> >
> > Message: 16
> >
> > Date: Sat, 4 Dec 2004 12:45:53 +0200
> >
> > From: "Walid Azab" <[EMAIL PROTECTED]>
> >
> > Subject: [Asterisk-Users] Asterisk and Cisco IP Phones
> >
> > To: <[EMAIL PROTECTED]>
> >
> > Message-ID: <[EMAIL PROTECTED]>
> >
> > Content-Type: text/plain; charset="us-ascii"
> >
> >
> >
> > Hello Everyone,
> >
> >
> >
> > I want to start using Asterisk with Cisco IP Phones 7960 / 7940/ and
> 7905.
> >
> > Any info or help is appreciated.
> >
> >
> >
> > I know I'll have to use SIP but if I want to use the phones off site
> > meaning
> >
> > from my home for example, how can this be done?
> >
> > Also, regarding external lines what are the options for Asterisk?
> >
> >
> >
> > Thanks
> >
> > Walid
> >
> >
> 
> 
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RE: [Asterisk-Users] Voicemail for Current Extension?

[Brian West]

Forgot the s

VoiceMailMain(s${CALLERIDNUM})

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Yair Hakak
> Sent: Saturday, December 04, 2004 2:08 PM
> To: Ian Chilton; Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Voicemail for Current Extension?
> 
> Hello Ian,
> 
> 
> VoiceMailMain(${CALLERIDNUM})
> 
> should do the trick (unless you have the blocked number problem a
> previous poster had)
> -yair
> 
> 
> On Sat, 4 Dec 2004 20:01:58 +, Ian Chilton
> <[EMAIL PROTECTED]> wrote:
> > Hi,
> >
> > Is it possible to create an extension (say *1) that will give access to
> > the voicemail for the current extension without entering the mailbox or
> > password?
> >
> > (or if this is not possible, at least not have to enter the mailbox -
> > only the password?)
> >
> > Thanks!
> >
> > --ian
> >
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RE: [Asterisk-Users] G.729 algorithm?

[Brian West]

You can get example code from the ITU.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Christopher Vance
> Sent: Monday, December 06, 2004 6:33 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Cc: Asterisk Developer Mailing List
> Subject: Re: [Asterisk-Users] G.729 algorithm?
> 
> On Sun, Dec 05, 2004 at 12:53:31PM +0100, Roy Sigurd Karlsbakk wrote:
> >does anyone know where I can find the algorithm?
> 
> If it's not disclosed sufficiently in the patent, you could argue the
> patent is invalid.  :-)
> 
> --
> Christopher Vance
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[Asterisk-Users] ChanSpy by anthm and more...

[Brian West]

Everyone we have a few new things to give back to the asterisk community.

http://bugs.digium.com/bug_view_page.php?bug_id=0002379
http://bugs.digium.com/bug_view_page.php?bug_id=0002380
http://bugs.digium.com/bug_view_page.php?bug_id=0002381

These include app_chanspy, the ability to spy on ANY bridged call taking
place inside asterisk.  NOT just ZAP as with ZapScan/Barge.

Native format_* files being used for moh.  Reload enabled res_musiconhold.  

format_mp3.c that produces SLNR output to asterisk, format_slinear.c for raw
headerless audio, format_base65_wav_gsm.c aka wav49 held in a base64
containers(it can read and playback from these .b64 files)

All this is thanks to my employer asterlink.com and anthm.

So everyone please test and provide feedback.

Thanks,
Brian
Asterlink.com
PS: More to come at a later date.

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RE: [Asterisk-Users] ChanSpy by anthm and more...

[Brian West]

Also don't forget to visit us at Astricon... :)

Brian
Asterlink.com

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Brian West
> Sent: Sunday, September 05, 2004 12:26 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] ChanSpy by anthm and more...
> 
> Everyone we have a few new things to give back to the asterisk community.
> 
> http://bugs.digium.com/bug_view_page.php?bug_id=0002379
> http://bugs.digium.com/bug_view_page.php?bug_id=0002380
> http://bugs.digium.com/bug_view_page.php?bug_id=0002381
> 
> These include app_chanspy, the ability to spy on ANY bridged call taking
> place inside asterisk.  NOT just ZAP as with ZapScan/Barge.
> 
> Native format_* files being used for moh.  Reload enabled res_musiconhold.
> 
> format_mp3.c that produces SLNR output to asterisk, format_slinear.c for
> raw
> headerless audio, format_base65_wav_gsm.c aka wav49 held in a base64
> containers(it can read and playback from these .b64 files)
> 
> All this is thanks to my employer asterlink.com and anthm.
> 
> So everyone please test and provide feedback.
> 
> Thanks,
> Brian
> Asterlink.com
> PS: More to come at a later date.
> 
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RE: [Asterisk-Users] ChanSpy by anthm and more...

[Brian West]

http://bugs.digium.com/bug_view_page.php?bug_id=0002384

Also res_sqlite is out... :)

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Brian West
> Sent: Sunday, September 05, 2004 12:57 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] ChanSpy by anthm and more...
> 
> Also don't forget to visit us at Astricon... :)
> 
> Brian
> Asterlink.com
> 
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Brian West
> > Sent: Sunday, September 05, 2004 12:26 PM
> > To: [EMAIL PROTECTED]
> > Subject: [Asterisk-Users] ChanSpy by anthm and more...
> >
> > Everyone we have a few new things to give back to the asterisk
> community.
> >
> > http://bugs.digium.com/bug_view_page.php?bug_id=0002379
> > http://bugs.digium.com/bug_view_page.php?bug_id=0002380
> > http://bugs.digium.com/bug_view_page.php?bug_id=0002381
> >
> > These include app_chanspy, the ability to spy on ANY bridged call taking
> > place inside asterisk.  NOT just ZAP as with ZapScan/Barge.
> >
> > Native format_* files being used for moh.  Reload enabled
> res_musiconhold.
> >
> > format_mp3.c that produces SLNR output to asterisk, format_slinear.c for
> > raw
> > headerless audio, format_base65_wav_gsm.c aka wav49 held in a base64
> > containers(it can read and playback from these .b64 files)
> >
> > All this is thanks to my employer asterlink.com and anthm.
> >
> > So everyone please test and provide feedback.
> >
> > Thanks,
> > Brian
> > Asterlink.com
> > PS: More to come at a later date.
> >
> > ___
> > 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
> 
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RE: [Asterisk-Users] Pause or Wait character in Dial command?

[Brian West]

Lowercase w for wait.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Arick Davis
> Sent: Sunday, September 05, 2004 1:20 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Pause or Wait character in Dial command?
> 
> Is there a Pause or Wait character for dial tone command for the Dial
> command? Like in a modem Dial string? I'm having issues with the FXO
> Analog line not returning dial tone quick enough for * to recognize.
> 
> Arick
> 

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RE: [Asterisk-Users] Wildcards and variable number of digits

[Brian West]

Just to clarify the usage of the . wildcard in your dialplan.

Here is the proper usage of this feature which seems to not be documented
ANYWHERE very well.

[default]
include => other
exten => _712XXX,1,NoOp,Blah

[other]
exten => _7.,1,NoOp,somethingelse


The extensions in the current context win over an include.. only if
something doesn't specifically match in [default] but does as a wildcard as
an include then it will work.  Remember includes are your friend.

bkw


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Karl Brose
> Sent: Sunday, September 05, 2004 1:50 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Wildcards and variable number of digits
> 
> 
> The problem you are having is due to the way chan_phone was designed.
> The distributed driver does not buffer the entire phone number dialed
> and then send it on to the PBX,
> like a SIP phone would, but instead scans the dial plan after every
> digit is entered to look for a match.
> The solution is to only use fixed length extension patterns, but at the
> same time requires different dial plans
> for the Phone/phoneX devices.  I you're only dialing PSTN numbers it's
> not so bad, but many VOIP providers
> have all kinds of numbering plans. On the other hand, fixed patterns are
> nice since you don't have to
> press any "call" or "dial" buttons to make the call.
> 
> I have a new chan_phone driver which solves this issue by buffering the
> dial string until the user presses
> the pound (#) key to send the phone number to the pbx.  The features can
> be toggled on/off any time by dialing
> *1#  or  *0#  or in the config file with a mode "buffered"  which is
> otherwise the same as "dialtone"
> 
> 
> 
> 
> Eric Jacksch wrote:
> 
> >Greetings,
> >
> >I'm having a miserable time getting Asterisk working with FWD.  All the
> >samples show something like...
> >
> > exten => _7., 
> >
> >How do I get Asterisk to wait until the user is finished dialing instead
> of
> >trying as soon as it gets the second digit?
> >
> >I can use _7XXX, and dial the FWD 3-digit test numbers fine, but I'd like
> to
> >be able to dial others...
> >
> >Same problem for outside analog line...how do I convince Asterisk to send
> >anything that starts with a "9" to it?
> >
> >If it makes a difference, I'm playing with some QuickNet cards to learn
> the
> >system...then I'll likely buy some other cards with higher capacity.
> >
> >Thanks,
> >Eric
> >
> >
> >
> >
> >___
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> >[EMAIL PROTECTED]
> >http://lists.digium.com/mailman/listinfo/asterisk-users
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> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
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RE: [Asterisk-Users] Wildcards and variable number of digits

[Brian West]

Actually it does the proper usage of the "." char in your dial plan should
solve this problem.  It's not the channel driver that's doing this its
asterisk.  You need to sandbox a wildcard into its own context then include
it.  Otherwise it wins NO MATER WHAT.  This way an extension defined within
the current context wins over the included wildcard context.
S=
bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Eric Jacksch
> Sent: Sunday, September 05, 2004 2:50 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Wildcards and variable number of digits
> 
> Not sure I understand..does that help my problem of not being able to
> enter
> sufficient digits, or is that a consideration once I get a driver that
> allows me to # terminate the dialing string?
> 
> 
> On 2004-09-05 15:00, "Brian West" <[EMAIL PROTECTED]> wrote:
> 
> > Just to clarify the usage of the . wildcard in your dialplan.
> >
> > Here is the proper usage of this feature which seems to not be
> documented
> > ANYWHERE very well.
> >
> > [default]
> > include => other
> > exten => _712XXX,1,NoOp,Blah
> >
> > [other]
> > exten => _7.,1,NoOp,somethingelse
> >
> >
> > The extensions in the current context win over an include.. only if
> > something doesn't specifically match in [default] but does as a wildcard
> as
> > an include then it will work.  Remember includes are your friend.
> >
> > bkw
> >
> >
> >> -Original Message-
> >> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> >> [EMAIL PROTECTED] On Behalf Of Karl Brose
> >> Sent: Sunday, September 05, 2004 1:50 PM
> >> To: Asterisk Users Mailing List - Non-Commercial Discussion
> >> Subject: Re: [Asterisk-Users] Wildcards and variable number of digits
> >>
> >>
> >> The problem you are having is due to the way chan_phone was designed.
> >> The distributed driver does not buffer the entire phone number dialed
> >> and then send it on to the PBX,
> >> like a SIP phone would, but instead scans the dial plan after every
> >> digit is entered to look for a match.
> >> The solution is to only use fixed length extension patterns, but at the
> >> same time requires different dial plans
> >> for the Phone/phoneX devices.  I you're only dialing PSTN numbers it's
> >> not so bad, but many VOIP providers
> >> have all kinds of numbering plans. On the other hand, fixed patterns
> are
> >> nice since you don't have to
> >> press any "call" or "dial" buttons to make the call.
> >>
> >> I have a new chan_phone driver which solves this issue by buffering the
> >> dial string until the user presses
> >> the pound (#) key to send the phone number to the pbx.  The features
> can
> >> be toggled on/off any time by dialing
> >> *1#  or  *0#  or in the config file with a mode "buffered"  which is
> >> otherwise the same as "dialtone"
> >>
> >>
> >>
> >>
> >> Eric Jacksch wrote:
> >>
> >>> Greetings,
> >>>
> >>> I'm having a miserable time getting Asterisk working with FWD.  All
> the
> >>> samples show something like...
> >>>
> >>> exten => _7., 
> >>>
> >>> How do I get Asterisk to wait until the user is finished dialing
> instead
> >> of
> >>> trying as soon as it gets the second digit?
> >>>
> >>> I can use _7XXX, and dial the FWD 3-digit test numbers fine, but I'd
> like
> >> to
> >>> be able to dial others...
> >>>
> >>> Same problem for outside analog line...how do I convince Asterisk to
> send
> >>> anything that starts with a "9" to it?
> >>>
> >>> If it makes a difference, I'm playing with some QuickNet cards to
> learn
> >> the
> >>> system...then I'll likely buy some other cards with higher capacity.
> >>>
> >>> Thanks,
> >>> Eric
> >>>
> >>>
> >>> --
> --
> >>>
> >>> ___
> >>> Asterisk-Users mailing list
> >>> [EMAIL PROTECTED]
> >>> http://lists.digium.com/mailman/listinfo/aste

RE: [Asterisk-Users] ChanSpy by anthm and more...

[Brian West]

No it doesn't its just a nice standalone res that allows you to use SQLite
from the dialplan, cli and as a CDR engine and sqlite_Switch.

Its great for a standalone pbx because you can do something like this:

exten => s,1,SQL(SELECT total,balance,lastpaydate FROM customers WHERE
callerid=\'${CALLERIDNUM}\')
exten => s,2,Playback(your-balance-is)
exten => s,3,SayNumber(${balance})
exten => s,4,Playback(dollars)

or something crazy like that.. It turns the first row into channel vars and
you cam use those.  This is a lot more powerful and flexible than
DBget/DBput.


Also say you want to do the same query from the CLI 

asterisk*CLI> sql use cdr
asterisk*CLI>
now using /var/lib/asterisk/sqlite/cdr.db

asterisk*CLI> select count(1) from cdr;
asterisk*CLI>

|count(1)|
|131|

Brian
Asterlink.com


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Kannaiyan Natesan
> Sent: Sunday, September 05, 2004 3:41 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] ChanSpy by anthm and more...
> 
> Does it removes the need of external databases (mysql, postgres) or it
> will
> work with existing databases?
> 
> -Kannaiyan
> 
> 
> - Original Message -
> From: "Brian West" <[EMAIL PROTECTED]>
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> <[EMAIL PROTECTED]>
> Sent: Sunday, September 05, 2004 7:04 PM
> Subject: RE: [Asterisk-Users] ChanSpy by anthm and more...
> 
> 
> > http://bugs.digium.com/bug_view_page.php?bug_id=0002384
> >
> > Also res_sqlite is out... :)
> >
> > bkw
> >
> >> -Original Message-
> >> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> >> [EMAIL PROTECTED] On Behalf Of Brian West
> >> Sent: Sunday, September 05, 2004 12:57 PM
> >> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> >> Subject: RE: [Asterisk-Users] ChanSpy by anthm and more...
> >>
> >> Also don't forget to visit us at Astricon... :)
> >>
> >> Brian
> >> Asterlink.com
> >>
> >> > -Original Message-
> >> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> >> > [EMAIL PROTECTED] On Behalf Of Brian West
> >> > Sent: Sunday, September 05, 2004 12:26 PM
> >> > To: [EMAIL PROTECTED]
> >> > Subject: [Asterisk-Users] ChanSpy by anthm and more...
> >> >
> >> > Everyone we have a few new things to give back to the asterisk
> >> community.
> >> >
> >> > http://bugs.digium.com/bug_view_page.php?bug_id=0002379
> >> > http://bugs.digium.com/bug_view_page.php?bug_id=0002380
> >> > http://bugs.digium.com/bug_view_page.php?bug_id=0002381
> >> >
> >> > These include app_chanspy, the ability to spy on ANY bridged call
> >> > taking
> >> > place inside asterisk.  NOT just ZAP as with ZapScan/Barge.
> >> >
> >> > Native format_* files being used for moh.  Reload enabled
> >> res_musiconhold.
> >> >
> >> > format_mp3.c that produces SLNR output to asterisk, format_slinear.c
> >> > for
> >> > raw
> >> > headerless audio, format_base65_wav_gsm.c aka wav49 held in a base64
> >> > containers(it can read and playback from these .b64 files)
> >> >
> >> > All this is thanks to my employer asterlink.com and anthm.
> >> >
> >> > So everyone please test and provide feedback.
> >> >
> >> > Thanks,
> >> > Brian
> >> > Asterlink.com
> >> > PS: More to come at a later date.
> >> >
> >> > ___
> >> > 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
> >
> >
> 
> 
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[Asterisk-Users] res_perl

[Brian West]

Latest version of res_perl is up also.

http://www.bkw.org/~brian/res_perl.tar.gz

Brian
Asterlink.com

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RE: [Asterisk-Users] Wildcards and variable number of digits

[Brian West]

And your using chan_phone?

