Re: [Asterisk-Users] Mutex problem in sip?

[Martin Pycko]

It doesn't look like a problem. It's that when you have so many calls ...
execution of some piece of code protected by mutex takes longer so it
happens that some calls wait for their time . I guess if you have too
many of those messages you should disable them.

regards
Martin

On Thu, 31 Jul 2003, Alex Zarubin wrote:

> Hello,
>
> CVS 07/31/03. Test with 130+ PSTN-to-SIP calls. Asterisk gets locked ...
>
> grep -e "Error" -e "eventually" p-console
>
> chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource
> busy
> chan_sip.c line 1453 (sip_alloc): Got it eventually...
> chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource
> busy
> chan_sip.c line 1453 (sip_alloc): Got it eventually...
> chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource
> busy
> chan_sip.c line 1453 (sip_alloc): Got it eventually...
> chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource
> busy
> chan_sip.c line 1453 (sip_alloc): Got it eventually...
> chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource
> busy
> chan_sip.c line 1453 (sip_alloc): Got it eventually...
> chan_sip.c line 4980 (do_monitor): Error obtaining mutex: Device or resource
> busy
> chan_sip.c line 4980 (do_monitor): Got it eventually...
> chan_sip.c line 4980 (do_monitor): Error obtaining mutex: Device or resource
> busy
> chan_sip.c line 4980 (do_monitor): Got it eventually...
> chan_sip.c line 4980 (do_monitor): Error obtaining mutex: Device or resource
> busy
> chan_sip.c line 4980 (do_monitor): Got it eventually...
> chan_sip.c line 4980 (do_monitor): Error obtaining mutex: Device or resource
> busy
> chan_sip.c line 4980 (do_monitor): Got it eventually...
> chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource
> busy
> chan_sip.c line 1453 (sip_alloc): Got it eventually...
> chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource
> busy
> chan_sip.c line 1453 (sip_alloc): Got it eventually...
> chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource
> busy
> chan_sip.c line 1453 (sip_alloc): Got it eventually...
> chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource
> busy
> chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource
> busy
> chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource
> busy
> chan_sip.c line 1453 (sip_alloc): Got it eventually...
> chan_sip.c line 1453 (sip_alloc): Got it eventually...
> chan_sip.c line 1453 (sip_alloc): Got it eventually...
>
> .
>
> chan_sip.c line 1453 (sip_alloc): Got it eventually...
> chan_sip.c line 4980 (do_monitor): Error obtaining mutex: Device or resource
> busy
> chan_sip.c line 4980 (do_monitor): Got it eventually...
> chan_sip.c line 948 (sip_hangup): Error obtaining mutex: Device or resource
> busy
> channel.c line 370 (ast_queue_frame): Error obtaining mutex: Device or
> resource busy
> chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or
> resource busy
> chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or
> resource busy
> chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or
> resource busy
> chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or
> resource busy
> chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or
> resource busy
> chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or
> resource busy
> chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or
> resource busy
> chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or
> resource busy
> chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or
> resource busy
> chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or
> resource busy
> chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or
> resource busy
> chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or
> resource busy
> chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or
> resource busy
> chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or
> resource busy
>
> Thank you.
>
> Alex Zarubin
> Webley Systems, Inc.
>
>
>

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Re: [Asterisk-Users] SIP Registration

[Martin Pycko]

sip show registry is when asterisk registers with some gateway.
you want to look at sip show peers or sip show users.

regards
Martin

On Thu, 31 Jul 2003, Steve Woolley wrote:

> I am trying to get SIP registrations to work within Asterisk. From my
> snom 200 phone (and on my SJPhone soft client) I can dial via extension.
> Example:
>
> To Dial extension 1110 on my asterisk1 server:
>
> I can simply enter SIP:[EMAIL PROTECTED] and the call goes through just
> like it should.
>
> As I understand it (and I probably don't), once my SIP device has
> established communication with the asterisk server, it registers the
> device name (in the sip registry) and thus I can dial the phone by
> entering:
>
> SIP:[EMAIL PROTECTED]
>
> (providing of course snom1 is the context for my sip phone in sip.conf)
>
> In fact I do see the following on the sip console when I make a call
> from snom1:
>
> asterisk1*CLI>
> -- Registered SIP 'snom1' at 172.16.14.11 port 5060 expires 3600
> -- Executing Macro("SIP/snom1-a17d", "oneline|Zap/4") in new stack
> -- Executing Dial("SIP/snom1-a17d", "Zap/4|20") in new stack
> -- Called 4
> -- Zap/4-1 is ringing
> -- Zap/4-1 is ringing
> -- Zap/4-1 is ringing
> -- Zap/4-1 is ringing
>
> I haven't found much documentation on sip registration in asterisk, but
> I kind of assumed that entering "sip show registry" on the console would
> show me the registrations, but only the following is returned by this
> command:
>
>  asterisk1*CLI> sip show registry
> Host  Username Refresh State
>
>
> Anyone have any ideas?
>
>
> --
> Steve Woolley
> ADS Telecom, Inc.
> 59 Skyline Drive
> Suite 1250
> Lake Mary, FL  32746
> (407)682-6226 x1110
> http://www.adstelecom.com
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Re: [Asterisk-Users] 'System' application exit with error even ifit performs the job as expected - partially solved

[Martin Pycko]

System should return with 0 when it's successfull. You have to have
something wrong with your system. Read "man system"

Martin

On Thu, 31 Jul 2003, Dan wrote:

> Hi Martin,
>
> I have modified the 'app_system.c' file like that and then recompile
> asterisk:
>
> /* Do our thing here */
> res = system((char *)data);
> //  if (res < 0) {
> //  ast_log(LOG_WARNING, "Unable to execute '%s'\n", (char
> *)data);
> //  res = -1;
> //  } else if (res == 127) {
> //  ast_log(LOG_WARNING, "Unable to execute '%s'\n", (char
> *)data);
> //  res = -1;
> //  } else {
> //  if (res && ast_exists_extension(chan, chan->context,
> chan->exten, chan->priority + 101, chan->callerid))
> //  chan->priority+=100;
> //  res = 0;
> //  }
> res = 0;
> LOCAL_USER_REMOVE(u);
> return res;
>
>
> Now everything work as expected.
> It seems to be a problem with the value returned by the 'system((char
> *)data)' function which is -1 even if the command is executed successfully.
>
> There is any reason to exit the System() application with -1 if the command
> is unable to execute? Maybe a parameter somewhere can prevent this type of
> behaviour.
>
> BR,
> Dan
>
>
> - Original Message -
> From: "Martin Pycko" <[EMAIL PROTECTED]>
> To: "Asterisk Users" <[EMAIL PROTECTED]>
> Sent: Thursday, July 31, 2003 7:49 PM
> Subject: Re: [Asterisk-Users] 'System' application exit with error even if
> it performs the job as expected
>
>
> > Try to do the same in shell. Does it work ?
> >
> > Martin
> >
> > On Thu, 31 Jul 2003, Dan wrote:
> >
> > > Hi,
> > >
> > > When I try to run the command wmix to mix two WAV files recorded by the
> > > Monitor application I get the following warning in the console and the
> macro
> > > exit at that point.
> > > Running the command from a standard system console it works. More, even
> from
> > > this macro it works and produce a valid mixed file, but still get that
> error
> > > and the macro cannot continue.
> > >
> > > Why?
> > > I have tried even with a simple wmix without any parameter and I get the
> > > same error.
> > >
> > > .
> > > -- Executing System("SIP/103-b7c0", "/usr/local/bin/wmix
> > > /var/spool/asterisk/monitor/31072003-19:08:11-103-in.wav
> > > /var/spool/asterisk/monitor/31072003-19:08:11-103-out.wav >
> > > /var/spool/asterisk/monitor/31072003-19:08:11-103.wav") in new stack
> > > WARNING[1200825920]: File app_system.c, Line 57 (skel_exec): Unable to
> > > execute '/usr/local/bin/wmix
> > > /var/spool/asterisk/monitor/31072003-19:08:11-103-in.wav
> > > /var/spool/asterisk/monitor/31072003-19:08:11-103-out.wav >
> > > /var/spool/asterisk/monitor/31072003-19:08:11-103.wav'
> > >   == Spawn extension (macro-record-cleanup, s, 3) exited non-zero on
> > > 'SIP/103-b7c0' in macro 'record-cleanup'
> > >   == Spawn extension (fullaccess, h, 1) exited non-zero on
> 'SIP/103-b7c0'
> > >
> > >
> > > Thanks,
> > > Dan
> > >
> > >
> > > ___
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> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> > ___
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> >
> >
>
>
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Re: [Asterisk-Users] 'System' application exit with error even ifit performs the job as expected

[Martin Pycko]

One thing is sure: the system should return with 0 if it's successful.
Read "man system"

regards
Martin

On Thu, 31 Jul 2003, Dan wrote:

> Something even more interesting.
> I have tried to execute the command 'ls' in the following line:
> ...
> exten => s,3,System(ls)
> ...
>
> And this is the result from the console:
>
> -- Executing System("SIP/103-2259", "ls") in new stack
> adsi.confasterisk.conf iax.conf  modem.conf
> oss.conf  privacy.conf~  tcpdmp.log
> adtranvofr.conf  cdr_mysql.confindications.conf  modules.conf
> oss.conf~ queues.conftelcordia-1.adsi
> agents.conf  enum.conf logger.conf   modules.conf~
> parking.conf  rpt.conf   voicemail.conf
> alsa.confextensions.conf   manager.conf  musiconhold.conf
> phone.confrtp.conf   vpb.conf
> alsa.conf~   extensions.conf~  meetme.conf   musiconhold.conf~
> phone.conf~   sip.conf   zapata.conf
> asterisk.adsifestival.conf mgcp.conf oh323.conf
> privacy.conf  sip.conf~
> WARNING[1200825920]: File app_system.c, Line 57 (skel_exec): Unable to
> execute 'ls'
>   == Spawn extension (macro-record-cleanup, s, 3) exited non-zero on
> 'SIP/103-2259' in macro 'record-cleanup'
>   == Spawn extension (fullaccess, h, 1) exited non-zero on 'SIP/103-2259'
>
>
> As you can see, the command was executed successfully, but still that
> warning who make the macro to exit .
>
> Dan
>
>
> - Original Message -
> From: "Dan" <[EMAIL PROTECTED]>
> To: "Asterisk Users" <[EMAIL PROTECTED]>
> Sent: Thursday, July 31, 2003 7:23 PM
> Subject: [Asterisk-Users] 'System' application exit with error even if it
> performs the job as expected
>
>
> > Hi,
> >
> > When I try to run the command wmix to mix two WAV files recorded by the
> > Monitor application I get the following warning in the console and the
> macro
> > exit at that point.
> > Running the command from a standard system console it works. More, even
> from
> > this macro it works and produce a valid mixed file, but still get that
> error
> > and the macro cannot continue.
> >
> > Why?
> > I have tried even with a simple wmix without any parameter and I get the
> > same error.
> >
> > .
> > -- Executing System("SIP/103-b7c0", "/usr/local/bin/wmix
> > /var/spool/asterisk/monitor/31072003-19:08:11-103-in.wav
> > /var/spool/asterisk/monitor/31072003-19:08:11-103-out.wav >
> > /var/spool/asterisk/monitor/31072003-19:08:11-103.wav") in new stack
> > WARNING[1200825920]: File app_system.c, Line 57 (skel_exec): Unable to
> > execute '/usr/local/bin/wmix
> > /var/spool/asterisk/monitor/31072003-19:08:11-103-in.wav
> > /var/spool/asterisk/monitor/31072003-19:08:11-103-out.wav >
> > /var/spool/asterisk/monitor/31072003-19:08:11-103.wav'
> >   == Spawn extension (macro-record-cleanup, s, 3) exited non-zero on
> > 'SIP/103-b7c0' in macro 'record-cleanup'
> >   == Spawn extension (fullaccess, h, 1) exited non-zero on 'SIP/103-b7c0'
> >
> >
> > Thanks,
> > Dan
> >
> >
> > ___
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> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
>
>
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Re: [Asterisk-Users] 'System' application exit with error even ifit performs the job as expected

[Martin Pycko]

Try to do the same in shell. Does it work ?

Martin

On Thu, 31 Jul 2003, Dan wrote:

> Hi,
>
> When I try to run the command wmix to mix two WAV files recorded by the
> Monitor application I get the following warning in the console and the macro
> exit at that point.
> Running the command from a standard system console it works. More, even from
> this macro it works and produce a valid mixed file, but still get that error
> and the macro cannot continue.
>
> Why?
> I have tried even with a simple wmix without any parameter and I get the
> same error.
>
> .
> -- Executing System("SIP/103-b7c0", "/usr/local/bin/wmix
> /var/spool/asterisk/monitor/31072003-19:08:11-103-in.wav
> /var/spool/asterisk/monitor/31072003-19:08:11-103-out.wav >
> /var/spool/asterisk/monitor/31072003-19:08:11-103.wav") in new stack
> WARNING[1200825920]: File app_system.c, Line 57 (skel_exec): Unable to
> execute '/usr/local/bin/wmix
> /var/spool/asterisk/monitor/31072003-19:08:11-103-in.wav
> /var/spool/asterisk/monitor/31072003-19:08:11-103-out.wav >
> /var/spool/asterisk/monitor/31072003-19:08:11-103.wav'
>   == Spawn extension (macro-record-cleanup, s, 3) exited non-zero on
> 'SIP/103-b7c0' in macro 'record-cleanup'
>   == Spawn extension (fullaccess, h, 1) exited non-zero on 'SIP/103-b7c0'
>
>
> Thanks,
> Dan
>
>
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Re: [Asterisk-Users] TE410P startup

[Martin Pycko]

It's fixed now

On Sun, 27 Jul 2003, Michael Bielicki wrote:

> we have now perfect results with yesterdays cvs and the te410p
> todays cvs allways thinks that immediate is set to yes in zapata.conf. weird
> ...
>
> cheers
> Michael
>
> On Sunday 27 July 2003 7:12 pm, Mark Spencer wrote:
> > > I put a TE410P card in a machine (a Tyan 2665 with 2x2.4GHz Xeons). A
> > > red flashing light circles around the 4 RJ48C sockets. I load the
> > > wct4xxp driver, and the flashing light stops. Whether I connect an E1
> > > signal or not, no lights are shown, and no alarms are reports in the
> > > /proc/zaptel/XXX files.  What is supposed to happen? I expected all the
> > > ports to show a continuous red LED until I plugged in an E1, and then to
> > > go green. That is what the other Digium cards do.
> >
> > First, if you want to run in E1 mode you either need to put jumpers on the
> > T1/E1 select jumpers located just south-west of the middle of the board
> > (and labeled).  Place a jumper on each span you want to be E1.
> >
> > Then you'll need to edit your /etc/zaptel.conf and make sure you have E1
> > spans defined.
> >
> > Finally, if the lights don't go red, then you might need to manually run
> > ztcfg.
> >
> > If all else fails, be *sure* you're running latest CVS :)
> >
> > Mark
> >
> > ___
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Re: [Asterisk-Users] Instant hangup on busy Zap channel.

[Martin Pycko]

Do 'iax2 debug' to see more.

Martin

On Fri, 25 Jul 2003, Richard Scobie wrote:

> A call is placed via IAX2 from one asterisk to another, to a TDM400
> channel whose extensions.conf entry is
>
> exten => 502,1,Dial(${COLIN})
> exten => 502,2,Congestion
>
> If  this channel is already busy when called, the call is instantly
> hungup, without the caller hearing the congestion tone.
>
> The log from the callers asterisk shows:
>
>  -- Executing Dial("Zap/1-1", "IAX2/192.168.3.223/502|30") in new stack
> -- Called 192.168.3.223/502
> -- Call accepted by 192.168.3.223 (format 4)
> -- Format for call is 4
> -- IAX2[pbxak]/2 is circuit-busy
>   == Everyone is busy at this time
> -- Hungup 'IAX2[pbxak]/2'
> -- Executing Hangup("Zap/1-1", "") in new stack
>   == Spawn extension (incoming, 502, 2) exited non-zero on 'Zap/1-1'
> -- Hungup 'Zap/1-1'
>
> Is this expected behavior, or have I missed something in the configuration?
>
> Thanks,
>
> Richard
>
>
>
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Re: [Asterisk-Users] Adtran TSU 600E

[Martin Pycko]

Sure, you just have to have the mailbox keyword before you define each
"channel => a"

You should have a stutter tone & MWI

regards
Martin

On Thu, 24 Jul 2003, Jerk Face wrote:

> Does the T100P support message waiting on an Adtran TSU 600E with FXS cards
> installed?
> So basically:
>
> Asterisk w/T100P  -> Adtran TSU 600E -> Analog phone
>
> Will I be able to receive the stutter dial tone on the analog phone?
>
> Thank you for your time
>
> _
> The new MSN 8: smart spam protection and 2 months FREE*
> http://join.msn.com/?page=features/junkmail
>
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Re: [Asterisk-Users] Problems with g729

[Martin Pycko]

Try the new_codec_binary/codec_g729b.so from the digium ftp site.

regards
Martin

On Wed, 23 Jul 2003, Dan Fernandez wrote:

> I am having some problems with g729 with SIP and ZAP channels.
>
> 1)
> I have two g729 licences. Very frequetnly (I don´t know what triggers the error)  I 
> get the following warnings and error when I try to place a call via SIP to my X100P. 
> The only way to get out of this is through a restart of *. When the error ocurrs 
> there are no other calls in place. Any ideas?
>
>
> Error Opening channel:2 not available, see va_g729_init_global(..) 
> WARNING[71694]:File codec_g729b.c line 102 (g729lin_new): No available g729b 
> resource for channel 2
> WARNING:[71694] File translate.c Line 111 (ast_translator_build_path):Failed to 
> build translator path from 8 to 6 Zap1-1 answered SIP/105-ce3c
> WARNING[71694]: File chan_zap.c Line 3367 (zt_write):Cannot handle frames in 256 
> format
> Hangup Zap/1-1
>
>
> 2)
>  have discovered a problem when using g729 under the following setup:
>
> SIP call between a Budgetone 102 and ATA 186  (configured without silence 
> suppresion). Both ends have a ADSL 64kbps. Both ends are behind Linksys routers. The 
> pings between them are aprox. 100ms. No other local users on each end.  * is being 
> hosted on a PIII,128MB. No other calls are being handled at the time of the test.
>
>  Basically, after a few minutes, with g729, both ends consistently start getting 
> delays up to a point where it becomes almost unbearable to speak.  If we switch to 
> g723 the problem goes away.
>
> ANy ideas what´s going on?
>
>

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Re: [Asterisk-Users] Asterisk Compile error: make: *** [subdirs]Error 1

[Martin Pycko]

It's fixed now. Aparently Mark forgot to compile before commiting.