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Eric Jacksch
> Sent: Sunday, September 05, 2004 4:17 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Wildcards and variable number of digits
> 
> Here are the snippets...I changed things to "9" just in case...
> 
> No matter what I do, I get to dial 9 plus two more digits...
> 
> 
> [internal]
> ;include => extensions
> ;include => tovpc
> include => tofwd
> ; this should just dial myself
> exten => _999XXX,1,Dial,${P1}
> 
> 
> 
> [macro-dialwfd]
> exten => s,1,SetCallerID(${FWDCIDNAME})
> exten =>
> s,2,Dial(IAX2/${FWDNUMBER}:[EMAIL PROTECTED]/${ARG1},${ARG2},
> r)
> exten => s,3,Hangup
> 
> ; Prefix 9 to dial out to Free World Dial
> [tofwd]
> ; when I do this, it gives me a ring (and then busy) as soon as I
> ; dial the second digit
> exten => _9X.,1,Macro(dialwfd,${EXTEN:1},60)
> 
> 
> 
> -Original Message-
> From: Brian West [mailto:[EMAIL PROTECTED]
> Sent: Sun 2004-09-05 15:55
> To:   'Asterisk Users Mailing List - Non-Commercial Discussion'
> Cc:
> Subject:  RE: [Asterisk-Users] Wildcards and variable number of digits
> Actually it does the proper usage of the "." char in your dial plan should
> solve this problem.  It's not the channel driver that's doing this its
> asterisk.  You need to sandbox a wildcard into its own context then
> include
> it.  Otherwise it wins NO MATER WHAT.  This way an extension defined
> within
> the current context wins over the included wildcard context.
> S=
> bkw
> 
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Eric Jacksch
> > Sent: Sunday, September 05, 2004 2:50 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] Wildcards and variable number of digits
> >
> > Not sure I understand..does that help my problem of not being able to
> > enter
> > sufficient digits, or is that a consideration once I get a driver that
> > allows me to # terminate the dialing string?
> >
> >
> > On 2004-09-05 15:00, "Brian West" <[EMAIL PROTECTED]> wrote:
> >
> > > Just to clarify the usage of the . wildcard in your dialplan.
> > >
> > > Here is the proper usage of this feature which seems to not be
> > documented
> > > ANYWHERE very well.
> > >
> > > [default]
> > > include => other
> > > exten => _712XXX,1,NoOp,Blah
> > >
> > > [other]
> > > exten => _7.,1,NoOp,somethingelse
> > >
> > >
> > > The extensions in the current context win over an include.. only if
> > > something doesn't specifically match in [default] but does as a
> wildcard
> > as
> > > an include then it will work.  Remember includes are your friend.
> > >
> > > bkw
> > >
> > >
> > >> -Original Message-
> > >> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > >> [EMAIL PROTECTED] On Behalf Of Karl Brose
> > >> Sent: Sunday, September 05, 2004 1:50 PM
> > >> To: Asterisk Users Mailing List - Non-Commercial Discussion
> > >> Subject: Re: [Asterisk-Users] Wildcards and variable number of digits
> > >>
> > >>
> > >> The problem you are having is due to the way chan_phone was designed.
> > >> The distributed driver does not buffer the entire phone number dialed
> > >> and then send it on to the PBX,
> > >> like a SIP phone would, but instead scans the dial plan after every
> > >> digit is entered to look for a match.
> > >> The solution is to only use fixed length extension patterns, but at
> the
> > >> same time requires different dial plans
> > >> for the Phone/phoneX devices.  I you're only dialing PSTN numbers
> it's
> > >> not so bad, but many VOIP providers
> > >> have all kinds of numbering plans. On the other hand, fixed patterns
> > are
> > >> nice since you don't have to
> > >> press any "call" or "dial" buttons to make the call.
> > >>
> > >> I have a new chan_phone driver which solves this issue by buffering
> the
> > >> dial string until the user presses
> > >> the pound (#) key to send the phone number t

RE: [Asterisk-Users] Wildcards and variable number of digits

[Brian West]

No, newer code does exactly how I described it.  Specific matches in the
current context override wildcards in any included context.  I have tested
this and that's how Mark himself says it works.  This is how it should work
if I understand it correctly and I usually do, ast_matchmore_extension is
what makes that possible.   I don't have any zap hooked up to double check
this.   Jeremy did that in chan_skinny so it may very well be the channel
driver at fault.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Karl Brose
> Sent: Sunday, September 05, 2004 4:33 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Wildcards and variable number of digits
> 
> 
> No Brian,
> The old driver scans the ENTIRE dial plan on EVERY digit dialed so no
> matter where, if you have a
> "." wildcard in the plan, it will match always on the first digit dialed.
> It is the driver that does this.
> If you use a SIP phone, or any technology that presents a complete dial
> string, then you are correct
> with your examples.
> 
> 
> Brian West wrote:
> 
> >Actually it does the proper usage of the "." char in your dial plan
> should
> >solve this problem.  It's not the channel driver that's doing this its
> >asterisk.  You need to sandbox a wildcard into its own context then
> include
> >it.  Otherwise it wins NO MATER WHAT.  This way an extension defined
> within
> >the current context wins over the included wildcard context.
> >S=
> >bkw
> >
> >
> >
> >>-Original Message-
> >>From: [EMAIL PROTECTED] [mailto:asterisk-users-
> >>[EMAIL PROTECTED] On Behalf Of Eric Jacksch
> >>Sent: Sunday, September 05, 2004 2:50 PM
> >>To: Asterisk Users Mailing List - Non-Commercial Discussion
> >>Subject: Re: [Asterisk-Users] Wildcards and variable number of digits
> >>
> >>Not sure I understand..does that help my problem of not being able to
> >>enter
> >>sufficient digits, or is that a consideration once I get a driver that
> >>allows me to # terminate the dialing string?
> >>
> >>
> >>On 2004-09-05 15:00, "Brian West" <[EMAIL PROTECTED]> wrote:
> >>
> >>
> >>
> >>>Just to clarify the usage of the . wildcard in your dialplan.
> >>>
> >>>Here is the proper usage of this feature which seems to not be
> >>>
> >>>
> >>documented
> >>
> >>
> >>>ANYWHERE very well.
> >>>
> >>>[default]
> >>>include => other
> >>>exten => _712XXX,1,NoOp,Blah
> >>>
> >>>[other]
> >>>exten => _7.,1,NoOp,somethingelse
> >>>
> >>>
> >>>The extensions in the current context win over an include.. only if
> >>>something doesn't specifically match in [default] but does as a
> wildcard
> >>>
> >>>
> >>as
> >>
> >>
> >>
> >
> >
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
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RE: [Asterisk-Users] Wildcards and variable number of digits

[Brian West]

Well then chan_phone is broken and shouldn't take much work to fix it.

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Karl Brose
> Sent: Sunday, September 05, 2004 7:36 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Wildcards and variable number of digits
> 
> Amazing,
> We are talking about CURRENT CVS CODE chan_phone the way it DOES work,
> not zap, not anything else, not the way it SHOULD work.
> 
> CHAN_PHONE scans the ENTIRE dial plan on EVERY digit dialed
> in dialtone mode and what you describe does not work for chan_phone.
> Why is it so hard to accept the facts?
> 
> 
> 
> 
> Brian West wrote:
> 
> >No, newer code does exactly how I described it.  Specific matches in the
> >current context override wildcards in any included context.  I have
> tested
> >this and that's how Mark himself says it works.  This is how it should
> work
> >if I understand it correctly and I usually do, ast_matchmore_extension is
> >
> Sorry to ruin your theory.
> 
> 
> >what makes that possible.   I don't have any zap hooked up to double
> check
> >this.   Jeremy did that in chan_skinny so it may very well be the channel
> >driver at fault.
> >
> >bkw
> >
> >
> >
> >>-Original Message-
> >>From: [EMAIL PROTECTED] [mailto:asterisk-users-
> >>[EMAIL PROTECTED] On Behalf Of Karl Brose
> >>Sent: Sunday, September 05, 2004 4:33 PM
> >>To: Asterisk Users Mailing List - Non-Commercial Discussion
> >>Subject: Re: [Asterisk-Users] Wildcards and variable number of digits
> >>
> >>
> >>No Brian,
> >>The old driver scans the ENTIRE dial plan on EVERY digit dialed so no
> >>matter where, if you have a
> >>"." wildcard in the plan, it will match always on the first digit
> dialed.
> >>It is the driver that does this.
> >>If you use a SIP phone, or any technology that presents a complete dial
> >>string, then you are correct
> >>with your examples.
> >>
> >>
> >>
> >
> >
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
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RE: [Asterisk-Users] Wildcards and variable number of digits

[Brian West]

Why do I even bother trying to help...  I even pointed out that the channel
driver is at fault... I only pointed out how it should work and you get an
attitude about it GOOD JOB... Check out line 749 in chan_phone.c you'll see
and compare it to chan_skinny.c and see if maybe that fixes it(hint
chanmatch vs matchmore).  I'll have a friend with the hardware next Tuesday
and maybe we'll have a patch to fix this.

bkw



> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Karl Brose
> Sent: Sunday, September 05, 2004 7:36 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Wildcards and variable number of digits
> 
> Amazing,
> We are talking about CURRENT CVS CODE chan_phone the way it DOES work,
> not zap, not anything else, not the way it SHOULD work.
> 
> CHAN_PHONE scans the ENTIRE dial plan on EVERY digit dialed
> in dialtone mode and what you describe does not work for chan_phone.
> Why is it so hard to accept the facts?
> 
> 
> 
> 
> Brian West wrote:
> 
> >No, newer code does exactly how I described it.  Specific matches in the
> >current context override wildcards in any included context.  I have
> tested
> >this and that's how Mark himself says it works.  This is how it should
> work
> >if I understand it correctly and I usually do, ast_matchmore_extension is
> >
> Sorry to ruin your theory.
> 
> 
> >what makes that possible.   I don't have any zap hooked up to double
> check
> >this.   Jeremy did that in chan_skinny so it may very well be the channel
> >driver at fault.
> >
> >bkw
> >
> >
> >
> >>-Original Message-
> >>From: [EMAIL PROTECTED] [mailto:asterisk-users-
> >>[EMAIL PROTECTED] On Behalf Of Karl Brose
> >>Sent: Sunday, September 05, 2004 4:33 PM
> >>To: Asterisk Users Mailing List - Non-Commercial Discussion
> >>Subject: Re: [Asterisk-Users] Wildcards and variable number of digits
> >>
> >>
> >>No Brian,
> >>The old driver scans the ENTIRE dial plan on EVERY digit dialed so no
> >>matter where, if you have a
> >>"." wildcard in the plan, it will match always on the first digit
> dialed.
> >>It is the driver that does this.
> >>If you use a SIP phone, or any technology that presents a complete dial
> >>string, then you are correct
> >>with your examples.
> >>
> >>
> >>
> >
> >
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RE: [Asterisk-Users] OT - Experience using Gmail for AsteriskMailingList

[Brian West]

STOP THIS NOW.. god you people amaze me.  Its just a sig.  GET OVER IT!!!

NEXT!!!

bkw

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RE: [Asterisk-Users] iaxy vs sipura

[Brian West]

ADPCM != G.726 they are a little bit different.  codec_adpcm.c and
codec_g726.c

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Benjamin on Asterisk Mailing Lists
> Sent: Tuesday, September 07, 2004 10:24 PM
> To: Lyle Giese
> Cc: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] iaxy vs sipura
> 
> On Tue, 7 Sep 2004 15:40:00 -0500, Lyle Giese <[EMAIL PROTECTED]>
> wrote:
> > Hmmm, I thought 56k modems were 56k outbound only and maxs at 33k
> inbound
> > or did the standard change again when I wasn't looking?
> >
> > And besides when did you get better than 28.8 through a hotel PBX?  >33k
> > with a 56k modem in the real world is not that common.
> 
> Don't mix up the content of different posts.
> 
> In one post, I explained how I used IAX and ILBC successfully on a sub
> 20k dialup link in Egypt (that was not a hotel but an office, BTW) and
> in a different paragraph I mentioned hotels in foreign countries as
> one of many locations where things can go funny.
> 
> In another post, I responded to a question whether or not it might be
> possible to use an IAXy on a 33k or 56k dialup connection. There was
> no mentioning of any hotels and I did say that I have not used an IAXy
> myself. My comment that the IAXy *should* work on a 56k link was based
> on the fact that it supports ADPCM and if you check out the specs of
> most DSPs which do ADPCM you will find that they do 16, 24, 32 and 40
> kbps, although the ITU ADPCM recommendation (G.726) talks about 32
> kbps. Which chip the IAXy uses and which modes that chip supports, I
> guess only Digium can tell you. If you want to be absolutely sure,
> then you may just want to try it out.
> 
> In fact, I think it would be very welcome if somebody with actual
> experience testing the IAXy on the road posted some feedback here.
> There is nothing that beats testing in the field.
> 
> rgds
> benjk
> 
> --
> Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
> Tokyo, Japan.
> 
> NB: Spam filters in place. Messages unrelated to the * mailing lists
> may get trashed.
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RE: [Asterisk-Users] SIP and */#

[Brian West]

After small review of the chan_sip.c you should turn on pedantic sipchecking


pedantic=yes in sip.conf [general]

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Roy Sigurd Karlsbakk
> Sent: Wednesday, September 08, 2004 8:16 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] SIP and */#
> 
> filed as 0002399
> 
> On 8. sep. 2004, at 14.25, [EMAIL PROTECTED] wrote:
> 
> >
> >
> > On Wed, 8 Sep 2004, Roy Sigurd Karlsbakk wrote:
> >
> >> hi all
> >>
> >> I'm trying to setup call divertion with the standard
> >>
> >> *21*numbertodivertto#
> >>
> >> etc
> >>
> >> but...
> >> When I dial such a number from a SIP client, it generally works quite
> >> badly
> >> most of the ones I've tried can handle *, but none, or at least few,
> >> can handle #
> >>
> >> Is this a SIP protocol weakness, or what is this?
> >
> >
> > I noticed that X-Lite sends the # like "URL-encoded"
> >
> > For example to dial $, X-Lite sends
> >
> > "To:  ;tag=as3355637e"
> >
> > Asterisk obviously doesn't convert that back.
> >
> > I don't know whether it should, or whether X-Lite shouldn't encode like
> > that.
> >
> > Probably, Asterisk should be fixed.
> >
> > Steve
> >
> > ___
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RE: [Asterisk-Users] New ChanSpy and MOH Patch

[Brian West]

Updated patch.. with README 

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Michael Workman
> Sent: Wednesday, September 08, 2004 8:01 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] New ChanSpy and MOH Patch
> 
> 
> I was wondering if anyone was able to get this patch to work...
> I tried everything and cant get the hold music to work
> 
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RE: [Asterisk-Users] New ChanSpy and MOH Patch

[Brian West]

You just setup

[moh_files]

someclass => /path/to/dir/with/files


The files in that dir can be ANY files asterisk can play via the playback ..
if you get the latest format_mp3 from asterisk-addons it can do MP3 also..
in addition playback can play MP3's then too.

bkw


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Michael Workman
> Sent: Wednesday, September 08, 2004 9:09 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] New ChanSpy and MOH Patch
> 
> Yes the one WITH README that does not tell you what file Extension to name
> your files nor does it tell you how to set it up and use the ulaw part...
> 
> I get nothing... Says its starting music on hold but I hear nothing
> 
> 
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Brian West
> Sent: Wednesday, September 08, 2004 10:00 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] New ChanSpy and MOH Patch
> 
> Updated patch.. with README
> 
> bkw
> 
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Michael Workman
> > Sent: Wednesday, September 08, 2004 8:01 PM
> > To: [EMAIL PROTECTED]
> > Subject: [Asterisk-Users] New ChanSpy and MOH Patch
> >
> >
> > I was wondering if anyone was able to get this patch to work...
> > I tried everything and cant get the hold music to work
> >
> > ___
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RE: [Asterisk-Users] Asterisk-Addons Changes

[Brian West]

-I../asterisk seems to be the key

cd /usr/src
cvs checkout asterisk asterisk-addons

cd asterisk-addons
make


bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Brancaleoni Matteo
> Sent: Thursday, September 09, 2004 12:05 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Asterisk-Addons Changes
> 
> seems that asterisk isn't installed
> 
> Il gio, 2004-09-09 alle 18:48, Michael Workman ha scritto:
> > Well this is what I am getting
> >
> >
> >
> > [EMAIL PROTECTED] asterisk-addons]$ make
> > ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE  -I/usr/include/mysql   `ls
> *.c`
> > cdr_addon_mysql.c:17:29: asterisk/config.h: No such file or directory
> > cdr_addon_mysql.c:18:30: asterisk/options.h: No such file or directory
> > cdr_addon_mysql.c:19:30: asterisk/channel.h: No such file or directory
> > cdr_addon_mysql.c:20:26: asterisk/cdr.h: No such file or directory
> > cdr_addon_mysql.c:21:29: asterisk/module.h: No such file or directory
> > cdr_addon_mysql.c:22:29: asterisk/logger.h: No such file or directory
> > cdr_addon_mysql.c:23:26: asterisk/cli.h: No such file or directory
> > make -C format_mp3 all
> > make[1]: Entering directory `/home/asterisk/asterisk-addons/format_mp3'
> > gcc -pipe  -Wall -Wstrict-prototypes -Wmissing-prototypes
> > -Wmissing-declarations   -D_REENTRANT -D_GNU_SOURCE  -O6-c -o
> common.o
> > common.c
> > common.c:1:29: asterisk/logger.h: No such file or directory
> > common.c: In function `decode_header':
> > common.c:93: warning: implicit declaration of function `ast_log'
> > common.c:93: error: `LOG_WARNING' undeclared (first use in this
> function)
> > common.c:93: error: (Each undeclared identifier is reported only once
> > common.c:93: error: for each function it appears in.)
> > make[1]: *** [common.o] Error 1
> > make[1]: Leaving directory `/home/asterisk/asterisk-addons/format_mp3'
> > make: *** [format_mp3/format_mp3.so] Error 2
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of
> Brancaleoni
> > Matteo
> > Sent: Thursday, September 09, 2004 12:43 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] Asterisk-Addons Changes
> >
> > Hi
> > Il gio, 2004-09-09 alle 18:18, Michael Workman ha scritto:
> > > I just downloaded it now off the CVS and it will no longer compile
> >
> > this kind of messages are only waste on bandwidth & space.
> >
> > please:
> > * don't send a message like this
> > OR
> > * paste the error into the email, if you need support OR
> > * try to resolve the issue and inform the ml
> >
> > --
> > Brancaleoni Matteo <[EMAIL PROTECTED]> Espia Srl
> >
> > ___
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
> --
> Brancaleoni Matteo <[EMAIL PROTECTED]>
> Espia Srl
> 
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RE: [Asterisk-Users] Valet Park Application

[Brian West]

No it doesn't/shouldn't..  If a call is already parked in that location you
shouldn't be able to complete the transfer and you'll have to press resume
and try again.  

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Kevin
> Sent: Friday, September 10, 2004 3:47 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: [Asterisk-Users] Valet Park Application
> 
> I love the functionality of the Valet Park Application.  I have a
> question regarding its operation.  The problem I am having when there is
> a call already parked on specific park extension.  If a caller uses
> 'blind' transfer on a Cisco Phone the caller gets disconnected.  Can any
> offer any suggestions on how to prevent the transfer from taking place?
> 
> 
> 
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RE: [Asterisk-Users] Valet Park Application

[Brian West]

It must be a bug with the 7905 I'll set it up later with my 7960 and see.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Kevin
> Sent: Friday, September 10, 2004 7:19 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Valet Park Application
> 
> Brian,
> 
> Thanks for your response and great efforts for the Valet Parking App.  I
> am using a Cisco 7905G phone and 1.0-RC2. I am using the following
> command:
> exten =>
> _376XX,1,ValetParkCall(${EXTEN:1}|mylot|60|63${EXTEN:1}|1|extensions)
> 
> Everything works fine unless the there is already a call parked. If I
> blind transfer, the call gets transferred and this is what I get:
> 
>-- Executing ValetParkCall("IAX2/192.168.2.75:4569/1",
> "7621|mylot|60|637621|1|extensions") in new stack
> Sep 10 20:13:54 WARNING[1117358896]: apps/app_valetparking.c:502
> valetpark_call: Call is already Valet Parked Here [7621]
>   == Spawn extension (internal, 37621, 1) exited non-zero on
> 'IAX2/192.168.2.75:4569/1'
> -- Hungup 'IAX2/192.168.2.75:4569/1'
>   == Spawn extension (internal, 8212038696030, 1) exited non-zero on
> 'SIP/7621-deea'
> localhost*CLI>
> 
> Thanks again for your assistance in advance.  You help is greatly
> appreciated.
> 
> Kevin
> 
> 
> 
> -Original Message-
> From: Brian West [mailto:[EMAIL PROTECTED]
> Sent: Friday, September 10, 2004 4:57 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Valet Park Application
> 
> No it doesn't/shouldn't..  If a call is already parked in that location
> you
> shouldn't be able to complete the transfer and you'll have to press
> resume
> and try again.
> 
> bkw
> 
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Kevin
> > Sent: Friday, September 10, 2004 3:47 PM
> > To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> > Subject: [Asterisk-Users] Valet Park Application
> >
> > I love the functionality of the Valet Park Application.  I have a
> > question regarding its operation.  The problem I am having when there
> is
> > a call already parked on specific park extension.  If a caller uses
> > 'blind' transfer on a Cisco Phone the caller gets disconnected.  Can
> any
> > offer any suggestions on how to prevent the transfer from taking
> place?
> >
> >
> >
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> 
> 
> 
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RE: [Asterisk-Users] Playback Fileformats

[Brian West]

At the cli do

show file formats

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Christian Victor
> Sent: Monday, September 13, 2004 9:42 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Playback Fileformats
> 
> Hi!
> 
> I wonder what sondfile formats Playback() could play. I know it plays
> GSM but to save the CPU time I will avoid converting GSM > alaw for my
> E1 and user alaw compressed wavs.
> 
> Unfortunately the wiki does not list the supported filetypes but I know
> that there are a few.
> 
> Thanks in davance
> Christian
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RE: [Asterisk-Users] Playback Fileformats

[Brian West]

Update your asterisk install them because you must have an old one

asterisk*CLI> show file formats
Format Name   Extensions
SLINR  slnsln|raw
ILBC   iLBC   ilbc
G726   g726-16g726-16
G726   g726-24g726-24
G726   g726-32g726-32
G726   g726-40g726-40
H263   h263   h263
ALAW   alaw   alaw|al
G729A  g729   g729
ULAW   pcmpcm|ulaw|ul|mu
GSMwav49  WAV|wav49
SLINR  wavwav
GSMgsmgsm
SLINR  mp3mp3

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Christian Victor
> Sent: Monday, September 13, 2004 10:05 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Playback Fileformats
> 
> Brian West schrieb:
> 
> > At the cli do
> >
> > show file formats
> 
> Sorry - that does not work. And SHOW AUDIO CODECS shows me the codecs
> but not the file formats (.wav etc) supported.
> 
> Christian
> 
> >>I wonder what sondfile formats Playback() could play. I know it plays
> >>GSM but to save the CPU time I will avoid converting GSM > alaw for my
> >>E1 and user alaw compressed wavs.
> >>
> >>Unfortunately the wiki does not list the supported filetypes but I know
> >>that there are a few.
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RE: [Asterisk-Users] Playback Fileformats

[Brian West]

Its correct.. they all result in SLINR output.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Eric Wieling
> Sent: Monday, September 13, 2004 10:51 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Playback Fileformats
> 
> On Mon, 2004-09-13 at 10:21, Brian West wrote:
> > Update your asterisk install them because you must have an old one
> >
> > asterisk*CLI> show file formats
> > Format Name   Extensions
> > SLINR  slnsln|raw
> > SLINR  wavwav
> > SLINR  mp3mp3
> 
> Is this correct or a bug in the formats list?
> 
> --
>   Eric Wieling * BTEL Consulting * 504-899-1387 x2111
> "In a related story, the IRS has recently ruled that the cost of Windows
> upgrades can NOT be deducted as a gambling loss."
> 
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[Asterisk-Users] Chanspy updated

[Brian West]

http://bugs.digium.com/bug_view_page.php?bug_id=0002379

Updated to try and fix the issue others were seeing.  Added a check in front
of ast_frfree(f);  Just to make sure we do the right thing(tm)

bkw 

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RE: [Asterisk-Users] Astricon

[Brian West]

I'm going to take my WRT54GS and set it up to see if that will work :)

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Rich Adamson
> Sent: Friday, September 17, 2004 6:04 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Astricon
> 
> > >>  It sure does.  On the astricon.net site you will find "High-speed
> > >>internet access provided by STSN" under hotel features.  I know from
> > >>experience in other Marriots that there is an adaptor in the room (an
> > >>xDSL I'm sure) with a 10/100 ethernet on it.  You register when you
> > >>connect for $9.99 per 24hr span to connect.  I am not sure about Wifi.
> > >>It would be bad if the hotel for Astricon didn't let us use our SIP
> phones!
> > >
> > >
> > > Assuming the equipment is the same stuff that's installed at many of
> > > the Marriott hotels, it talks a web browser to activate the service.
> > > The browser is auto-redirected to an internal hotel web page, and you
> > > authorize the access for the fee (for 24 hours as noted above).
> > >
> > > Be carefull with assumptions... some of these arrangements require the
> web
> > > browser access to open the channel again during that 24 hour period. I
> > > was at one last week, and used a hub in the room thinking I could use
> > > a snom 200 for making calls. Didn't work. PC worked fine has long as
> > > I started with the web browser, then x-lite would work; but the snom
> > > never did. (The snom has been used in lots of hotel rooms around the
> > > country, and I'm quit comfortable with its ability to handle nating,
> > > firewalls, etc.) I wouldn't make any assumptions relative to sip and
> > > iax2 though.
> > >
> > > Rich
> > >
> >
> > Rich,
> >
> > Experimenting at various Marriots in several states tells that STSN
> > binds that $9.99 charge your in-room box and MAC address. I register
> > with my laptop (RHEL3 WS & Mozilla) then do NAT firewalling with a
> > second NIC (or built in WiFi) to support other devices.  I would love to
> > try with one of the 802.11b WiFi phones for a nice cordless phone...
> 
> Cool. Only reason for mentioning the above on this list is that we can
> all guess at how many sip phones will be packed for the trip, laptops
> with *, etc, etc. So for those packing, understand the above
> requirements and plan accordingly. :)
> 
> 
> 
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RE: [Asterisk-Users] Asterisk stopped answering the calls

[Brian West]

Just an FYI

localmask is deprecated... 

As of latest CVS:

localnet=ip/mask

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Joseph
> Sent: Saturday, September 18, 2004 9:06 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Asterisk stopped answering the calls
> 
> Asterisk stopped answering the calls.
> I'm just experimenting with asterisk, upon setup there is a [demo]
> context.
> I have SPA-3000 with PSTN line:
> Dial plan 2: S0<:[EMAIL PROTECTED]>
> my sip.conf
> localnet = 10.0.0.101
> localmask = 255.255.255.0
> 
> [3000]
> type=friend
> host=dynamic
> username=3000
> secret=my_secret
> mailbox=3000
> context=from_pstn
> callerid="PSTN GW" <3000>
> deny=0.0.0.0
> permit=10.0.0.154  ;SPA-3000 IP address
> dtmfmode=rfc2833
> canreinvite=no
> cantransfer=yes
> 
> My extension.conf
> [globals]
> PSTN_GW=10.0.0.154:5062
> [from_pstn]
> exten => 1000,1,Goto(demo,s,1)
> [demo]
> exten => s,1,Answer
> exten => s,2,DigitTimeout,5
> exten => s,3,ResponseTimeout,10
> exten => s,4,BackGround(demo-congrats
> 
> When I type "show channels" I get "0 active channel(s)"
> 
> Why isn't asterisk answering the call?
> --
> #Joseph
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RE: [may-be-spam] Re: [Asterisk-Users] PBX CallTransfer

[Brian West]

Yes its possible from zap, sip and many other situations but not with the #
transfer feature without this patch.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
> Sent: Monday, September 20, 2004 9:05 AM
> To: [EMAIL PROTECTED]; [EMAIL PROTECTED]
> Subject: [may-be-spam] Re: [Asterisk-Users] PBX CallTransfer
> 
> > Yeah that works perfectly. Now trying to merge that with the
> > configurable transfer button patch :)
> 
> Should "Supervised call transfer" not already be possible without patching
> asterisk?
> 
> 
> >
> >
> > On Mon, 20 Sep 2004 09:41:15 -0300, Nicolás Gudiño <[EMAIL PROTECTED]>
> > wrote:
> >> Hello,
> >>
> >> On Mon, 20 Sep 2004 13:37:35 +0200 (CEST), [EMAIL PROTECTED]
> >> <[EMAIL PROTECTED]> wrote:
> >> > Hello List,
> >> > i have asked this a couple of times now on this list. But never got
> >> > an answer.Where can i find informations about a "Supervised call
> >> > transfer" ?
> >> >
> >>
> >> I have just found this on the bugtraker
> >>
> >> http://bugs.digium.com/bug_view_page.php?bug_id=0002460
> >>
> >> Best regards,
> >>
> >> --
> >> Nicolás Gudiño
> >> Buenos Aires - Argentina
> >>
> >>
> >> ___
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> >>http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >
> >
> >
> > --
> > Michael Bielicki
> > ___
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> 
> 
> 
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RE: [Asterisk-Users] Asterisk 1.0 released

[Brian West]

If you want the patches in CVS the best place is to put them on the bug
tracker at http://bugs.digium.com then everyone can benefit from it.

bkw


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Adam Goryachev
> Sent: Thursday, September 23, 2004 7:58 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Asterisk 1.0 released
> 
> On Fri, 2004-09-24 at 01:12, Kevin Walsh wrote:
> > I seem to be hoarding patches, and sending them out on request.  I
> should
> > set up a website to list and share them more easily.
> 
> I did, but nobody used it... http://www.websitemanagers.com.au/asterisk/
> 
> Anyone can register and upload *their* files, with whatever license they
> want. You can then add/upgrade/remove your files whenever you want.
> 
> Regards,
> Adam
> 
> 
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RE: [Asterisk-Users] Thank you Mr. Mark Spencer and Asterisk CommunityMembers

[Brian West]

Last but not least let's not forget to thank Anthony Minessale (anthm) on
IRC for all the work he has done on the project.  

Bridge config
Valetparking
res_perl
res_sqlite
ChanSpy
ControlPlayback

AND many many many more patches that have been commited.  I knew anthm and
worked with him on some of these projects before I started working for him
about two months ago.  I only wish he could have made it to Astricon this
year.

Thanks,
Brian

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Administrator
> Sent: Thursday, September 23, 2004 11:15 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Thank you Mr. Mark Spencer and Asterisk
> CommunityMembers
> 
> 
> Folks,
> 
> Today was great day, Asterisk 1.0.0 was released.
> I think today is a time to say Thank You Mark Spencer and thank you
> Asterisk community, so this project is  alive and project is booming and
> growing like crazy.
> It's amazing to have Asterisk with all features and components, it's just
> sometimes unbelivable.
> 
> Let's say Thank You to  Mark, please sign this message with your name and
> comments and we will redirect message to MARK. He deserves this.
> He needs to know that we appreciate all his affords and hard work for
> Asterisk. Please put your name and comments, give Asterisk project and
> MArk your FeedBack,  you MUST ! ;-)
> 
> There is no company in the world who is
> able to provide cvs changes at the rate that Mark does when there is
> a problem.  Huge kudos to him and everyone else.
> 
> And of course thanks have to go out to the person who decided to make
> Asterisk Open Source.  Without this decision I don't think Asterisk
> would be the software it is today!
> 
> Also our greetings to JerJEr, (even he thinks Radius is evil ;-),BKW
> (with all sip/nat dramas  ;-), Twisted, Lenny, Olle, Steve Sokol, Voip-
> info.org, Wsuff, etc.
> 
> Matt (www.sineapps.com/news.php) aka ZX81 - Asterisk Unofficial Daily
> News and Alexander aka Stealth_MAn (http://asterisk.xvoip.com)
> 
> Once again, please say and post your comments for Mark and our Asterisk
> Community. Congratulations to all of us !
> Also we have Astericon, thanks to Steve and Olle who organized it.
> 
> 
> Cheers!
> Matt (www.sineapps.com/news.php) aka ZX81 - Asterisk Unofficial Daily
> News and
> Alexander(http://asterisk.xvoip.com) aka Stealth_Man
> 
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RE: [Asterisk-Users] Thank you Mr. Mark Spencer and AsteriskCommunity Members

[Brian West]

THIS GUY!!! Yes without him and Mark Asterisk wouldn't be what it is today.