Martin

On Tue, 22 Jul 2003, Ashley Jones wrote:

> Hi all,
>
> I'm trying to compile Asterisk (checked out of CVS at aprox 3pm PST 7.22.03)
> on a 2U Compaq running Redhat 8 and 1 TCM400P(w/ 2 hot ports) and 2 X100P's.
> The error I get after running "make install"(in /usr/src/asterisk) is:
>
> app_dial.c: In function `wait_for_answer':
> app_dial.c:232: parse error before "o"
> app_dial.c:242: parse error before "o"
> app_dial.c:285: parse error before "o"
> make[1]: *** [app_dial.o] Error 1
> make[1]: Leaving directory `/usr/src/asterisk/apps'
> make: *** [subdirs] Error 1
>
> I was following the quick start instructions and this came up on the last
> half of the last line:
>
> cd zaptel
> make clean ; make install
> cd ../zapata
> make clean ; make install
> cd ../lipbri
> make clean ; make install
> cd ../asterisk
> make clean ; make install
>
> any thoughts?
>
> -adj
>
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Re: [Asterisk-Users] busydetect and random hangups

[Martin Pycko]

use BUSYDETECT_MARTIN in asteirsk/Makefile

Martin

On 22 Jul 2003, Brancaleoni Matteo wrote:

> increase busycount in zapata.conf
> busycount=6 is ok for me.
> the default is 3 , I think, and sometimes
> it hangsup on speaking (or some other moh ;) )
>
> Matteo.
>
> Il mar, 2003-07-22 alle 22:11, Paulo Mannheimer ha scritto:
> > Hi,
> >
> >
> >
> > Iÿm having random hangup problems with zap channels.
> >
> >
> >
> > If I turn busydetect off in Zapata.conf, * fails completely to detect
> > a user hangup in the middle of a script.
> >
> >
> >
> > On the other hand, if I turn it on, everything works much better, but
> > long calls tend to be hung up without a motive.
> >
> >
> >
> > Any other parameter that I can try? Any #define that I can tweak and
> > recompile?
> >
> >
> >
> >  Will callprogress be of any help, as Iÿm outside the US?
> >
> >
> >
> > Thanks!
> >
> >
> >
> > PauloHM
> >
> >
> --
> Brancaleoni Matteo <[EMAIL PROTECTED]>
> Espia - Emmegi Srl
>
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Re: [Asterisk-Users] enabling dtmf detection on zap channel?

[Martin Pycko]

Use application Background.

Martin

On 22 Jul 2003, Thilo Salmon wrote:

> Hi,
>
> is there a way to enable dtmf detection on zap channels? I am trying to
> pickup, play a ringtone and the dial out. I.e.
>
> exten => s,1,Wait,1
> exten => s,1,Answer
> exten => s,2,Playtones(dial)
> exten => s,3,DigitTimeout,5
> exten => s,4,ResponseTimeout,10
> exten => _X,1,StopPlaytones
> exten => _X,2,Dial,Zap/g8/BYEXTENSION|10
>
> I don't quite recall what I did last, but I remember this used to work.
> My current installation of asterisk does not detect dtmf tones as I can
> easily verify with debugging turned on. Just to be sure I also dialed in
> from different phones, did a fresh checkout of zaptel, libpri and
> asterisk and wrote my configuration from scratch. None of that made
> asterisk detect dtmf, though.
>
> Can anybody suggest what to try next?
>
> Thilo
>
>
>
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Re: [Asterisk-Users] g729 + oh323

[Martin Pycko]

But you need to edit h323/Makefile and uncomment -DWANT_G729

Martin

On Mon, 21 Jul 2003, Jeremy McNamara wrote:

> You should run chan_h323.  It is distributed with Asterisk and works
> with G.729 and any other codec asterisk supports TODAY.   There is no
> need to run a 3rd party driver.
>
> Jeremy McNamara
>
>
> Chee Foong wrote:
>
> >Thanks for the info mate.
> >Looking forward to the bug fix release. :)
> >
> >cheers
> >
> >Foong
> >
> >- Original Message -
> >From: "Michael Manousos" <[EMAIL PROTECTED]>
> >To: <[EMAIL PROTECTED]>
> >Sent: Tuesday, July 22, 2003 7:02 PM
> >Subject: Re: [Asterisk-Users] g729 + oh323
> >
> >
> >
> >
> >>Chee Foong wrote:
> >>
> >>
> >>>Hello,
> >>>
> >>>Is Oh323 supports g729 codec from digium? I saw an g729 option in the
> >>>oh323.conf but I have also read some post in the mailing list saying
> >>>
> >>>
> >that
> >
> >
> >>>oh323 doesn't support g729 codec from digium.
> >>>
> >>>
> >>>
> >>asterisk-oh323 had some problems with G.729 formats.
> >>I have fixed them and soon I 'll make a new bug-fix
> >>release. But I have't tested it with digium's G.729
> >>codec, just with some Cisco boxes.
> >>
> >>
> >>
> >>>Foong
> >>>
> >>>
> >>>
> >>
> >>Michael.
> >>
> >>
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> >>
> >>
> >>
> >
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> >
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>
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Re: [Asterisk-Users] Dynamically setting up/tearing down extensions

[Martin Pycko]

Yes, you can contact over the manager interface (you need to setup a
user/pass in /etc/asterisk/manager.conf). I've sent a short perl script
how to do that some time ago.

Now notice that "extensions reload" only renews extensions without
touching other modules.

regards
Martin

On Mon, 21 Jul 2003, Steve Sobol wrote:

> At 02:06 PM 7/21/2003 -0500, you wrote:
> >One can use the retrieve_extensions_from_mysql.pl script and then issue a
> >"extensions reload" command to asterisk. The pending calls are unaffected
> >and the final substitution of the new dialplan is done in a very short
> >time.
>
> I want to explore truly dynamic extensions as a long-term project, but this
> might
> be an excellent short-term solution.
>
> Can the reload be done without being root?
>
>
>
>   --
> Steven J. Sobol, Geek In Charge, JustThe.net
> POTS: Toll Free from anywhere in the USA or Canada, 888.480.4NET (4638)
> HTTP: www.JustTheNetLLC.com
> MAIL: 5686 Davis Drive, Mentor on the Lake, OH 44060-2752
>
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Re: [Asterisk-Users] anyone with X100P & Callerid working outsideUS ?

[Martin Pycko]

I don't know yet. However you should be able to hear some wierd signal
that is callerid codec in FSK mode.

regards
Martin

On Mon, 21 Jul 2003, Tamas Levente wrote:

> How can I use ztmonitor to figure out the caller id sent by the telco?
> Because it is not working for me in Chicago.
>
> - Original Message -
> From: "Martin Pycko" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Monday, July 21, 2003 9:03 PM
> Subject: Re: [Asterisk-Users] anyone with X100P & Callerid working outside
> US ?
>
>
> > It's possible that your telco first transmits the DID (your number) and
> > then later on the callerid ...
> >
> > Did you "listen" for it with ztmonitor ?  If my suspicion is right ?
> >
> > regards
> > Martin
> >
> > On Mon, 21 Jul 2003, Dan wrote:
> >
> > > Hi Martin,
> > >
> > > For me it just display my own PSTN number, extracted as caller id from
> the
> > > PSTN line.
> > > I am located in Romania.
> > >
> > > Best regards,
> > > Dan
> > > P.S. Some analog phones with internal caller id displays the same
> number,
> > > but others (especially some Siemens ones) display the correct caller id.
> > > I think that the X100P card does not extract the correct part of the
> > > callerid information.
> > >
> > >
> > > - Original Message -
> > > From: "Martin Pycko" <[EMAIL PROTECTED]>
> > > To: <[EMAIL PROTECTED]>
> > > Sent: Monday, July 21, 2003 8:25 PM
> > > Subject: [Asterisk-Users] anyone with X100P & Callerid working outside
> US ?
> > >
> > >
> > > > I'm just curious if anyone has the X100P & Callerid receiving working
> > > > outside US.
> > > >
> > > > Replies are appreciated. Also if it's not working for you in a certain
> > > > coutry you can respond too.
> > > >
> > > > regards
> > > > Martin
> > > >
> > > > ___
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> > > >
> > > >
> > >
> > >
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> > >
> >
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>
>
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Re: [Asterisk-Users] Dynamically setting up/tearing down extensions

[Martin Pycko]

One can use the retrieve_extensions_from_mysql.pl script and then issue a
"extensions reload" command to asterisk. The pending calls are unaffected
and the final substitution of the new dialplan is done in a very short
time.

regards
Martin

On Tue, 22 Jul 2003, Jeremy McNamara wrote:

>
> DynExtenDB is not even close to being the proper way to achieve dynamic
> extensions.
>
> Karl (klasstek) and myself (mainly Karl) has spent a few clock cycles
> figuring out how to make dynamic extensions happen, but we had no real
> motivation to finish the task.
>
> Find either one of us on IRC or search the mailing list archives.
>
>
>
> Jeremy McNamara
>
>
> Steven J. Sobol wrote:
>
> >Hello, * newbie here,
> >
> >I'm designing a setup that is to eventually be used in a production
> >virtual PBX/VoIP service.
> >
> >Customers need to be able to change their setups over the web - I want
> >them to be able to do simple things like setting up call forwarding, as
> >well as more intricate stuff that will require me to re-generate their
> >dialplans.
> >
> >Administration of the service is to be web-based.
> >
> >I'm looking at DynExtenDB (and have played with it). I love that it reads
> >the dialplans out of a MySQL database - that is a critical issue for me.
> >But it has some issues.
> >
> >I have a test dialplan with one call to Playback() - just plays a wav file
> >then exits. When DynExtenDB() is called, it adds one extra step that calls
> >DynExtenDB_Free()...
> >
> >--If I let the wav file play to the end, DynExtenDB_Free() is called
> >properly. If I hang up prematurely, it isn't, and it also isn't called if
> >I set the dialplan to dial out (for example, to forward the call to my
> >cell phone).
> >
> >--If DynExtenDB_Free() *is* called properly, and I then make another call,
> >DynExtenDB() doesn't seem to be called again.
> >
> >--I'm not sure that setting up a dialplan for extension 'h' is a good
> >idea. What if I call, and then someone else calls and I hang up in the
> >middle of the call?
> >
> >I am ready and willing to make changes to the source to DynExtenDB. In
> >fact, I'd like to get it to a point where it could be used in a production
> >environment. But I have a lot of questions before I can do that.
> >
> >BTW, I have looked in the archives, and it's been suggested that maybe AGI
> >is a better way to handle this sort of thing - but wouldn't the same
> >issues still exist??
> >
> >Thanks
> >   SJS
> >
> >
> >
>
>
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Re: [Asterisk-Users] anyone with X100P & Callerid working outsideUS ?