GREAT JOB GUYS... we can only get better from here... Now who wants DS3 and
Hardware DSP cards to scale asterisk to the DS3 level? :)

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Benjamin on Asterisk Mailing Lists
> Sent: Friday, September 24, 2004 6:00 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Thank you Mr. Mark Spencer and
> AsteriskCommunity Members
> 
> On Fri, 24 Sep 2004 00:14:42 -0400, Administrator <[EMAIL PROTECTED]> wrote:
> > I think today is a time to say Thank You Mark Spencer and thank you
> > Asterisk community, so this project is  alive and project is booming and
> > growing like crazy.
> 
> And let's not forget to give credit to Jim Dixon, who created Zaptel
> and gave it away as open source (both hardware designs and drivers).
> 
> rgds
> benjk
> 
> --
> Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
> Tokyo, Japan.
> 
> NB: Spam filters in place. Messages unrelated to the * mailing lists
> may get trashed.
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RE: [Asterisk-Users] Zaptel 1.0.0. will not compile

[Brian West]

I'm the same way... Asterisk, zaptel and libpri are all done from src and
not portage.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Karl Dyson
> Sent: Sunday, October 10, 2004 3:54 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Zaptel 1.0.0. will not compile
> 
> Personally I get zaptel, zapata etc from cvs rather than portage.
> 
> Check your /usr/src/linux symlink points to the correct place... I got
> all sorts of grief with gentoo when I forgot to put it back after some
> "playing".
> 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of
> > Remco Barende
> > Sent: 10 October 2004 21:48
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] Zaptel 1.0.0. will not compile
> >
> > OK, I got a little further. I don't know why but after
> > re-emerging zapata finally zaptel will build. It is not
> > working however and I still get too many erros during the
> > build. It's now complaining about:
> >
> >CC [M]  /var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/zaptel.o
> > /var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/zaptel.c:163:
> > warning:
> > `fcstab' defined but not used
> >CC [M]  /var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/tor2.o
> >CC [M]  /var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/torisa.o
> > /var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/torisa.c:1139:
> >  warning:
> > `set_tor_base' defined but not used
> >
> >
> > *** Warning: "zt_register"
> > [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztdynamic.ko]
> >  has no CRC!
> > *** Warning: "zt_transmit"
> > [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztdynamic.ko]
> >  has no CRC!
> > *** Warning: "zt_receive"
> > [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztdynamic.ko]
> >  has no CRC!
> > *** Warning: "zt_ec_chunk"
> > [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztdynamic.ko]
> >  has no CRC!
> > *** Warning: "zt_set_dynamic_ioctl"
> > [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztdynamic.ko]
> >  has no CRC!
> > *** Warning: "zt_unregister"
> > [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztdynamic.ko]
> >  has no CRC!
> > *** Warning: "zt_alarm_notify"
> > [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztdynamic.ko]
> >  has no CRC!
> > *** Warning: "zt_rbsbits"
> > [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztdynamic.ko]
> >  has no CRC!
> > *** Warning: "zt_transmit"
> > [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztdummy.ko]
> > has no CRC!
> > *** Warning: "zt_receive"
> > [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztdummy.ko]
> > has no CRC!
> > *** Warning: "zt_unregister"
> > [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztdummy.ko]
> > has no CRC!
> > *** Warning: "zt_register"
> > [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztdummy.ko]
> > has no CRC!
> > *** Warning: "zt_dynamic_unregister"
> > [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztd-eth.ko]
> > has no CRC!
> > and many many more
> >
> >
> >
> > On Sun, 10 Oct 2004, Remco Barende wrote:
> >
> > > Hi list
> > >
> > > I am trying to install asterisk on a gentoo box running
> > kernel version
> > > Linux version 2.6.8-gentoo-r7 (gcc version 3.3.4 20040623 (Gentoo
> > > Linux 3.3.4-r1, ssp-3.3.2-2, pie-8.7.6)) #1 Thu Oct 7 20:24:31 CEST
> > > 2004
> > >
> > > I tried to install from the provided ebuild for Asterisk
> > 1.0.0 but the
> > > compile of the zaptel module fails miserably.
> > >
> > > Compilation seems to start at first but then generates
> > pages and pages
> > > of errors like :
> > >
> > > include/linux/types.h:18: error: syntax error before
> > "__kernel_dev_t"
> > > include/linux/types.h:18: warning: type defaults to `int' in
> > > declaration of `__kernel_dev_t'
> > > include/linux/types.h:18: warning: data definition has no type or
> > > storage class
> > > include/linux/types.h:21: error: syntax error before "dev_t"
> > >
> > > include/linux/types.h:152: warning: type defaults to `int' in
> > > declaration of `f_tinode'
> > > include/linux/types.h:152: warning: data definition has no type or
> > > storage class
> > > include/linux/types.h:155: error: syntax error before '}' token In
> > > file included from
> > > /var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/zaptel.c:42:
> > > include/linux/kernel.h:15:27: asm/byteorder.h: No such file or
> > > directory
> > > include/linux/kernel.h:16:21: asm/bug.h: No such file or
> > directory In
> > > file included from
> > > /var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/zaptel.c:42:
> > > include/linux/kernel.h:81: error: syntax error before "size_t"
> > > include/linux/kernel.h:82: warning: function declaration isn't a
> > > prototype
> > > include/linux/kernel.h:82: warning: conflicting types for built-in
> > > function `snprintf'
> > > include/linux/kernel.h:83: error: syntax error before "size_t"
> > > include/linux/kerne

RE: [Asterisk-Users] newbie question - app_realtime.so failed

[Brian West]

Because realtime isn't in 1.0 or 1.0.1 its ONLY in cvs-head.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of mihai iancu
> Sent: Sunday, October 10, 2004 9:05 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] newbie question - app_realtime.so failed
> 
> Hello,
> 
> Here are my info: asterisk version 1.0 with Redhat 8.0 kernel 2.4.18
> Everything was running nice and clean with an old version from Aug
> 2004.
> 
> Cleaned all source code and binaries - download and install version 1.0
> and this is what I get:
> 
> 
> Oct 10 22:44:36 WARNING[8192]:
> /usr/lib/asterisk/modules/app_realtime.so: undefined symbol:
> ast_load_realtime
> Oct 10 22:44:36 WARNING[8192]: Loading module app_realtime.so failed!
> 
> Any ideas?
> 
> Thank you.
> 
> 
> 
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RE: [Asterisk-Users] Re: SIP peers in MySQL Database

[Brian West]

> Somebody seems start a mysql drivers for realtime external configuration
> instead of ODBC.

You can speak to MySQL with ODBC.

bkw

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RE: [Asterisk-Users] Re: SIP peers in MySQL Database

[Brian West]

The Makefile isn't gonna help with cvs-head since the code was ripped out.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of harry gaillac
> Sent: Monday, October 11, 2004 7:43 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Re: SIP peers in MySQL Database
> 
> Here is the Makefile from asterisk-1.0.0
> 
>  --- Tomica Crnek <[EMAIL PROTECTED]> a écrit :
> >
> > >From few days ago there is no USE_MYSQL_FRIENDS in
> > channels/Makefile. That is why I am asking this.
> >
> > > -Original Message-
> > > From: [EMAIL PROTECTED]
> > > [mailto:[EMAIL PROTECTED]
> > On Behalf Of
> > > harry gaillac
> > > Sent: Monday, October 11, 2004 12:19 PM
> > > To: Asterisk Users Mailing List - Non-Commercial
> > Discussion
> > > Subject: RE: [Asterisk-Users] Re: SIP peers in
> > MySQL Database
> > >
> > > look at ../channels/Makefile
> > >
> > > try USE_MYSQL_FRIENDS=1
> > >
> > > Harry
> > >
> > 
> > > #
> > > # Asterisk -- A telephony toolkit for Linux.
> > > #
> > > # Makefile for Channel backends (dynamically
> > loaded) # #
> > > Copyright (C) 1999, Mark Spencer # # Mark Spencer
> > > <[EMAIL PROTECTED]> # # Edited By
> > Belgarath <> Aug
> > > 28 2004 # Added bare bones ultrasparc-linux
> > support.
> > > #
> > > # This program is free software, distributed under
> > the terms
> > > of # the GNU General Public License #
> > >
> > > OSARCH=$(shell uname -s)
> > > PROC=$(shell uname -m)
> > >
> > > USE_MYSQL_FRIENDS=0
> > > USE_SIP_MYSQL_FRIENDS=0
> > >
> > 
> > >
> > >
> > >  --- Tomica Crnek <[EMAIL PROTECTED]> a écrit :
> >
> > > >
> > > > It says "To enable this, you need to edit the
> > Makefile in
> > > the channels
> > > > directory of your source tree and enable
> > MYSQL_FRIENDS.",
> > > but there is
> > > > no MYSQL_FRIENDS in channels/Makefile any more.
> > > >
> > > > > -Original Message-
> > > > > From: [EMAIL PROTECTED]
> > > > >
> > [mailto:[EMAIL PROTECTED]
> > > > On Behalf Of
> > > > > harry gaillac
> > > > > Sent: Monday, October 11, 2004 11:45 AM
> > > > > To: Glynn Condez
> > > > > Cc: [EMAIL PROTECTED]
> > > > > Subject: [Asterisk-Users] Re: SIP peers in
> > MySQL
> > > > Database
> > > > >
> > > > > Hi,
> > > > >
> > > > > Look at:
> > > > >
> > > >
> > >
> >
> http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers
> > > > >
> > > >
> > >
> >
> http://www.voip-info.org/wiki-Asterisk+configuration+from+database
> > > > >
> > > > > Is it working well? I don't know because of
> > i'm
> > > > waiting  a
> > > > > reply in order to use sql database for all sip
> > > > clients from
> > > > > small offices asterisk box with nat context.
> > > > >
> > > > >
> > > > > May I use autocreatepeer in all asterisk
> > sip.conf
> > > > file with
> > > > > nat=yes in general option ???
> > > > >
> > > > > [general]
> > > > > dbname= Name of database in your Mysql server
> > > > dbhost=
> > > > > Hostname of server dbuser= Username in MySQL
> > > > dbpass= Password
> > > > > for user in MySQL autocreatepeer=yes nat=yes
> > 
> > > > > ---   --
> > > > > |Asterisk |-- |nat/firewall box |
> > > > > ---   --
> > > > > |
> > > > > |
> > > > >   --
> > > > >| Internet |-- |nat/firewall
> > > > box|--Asterisk
> > > > >
> > > > >   --
> >
> > > > +
> > > > > |  SIP
> > > > peers
> > > > > in
> > > > > |mysql
> > > > > database
> > > > > ---   --
> > > > > |Asterisk |-- |nat/firewall box |
> > > > > ---   --
> > > > >
> > > > > Harry
> > > > >
> > > > >  --- Glynn Condez <[EMAIL PROTECTED]> a écrit
> > :
> > > > > > Hi Harry,
> > > > > >
> > > > > > how did you make sip peers on mysql
> > database? is
> > > > it working well?
> > > > > > where can I find a documentation so I could
> > > > migrate my Asterisk sip
> > > > > > config to use Mysql also.
> > > > > >
> > > > > > Regards
> > > > > >
> > > > > >
> > > > >
> > > > >
> > > > >
> > > > >
> > > > >
> > > > >
> > > > > Vous manquez d'espace pour stocker vos mails ?
> >
> > > > > Yahoo! Mail vous offre GRATUITEMENT 100 Mo !
> > > > > Créez votre Yahoo! Mail sur
> > > > http://fr.benefits.yahoo.com/
> > > > >
> > > > > Le nouveau Yahoo! Messenger est arrivé !
> > Découvrez
> > > > toutes les
> > > > > nouveautés pour dialoguer instantanément avec
> > vos
> > > > amis. A
> > > > > télécharger gratuitement sur
> > > > http://fr.messenger.yahoo.com
> > > > >
> > ___
> > > > > Asterisk-Users mailing list
> > > > > [EMAIL PROTECTED]
> > > > >
> > > >
> > >
> >
> http://lists.digium.com/mailman/lis

RE: [Asterisk-Users] Can't compile chan_h323 in latest CVS...

[Brian West]

Use Asterisk v1-0 and please you're using chan_oh323 NOT chan_h323 they are
two totally different channel drivers.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Pablo Endres
> Sent: Monday, October 11, 2004 7:49 AM
> To: deimios; Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Can't compile chan_h323 in latest CVS...
> 
> On Sat, 2004-10-09 at 02:16, deimios wrote:
> > On Sat, 9 Oct 2004 12:56:44 +0800, Walter Klomp <[EMAIL PROTECTED]>
> wrote:
> > > Hi,
> > >
> > > In the latest CVS I am trying to compile chan_h323, but it doesn't
> want to.
> > >
> > > chan_h323.c: In function `oh323_call':
> > > chan_h323.c:453: error: structure has no member named `callerid'
> > > chan_h323.c:455: error: structure has no member named `callerid'
> > > chan_h323.c:455: error: structure has no member named `callerid'
> > > chan_h323.c: In function `oh323_new':
> > > chan_h323.c:756: error: structure has no member named `callerid'
> > > make[1]: *** [chan_h323.o] Error 1
> > >
> > > ...if I unremark the line
> > > #CHANNEL_LIBS+=$(shell [ -f h323/libchanh323.a ] && echo chan_h323.so)
> > > In channels/Makefile...
> > >
> > > I have successfully made the channels/h323 with the openh323 and the
> > > pwlib...
> > >
> > > But for some reason asterisk is not making the chan_h323.
> > >
> > > Am I missing something? (I have asked this question a few days ago and
> > > nobody responded, am I alone in this?)
> > >
> > > ~help~please~
> > > Walter
> > >
> >
> > I am having similiar problems, I have searched for an answer but
> > nothing has sprung forward. Cruising the code tonight looking for a
> > fix... If someone has a fix already or information on how to fix it
> > please please let us know.
> >
> > -Regards
> > Deimios
> 
> Hi I walked that path last week.  What I got from the IRC channel (don't
> exactly remember who) is that the is mayor redo on the
> caller ID structure, you have to wait until the chan_oh323 guys catch
> up.
> 
> Use the cvs from 2004-08-30 that compiles OK.
> 
> 
> > ___
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> --
> Pablo Endres <[EMAIL PROTECTED]>
> ComVoz Communications
> 
> USA:   +1 954 343 2085 Ext 199
> Venezuela: +58 212 771 3100 Ext 199
> Colombia:  +57 1 325 6900 Ext 199
> 
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RE: [Asterisk-Users] Re: SIP peers in MySQL Database

[Brian West]

You must be one of those people that doesn't know much about ODBC and is
under the impression it's SLOW!

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Arnaud Pignard
> Sent: Monday, October 11, 2004 8:23 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Re: SIP peers in MySQL Database
> 
> Yes but it's will be better to have mysql driver
> 
> At 14:20 11/10/2004, you wrote:
> > > Somebody seems start a mysql drivers for realtime external
> configuration
> > > instead of ODBC.
> >
> >You can speak to MySQL with ODBC.
> >
> >bkw
> >
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RE: [Asterisk-Users] Re: SIP peers in MySQL Database

[Brian West]

> Why asterisk use ODBC(Microsoft?) to connect to SQL
> database?

1. It's not Microsoft at all.
2. It's unixODBC (I don't see Microsoft here at all)
3. Wider database support without having to know each database type.
4. It's not much slower than native DB drivers. (15-33% slower)

But In my tests you would never see this unless you're doing 10k
selects and 5k inserts and that's on a 1ghz box.

> Anybody could answer to my first question ?

Not really sure what your first question is and I'm not gonna dig for it.

bkw

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RE: [Asterisk-Users] Re: SIP peers in MySQL Database

[Brian West]

> look at unixODBC or iodbc for more information
> 
> Also the reason (i guess) why they move to ODBC is that's ODBC have many
> connector to most SQL database.

Bingo

bkw

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RE: [Asterisk-Users] Re: SIP peers in MySQL Database

[Brian West]

Per second.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Andreas Sikkema
> Sent: Monday, October 11, 2004 8:59 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Re: SIP peers in MySQL Database
> 
> [EMAIL PROTECTED] wrote:
> 
> > But In my tests you would never see this unless you're doing 10k
> > selects and 5k inserts and that's on a 1ghz box.
> 
> Per seconds? Per day?
> 
> --
> Andreas SikkemaRits tele.com
> Scheepmakersstraat 11  3011 VH Rotterdam
> t: +31 (0)10 2245544f: +31 (0)10 2245540
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RE: [Asterisk-Users] voicemail attachment volume

[Brian West]

> Is there a fix/patch that can be applied to allow the voicemails to be
> recorded LOUDER?  I would like to go live with my Asterisk system, but
> this is a major problem.

Its not asterisk that's the problem I suspect.  If you get low recordings
you need to look into using app_test to help find them.

bkw

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RE: [Asterisk-Users] reading global vars from AGI

[Brian West]

Those are not global vars.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of shabanip
> Sent: Monday, October 11, 2004 10:17 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] reading global vars from AGI
> 
> is there any way to read global vars like ${EXTEN},  ${GROUPCOUNT} from an
> AGI?
> 
> 
> 
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RE: [Asterisk-Users] G726 Codec Question

[Brian West]

Also at the time Mark wrote it.  G726-32 was VERY in the clear on patents
because G726-32 was really the old G721 standard.  The old G723 and Old G721
equals todays G726.  (Also G723 and G723.1 are two different things)

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Steve Underwood
> Sent: Monday, October 11, 2004 7:34 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] G726 Codec Question
> 
> Darren Sessions wrote:
> 
> > What is the rational for only supporting 32kbps G726 and not 16kbps?
> >
> > Thanks,
> 
> G.726 32K is widely used. The other bit rates are not. That seems one
> sensible rationale for only supporting 32K.
> 
> Regards,
> Steve
> 
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RE: [Asterisk-Users] cvsup options file for v1-0

[Brian West]

Tag=v1-0 but I don't think we have it setup to pull X or Y tag.  Cvsup was
setup so you could create your own local mirror with ease then you can
checkout from that all you want with any tag or branch.


*default host=cvs.digium.com
*default base=/usr/cvsroot
*default prefix=/usr/cvsroot
*default release=cvs
*default delete use-rel-suffix
asterisk
zaptel
libpri


bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Glenn Dalgliesh
> Sent: Tuesday, October 12, 2004 9:52 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] cvsup options file for v1-0
> 
> I want to dowload cvs of v1-0 with cvsup and was wondering what the
> options
> file will look like to make this happen.
> 
> I am assuming the some thing on the line *default release=cvs tag=.
> 
> - options file for cvsup to download cvs head 
> *default host=cvs.digium.com
> *default base=/usr/src
> *default release=cvs tag=.
> *default delete use-rel-suffix
> asterisk
> libpri
> zaptel
> 
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RE: [Asterisk-Users] How big .CONF files can be?

[Brian West]

Not really it just takes about 2 min for it to settle down and actually
start working :P (and don't reload)

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Michael Bielicki
> Sent: Tuesday, October 12, 2004 1:33 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] How big .CONF files can be?
> 
> 30.000 entries in iax.conf will kill * on startup
> 
> 
> On Tue, 12 Oct 2004 14:39:23 +0200, Jens Kübler <[EMAIL PROTECTED]>
> wrote:
> > Am Dienstag, 12. Oktober 2004 14:04 schrieb Goran Dj:
> > > I'm new to Asterisk.
> > > How big can be sip.conf (and other: iax.conf, extensions.conf...)
> > > Is there point when I must use DB (MySQL...) instead of pure .conf?
> > >
> > There is no real answer to this question.
> > Databases are always a good choice as the search strategies are
> substantially
> > faster and the administration is much easier.
> > If you only got few extensions don't use a database as it's not worth
> the work
> > for the interfaces to setup.
> >
> > For example if a peer is searched in the conf file the average time will
> be
> > N/2 to find the peer where N is the number of peers listed.
> >
> > Most databases should be able to search for this entry in
> > log (N) time which for big N is far better than the above.
> >
> > Generally speaking a sip.conf can be as large as your ram but you
> certainly
> > don't want to check that out.
> >
> > Jens
> >
> >
> >
> > ___
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
> 
> --
> Michael Bielicki
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RE: [Asterisk-Users] Cisco 7960G "disk full error"

[Brian West]

> I upgraded 6 7960G's the other week, all at the same time, all starting
> with
> year 2000 SCCP firmware on them.
> 
> You cannot upgrade these phones directly to SIP7.2.

Actually you can.  You put in your OS79XX.TXT P003-07-2-00 which is the UAL.
The UAL is around 125k, then in your .cnf files ie SIPDefault.cnf you put
P0S3-07-2-00.  The reason you can't upgrade some older SCCP phones is the
buffer in the phone isn't large enough to take a load that's larger than
about 400K.  You should be able to load the UAL without a problem on most
phones.  I have done it more than a dozen times without a problem.

bkw

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RE: [Asterisk-Users] Cisco 7960G "disk full error"

[Brian West]

As of 7.x the UAL is what the phone loads..  Nothing to be CAREFUL of.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Shilliday, Jim
> Sent: Tuesday, October 12, 2004 5:32 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Cisco 7960G "disk full error"
> 
> Careful - in the later versions, P00* no longer specifies a protocol, it's
> actually the UAL.
> 
> 
> 
> Jim Shilliday
> 
> 
> 
> -Original Message-
> From: Andrew Edmond [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, October 12, 2004 5:59 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Cisco 7960G "disk full error"
> 
> 
> 
> Henry,
> 
> 
> 
> Thanks!  This is a step in the right direction...
> 
> 
> 
> However, since it has P00* firmware on it, it never downloads
> SIPDefault.cnf.  I guess I'll have to make SEPDefault.cnf as small as
> possible then...?
> 
> 
> 
> Andrew
> 
> 
> 
>   -Original Message-
>   From: [EMAIL PROTECTED] [mailto:asterisk-
> [EMAIL PROTECTED] On Behalf Of Henry Devito
>   Sent: Tuesday, October 12, 2004 2:43 PM
>   To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
>   Subject: RE: [Asterisk-Users] Cisco 7960G "disk full error"
> 
>   The buffer is full on the 7960 because it keeps the old software
> along with downloading the new software just incase the new software
> fails.  What I have been doing is using a very generic SIPDefault.cnf and
> that allows enough space to download the upgrades.
> 
> 
> 
>   This is the SIPDefault.cnf I use for upgrades.  Once the upgrade is
> done use your original SIPDefault.
> 
> 
> 
>   image_version: "P0S3-04-4-00"
> 
>   # Proxy Server
> 
>   proxy1_address: "192.168.254.4"
> 
>   proxy6_port:"5060"
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
>   From: [EMAIL PROTECTED] [mailto:asterisk-
> [EMAIL PROTECTED] On Behalf Of Andrew Edmond
>   Sent: Tuesday, October 12, 2004 4:26 PM
>   To: [EMAIL PROTECTED]
>   Subject: [Asterisk-Users] Cisco 7960G "disk full error"
> 
> 
> 
>   Hello,
> 
> 
> 
>   I'm trying to upgrade my 7960G that I bought off ebay to SIP 7.2
> firmware.  Apparently this phone had pretty old firmware on it... it's
> using Application Load ID P003H310 (version 3.0.12 Call Manager).
> 
> 
> 
>   I've setup both a UNIX and Windows (SolarWinds) tftp server, and
> acquired firmware images for SIP 3.2, 4.4, 5.3, 6.3, and 7.2.
> 
> 
> 
>   I set my OS79XX.txt contents to 8.3 filenames to download P0S3-03-2-
> 00.bin (renamed to 8.3 filename P0S30320.bin).
> 
> 
> 
>   The tftp server successfully logs the connection from the phone,
> succesfully downloads OS79XX.txt, and then starts to download
> P0S30320.bin.  After 391168 bytes of download (the file is 409k), the
> phone dumps the transfer.
> 
> 
> 
>   So I installed ethereal on my windows server.  It shows that at
> packet 769 the cisco phone reports "Disk Full".
> 
> 
> 
>   I looked into this and came up with these previous posts:
> 
> 
> 
>   http://www.loligo.com/asterisk/Cisco/79xx/upgrading.79xx.phones
> 
>   http://lists.digium.com/pipermail/asterisk-users/2003-
> October/023021.html
> 
> 
> 
>   However, since my phone firmware is so old I assume that I might
> need SIP 2.3 firmware to get this going.
> 
> 
> 
>   In actuality, this is taking WAY more time than I had anticipated,
> so finally I'm reaching out to anyone who has had old firmware like mine
> (P003H310) and successfully upgraded it.
> 
> 
> 
>   Thank you!
> 
> 
> 
>   Andrew Edmond


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RE: [Asterisk-Users] Cisco 7960G "disk full error"

[Brian West]

Also if you do a config reset.. the UAL will load what ever firmware it can
find.  SIP, SCCP or MGCP watch your tftp logs.. it goes with what it can
find.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Andrew Edmond
> Sent: Tuesday, October 12, 2004 5:42 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Cisco 7960G "disk full error"
> 
> Right you are!
> 
> >From my dated 2000 firmware, using the UAL (at 125k-ish), the phone
> updated
> GREAT to 7.1 (and about to go to 7.2).
> 
> Thanks bkw!
> 
> Andrew
> 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of
> > Brian West
> > Sent: Tuesday, October 12, 2004 3:21 PM
> > To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> > Subject: RE: [Asterisk-Users] Cisco 7960G "disk full error"
> >
> >
> > > I upgraded 6 7960G's the other week, all at the same time, all
> > > starting with year 2000 SCCP firmware on them.
> > >
> > > You cannot upgrade these phones directly to SIP7.2.
> >
> > Actually you can.  You put in your OS79XX.TXT P003-07-2-00
> > which is the UAL. The UAL is around 125k, then in your .cnf
> > files ie SIPDefault.cnf you put P0S3-07-2-00.  The reason you
> > can't upgrade some older SCCP phones is the buffer in the
> > phone isn't large enough to take a load that's larger than
> > about 400K.  You should be able to load the UAL without a
> > problem on most phones.  I have done it more than a dozen
> > times without a problem.
> >
> > bkw
> >
> > ___
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> > UNSUBSCRIBE or update options visit:
> >
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RE: [Asterisk-Users] How many running instances (jobs) of asterisk

[Brian West]

It depends on how its configured and how many threads you need for the
current load.  If your box is loaded a lot that number will go up.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Roger Schreiter
> Sent: Tuesday, October 12, 2004 6:16 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] How many running instances (jobs) of asterisk
> 
> Hi,
> 
> when I do "ps aux | grep asterisk" I can find very different
> results on my various machines I'm running asterisk on.
> 
> One of my machines just shows one asterisk job running,
> others are shown approx 12.
> 
> Is it defined by a parameter?
> Does it affect quality or reliablity?
> 
> 
> Roger.
> 
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RE: [Asterisk-Users] G729 to G711 bridge

[Brian West]

> Called 2001
> -- SIP/2001-0a50 is ringing
> -- SIP/2001-0a50 answered SIP/2008-24fc
> -- Attempting native bridge of SIP/2008-24fc and SIP/2001-0a50
> Oct 12 17:15:59 WARNING[76823]: rtp.c:1392 ast_rtp_bridge: codec0 = 277 is
> not codec1 = 8, cannot native bridge.
>   == Spawn extension (from-sip, 2001, 1) exited non-zero on 'SIP/2008-
> 24fc'
> 
> 
> the g729 codecs is  showed as number 256 and not 277 in show codecs, what
> can be wrong?

That's because the codec is a bitmask.

bkw

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RE: [Asterisk-Users] G729 to G711 bridge

[Brian West]

I already sent you an email to your account off list that explained it.

One side is speaking g.711ulaw, g.726, g.729 and g.723.1 while the other leg
can only speak g.711alaw it seems (codec 8 is g.711alaw). 

G723.1 = 1
G711 = 4
G726 = 16
G729 = 256

Add it up and you have 277.

Thanks,
bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Miranda Gomez Miguel Angel
> Sent: Tuesday, October 12, 2004 6:55 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] G729 to G711 bridge
> 
> Excuse me, but can you explain a little more, what is the problem?  so i
> can
> search for the solutions,
> regards
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Brian West
> Sent: Tuesday, October 12, 2004 5:43 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] G729 to G711 bridge
> 
> 
> > Called 2001
> > -- SIP/2001-0a50 is ringing
> > -- SIP/2001-0a50 answered SIP/2008-24fc
> > -- Attempting native bridge of SIP/2008-24fc and SIP/2001-0a50
> > Oct 12 17:15:59 WARNING[76823]: rtp.c:1392 ast_rtp_bridge: codec0 = 277
> is
> > not codec1 = 8, cannot native bridge.
> >   == Spawn extension (from-sip, 2001, 1) exited non-zero on 'SIP/2008-
> > 24fc'
> >
> >
> > the g729 codecs is  showed as number 256 and not 277 in show codecs,
> what
> > can be wrong?
> 
> That's because the codec is a bitmask.
> 
> bkw
> 
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RE: [Asterisk-Users] mwi over serial port

[Brian West]

Can someone please get me the SMDI specs?

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Jim Van Meggelen
> Sent: Tuesday, October 12, 2004 7:51 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] mwi over serial port
> 
> The bounty seems to be at $3000 so far:
> 
> http://www.voip-info.org/tiki-index.php?page=Asterisk%20bounty%20SMDI
> 
> 
> 
> 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of
> > Mike Cathey
> > Sent: October 12, 2004 8:46 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] mwi over serial port
> >
> >
> > On Tue, 2004-10-12 at 20:25, Clay Zevely wrote:
> > > I am trying to interface to a nortel dms100
> >
> > It's called SMDI.
> >
> > Cheers,
> >
> > Mike
> >
> > ___
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> > 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).
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RE: [Asterisk-Users] mwi over serial port

[Brian West]

Can you send them?