[Martin Pycko]

It's possible that your telco first transmits the DID (your number) and
then later on the callerid ...

Did you "listen" for it with ztmonitor ?  If my suspicion is right ?

regards
Martin

On Mon, 21 Jul 2003, Dan wrote:

> Hi Martin,
>
> For me it just display my own PSTN number, extracted as caller id from the
> PSTN line.
> I am located in Romania.
>
> Best regards,
> Dan
> P.S. Some analog phones with internal caller id displays the same number,
> but others (especially some Siemens ones) display the correct caller id.
> I think that the X100P card does not extract the correct part of the
> callerid information.
>
>
> - Original Message -
> From: "Martin Pycko" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Monday, July 21, 2003 8:25 PM
> Subject: [Asterisk-Users] anyone with X100P & Callerid working outside US ?
>
>
> > I'm just curious if anyone has the X100P & Callerid receiving working
> > outside US.
> >
> > Replies are appreciated. Also if it's not working for you in a certain
> > coutry you can respond too.
> >
> > regards
> > Martin
> >
> > ___
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> >
> >
>
>
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[Asterisk-Users] anyone with X100P & Callerid working outside US ?

[Martin Pycko]

I'm just curious if anyone has the X100P & Callerid receiving working
outside US.

Replies are appreciated. Also if it's not working for you in a certain
coutry you can respond too.

regards
Martin

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Re: [Asterisk-Users] E911 and asterisk

[Martin Pycko]

Isn't that enough to set up a proper Caller ID NAME ?

Martin

On Mon, 21 Jul 2003, Alex Lopez wrote:

> I have a client that would like to use asterisk to link their multiples locations 
> together.  However, if a person in the remote office dials 911, How can the 911 
> operator determine WHERE the emergency is??  Since all calss would be going out of 
> the PRI in the main location, the police/fire trucks will show up at our COLO!!
>
> I know that there are some that are doing this multi site setup, how did they handle 
> 911 services???  For now I am quoting a one port FXO card to be placed in each 
> location, that will in turn connect to a POTS line. However, even though we can use 
> it for the alarm system and it is a kind of insurance I would like to do away with 
> it!
>
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Re: [Asterisk-Users] Asterisk crashes when trying to load G.729module.

[Martin Pycko]

Try to install the new codec code that is available in

ftp://ftp.digium.com/pub/telephony/asterisk/g729/new_codec_binary/codec_g729b.so
place it in /usr/lib/asterisk/modules and restart asterisk (or try to
start it).

There is also a new command available "g.729 show license usage" and a few
fixes to the code.

Write back about the results.

regards
Martin

On Sun, 20 Jul 2003, Anton Tinchev wrote:

> Before few days i bought few g.729 licenses.
> When i try to load the codec, asterisk crahses.
> I tried with and without oh323 module, same result:
> --
> Jul 20 07:06:49 WARNING[589851]: File codec_g729b.c, Line 413 (load_module): Unable 
> to initialize va stuff: -1
> --
>
> Here the ldd result:
> --
> [EMAIL PROTECTED]:~$ ldd -v /usr/lib/asterisk/modules/codec_g729b.so
> libc.so.6 => /lib/libc.so.6 (0x40039000)
> /lib/ld-linux.so.2 => /lib/ld-linux.so.2 (0x8000)
>
> Version information:
> /usr/lib/asterisk/modules/codec_g729b.so:
> libc.so.6 (GLIBC_2.1.3) => /lib/libc.so.6
> libc.so.6 (GLIBC_2.2) => /lib/libc.so.6
> libc.so.6 (GLIBC_2.1) => /lib/libc.so.6
> libc.so.6 (GLIBC_2.0) => /lib/libc.so.6
> /lib/libc.so.6:
> ld-linux.so.2 (GLIBC_2.1) => /lib/ld-linux.so.2
> ld-linux.so.2 (GLIBC_2.0) => /lib/ld-linux.so.2
> ld-linux.so.2 (GLIBC_PRIVATE) => /lib/ld-linux.so.2
> ---
>
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Re: [Asterisk-Users] Asterisk NOOB

[Martin Pycko]

You need to have installed the dev version of those packages

libncurses-dev , etc...

Martin

On Fri, 18 Jul 2003, Kyle Hagan wrote:

> Tried that. It says everything already installed.
>
>
> Kyle
>
> - Original Message -
> From: "Wade Weppler" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Friday, July 18, 2003 3:04 PM
> Subject: RE: [Asterisk-Users] Asterisk NOOB
>
>
> > As per your error, you need to install termcap.
> >
> > In Mandrake:
> >
> > urpmi termcap
> >
> > Should work...
> >
> > There might be other dependencies as you try to compile.  Install them
> (and
> > their -devel counterparts if necessary) as you find them.
> >
> > -wade
> >
> >
> > > -Original Message-
> > > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > > [EMAIL PROTECTED] On Behalf Of Kyle Hagan
> > > Sent: Friday, July 18, 2003 5:58 PM
> > > To: [EMAIL PROTECTED]
> > > Subject: [Asterisk-Users] Asterisk NOOB
> > >
> > > Im new to asterisk and just ordered the Dev Kit from Digium.
> > >
> > > Im vaguely familiar with Linux. So bare with me.  (But learing quickly)
> > >
> > > Im running Mandrake 9.1.
> > >
> > >
> > > I used CVS to dload the system.
> > >
> > > But when I do Make Install with asterisk i get:
> > >
> > > checking for tgetent in -ltermcap... no
> > > checking for tgetent in -ltinfo... no
> > > checking for tgetent in -lcurses... no
> > > checking for tgetent in -lncurses... no
> > > configure: error: termcap support not found.
> > > make:*** [editline/config.h] Error 1
> > >
> > >
> > > I have on the web site it appears I should be able to run it without the
> > > wildcard cards as long as I have a sound card,which I do and works.
> > >
> > > I have installed openssl and readline.
> > >
> > > anyhelp would be appreciated.
> > >
> > >
> > > Kyle
> > >
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Re: [Asterisk-Users] Chan_H323, G729 (minor problem)

[Martin Pycko]

dtmfmode=rfc2833

or

dtmfmode=info

try that instead

Martin

On Thu, 17 Jul 2003, Dave Alan Caruana wrote:

> ok ...
> I removed the dtmfmode=inband
> from the h323.conf file which resulted in the error messages vanishing ..
> ya I thought ...
>
> alas DTMF tones sent to an IVR at the other end of the connection
> do not work either!!!
>
> My incoming calls are coming from PSTN lines through an E1
> so DTMF must be inline .. THe (thousands of) error messages
> aren't really a problem, just annoying.
>
> Dave
>
> - Original Message -
> From: "Martin Pycko" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Tuesday, July 15, 2003 4:28 PM
> Subject: Re: [Asterisk-Users] Chan_H323, G729 (minor problem)
>
>
> > You're trying to detect inband dtmfs from the codec stream.
> >
> > Martin
> >
> > On Tue, 15 Jul 2003, Dave Alan Caruana wrote:
> >
> > > hi ..
> > >
> > > I have finally managed to get Chan_H323 & G729 working
> > > flawlessly, thanks to some help from Jerry McNamara.
> > > For those out there who are stuck with the same problem
> > > the procedure is :
> > > 1. install on RedHat 8.0 and nothing else (RH9 doesn't work!)
> > > 2. Install asterisk, zaptel etc. the normal way
> > > 3. Compile Pwlib & oH323 with versions taken from nufone's
> > > site (http://www.nufone.net/downloads) since the latest versions
> > > do not have support for G729. Remember to set the environment
> > > versions as described in the Readme files.
> > > 4. Modify the makefile of chan_h323 (which is in
> > > /usr/src/asterisk/channels/h323)
> > > to re-enable the G729 code.
> > > 5. in h323.conf put in "allow=g729"
> > > and you should have a working configuration ..
> > >
> > > now for my question ..
> > > during G729 calls I am getting repeatedly the message
> > >
> > > WARNING[311314]: File dsp.c, Line 1198 (ast_dsp_process): Unable to
> detect
> > > process 256 frames
> > >
> > > this scrolls up the screen at a very high rate of knots.. the call is
> > > unaffected and goes through normally.
> > > Is this something wrong? normal? can it be fixed/suppressed?
> > >
> > > cheers
> > > Dave
> > >
> > >
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> >
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>
>
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Re: [Asterisk-Users] random hangups

[Martin Pycko]

do you have in zapata.conf

busydetect=yes
or
callprogress=yes ?

Martin

On Thu, 17 Jul 2003, Paulo H. Mannheimer wrote:

> Hi ,
>
> I''m getting random hangups on zap channels with long calls. It seems that the
> hungup happens after 10 minutes or so. AbsoluteTimeout is set to 0. Any other
> thing I should be configuring?
>
> Thanks!
>
> PHM
>
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Re: [Asterisk-Users] Call Pickup

[Martin Pycko]

You need to have a pending call in the system (some extensions that is
ringing to test that). If you have 3 FXS ports try to place a call from
the first one to the 2nd and then instead of taking the 2nd off hook dial
*8 on the 3rd phone

Martin

On Thu, 17 Jul 2003, Jay Tyndall wrote:

>   Hi,
>
> I have been trying to workout how to use the call pickup.
>
> So Far, I have the following in zapata.conf
> [channels]
> signalling => fxo_ks
> context => local
> pickupgroup=1
> callgroup=1
> channel => 1-3
>
>
> When I dial *8# all I hear is busy tone.
>
> What have I missed?
>
> thanks
> Jay.
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[Asterisk-Users] Re: Alphanumerical digits

[Martin Pycko]

Your telco doesn't send you this IE

-- Processing IE 112 (Called Party Number)

Martin

On Tue, 15 Jul 2003, Cristi wrote:

> I see the following line into debug (pri debug span 1):
> 1. Progress Description: Calling equipment is non-ISDN. (3) ]
> 2.  Calling Number (len=14) [ Ext: 0  TON: Unknown Number Type (0)  NPI:
> Unknown Number Plan (0)
>
> Into a ISDN that is working I got this :
>
> -- Making new call for cr 23274
> -- Processing Q.931 Call Setup
> -- Processing IE 33 (Sending Complete)
> -- Processing IE 4 (Bearer Capability)
> -- Processing IE 24 (Channel Identification)
> -- Processing IE 30 (Progress Indicator)
> -- Processing IE 108 (Calling Party Number)
> -- Processing IE 112 (Called Party Number)
>  -- Extension '' in context 'outbound'
>
> Into this one I got only :
> -- Processing Q.931 Call Setup
> -- Processing IE 33 (Sending Complete)
> -- Processing IE 4 (Bearer Capability)
> -- Processing IE 24 (Channel Identification)
> -- Processing IE 30 (Progress Indicator)
> -- Processing IE 108 (Calling Party Number)
>
> What parameters they not send!
>
>
> The complet debugging is this:
> < Protocol Discriminator: Q.931 (8)  len=34
> < Call Ref: len= 2 (reference 162/0xA2) (Originator)
> < Message type: SETUP (5)
> < Sending Complete (len= 4)
> < Bearer Capability (len= 3) [ Ext: 1  Q.931 Std: 0  Info transfer
> capability: 3.1kHz audio (16)
> <  Ext: 1  Trans mode/rate: 64kbps,
> circuit-mode (16)
> <  Ext: 1  User information layer 1: A-Law (35)
> < Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive
> Dchan: 0
>  <   Ext: 1  Coding: 0   Number Specified   Channel
> Type: 3
> <   Ext: 1  Channel: 1 ]
> < Progress Indicator (len= 2) [ Ext: 1  Coding: CCITT (ITU) standard (0)
> 0: 0   Location: Public network serving the remote user (4)
> <   Ext: 1  Progress Description: Calling
> equipment is non-ISDN. (3) ]
> < Calling Number (len=14) [ Ext: 0  TON: Unknown Number Type (0)  NPI:
> Unknown Number Plan (0)
> <   Presentation: Unknown (1) '0788401422' ]
> -- Making new call for cr 162
> -- Processing Q.931 Call Setup
> -- Processing IE 33 (Sending Complete)
> -- Processing IE 4 (Bearer Capability)
> -- Processing IE 24 (Channel Identification)
> -- Processing IE 30 (Progress Indicator)
> -- Processing IE 108 (Calling Party Number)
>  > Protocol Discriminator: Q.931 (8)  len=14
>  > Call Ref: len= 2 (reference 32930/0x80A2) (Terminator)
>  > Message type: SETUP ACKNOWLEDGE (13)
>  > Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0,
> Exclusive Dchan: 0
>  >ChanSel: Reserved
>  >   Ext: 1  Coding: 0   Number Specified   Channel
> Type: 3
>  >   Ext: 1  Channel: 1 ]
>  > Progress Indicator (len= 2) [ Ext: 1  Coding: CCITT (ITU) standard
> (0) 0: 0   Location: Private network serving the local user (1)
>  >   Ext: 1  Progress Description: Called
> equipment is non-ISDN. (2) ]
> < Protocol Discriminator: Q.931 (8)  len=12
> < Call Ref: len= 2 (reference 162/0xA2) (Originator)
> < Message type: STATUS (125)
> < Cause (len= 2) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0
> Location: Public network serving the local user (2)
> <  Ext: 1  Cause: Protocol error, unspecified (111),
> class = Protocol Error (6) ]
> < Call State (len= 1) [ Ext: 0  Coding: CCITT (ITU) standard (0) Call
> state: Call Present (6)
> -- Processing IE 8 (Cause)
> -- Processing IE 20 (Call State)
> < Protocol Discriminator: Q.931 (8)  len=13
> < Call Ref: len= 2 (reference 162/0xA2) (Originator)
> < Message type: DISCONNECT (69)
> < Cause (len= 2) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0
> Location: Public network serving the local user (2)
> <  Ext: 1  Cause: Recover on timer expiry (102), class =
> Protocol Error (6) ]
> < Progress Indicator (len= 2) [ Ext: 1  Coding: CCITT (ITU) standard (0)
> 0: 0   Location: Public network serving the remote user (4)
> <   Ext: 1  Progress Description: Calling
> equipment is non-ISDN. (3) ]
> -- Processing IE 8 (Cause)
> -- Processing IE 30 (Progress Indicator)
> WARNING[9226]: File chan_zap.c, Line 5275 (pri_fixup): Call specified,
> but not found?
> WARNING[9226]: File chan_zap.c, Line 5830 (pri_dchannel): Hangup on bad
> channel 1
> < Protocol Discriminator: Q.931 (8)  len=9
> < Call Ref: len= 2 (reference 162/0xA2) (Originator)
> < Message type: RELEASE (77)
> < Cause (len= 2) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0
> Location: Pub
> lic network serving the local user (2)
> <  Ext: 1  Cause: Recover on timer expiry (102), class =
> Protoco
> l Error (6) ]
> -- Processing IE 8 (Cause)
>  > Protocol Discriminator: Q.931 (8)  len=5
>  > Call Ref: len= 2 (reference 32930/0x80A2) (

Re: [Asterisk-Users] Chan_H323, G729 (minor problem)

[Martin Pycko]

You're trying to detect inband dtmfs from the codec stream.

Martin

On Tue, 15 Jul 2003, Dave Alan Caruana wrote:

> hi ..
>
> I have finally managed to get Chan_H323 & G729 working
> flawlessly, thanks to some help from Jerry McNamara.
> For those out there who are stuck with the same problem
> the procedure is :
> 1. install on RedHat 8.0 and nothing else (RH9 doesn't work!)
> 2. Install asterisk, zaptel etc. the normal way
> 3. Compile Pwlib & oH323 with versions taken from nufone's
> site (http://www.nufone.net/downloads) since the latest versions
> do not have support for G729. Remember to set the environment
> versions as described in the Readme files.
> 4. Modify the makefile of chan_h323 (which is in
> /usr/src/asterisk/channels/h323)
> to re-enable the G729 code.
> 5. in h323.conf put in "allow=g729"
> and you should have a working configuration ..
>
> now for my question ..
> during G729 calls I am getting repeatedly the message
>
> WARNING[311314]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect
> process 256 frames
>
> this scrolls up the screen at a very high rate of knots.. the call is
> unaffected and goes through normally.
> Is this something wrong? normal? can it be fixed/suppressed?
>
> cheers
> Dave
>
>
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Re: [Asterisk-Users] Making Analog Phones Work

[Martin Pycko]

Check if the board is still getting interrupts.
"grep wcfxs /proc/interrupts; sleep 10; grep wcfxs /proc/interrupts"
should show two numbers that differ by ~1.

regards
Martin

On Tue, 15 Jul 2003, Jay Tyndall wrote:

>
> Hi,
>
> I have got my TDM400P working.(3 modules), asterisk dials Zap/1 and says
> "Ringing" but the analogue phone plugged in, does not ring, or does not
> have any tone when I pickup the handpiece.
>
> Here are by configs:
> zapata.conf:
> [channels]
> signalling => fxo_ks
> context=internal
> channel => 1-3
>
>
> zaptel.conf:
> fxoks=1-3
>
>
> Any ideas would be greatly appriciated Thanks
> Jay
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Re: [Asterisk-Users] Call Recording

[Martin Pycko]

Sure you just need to use Monitor and Changemonitor apps.
A little bit of scripting is a must though to get a unique id 
eg a current date in seconds. I'm not sure if asterisk has it already.

regards
Martin

On Fri, 11 Jul 2003, Erik Kendall wrote:

> Can Asterisk automatically record all calls to unique
> files, like voicemail does with the messages?
>
>
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Re: [Asterisk-Users] ISDN PRI E1 configuration with E100P

[Martin Pycko]

Look in zttool or "head /proc/zaptel/[1-5]
to see if the spans are in alarms. The leds on your boards might not lit
properly.

regards
Martin

On Fri, 11 Jul 2003 [EMAIL PROTECTED] wrote:

> yes, now i got the that problem solved, with 'immediate=yes', but now i've
> faced with another,
> When I connect the E1 to the cards, the LED lights does not change to green,
> in E100P, its blinking red (even when there is no E1 plugged, its blinking
> red)
> and in E400P, its solid red. (this was the same even without
> 'immediate=yes')
>
> When i start asterisk, the cards starts fine (as we see), givin the
> following output,
> *CLI>
>   == D-Channel on span 1 up
> -- B-channel 1 successfully restarted on span 1
> -- B-channel 2 successfully restarted on span 1
>. . .
> -- B-channel 31 successfully restarted on span 1
>
> after i put 'immediate=yes', i can even call from outside to the PRI E1, and
> get connected to asterisk, and listen to the prompts played by asterisk
>
> -- Accepting call from '062279955' to 's' on channel 1, span 1
> -- Executing Wait("Zap/1-1", "2") in new stack
> -- Executing Answer("Zap/1-1", "") in new stack
> -- Executing DigitTimeout("Zap/1-1", "5") in new stack
> -- Set Digit Timeout to 5
> -- Executing ResponseTimeout("Zap/1-1", "10") in new stack
> -- Set Response Timeout to 10
> -- Executing BackGround("Zap/1-1", "demo-congrats") in new stack
> -- Playing 'demo-congrats'
>
> But calling to outside from asterisk fails (with the following errors), we
> don't know whether this related to the above problem
>
> Executing Dial("SIP/802-8a26", "Zap/g2/0129063800|20|t") in new stack
> NOTICE[262160]: File app_dial.c, Line 481 (dial_exec): Unable to create
> channel of type 'Zap'
>   == Everyone is busy at this time
> -- Executing Congestion("SIP/802-8a26", "") in new stack
>   == Spawn extension (sip, 980129063800, 2) exited non-zero on
> 'SIP/802-8a26'
>
> Following are new zaptel and zapata conf files,
>
> zapata.conf
>
> [channels]
> transfer=yes
> echocancel=yes
> callprogress=yes
> immediate=yes
>
> ;E100p card
> switchtype=EuroISDN
> signalling=pri_cpe
> pridialplan=unknown
> context=inbound-pstn
> group=2
> channel => 1-15,17-31
>
> zaptel.conf
>
>
> #E100p card
> span=1,0,0,ccs,hdb3,crc4
> bchan=1-15,17-31
> dchan=16
>
> defaultzone=us
> loadzone=us
>
>
> Thank you,
>
> Surajee
>
> - Original Message -
>
> From: "Cristi" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Friday, July 11, 2003 10:39 PM
> Subject: Re: [Asterisk-Users] ISDN PRI E1 configuration with E100P
>
>
> > [EMAIL PROTECTED] wrote:
> >
> > > hi Everyone,
> > >
> > > We are configuring an ISDN PRI E1 with an E100P card, when you load
> > > the drivers, and starts the asterisk, cards also starts fine, givin
> > > following output,
> > >
> > > *CLI>
> > >   == D-Channel on span 1 up
> > > -- B-channel 1 successfully restarted on span 1
> > > -- B-channel 2 successfully restarted on span 1
> > >. . .
> > > -- B-channel 31 successfully restarted on span 1
> > >
> > > but, when we make a call to this E1 from outside, it gives the
> > > following error,
> > >
> > > WARNING[213006]: File chan_zap.c, Line 5248 (pri_fixup): Call
> > > specified, but not found?
> > > WARNING[213006]: File chan_zap.c, Line 5786 (pri_dchannel): Hangup on
> > > bad channel 1
> > > WARNING[213006]: File chan_zap.c, Line 5248 (pri_fixup): Call
> > > specified, but not found?
> > > WARNING[213006]: File chan_zap.c, Line 5786 (pri_dchannel): Hangup on
> > > bad channel 2
> > >
> > > does anybody hav an idea on this?
> > >
> > > our zaptel.conf is,
> > >
> > > #E100p card
> > > span=1,0,0,ccs,hdb3,crc4
> > > bchan=1-15,17-31
> > > dchan=16
> > >
> > > zapata.conf,
> > >
> > > ;E100p card
> > > switchtype=EuroISDN
> > > signalling=pri_cpe
> > > pridialplan=unknown
> > > context=incoming
> > > group = 2
> > > channel => 1-15,17-31
> > >
> > > Thanks inadvance,
> > >
> > > Surajee
> > >
> > >
> > > ---

Re: [Asterisk-Users] ISDN PRI E1 configuration with E100P

[Martin Pycko]

Unfortunatelly if your telco doesn't send you any DID along with the SETUP
message you need to have immediate=yes in zapata.conf for those channels.

regards
Martin

On Fri, 11 Jul 2003 [EMAIL PROTECTED] wrote:

> Very sorry about the previous mail,
> heres the mail again,
>
> hi Everyone,
>
> We are configuring an ISDN PRI E1 with an E100P card, when you load the drivers, and 
> starts the
> asterisk, cards also starts fine, givin following output,
>
> *CLI>
>   == D-Channel on span 1 up
> -- B-channel 1 successfully restarted on span 1
> -- B-channel 2 successfully restarted on span 1
>. . .
> -- B-channel 31 successfully restarted on span 1
>
> but, when we make a call to this E1 from outside, it gives the following error,
>
> WARNING[213006]: File chan_zap.c, Line 5248 (pri_fixup): Call specified, but not 
> found?
> WARNING[213006]: File chan_zap.c, Line 5786 (pri_dchannel): Hangup on bad channel 1
> WARNING[213006]: File chan_zap.c, Line 5248 (pri_fixup): Call specified, but not 
> found?
> WARNING[213006]: File chan_zap.c, Line 5786 (pri_dchannel): Hangup on bad channel 2
>
> does anybody hav an idea on this?
>
> our zaptel.conf is,
>
> #E100p card
> span=1,0,0,ccs,hdb3,crc4
> bchan=1-15,17-31
> dchan=16
>
> zapata.conf,
>
> ;E100p card
> switchtype=EuroISDN
> signalling=pri_cpe
> pridialplan=unknown
> context=incoming
> group = 2
> channel => 1-15,17-31
>
> Thanks inadvance,
>
> Surajee
>
>
>
> --This mail sent through OmniBIS.com--
>
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Re: [Asterisk-Users] T1 config for robbed-bit E&M AMI

[Martin Pycko]

T100P handles the E&M wink start signalling as well as D4AMI
framing/coding.

The config in /etc/zaptel.conf

span=1,0,0,d4,ami
e&m=1-24

in /etc/zapata.conf
[channels]
signalling=em_w
context=incoming
group = 1
channel => 1-24

But read more and have all the keywords/options that you need added to
config files.

regards
Martin

On Thu, 10 Jul 2003, mattf wrote:

> I have a couple of live T1s sitting around and they are not ISDN(like most
> of the people that are using Asterisk seem to be using), they are regular
> old 24 channel, robbed-bit, E&M wink start, D4AMI T1 circuits.
> Can I get these T1s to work with a T100P Digium card and asterisk?
>
> Searching through the lists and the documentation I haven't seen any
> examples of how to configure this kind of T1.
>
>  Is anyone currently using these kind of T1s on their Asterisk system?
>
> Any help would be appreciated.
>
> Thanks,
>
> MATT---
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Re: [Asterisk-Users] IAX2 Warning

[Martin Pycko]

IAX2 uses hardcoded 4569 port so it's not looking for port keyword.
Nothing to worry about.

Martin

On Thu, 10 Jul 2003, Richard Scobie wrote:

> When starting *, I get the following when the chan_iax2.so loads:
>
>  [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2))
>   == Manager registered action IAXpeers
>   == Parsing '/etc/asterisk/iax.conf': Found
> WARNING[16384]: File chan_iax2.c, Line 4980 (set_config): Ignoring port
> for now
>   == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2))
>   == Using TOS bits 16
>   == IAX Ready and Listening on 0.0.0.0 port 4569
>
> As I am not loading chan_iax.so and IAX is working, it is obviously not
> fatal, but after a reasonable amount of checking docs, archives etc. I
> am curious to know why it occurs and it is not using port 5036 as
> specified in iax.conf.
>
> Regards,
>
> Richard
>
>
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Re: [Asterisk-Users] How to modify dialed number?

[Martin Pycko]

You forgot about "_" in front of 0X

Martin

On Thu, 10 Jul 2003, BK [address only for mailing lists] wrote:

>
> On Thursday, July 10, 2003, at 06:37 AM, Petr Michálek wrote:
>
> > Is there simple way how to add prefix to dialed number?
> > I need change 0X. to 0X.
>
> how about this
>
> exten => 0X.,1,Dial(0{EXTEN:1})
>
> rgds
> bk
>
>
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Re: [Asterisk-Users] How to modify dialed number?

[Martin Pycko]

exten => _0X,1,Dial,Zap/g1/0${EXTEN:1}

Martin

On Wed, 9 Jul 2003, Petr Michálek wrote:

> Hi!
>
> Is there simple way how to add prefix to dialed number?
> I need change 0X. to 0X.
>
> Regards
>
> Petr Michálek
>
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Re: [Asterisk-Users] PRI with variable length numbers

[Martin Pycko]

Why do you see the problem with that ?

How would you use that functionality that analog channels have ?