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Clay Zevely
> Sent: Tuesday, October 12, 2004 8:42 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] mwi over serial port
> 
> I have the belcore standard
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Jim Van
> Meggelen
> Sent: Tuesday, October 12, 2004 8:22 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] mwi over serial port
> 
> 
> The trick is finding a published source for the SMDI protocol.
> 
> 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of
> > [EMAIL PROTECTED]
> > Sent: October 12, 2004 7:51 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: RE: [Asterisk-Users] mwi over serial port
> >
> >
> > On Tue, 12 Oct 2004, Jim Van Meggelen wrote:
> >
> > > The bounty seems to be at $3000 so far:
> > >
> > >
> > http://www.voip-info.org/tiki-index.php?>
> page=Asterisk%20bounty%20SMDI
> > >
> > >
> > I do not know for sure, but it seems easier enough to write this as
> > another application that just scans the /var/spool/asterisk/voicemail
> >
> > Michael
> >
> > ___
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> > [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: 6.0.775 / Virus Database: 522 - Release Date: 08/10/2004
> 
> 
> ---
> Outgoing mail is certified Virus Free.
> Checked by AVG anti-virus system (http://www.grisoft.com).
> Version: 6.0.775 / Virus Database: 522 - Release Date: 08/10/2004
> 
> 
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RE: [Asterisk-Users] mwi over serial port

[Brian West]

Also check out www.bkw.org/cdr_serial.c

It might not be totally right.. but hey I did it in like 20 min.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Clay Zevely
> Sent: Tuesday, October 12, 2004 8:42 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] mwi over serial port
> 
> I have the belcore standard
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Jim Van
> Meggelen
> Sent: Tuesday, October 12, 2004 8:22 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] mwi over serial port
> 
> 
> The trick is finding a published source for the SMDI protocol.
> 
> 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of
> > [EMAIL PROTECTED]
> > Sent: October 12, 2004 7:51 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: RE: [Asterisk-Users] mwi over serial port
> >
> >
> > On Tue, 12 Oct 2004, Jim Van Meggelen wrote:
> >
> > > The bounty seems to be at $3000 so far:
> > >
> > >
> > http://www.voip-info.org/tiki-index.php?>
> page=Asterisk%20bounty%20SMDI
> > >
> > >
> > I do not know for sure, but it seems easier enough to write this as
> > another application that just scans the /var/spool/asterisk/voicemail
> >
> > 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: 6.0.775 / Virus Database: 522 - Release Date: 08/10/2004
> 
> 
> ---
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> 
> 
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RE: [Asterisk-Users] RxFax multiple pages

[Brian West]

> You should either not convert (IMHO, this
> is not the best solution, as it is difficult to get a decent TIFF
> viewer) or convert to another format which does support multiple pages
> in a single file (think pdf).

http://www.hylafax.org/links.html#viewers

A tiff viewer is like standard on any SANE operating system.  XP and OS X
have no issues with viewing tiff files.

bkw

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RE: [Asterisk-Users] SIP peers in MySQL Database

[Brian West]

> > include => sip_additional.conf in [general]

Ok Just for the sake of some poor soul 4 months from now that keeps trying
to do an include and reads this info thinking its correct.

The proper way is:

#include somefile.conf

Or

#include /full/path/to/a/file.conf

No => and you MUST have a # in front.

bkw

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RE: [Asterisk-Users] ValetParking

[Brian West]

NO it won't go in CVS.  We have a few options ... 1. Try to work in most (if
not all) the features into the internal parking.  2. Keep it up to date with
latest cvs which is what we do now.  www.asterlink.com/svp

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Glenn Dalgliesh
> Sent: Wednesday, October 13, 2004 8:47 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] ValetParking
> 
> First Thanks to brian for work on valetpark it seems to work really well
> 
> I was working on some apps using ValetParking and having good success but
> was wondering when you think valetparking will make it into the
> CVS/releases? So, I can build around it with a little more confidence.
> Thanks
> 
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RE: [Asterisk-Users] Prerelease of DIAX 0.9.9a

[Brian West]

Anyway we could talk you into releasing the source?  I would love to see
wider codec support. And the ability to launch the URL sent with the IAX
call.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Dan
> Sent: Wednesday, October 13, 2004 10:02 AM
> To: Jon Bebeau; Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Prerelease of DIAX 0.9.9a
> 
> Hi Jon,
> 
> >- Original Message -
> >From: "Jon Bebeau" <[EMAIL PROTECTED]>
> >Sent: Wednesday, October 13, 2004 5:52 PM
> 
> 
> > Hi Dan,
> >
> > Did you release the source for DIAX?  I'm trying to build a drop-on
> > component for MS .NET (2005) and I've been looking for a good starting
> > place.  I spent some time with IAXClient and a few other from wiki, but
> > most
> > are Linux specific..then there's X10, but it's commercial.
> >
> 
> The application is distributed as a freeware, source code not included.
> 
> Sorry for the inconvenience.
> If you need some specific help, please send me a mail directly.
> 
> Best  regards,
> Dan
> 
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RE: [Asterisk-Users] SIP peers in MySQL Database

[Brian West]

> The perl script will overwrite the existing conf file. I've had bad
> experiences with constant reloading. Maybe you want to schedule your
> updates
> through a crontab.

Problem?  We do HUNDREDS of reloads a day without problem.  Just an FYI

bkw

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RE: [Asterisk-Users] SIP peers in MySQL Database

[Brian West]

:) res_perl/res_config .. Use the perl config handler in there and
you're reloads will not be painful at all.  We build an almost 20,000 line
extensions.conf with 800+ contexts that gets reloaded MANY MANY times a day
without a problem.   Did you happen to seem me talk about res_config?
Because it can save your ass!

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Brian Wilkins
> Sent: Wednesday, October 13, 2004 7:51 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] SIP peers in MySQL Database
> 
> I've had bad experiences with reloads of thousands of extensions several
> times
> throughout the day. I've spoken with others at AstriCon about the same
> issue,
> fyi
> 
> On Wednesday 13 October 2004 04:35 pm, Brian West wrote:
> > > The perl script will overwrite the existing conf file. I've had bad
> > > experiences with constant reloading. Maybe you want to schedule your
> > > updates
> > > through a crontab.
> >
> > Problem?  We do HUNDREDS of reloads a day without problem.  Just an FYI
> >
> > bkw
> >
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> 
> --
> --
> Heritage Communications Corporation
>   Melbourne, FL USA 32935
> http://www.hcc.net
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RE: [Asterisk-Users] OpenSwitch12 install problems

[Brian West]

Don't know if it works on the 2.6 yet.  I did get the 6 working on 2.4
without a problem.  I helped them trouble shoot the driver for it a bit.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Nichols, Andrew
> Sent: Wednesday, October 13, 2004 2:38 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] OpenSwitch12 install problems
> 
> I recently purchased an OpenSwitch12. I am having some trouble with the
> install. I am running Fedora 2 with the 2.6.8-1.521 kernel. The
> OpenSwitch12 is the only device using PCI slots.
> 
> When I unzip the 2.3.2 driver files that I got from www.voicetronix.com
> and do a make, I get the following error (I have pasted the full text that
> is output at the end of this email):
> 
> vpb.c:162: error: storage size of `vpb_fops' isn't known
> make[1]: *** [vpb.o] Error 1
> make[1]: Leaving directory `/usr/local/vpb-driver-2.3.2/src'
> make: *** [all] Error 2
> 
> I read the FAQ that is included with the drivers and made a symbolic link
> called "linux" in /usr/src, tested and received the same error. I then
> edited src/Makefile to point cc -O6 -m486 -c vpb.c -
> I/usr/src/linux/include -Wall to my kernel source, tested and received the
> same error. I then edited src/vpb.c to:
> 
> /*#ifdef MODVERSIONS*/
> #include 
> #include 
> /*#endif*/
> 
> I then tested and received the same error. I do have a file called
> "modversions.h" in linux, but I do not have one called "module.h".
> Instead, I have "modules.h". I tried using "modules.h" and this failed. I
> then found module.h in a subdirectory of linux and pointed to that file. I
> tested again and the make failed.
> 
> I have tried all of this with the 2.3.1 drivers and with kernel 2.6.5-
> 1.358 with the same results. Has anyone else run into these errors? Do you
> know if this card is compatible with Asterisk 1.0 and this Kernel?
> 
> Thanks,
> 
> Andrew
> 
> 
> 
> 
> 
> 
> 
> [EMAIL PROTECTED] root]# cd /usr/local/vpb-driver-2.3.2
> [EMAIL PROTECTED] vpb-driver-2.3.2]# make
> echo LINUX
> LINUX
> cd src; echo "#define VERSION \"2.3.2\" " > version.h ;make
> make[1]: Entering directory `/usr/local/vpb-driver-2.3.2/src'
> cc -O6 -mcpu=i486 -c vpb.c -o vpb-tmp.o -I/lib/modules/2.6.8-1.521/build -
> Wall
> In file included from vpb.c:95:
> /usr/include/linux/autoconf.h:1:2: #error Invalid kernel header included
> in userspace
> In file included from vpb.c:102:
> /usr/src/linux/include/asm/module.h:54:2: #error unknown processor family
> vpb.c:108:22: linux/mm.h: No such file or directory
> In file included from /usr/include/linux/fs.h:9,
>  from vpb.c:109:
> /usr/include/linux/config.h:5:2: #error Incorrectly using glibc headers
> for a kernel module
> vpb.c:114:25: asm/uaccess.h: No such file or directory
> In file included from vpb.c:117:
> /usr/include/asm/io.h:4:2: warning: #warning  is deprecated, use
>  instead
> vpb.c:119:27: linux/vmalloc.h: No such file or directory
> vpb.c:121:25: linux/delay.h: No such file or directory
> vpb.c:122:25: asm/uaccess.h: No such file or directory
> vpb.c:143: warning: `struct file' declared inside parameter list
> vpb.c:143: warning: its scope is only this definition or declaration,
> which is probably not what you want
> vpb.c:143: warning: `struct inode' declared inside parameter list
> vpb.c:162: error: variable `vpb_fops' has initializer but incomplete type
> vpb.c:164: error: unknown field `owner' specified in initializer
> vpb.c:164: error: `__this_module' undeclared here (not in a function)
> vpb.c:164: warning: excess elements in struct initializer
> vpb.c:164: warning: (near initialization for `vpb_fops')
> vpb.c:165: error: unknown field `llseek' specified in initializer
> vpb.c:165: warning: excess elements in struct initializer
> vpb.c:165: warning: (near initialization for `vpb_fops')
> vpb.c:166: error: unknown field `read' specified in initializer
> vpb.c:166: warning: excess elements in struct initializer
> vpb.c:166: warning: (near initialization for `vpb_fops')
> vpb.c:167: error: unknown field `write' specified in initializer
> vpb.c:167: warning: excess elements in struct initializer
> vpb.c:167: warning: (near initialization for `vpb_fops')
> vpb.c:168: error: unknown field `poll' specified in initializer
> vpb.c:168: warning: excess elements in struct initializer
> vpb.c:168: warning: (near initialization for `vpb_fops')
> vpb.c:169: error: unknown field `ioctl' specified in initializer
> vpb.c:169: warning: excess elements in struct initializer
> vpb.c:169: warning: (near initialization for `vpb_fops')
> vpb.c:170: error: unknown field `open' specified in initializer
> vpb.c:170: warning: excess elements in struct initializer
> vpb.c:170: warning: (near initialization for `vpb_fops')
> vpb.c:171: error: unknown field `release' specified in initializer
> vpb.c:171: warning: excess elements in struct initialize

[Asterisk-Users] GPL Violations (Was: Advice on OS Choice)

[Brian West]

> The EULA is where the real teeth are -- prohibiting even people who
> have purchased RHEL from using it in ways that RedHat prohibits.  For
> example, it is not possible to purchase one copy of RHEL and install it
> on two machines.  Nor are you allowed to run RHEL on a machine without
> having purchased support.  I am unclear on how this is not a further
> restriction on the code (and therefore prohibited by the GPL) but the
> FSF appears unwilling to pursue the point.

I do feel that those are violations of the GPL.  They can't place more
restrictions on software that is already free via the GPL.  This is the
exact reason I told RedHat to f$%k off.  They used the community to build a
brand then said F$%K YOU, but here is Fedora which we will use to test new
stuff that might make it into RedHat's high end products.  So basically
you're a test community for future RedHat products if you run Fedora. 

Personally if the copyright holder of a GPL software such as Asterisk or any
GPL software allows someone to violate the GPL then allow them to skid by is
setting a bad precedent for the GPL.  I think if you find a GPL violation
you must stand up and make an example out of said company.  If a copyright
holder doesn't standup and say something that only weakens the GPL's power.
The GPL is still untested in the court system.

That's my opinion.  Flame away!  Software only wants to be free.

bkw


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RE: [Asterisk-Users] GPL Violations (Was: Advice on OS Choice)

[Brian West]

Good point.  But that doesn't mean anything in this silly US Court system.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Stefan de Konink
> Sent: Thursday, October 14, 2004 7:13 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] GPL Violations (Was: Advice on OS Choice)
> 
> On Thu, 14 Oct 2004, Brian West wrote:
> 
> > The GPL is still untested in the court system.
> 
> It stands in Germany now, Netfilter vs Sitecom GmbH, that is one of the
> reasons why OpenXchange got Open Source & GPL.
> 
> 
> Stefan de Konink
> 
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RE: [Asterisk-Users] Dialogic D/300JCT-E1 support

[Brian West]

> You would do well to ebay the card if you don't otherwise need it and
> then buy a Digium card.

And you have to sign an NDA to get the drivers for a Dialogic card from
Digium.

bkw

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RE: [Asterisk-Users] GPL Violations (Was: Advice on OS Choice)

[Brian West]

> http://www.taolinux.org/
> 
> http://www.whiteboxlinux.org/

I might have to check them out... I'm still a hardcore gentoo fan.

bkw

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RE: [Asterisk-Users] GPL Violations (Was: Advice on OS Choice)

[Brian West]

> IANAL but you can make any changes you want provided you do not
> distribute/sell the changed software without making the changed code
> available. And AFAIK *Available* does not mean "ship with". Available
> can mean download, CVS or other means. Distribution is what triggers the
> GPL. And use within a company does not, I believe, constitute
> distribution.
> 
> I also think GPL violations are rare. But there was a highly publicized
> alleged violation by Cisco/Linksys:

I hate to say this but it's not as rare as you think.  

bkw

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RE: [Asterisk-Users] res_odbc app_realtime

[Brian West]

You must still have a sip.conf with the [general] section.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Michael Baird
> Sent: Friday, October 15, 2004 2:30 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] res_odbc app_realtime
> 
> Hello, I've been trying to move some of my static configs into a MySQL
> database. I have been trying with sip.conf and voicemail.conf, can't
> seem to get it to work although it says it is loading the config via
> res_odbc fine. Here's what I did, if someone can't point out my error, I
> would be grateful.
> 
> Setup up UnixODBC, verified I could connect to my database through it,
> with the test application, all worked fine.
> 
> I then setup res_odbc.conf in /etc/asterisk, like so
> [mysql]
> dsn => MySQL-voicemail
> username => dbuser
> password => dbpass
> pre-connect => yes
> 
> I setup /etc/asterisk/extconfig.conf with the following info.
> [settings]
> ; file.conf => driver,database[,table]
> sip.conf => odbc,accounting,sipusers
> 
> I removed sip.conf from /etc/asterisk
> 
> I setup my database called accounting, with data in the table sipusers
> 
> sipusers table layout
> 
> name | host | context | ipaddr | port | regseconds | callerid | username
> | md5secret
> 
> Then I start up asterisk with asterisk - and see the following
> 
> Oct 15 15:18:54 NOTICE[1077037376]: config.c:556 ast_config_register:
> Registered Config Engine odbc
>   == Parsing '/etc/asterisk/extconfig.conf': Found
>   == Binding sip.conf to mysql/accounting/sipusers
> res_config_odbc loaded.
> 
> Which looks good I think, until the end, and chan_sip.so fails to load
> and sip doesn't run.
> 
> [chan_sip.so] => (Session Initiation Protocol (SIP))
>   == Parsing '/etc/asterisk/sip.conf': Not found (No such file or
> directory)
> Oct 15 15:23:21 NOTICE[1077037376]: chan_sip.c:8432 reload_config:
> Unable to load config sip.conf, SIP disabled
> 
> I tried setting up voicemail.conf with similar errors, both voicemail
> and sip work fine when accessed from flatfiles.
> 
> Regards
> Michael Baird
> 
> 
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RE: [Asterisk-Users] res_odbc app_realtime

[Brian West]