With PRI you're going to always receive the full number that was dialed.
So since you have a limited number of DID's do the matching one for one
and it'll work.

regards
Martin

On Wed, 9 Jul 2003, The Traveller wrote:

> Hey all,
>
> I have an Asterisk-box with an E100P and a PRI (Euro-ISDN) coming
> into it from a Meridian-switch.  The incoming numbers on this PRI all start
> with the same digit and the last part of the dialled number is signalled to
> Asterisk digit by digit, until Asterisk signals that the number is
> complete and the call rings.
>
> All works well, unless I have 2 or more numbers which start with the same
> digits.  In that case, dialling will be signalled as complete as soon
> as the shortest of the numbers is dialed.  An example:
>
> exten => 31801,1, ...
> exten => 3180,1, ...
>
> In this case, dialling "3180" will immediately start ringing, while
> on similar setups, with TDM40B's and analog phones, Asterisk will
> wait for the duration of "digittimeout" for more digits, if it can't
> be absolutely certain that the dialled number is complete.  If I remove
> the "3180" in the above example, ringing will only start after "31801"
> is fully dialled.
>
>
> Relevant configs:
>
> /etc/zaptel.conf:
>
> bchan=1-15
> dchan=16
> bchan=17-31
>
> /etc/asterisk/zapata.conf:
>
> switchtype=euroisdn
> signalling=pri_cpe
> immediate=no
>
>
> Any ideas?
>
>
>
> Grtz,
>
>   Oliver
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Re: [Asterisk-Users] callerid= being ignored

[Martin Pycko]

At the moment asterisk can get the callerid from the "From: " field.
regards
Martin

On Thu, 10 Jul 2003, BK [address only for mailing lists] wrote:

>
> Hi
>
> I have defined my SIP phones like this ...
>
> [Sip1]
> username=gs1
> callerid= "Full name" <1001>
>
> etc etc
>
> Now, when I do this in a given extension
>
> exten => ,1,NoOp(${CALLERIDNUM})
>
> then I get "" as callerid and not "<1001>" as defined with callerid=
>
> Sure, I could set the usernames to their respective extensions, but I
> don't want to do that. I'd like to keep login names independent of
> extensions.
>
> Is there any fix for this so that the real callerid shows up?
>
> thanks in advance
> regards
> bk
>
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Re: [Asterisk-Users] Using multiple iconnecthere accounts

[Martin Pycko]

On your place I would check separately if you can use both accounts. I
think that one of your accounts in disabled ...

Martin

On Tue, 8 Jul 2003, Derek Beaumont wrote:

> First off, sorry for using a mail client without the "in-reply-to"
> function.
>
> Second:  I still can't make two calls using iconnecthere at the same
> time.
> Here is what I have tried:
>
> Attempt 1:
> >>exten=>_91NXXNXX,1,Dial,StripMSD
> >>exten=>_1NXXNXX,2,Dial,SIP/[EMAIL PROTECTED]
> >>exten=>_1NXXNXX,3,Dial,SIP/[EMAIL PROTECTED]
>
> Attempt 2:
> >>exten=>_91NXXNXX,1,Dial,StripMSD
> >>exten=>_1NXXNXX,2,Dial,SIP/[EMAIL PROTECTED]
> >>exten=>_1NXXNXX,103,Dial,SIP/[EMAIL PROTECTED]
>
> Attempt 3:
> >>exten => _91NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
> >>exten => _91NXXNXX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
>
> Attempt 4:
> exten =>
> _91NXXNXX,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED]&SIP/${EXTEN:[EMAIL PROTECTED]
>
>
> So far nothing has worked.
>
> Another question I have is about jitter buffer.  Is there a way to
> create a
> Jitter buffer in sip.conf?
> When I type sip show channels I get the following output:
>
> sip show channels
> Peer UsernameCall ID  Seq (Tx/Rx)  Lag  Jitter
> Format
> 213.137.73.176   xx  5752cb7a55f  00103/0  0ms  ms
> 4
>
> There is a section for Jitter, so I would imagine that there is some way
> to do it.
>
> Thank you for your time.
> Also, if anybody could suggest a good mail client for windows that is
> able to
> use the in-reply-to function, it would be helpful.
>
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Re: [Asterisk-Users] Using multiple iconnecthere accounts

[Martin Pycko]

How about that:

exten => _91NXXNXX,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED]&SIP/${EXTEN:[EMAIL 
PROTECTED]

Martin

On Tue, 8 Jul 2003, Derek Beaumont wrote:

> Asterisk has registered with both accounts:
>
> sip show registry
> Host  Username Refresh State
> 213.137.73.178:5060    120 Registered
> 213.137.73.178:5060    120 Registered
>
> I can make one call just fine, but when I try to make the second call,
> I get an invalid extension error.  When using the following
> configuration:
> >exten=>_91NXXNXX,1,StripMSD,1
> >exten=>_1NXXNXX,2,Dial,SIP/[EMAIL PROTECTED]
> >exten=>_1NXXNXX,3,Dial,SIP/[EMAIL PROTECTED]
> I get the following output
>
> Executing Dial("Zap/4-1", "SIP/[EMAIL PROTECTED]") in new stack
> -- Called [EMAIL PROTECTED]
> -- SIP/iconnect-cd45 is making progress passing it to Zap/4-1
>
> >show channels
> >Peer UsernameCall ID  Seq (Tx/Rx)  Lag  Jitter
> Format
> >213.137.73.178   xx  6631b1e766b  00103/0  0ms  ms
> 4
> >1 active SIP channel(s)
>
> This appears when I make the first call.  I notice that I have a 0ms
> Jitter buffer. I am now curious as to how I create a jitter buffer
> in sip.conf?  I have the following in the [general] section of sip.conf
>
> >jitterbuffer=yes
> >dropcount=3
> >maxjitterbuffer=2500
> >maxexccessbuffer=100
>
>
> Below is the output when I tried to call a second long distance number
>
> -- Executing Dial("Zap/4-2", "SIP/[EMAIL PROTECTED]") in new
> stack
> -- Called [EMAIL PROTECTED]
> -- Got SIP response 480 "Temporarily not available" back from
> 213.137.73.178
> -- SIP/iconnect-fde9 is circuit-busy
>   == Everyone is busy at this time
> -- Executing Dial("Zap/4-2", "SIP/[EMAIL PROTECTED]") in new
> stack
> -- Called [EMAIL PROTECTED]
> sip show channels
> Peer UsernameCall ID  Seq (Tx/Rx)  Lag  Jitter
> Format
> 213.137.37.178   xx  7047ee1a76b  00102/0  0ms  ms
> 2
> 213.137.73.176   xx  7b782a7b3dd  00103/0  0ms  ms
> 4
> 2 active SIP channel(s)
> *CLI> WARNING[98311]: File chan_sip.c, Line 410 (retrans_pkt): Maximum
> retries exceeded on call
> [EMAIL PROTECTED] for seqno 102 (Request)
>   == No one is available to answer at this time
> -- Sent into invalid extension 'xxx' in context 'outgoing'
> on Zap/4-2
> -- Executing Playback("Zap/4-2", "TelError") in new stack
> -- Playing 'TelError'
> WARNING[98311]: File chan_sip.c, Line 410 (retrans_pkt): Maximum retries
> exceeded on call [EMAIL PROTECTED] for
> seqno 102 (Request)
>
> Any help is appreciated.  Thank you for your time.
>
> OLD MESSAGE===
>
> >>Did asterisk register with both accounts ?
> >>"sip show registry"
> >>
> >>Can you post what happens on the console along with 'sip debug' ?
> >>
> >>Martin
>
> On Tue, 8 Jul 2003, Derek Beaumont wrote:
>
> > Has anybody out there tried to use two different iconnecthere accounts
> > with Asterisk?
> > What I want to do is use a second account if the first is busy.
> > I have tried the following:
> >
> > exten=>_91NXXNXX,1,StripMSD,1
> > exten=>_1NXXNXX,2,Dial,SIP/[EMAIL PROTECTED];iconnect is the
> > first account
> > exten=>_1NXXNXX,3,Dial,SIP/[EMAIL PROTECTED];iconnect2 is
> > the second account
> >
> > But that doesn't work.  Has anybody tried this before?
> >
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>
>
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Re: [Asterisk-Users] Using multiple iconnecthere accounts

[Martin Pycko]

Busy is n+1 if n+101 doesn't exist.

Martin

On 8 Jul 2003, Steven Critchfield wrote:

> On Tue, 2003-07-08 at 10:10, Derek Beaumont wrote:
> > Has anybody out there tried to use two different iconnecthere accounts
> > with Asterisk?
> > What I want to do is use a second account if the first is busy.
> > I have tried the following:
> >
> > exten=>_91NXXNXX,1,StripMSD,1
> > exten=>_1NXXNXX,2,Dial,SIP/[EMAIL PROTECTED];iconnect is the
> > first account
> > exten=>_1NXXNXX,3,Dial,SIP/[EMAIL PROTECTED];iconnect2 is
> > the second account
> >
> > But that doesn't work.  Has anybody tried this before?
>
> Isn't busy n+101 priority, or is it n+100? Basically you dial out
> similar to how you set up the busy portion of your voicemail.
> --
> Steven Critchfield  <[EMAIL PROTECTED]>
>
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Re: [Asterisk-Users] Using multiple iconnecthere accounts

[Martin Pycko]

Did asterisk register with both accounts ?
"sip show registry"

Can you post what happens on the console along with 'sip debug' ?

Martin

On Tue, 8 Jul 2003, Derek Beaumont wrote:

> Has anybody out there tried to use two different iconnecthere accounts
> with Asterisk?
> What I want to do is use a second account if the first is busy.
> I have tried the following:
>
> exten=>_91NXXNXX,1,StripMSD,1
> exten=>_1NXXNXX,2,Dial,SIP/[EMAIL PROTECTED]  ;iconnect is the
> first account
> exten=>_1NXXNXX,3,Dial,SIP/[EMAIL PROTECTED]  ;iconnect2 is
> the second account
>
> But that doesn't work.  Has anybody tried this before?
>
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Re: [Asterisk-Users] overlap dialing on a pri span

[Martin Pycko]

Well first of all if you set up DigitTimeout to 5 seconds so asterisk is
going to wait up to 5 seconds to retrieve the digits specially when you
have a match of _X. that is at least to digits but with the timeout of 5
you could imagine that asterisk will intercept all digits.