Also your database is [mysql] not accounting.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Brian West
> Sent: Friday, October 15, 2004 2:36 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] res_odbc app_realtime
> 
> You must still have a sip.conf with the [general] section.
> 
> bkw
> 
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Michael Baird
> > Sent: Friday, October 15, 2004 2:30 PM
> > To: [EMAIL PROTECTED]
> > Subject: [Asterisk-Users] res_odbc app_realtime
> >
> > Hello, I've been trying to move some of my static configs into a MySQL
> > database. I have been trying with sip.conf and voicemail.conf, can't
> > seem to get it to work although it says it is loading the config via
> > res_odbc fine. Here's what I did, if someone can't point out my error, I
> > would be grateful.
> >
> > Setup up UnixODBC, verified I could connect to my database through it,
> > with the test application, all worked fine.
> >
> > I then setup res_odbc.conf in /etc/asterisk, like so
> > [mysql]
> > dsn => MySQL-voicemail
> > username => dbuser
> > password => dbpass
> > pre-connect => yes
> >
> > I setup /etc/asterisk/extconfig.conf with the following info.
> > [settings]
> > ; file.conf => driver,database[,table]
> > sip.conf => odbc,accounting,sipusers
> >
> > I removed sip.conf from /etc/asterisk
> >
> > I setup my database called accounting, with data in the table sipusers
> >
> > sipusers table layout
> >
> > name | host | context | ipaddr | port | regseconds | callerid | username
> > | md5secret
> >
> > Then I start up asterisk with asterisk - and see the following
> >
> > Oct 15 15:18:54 NOTICE[1077037376]: config.c:556 ast_config_register:
> > Registered Config Engine odbc
> >   == Parsing '/etc/asterisk/extconfig.conf': Found
> >   == Binding sip.conf to mysql/accounting/sipusers
> > res_config_odbc loaded.
> >
> > Which looks good I think, until the end, and chan_sip.so fails to load
> > and sip doesn't run.
> >
> > [chan_sip.so] => (Session Initiation Protocol (SIP))
> >   == Parsing '/etc/asterisk/sip.conf': Not found (No such file or
> > directory)
> > Oct 15 15:23:21 NOTICE[1077037376]: chan_sip.c:8432 reload_config:
> > Unable to load config sip.conf, SIP disabled
> >
> > I tried setting up voicemail.conf with similar errors, both voicemail
> > and sip work fine when accessed from flatfiles.
> >
> > Regards
> > Michael Baird
> >
> >
> > ___
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RE: [Asterisk-Users] res_odbc app_realtime

[Brian West]

If your setting up realtime its only for friends.. so yes you would still
need sip.conf for the peers/friends that aren't listed in the conf file.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Michael Baird
> Sent: Friday, October 15, 2004 3:09 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] res_odbc app_realtime
> 
> Thanks, I had tried it with a basic sip.conf, does this mean I can still
> have settings in both a flatfile and a database for sip for example? I
> changed the database in my extconfig.conf to mysql, and it still failed,
> I ran a trace on odbc and it had an error message, I'm not sure what
> that means. I'm using todays asterisk CVS. Is there a better way for me
> to debug what's happening, asterisk - doesn't show me anything when
> a sip call comes in, other then it can't find the extension.
> 
> SQL Error
> >SQLError
> | enter: szErrorMsg: bfffd530
>  
> Regards
> Michael Baird
> > Also your database is [mysql] not accounting.
> >
> > bkw
> >
> > > -Original Message-
> > > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > > [EMAIL PROTECTED] On Behalf Of Brian West
> > > Sent: Friday, October 15, 2004 2:36 PM
> > > To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> > > Subject: RE: [Asterisk-Users] res_odbc app_realtime
> > >
> > > You must still have a sip.conf with the [general] section.
> > >
> > > bkw
> > >
> > > > -Original Message-
> > > > From: [EMAIL PROTECTED] [mailto:asterisk-
> users-
> > > > [EMAIL PROTECTED] On Behalf Of Michael Baird
> > > > Sent: Friday, October 15, 2004 2:30 PM
> > > > To: [EMAIL PROTECTED]
> > > > Subject: [Asterisk-Users] res_odbc app_realtime
> > > >
> > > > Hello, I've been trying to move some of my static configs into a
> MySQL
> > > > database. I have been trying with sip.conf and voicemail.conf, can't
> > > > seem to get it to work although it says it is loading the config via
> > > > res_odbc fine. Here's what I did, if someone can't point out my
> error, I
> > > > would be grateful.
> > > >
> > > > Setup up UnixODBC, verified I could connect to my database through
> it,
> > > > with the test application, all worked fine.
> > > >
> > > > I then setup res_odbc.conf in /etc/asterisk, like so
> > > > [mysql]
> > > > dsn => MySQL-voicemail
> > > > username => dbuser
> > > > password => dbpass
> > > > pre-connect => yes
> > > >
> > > > I setup /etc/asterisk/extconfig.conf with the following info.
> > > > [settings]
> > > > ; file.conf => driver,database[,table]
> > > > sip.conf => odbc,accounting,sipusers
> > > >
> > > > I removed sip.conf from /etc/asterisk
> > > >
> > > > I setup my database called accounting, with data in the table
> sipusers
> > > >
> > > > sipusers table layout
> > > >
> > > > name | host | context | ipaddr | port | regseconds | callerid |
> username
> > > > | md5secret
> > > >
> > > > Then I start up asterisk with asterisk - and see the following
> > > >
> > > > Oct 15 15:18:54 NOTICE[1077037376]: config.c:556
> ast_config_register:
> > > > Registered Config Engine odbc
> > > >   == Parsing '/etc/asterisk/extconfig.conf': Found
> > > >   == Binding sip.conf to mysql/accounting/sipusers
> > > > res_config_odbc loaded.
> > > >
> > > > Which looks good I think, until the end, and chan_sip.so fails to
> load
> > > > and sip doesn't run.
> > > >
> > > > [chan_sip.so] => (Session Initiation Protocol (SIP))
> > > >   == Parsing '/etc/asterisk/sip.conf': Not found (No such file or
> > > > directory)
> > > > Oct 15 15:23:21 NOTICE[1077037376]: chan_sip.c:8432 reload_config:
> > > > Unable to load config sip.conf, SIP disabled
> > > >
> > > > I tried setting up voicemail.conf with similar errors, both
> voicemail
> > > > and sip work fine when accessed from flatfiles.
> > > >
> > > > Regards
> > > > Michael Baird
> > > >
> > > >
> > > &

RE: [Asterisk-Users] res_odbc app_realtime

[Brian West]

Matthew,
If you have a chance can you look at the bug associated with having
both mysql and odbc loaded?  If one or the other is loaded its fine.. both
it goes kaboom.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Matthew Boehm
> Sent: Friday, October 15, 2004 3:10 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] res_odbc app_realtime
> 
> >From what I have seen, that is the wrong table layout for using res_odbc
> and
> storing *.conf files.
> Should be more like this:
> 
> http://www.voip-info.org/wiki-Asterisk+res_config
> 
> If you want to use RealTime and MySQL, I have developed (the only) a
> native
> RealTime driver for MySQL.
> 
> It bypasses ODBC and makes calls directly to mysql.
> 
> http://bugs.digium.com/bug_view_page.php?bug_id=0002613
> 
> It should be noted that RealTime is currently in development and not
> recommened for prod servers.
> 
> Matthew
> - Original Message -
> From: "Michael Baird" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Friday, October 15, 2004 2:29 PM
> Subject: [Asterisk-Users] res_odbc app_realtime
> 
> 
> > Hello, I've been trying to move some of my static configs into a MySQL
> > database. I have been trying with sip.conf and voicemail.conf, can't
> > seem to get it to work although it says it is loading the config via
> > res_odbc fine. Here's what I did, if someone can't point out my error, I
> > would be grateful.
> >
> > Setup up UnixODBC, verified I could connect to my database through it,
> > with the test application, all worked fine.
> >
> > I then setup res_odbc.conf in /etc/asterisk, like so
> > [mysql]
> > dsn => MySQL-voicemail
> > username => dbuser
> > password => dbpass
> > pre-connect => yes
> >
> > I setup /etc/asterisk/extconfig.conf with the following info.
> > [settings]
> > ; file.conf => driver,database[,table]
> > sip.conf => odbc,accounting,sipusers
> >
> > I removed sip.conf from /etc/asterisk
> >
> > I setup my database called accounting, with data in the table sipusers
> >
> > sipusers table layout
> >
> > name | host | context | ipaddr | port | regseconds | callerid | username
> > | md5secret
> >
> > Then I start up asterisk with asterisk - and see the following
> >
> > Oct 15 15:18:54 NOTICE[1077037376]: config.c:556 ast_config_register:
> > Registered Config Engine odbc
> >   == Parsing '/etc/asterisk/extconfig.conf': Found
> >   == Binding sip.conf to mysql/accounting/sipusers
> > res_config_odbc loaded.
> >
> > Which looks good I think, until the end, and chan_sip.so fails to load
> > and sip doesn't run.
> >
> > [chan_sip.so] => (Session Initiation Protocol (SIP))
> >   == Parsing '/etc/asterisk/sip.conf': Not found (No such file or
> > directory)
> > Oct 15 15:23:21 NOTICE[1077037376]: chan_sip.c:8432 reload_config:
> > Unable to load config sip.conf, SIP disabled
> >
> > I tried setting up voicemail.conf with similar errors, both voicemail
> > and sip work fine when accessed from flatfiles.
> >
> > Regards
> > Michael Baird
> >
> >
> > ___
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[Asterisk-Users] Asked to transmit frame type 64, while native formats is 8

[Brian West]

http://bugs.digium.com/bug_view_page.php?bug_id=0002519

If anyone has seen that error please come forward and report on this bug
please.  The original reporter is unwilling or unmotivated to even make an
effort to assist in correcting the issue.  So if anyone else has seen this
please post.

Thanks,
Brian

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RE: [Asterisk-Users] Asked to transmit frame type 64, whilenative formats is 8

[Brian West]

Care to post your findings to the bug note?

Thanks,
Brian

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Danny Froberg
> Sent: Sunday, October 17, 2004 11:53 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Asked to transmit frame type 64, whilenative
> formats is 8
> 
> Think i solved that one by the ordering of allow= in sip.conf
> 
> At 18:39 2004-10-17, you wrote:
> >http://bugs.digium.com/bug_view_page.php?bug_id=0002519
> >
> >If anyone has seen that error please come forward and report on this bug
> >please.  The original reporter is unwilling or unmotivated to even make
> an
> >effort to assist in correcting the issue.  So if anyone else has seen
> this
> >please post.
> >
> >Thanks,
> >Brian
> 
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RE: [Asterisk-Users] Re: can not compile chan_capi 0.3.5

[Brian West]

Or better yet fix it yourself its like all of a few lines to make it work.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Patrick
> Sent: Sunday, October 17, 2004 4:10 PM
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
> Discussion
> Subject: Re: [Asterisk-Users] Re: can not compile chan_capi 0.3.5
> 
> On Sun, 2004-10-17 at 22:58, Nicolas wrote:
> > Patrick wrote:
> >
> > > On Sun, 2004-10-17 at 22:12, Nicolas wrote:
> > >> Hello,
> > >>
> > >> i can not compile chan_capi 0.3.5 on a suse 9.1 plattform.
> > >> i run latest asterisk cvs build 14/10/04.
> > >
> 
> chan_capi uses header files from asterisk. Look in the chan_capi
> Makefile and you will see. Obviously chan_capi does not know about the
> new callerid code that is part of recent asterisk cvs. They are tied
> together. That is why you need to use v1-0 of asterisk or wait until
> kapejod releases an updated chan_capi (prepare for a wait afaik). Or fix
> it yourself off course...
> 
> Regards,
> Patrick
> 
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[Asterisk-Users] Calling all Users to check out bug 2655

[Brian West]

http://bugs.digium.com/bug_view_page.php?bug_id=0002655

Can anyone comment on what is proper or not.

Thanks,
Brian

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RE: [Asterisk-Users] mysql sipfriends and allowing individual codecsper user?

[Brian West]

Realtime in cvs-head allows this.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Jens Kübler
> Sent: Monday, October 18, 2004 7:33 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] mysql sipfriends and allowing individual
> codecsper user?
> 
> Am Montag, 18. Oktober 2004 14:21 schrieb Roy Sigurd Karlsbakk:
> > hi
> >
> > how can I, using mysql sipfriends, allow one user to use g.729 while
> > disallowing this codec on another user?
> >
> > thanks
> >
> > roy
> >
> There is currently no support for allow clauses unless you decide to add
> it.
> 
> Jens
> 
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RE: [Asterisk-Users] verisign immitate e164

[Brian West]

Lets all post DUNDi to /. 

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Matt Riddell
> Sent: Monday, October 18, 2004 7:03 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] verisign immitate e164
> 
> dean collins wrote:
> >
> >
> > http://www.eweek.com/article2/0,1759,1678531,00.asp
> >
> > Verisign announce at VON that they will try to imitate www.e164.org
> >  , no comment on them all growing beards.
> >
> > Cheers,
> >
> > Dean
> >
> Quickly followed by Digium's release of DUNDi (tm) (Distributed
> Universal Number Directory):
> 
> http://www.sineapps.com/news.php?rssid=240
> 
> :-)
> 
> --
> Cheers,
> 
> Matt Riddell
> ___
> 
> http://www.sineapps.com/news.php (Daily Asterisk News - html)
> http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)
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RE: [Asterisk-Users] verisign immitate e164

[Brian West]

I still no see it in /.  

Bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Matt Riddell
> Sent: Monday, October 18, 2004 7:19 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] verisign immitate e164
> 
> Brian West wrote:
> > Lets all post DUNDi to /.
> >
> > bkw
> >
> Agreed.
> 
> --
> Cheers,
> 
> Matt Riddell
> ___
> 
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RE: [Asterisk-Users] GSM to g729 Conversion

[Brian West]

Record them from a phone that speaks g729 right to raw .g729 files.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Matthew Boehm
> Sent: Tuesday, October 19, 2004 10:21 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] GSM to g729 Conversion
> 
> States right here:
> 
> http://www.voip-info.org/tiki-
> index.php?page=Convert%20WAV%20audio%20files%20for%20use%20in%20Asterisk
> 
> Asterisk can play anything it has a format and codec for. Including wav,
> gsm, g729, g726, wav49 all of which can be used for Playback and
> Background.
> 
> So, how can you make g729 files for Playback and Background?
> 
> Thanks,
> Matthew
> 
> - Original Message -
> From: "Kanuri, Seshu (Company IT)" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <[EMAIL PROTECTED]>
> Sent: Tuesday, October 19, 2004 8:54 AM
> Subject: RE: [Asterisk-Users] GSM to g729 Conversion
> 
> 
> You are mixing oranges and apples here i guess. G729 is a Media
> Transmission
> Protocol Codec the other is a Compressed Audio File format.
> 
> There are no .g729 audio files as far as I know.
> 
> Seshu Kanuri
> 
> 
> 
>   _
> 
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Victor
> Cartes
> Sent: Monday, October 18, 2004 3:39 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] GSM to g729 Conversion
> 
> 
> Hi!
> 
> Does anybody know how to convert .gsm file format to .g729 in order to use
> it for an IVR system?
> 
> Thanks in advance.
> 
> Vïctor
> 
> 
> NOTICE: If received in error, please destroy and notify sender.  Sender
> does
> not waive confidentiality or privilege, and use is prohibited.
> 
> 
> 
> 
> --
> --
> 
> 
> 
> > ___
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RE: [Asterisk-Users] chan_mISDN

[Brian West]

Well the error does give you some clue on whats wrong and it's done that way
to give you exactly what you need to do:

Use AST_DEFINE_STATIC rather than AST_MUTEXT_INITIALIZER

Check out the other apps and compare them to chan_mISDN and you'll get what
you need to change.. its only one line if I recall.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Erwan DESVERGNES
> Sent: Tuesday, October 19, 2004 10:57 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] chan_mISDN
> 
> Did someone have succeed to compile chan_misdn ???
> 
> 
> 
> I’ve got an error when in try to compile
> 
> 
> 
> chan_misdn.c:68: error:
> `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__'
> undeclared here (not in a function)
> 
> 
> 
> 
> 
> thanks
> 
> 
> 
> _
> 
> Erwan Desvergnes - ANDIUM -
> 
> 82/86 rue Château Gaillard
> 
> 69100 Villeurbanne
> 
> 
> 
> Tel. 04 37 43 44 45 / Fax 04 37 43 44 44
> 
> E-mail: [EMAIL PROTECTED] 
> 
> 


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RE: [Asterisk-Users] DUNDi on Slashdot

[Brian West]

Head as of now is pretty damn stable.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Robert Jackson
> Sent: Tuesday, October 19, 2004 1:56 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] DUNDi on Slashdot
> 
> DUNDi made /.  Check it out at:
> 
> http://www.dundi.com
> 
> Yet, another great idea!!  Thanks Mark!!
> 
> I wish it was in v1.0, but I guess I'll have to update to head.
> 
> Robert Jackson
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RE: [Asterisk-Users] DUNDi in stable? (New subject)

[Brian West]

I totally agree... if you want DUNDi get cvs-head NOW and I mean NOW.. next
week lots of stuff starts to change and it will NOT be something you will
wanna run in production.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Olle E. Johansson
> Sent: Tuesday, October 19, 2004 3:42 PM
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
> Discussion
> Subject: Re: [Asterisk-Users] DUNDi in stable? (New subject)
> 
> Ryan Courtnage wrote:
> 
> >>http://www.dundi.com
> >>
> >>Yet, another great idea!!  Thanks Mark!!
> >>
> >>I wish it was in v1.0, but I guess I'll have to update to head.
> >
> >
> > I wish it were in v1.0 as well. Would creating a patch for 1.0 be pretty
> > simple, or do the code changes run deep?
> It will eventuall get into a release.
> 
> But please, as a community, we have to refrain from temptation of adding
> new functions to the stable tree. It has to be kept stable - and that is
> boring. If you want to walk on the wild side, run CVS head.
> 
> CVS head is the development tree. Please don't encourage people to use it
> in production environments, even if it from time to time seems to work
> well.
> We need to be able to include new untested functions. Some days, it
> doesn't
> compile properly after you download it to your system. That is okey, it
> means
> that we have new code, new functions and new bugs to fix.
> 
> We need a development CVS as fertile soil for new major Asterisk
> breakthroughs.
> I've been waiting long for that to happen, and if you check the -cvs list,
> you'll find that there's been a lot of changes to the development branch
> that would never have happened if we didn't have a less strict environment
> to play around with. Keeps the bug tracker alive :-)
> 
> /Olle
> 
> 
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RE: [Asterisk-Users] DUNDi in stable? (New subject)

[Brian West]

Yes but I'm just saying that if you want it get a checkout it from now as in
"THIS POINT IN TIME" otherwise you're gonna have fun in the next few weeks
once the major changes start going in.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Kevin P. Fleming
> Sent: Tuesday, October 19, 2004 5:18 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] DUNDi in stable? (New subject)
> 
> Deon Rodden wrote:
> > When do you think the last stable CVS will be available before "lots of
> > stuff" begins to change? I want to find the best possible Asterisk and
> stick
> > with it, for some time, maybe until 2.0; If I get CVS right now, what if
> > tomorrow or the day after he comes out with a better CVS.
> 
> There is no need to rush and "pull it now"... you can always pull a
> snapshot of the tree as of any past date if you want, it's easy to do.
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[Asterisk-Users] Anyone else seeing this?

[Brian West]

Anything after these versions:

zaptel.c version 1.95 (known working)
chan_zap.c version 1.357 (known working)

with a tor2 card... causes kernel panic... 

Can anyone else confirm this?  I honestly think it's a combo issue with the
new zap reload and that zaptel change.  But I have spent hours trying to
narrow it down to those two files and those changes.

Has anyone else seen strange issues when using PRI?

If we have zaptel.c 1.95 and latest chan_zap.c you can place and take calls
but if you do something like show channels at the CLI you'll deadlock the
box.  I have no thread apply all bt since the glibc on this box didn't have
debug compiled in on it. (will retry this tomorrow)

If you have the latest zaptel.c and the latest chan_zap.c placing any call
out/in the zap interface will cause a kernel panic and kill the box. 

Use the above listed known working files and you have no problems.  

I would open a bug report but I would like to find more information before
doing so.

Thanks,
Brian

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RE: [Asterisk-Users] Anyone else seeing this?

[Brian West]

I'm going to do some more research and testing today to find out what's up.
Maybe I can get the darn thing figured out.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Steve Underwood
> Sent: Wednesday, October 20, 2004 9:05 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Anyone else seeing this?
> 
> Hi Brian,
> 
> I haven't reported this yet, as I don't have an overall picture of what
> is happening, but
> 
> A couple of weeks ago I had several machine lockups on the same day
> while testing MFC/R2 with a tor2. It hasn't happened any more here. I
> have no idea why it suddenly started or stopped. However, now people are
> starting to deploy R2, I have reports of occasional lockups with tor2
> cards. I have no idea if these lockups have the same cause as mine.
> 
> Regards,
> Steve
> 
> 
> Brian West wrote:
> 
> >Anything after these versions:
> >
> >zaptel.c version 1.95 (known working)
> >chan_zap.c version 1.357 (known working)
> >
> >with a tor2 card... causes kernel panic...
> >
> >Can anyone else confirm this?  I honestly think it's a combo issue with
> the
> >new zap reload and that zaptel change.  But I have spent hours trying to
> >narrow it down to those two files and those changes.
> >
> >Has anyone else seen strange issues when using PRI?
> >
> >If we have zaptel.c 1.95 and latest chan_zap.c you can place and take
> calls
> >but if you do something like show channels at the CLI you'll deadlock the
> >box.  I have no thread apply all bt since the glibc on this box didn't
> have
> >debug compiled in on it. (will retry this tomorrow)
> >
> >If you have the latest zaptel.c and the latest chan_zap.c placing any
> call
> >out/in the zap interface will cause a kernel panic and kill the box.
> >
> >Use the above listed known working files and you have no problems.
> >
> >I would open a bug report but I would like to find more information
> before
> >doing so.
> >
> >Thanks,
> >Brian
> >
> >
> 
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RE: [Asterisk-Users] DUNDi in stable? (New subject)

[Brian West]

cvs checkout zaptel libpri asterisk  <== HEAD

cvs checkout -r v1-0 zaptel libpri asterisk  <== STABLE with bug fixes.

bkw


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Deon Rodden
> Sent: Wednesday, October 20, 2004 9:58 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] DUNDi in stable? (New subject)
> 
> That's good to know. But, not to sound dumb, I'm not a heavy CVS user, how
> do I get the latest stable? As of now.
> 
> The way I'm used to doing it is:
> export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot
> cvs login
> cvs checkout zaptel libpri asterisk
> 
> 
> But that doesn't tell me if that's head or stable. The instructions say:
> cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-
> sounds
> 
> For stable. But my understanding is that will give me version 1.0; no bug
> fixes since the release of 1.0. I want the latest w/ bug fixes but no new
> features.
> 
> My voicemail right now is not rigged for database support and such, just
> the
> standard voicemail.conf; So if I go to the latest, I don't want to be
> forced
> to retrofit my current voicemail setup.
> 
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
> Fleming
> Sent: Tuesday, October 19, 2004 6:18 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] DUNDi in stable? (New subject)
> 
> Deon Rodden wrote:
> > When do you think the last stable CVS will be available before "lots of
> > stuff" begins to change? I want to find the best possible Asterisk and
> stick
> > with it, for some time, maybe until 2.0; If I get CVS right now, what if
> > tomorrow or the day after he comes out with a better CVS.
> 
> There is no need to rush and "pull it now"... you can always pull a
> snapshot of the tree as of any past date if you want, it's easy to do.
> ___
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RE: [Asterisk-Users] DUNDi in stable? (New subject)

[Brian West]

www.bkw.org/dundi.tar.gz should compile and install on stable

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Brian West
> Sent: Wednesday, October 20, 2004 10:03 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] DUNDi in stable? (New subject)
> 
> 
> cvs checkout zaptel libpri asterisk  <== HEAD
> 
> cvs checkout -r v1-0 zaptel libpri asterisk  <== STABLE with bug fixes.
> 
> bkw
> 
> 
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Deon Rodden
> > Sent: Wednesday, October 20, 2004 9:58 AM
> > To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> > Subject: RE: [Asterisk-Users] DUNDi in stable? (New subject)
> >
> > That's good to know. But, not to sound dumb, I'm not a heavy CVS user,
> how
> > do I get the latest stable? As of now.
> >
> > The way I'm used to doing it is:
> > export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot
> > cvs login
> > cvs checkout zaptel libpri asterisk
> >
> >
> > But that doesn't tell me if that's head or stable. The instructions say:
> > cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-
> > sounds
> >
> > For stable. But my understanding is that will give me version 1.0; no
> bug
> > fixes since the release of 1.0. I want the latest w/ bug fixes but no
> new
> > features.
> >
> > My voicemail right now is not rigged for database support and such, just
> > the
> > standard voicemail.conf; So if I go to the latest, I don't want to be
> > forced
> > to retrofit my current voicemail setup.
> >
> >
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
> > Fleming
> > Sent: Tuesday, October 19, 2004 6:18 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] DUNDi in stable? (New subject)
> >
> > Deon Rodden wrote:
> > > When do you think the last stable CVS will be available before "lots
> of
> > > stuff" begins to change? I want to find the best possible Asterisk and
> > stick
> > > with it, for some time, maybe until 2.0; If I get CVS right now, what
> if
> > > tomorrow or the day after he comes out with a better CVS.
> >
> > There is no need to rush and "pull it now"... you can always pull a
> > snapshot of the tree as of any past date if you want, it's easy to do.
> > ___
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> >
> >
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RE: [Asterisk-Users] IP Phone that OFFICIALLY support Asterisk

[Brian West]

Good luck!  Personally I like my cisco 7960's

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Joseph
> Sent: Wednesday, October 20, 2004 12:59 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] IP Phone that OFFICIALLY support Asterisk
> 
> What IP Phones officially support Asterisk.  I know that most of them
> will work with * but I do not want to support companies that don't
> support OSS
> 
> --
> #Joseph
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RE: [Asterisk-Users] Manger API flag from dialplan

[Brian West]

asterisk*CLI> show application UserEvent
asterisk*CLI>
  -= Info about application 'UserEvent' =-

[Synopsis]:
Send an arbitrary event to the manager interface

[Description]:
  UserEvent(eventname[|body]): Sends an arbitrary event to the
manager interface, with an optional body representing additional
arguments.  The format of the event will be:
Event: UserEvent
Channel: 
Uniqueid: 
[body]
If the body is not specified, only Event, Channel, and Uniqueid fields
will be present.  Returns 0.

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of TELUX
> Sent: Wednesday, October 20, 2004 1:35 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Manger API flag from dialplan
> 
> Is there a way to flag the manager API on an event from the dialplan?
> 
> db
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Re: [Asterisk-Users] Quality of provider: VocTel

[Brian West]

Livevoip was HARDLY big.  They had one server total if you read their  
bankruptcy papers.


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

On Jun 29, 2005, at 10:36 PM, Michael Stahl wrote:


Any users of the VocTel VOIP service?  (Canadian)

How have you found the quality (Choppy / smooth audio)?
Any problems registering?  (I have been unable to register for hours)

After reading about the collapse of a big USA VOIP provider, I'm  
curious


Thanks,
OCG
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Re: [Asterisk-Users] Can't build cdr_addon_mysql.

[Brian West]

You could have just done "ln -s asterisk-1.0.9 asterisk" and it would  
have fixed that.  It should by default do -I../asterisk


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

On Jun 30, 2005, at 1:13 AM, Chris Mason (Lists) wrote:


Marcel van Kaam, Fonetica wrote:


I had the same problem with installing addons. I checked out in  
the file
cdr_addons_mysql.c what the location of the asterisk.h must be and  
changed

the cdr_addons_mysql.c to the location of the asterisk.h file.

After this it worked. Also to be sure do: locate asterisk.h to  
check or you

have the file on your system.

Marcel


Yes, that worked. For the record, it had to be

#include "../asterisk-1.0.9/asterisk.h"

--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED]
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Re: [Asterisk-Users] Resolving groupcalls

[Brian West]

You can also set anything you wish into the CDR variables.  We came  
up with the whole CDR variable thing for this exact purpose.  Check  
cdr_custom to log it like you want.


ie Set(CDR(GROUP)=${GROUPCALL})



/b
PS don't for get to come to cluecon!

On Jun 30, 2005, at 4:15 AM, Chris Coulthurst wrote:


Oops, sent that last one prematurely!

How about the accountcode setting?  You could get user information  
from

that, right?

Maybe you could send:

Asterisk -rx 'show channels'

..and when you get the data, you'd know which channels are up and  
alive

(full names).

You could then re-run the command with the channel information:

Asterisk -rx 'show channel SIP/201-ec69'

..you'd get a dump, with the end looking something like this:
  CDR Variables:
level 1: clid="Chris Office" <201>
level 1: src=201
level 1: dst=18009427433
level 1: dcontext=unlimited
level 1: channel=SIP/201-ec69
level 1: dstchannel=IAX2/provider-7
level 1: lastapp=Dial
level 1: lastdata=iax2/[EMAIL PROTECTED]/2047622726
level 1: start=2005-06-30 02:10:35
level 1: answer=2005-06-30 02:10:38
level 1: end=2005-06-30 02:10:38
level 1: duration=0
level 1: billsec=0
level 1: disposition=ANSWERED
level 1: amaflags=DOCUMENTATION
level 1: accountcode=019284718233 <<--account code
unique to the user
level 1: uniqueid=1120122635.400


Anyway, maybe something like that...

Chris Coulthurst
[EMAIL PROTECTED]



|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Martin Czarnowski
|Sent: Thursday, June 30, 2005 12:58 AM
|To: asterisk-users@lists.digium.com
|Subject: [Asterisk-Users] Resolving groupcalls
|
|
|Hi,
|
|I'm trying to write a tool, which shows me the state of the current
|calls. For this purpose I'm reading from Pipe the Asterisk output and
|parse it... asterisk -vr | mytool
|
|However, the problem ist how to get the information about who got  
this

|call in the group. The Zap channels are assigned dynamical.
|Only thing I
|can see which channel is connect to the caller but not who is
|using the
|channel.
|
|I know there is the CDR output in Master.csv. But it shows me
|the same.
|The other problem with CDR is, that it shows me the Info only
|after the
|call is finished. That's why I'm trying to parse the asterisk output.
|
|My extensions.conf looks like this..
|GROUPCALL => Zap/g2/1200021&Zap/g2/1200022&Zap/g2/1200023
|.
|.
|exten => s,1,Dial(${GROUPCALL})
|
|
|___
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|UNSUBSCRIBE or update options visit:
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[Asterisk-Users] Craig Southeren to speak at Cluecon!

[Brian West]

Through the generosity of our Premier sponsor, Sangoma Technologies  
we are proud to welcome Craig Southeren all the way from Australia.  
Mr. Southeren.s work has pioneered the development of open source  
telephony applications with his ground-breaking OpenH323 protocol  
stack that stood alone as the only open source VOIP software for  
quite some time. Today, Craig continues to raise the bar with the  
next generation OPAL VOIP abstraction layer and WOOMERA a brilliant  
approach to overcome software incompatibles. Hope to see you all at  
ClueCon.


Don't forget to register its less than a month away!!!  If you have  
problems registering please email me.


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

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Re: [Asterisk-Users] Asterisk-CVS-HEAD locks up on 'reload' from CLI (sometimes)

[Brian West]

rm -rf /usr/include/asterisk

do a fresh checkout and try again.

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

On Jul 5, 2005, at 12:02 PM, Chris Coulthurst wrote:

Lately when I issue a 'reload' from the CLI, I find that it will  
sometimes

hang forever, completely locked up.  I can press enter and see the CLI
prompt move, but no commands are taken.  "top" shows asterisk eating
everything up:

  PID USER PRI  NI  SIZE  RSS SHARE STAT %CPU %MEM   TIME CPU  
COMMAND
20669 root  25   0 10068 9.8M  5392 R88.4  1.9   1:02   0  
asterisk

20877 root  15   0  1124 1124   896 R 0.3  0.2   0:00   0 top
1 root  15   0   448  448   396 S 0.0  0.0   0:04   0 init
2 root  15   0 00 0 SW0.0  0.0   0:01   0  
keventd
3 root  15   0 00 0 SW0.0  0.0   0:00   0  
kapmd

4 root  34  19 00 0 SWN   0.0  0.0   0:00   0
ksoftirqd/0

Most recent add-ons have been Speex and h323.  I just installed  
h323 today
and this has been going on for about a week, so I know its not  
that, but I
can't remember if this was happening before Speex or not.  Anyone  
have any

similar happenings?

Chris Coulthurst
[EMAIL PROTECTED]



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Re: [Asterisk-Users] URGENT: hardware spesifications needed

[Brian West]

Why not do your research instead of asking the list to do it for  
you  lazy ass!


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

On Jul 6, 2005, at 2:09 AM, Erdem HAKİ wrote:


Hello;



I need to set up Asterisk to serve and register for 1000 users(not  
simultaneus). What kind of specifications do my server need.




For example:



Xenon processor

1 GB RAM

120 GB HDD  etc...



Thanks for your help..



Erdem HAKI – [EMAIL PROTECTED]

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