How about having a pattern _X (without a dot). The amount of digits that
asterisk is waiting for is set by you. _X is one digit, _X is 5 digits

Martin

On Tue, 8 Jul 2003 [EMAIL PROTECTED] wrote:

> Martin,
>
> I probably should have mentioned that: overlapdial=yes was set in
> zapata.conf (I take it this option is inherited through all the
> channels I configure in zapata.conf). I also did a fresh checkout today.
>
> My guess is that the main problem for now lies in the fact that asterisk
> won't execute a dial application once it received the first digit.
> Apparently, the extension _X. won't spawn dial before asterisk hits
> the timeout:
>
> exten => s,1,Wait,1 ; Wait a second, just for fun
> exten => s,2,Answer ; Answer the line
> exten => s,3,DigitTimeout,2 ; Set Digit Timeout to 5 seconds
> exten => s,4,ResponseTimeout,2  ; Set Response Timeout to 10 seconds
> exten => _X.,1,Dial,Zap/g8/BYEXTENSION
>
> I can see asterisk pick up:
>
> -- Executing Answer("Zap/159-1", "") in new stack
>
> the receive some digits
>
> DEBUG[22551]: File chan_zap.c, Line 3254 (zt_read): DTMF digit: 7 on Zap/159-1
> DEBUG[22551]: File chan_zap.c, Line 3254 (zt_read): DTMF digit: 8 on Zap/159-1
> <...>
>
> and seconds (!) later asterisk dials out
>
> -- Executing Dial("Zap/159-1", "Zap/g8/BYEXTENSION") in new stack
> -- Called g8/78997899
> -- Channel 1, span 8 got hangup
>
> Do you know why? Is there a minimum number of digits asterisk need for an
> inital setup message?
>
> Thilo
>
> > overlapdial=yes in zapata.conf
> > for those channels that you want the overlapdialing be activated.
> >
> > By default only incoming overlap dialing is enabled.
>
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Re: [Asterisk-Users] Transfert call

[Martin Pycko]

That got implemented recently ...

Martin

On Tue, 8 Jul 2003, carlos del mayor wrote:

> Hi Rattana,
>
> That kind of transfer is not yet implemented in *. The
> way it will be indicated is:
> exten =>111,dial,Zap/1,20,T
>
> The T indicate that transfer is permitted for calling
> party, but as I've said, that's not implemented at the
> moment.
>
> Regards
> cmayor
>
>  --- Rattana BIV <[EMAIL PROTECTED]> escribió: > Hi,
> >
> >
> > A question about transfert.
> >
> > How can I make transfert with the the person who
> > call.
> > X call Z and X transfert Z to Y.
> > I only succeed to do X call Z and Z transfert to Y.
> >
> > If someone have a solution it will be very good =)
> >
> >
> > regards
> > Rattana
>
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Re: [Asterisk-Users] Can't access outside voicemail services throughasterisk

[Martin Pycko]

Come on,

exten => *98,1,Dial,Zap/g1/BYEXTENSION
should work since it's old sytax.

It's more propable that you have that *98 in a diffrent context
than assigned for those channels or you don't have the group 1 defined
properly 

Martin

On 7 Jul 2003, Steven Critchfield wrote:

> On Mon, 2003-07-07 at 13:31, Derek Beaumont wrote:
> > I want to be able to check my Bell voicemail
> > (*98) using a phone attached to my asterisk box.
> >
> > In extensions.conf I have defined
> > exten=>*98,1,Dial,Zap/g1/BYEXTENSION
> >
> > However, when I dial *98, I just get a fast busy
> > signal.
> >
> > Is the * digit reserved for internal purposes?
>
> Not really, but likely you do not have a extension to deal with that
> option. Try...
> exten => *98,1,Dial(Zap/${TRUNK}/${EXTEN}
> --
> Steven Critchfield  <[EMAIL PROTECTED]>
>
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Re: [Asterisk-Users] overlap dialing on a pri span

[Martin Pycko]

overlapdial=yes in zapata.conf
for those channels that you want the overlapdialing be activated.

By default only incoming overlap dialing is enabled.

regards
Martin

On 7 Jul 2003, Thilo Salmon wrote:

> Hi,
>
> I am lost trying to figure out how to enable overlap dialing for calls
> coming in and coing out on a pri span. DISA looked promising at first,
> but does not seem to support overlap dialing. Just picking up a call by
> and trying to dial out does not seem the way to do it either. I tried:
>
> [dialincontext]
> exten => 12341234,1,Goto(dialoutcontext,s,1)
>
> [dialoutcontext]
> exten => s,1,Wait,1 ; Wait a second, just for fun
> exten => s,2,Answer ; Answer the line
> exten => s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
> exten => _X.,1,Dial,Zap/g8/BYEXTENSION
> exten => t,1,Dial,Zap/g8/BYEXTENSION
>
> in extensions.conf, hoping this would make asterisk dial out after
> reading a single digit. I can see asterisk detecting DTMF digits, but it
> won't dial out before the timeout kicks in and then hangs up right away.
>
> In the end I want to serve incoming callers with a dialtone, listen for
> DTMF digits and dial each digit seperately by submitting q.931
> information messages. According to Mark and the parts of libpri.h and
> app_dial.c I understood asterisk supports this way of "overlap dialing".
> Any ideas how to achieve this? A simple extensions.conf configured to
> handle overlap dialing could help me a lot.
>
> Thilo
>
>
>
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Re: [Asterisk-Users] System command..

[Martin Pycko]

The system at the moment can run some program/script but there is no way
to retrieve the results. Although you could have tried with
${ENV(VARIABLE)}

Martin

On Mon, 7 Jul 2003, WipeOut . wrote:

> Can the system command be used to retrieve a variable from a mysql database using 
> the mysql command line client??
> or would it be simpler to write some sort of AGI type application??
> --
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Re: [Asterisk-Users] Newbie Doubts

[Martin Pycko]

You plug a channel bank to a T1 in your PC connected either over T100P or
T400P.

regards
Martin

On Mon, 7 Jul 2003, Ricardo Saar Gemignani wrote:

> Hello everybody
>
> My doubt is about configuration. Can I use a channel bank like zplex-10 or 
> adtran, plug on it an T1, 24 POTS, an ethernet cable connected at the computer with 
> Asterisk installed? Will it work? Will asterisk be able to control the system? 
> Receive a call and work with all its functions(transfer, conference, voice mail)?
>
> Thanks in advance,
> Ricardo Saar Gemignani
>

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Re: [Asterisk-Users] zt_pri_errors: PRI got event: 8 / 6

[Martin Pycko]

Also you can have the wrong pridialplan in zapata.conf. Look at "pri debug
span 1" for q931 trace.

Martin

On Fri, 4 Jul 2003, Stefano Finetti wrote:

> Greetings.
>
> Today I've installed a fresh new E100P on a EuroISDN PRI.
>
> It seems to work well, accepting calls, but, when I start *, I have
> the screen flooded with this message:
>
> PRI got event: 8 (or  6)
>
> If i look into /var/log/asterisk/messages i've this Error:
>
> File chan_zap.c, Line 5341 (zt_pri_error): PRI: Read on 35 failed:
> Unknown error 500
>
> Repeated indefinitely.
>
> Also, i've a real strange problem.
>
> It seems that i'm perfectly able to call fixed phones, but If i try to
> call:
> - Mobile Phones
> - Toll Free numbers, i obtain this from the *CLI>
>
> -- Executing SetCallerID("SIP/Telefono-1-d8f2", "199770783") in
> new stack
> -- Executing Dial("SIP/Telefono-1-d8f2", "Zap/g1/348xxx") in
> new stack
> -- Called g1/348xxx
> -- Channel 4, span 1 got hangup
>   == No one is available to answer at this time
> -- Hungup 'Zap/4-1'
>
> (x are for privacy issues)
>
> Has anyone ideas?
>
> I've lurked the archive for the zt_pri_error but did'nt find answers.
>
> Another info that may be useful...
>
> Every 3 or 4 minutes, i obtain this from the *CLI>
>
>   == D-Channel on span 1 down
>   == D-Channel on span 1 up
> -- B-channel 1 successfully restarted on span 1
> -- B-channel 2 successfully restarted on span 1
> -- B-channel 3 successfully restarted on span 1
> -- B-channel 4 successfully restarted on span 1
> -- B-channel 5 successfully restarted on span 1
> -- B-channel 6 successfully restarted on span 1
> -- B-channel 7 successfully restarted on span 1
> -- B-channel 8 successfully restarted on span 1
> -- B-channel 9 successfully restarted on span 1
> -- B-channel 10 successfully restarted on span 1
> -- B-channel 11 successfully restarted on span 1
> -- B-channel 12 successfully restarted on span 1
> -- B-channel 13 successfully restarted on span 1
> -- B-channel 14 successfully restarted on span 1
> -- B-channel 15 successfully restarted on span 1
>
> Thanks
> --
> Stefano Finetti
> System Consultant
> Lynx Automotive srl
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Re: [Asterisk-Users] PCI Master abort (debian / S400P & X101P

[Martin Pycko]

You have one board on the same IRQ as VGA. Try to replace the board in a
diffrent PCI slot or maybe you can do the IRQ job in the BIOS.

Martin

On Fri, 4 Jul 2003 [EMAIL PROTECTED] wrote:

> Debian / kernel 2.4.18.
> Found in /proc/pci ::
> Bus  0, device   0, function  0:Host bridge: Intel Corp. 430HX -
> 82439HX TXC [Triton II] (rev 3).
> Bus  0, device   7, function  0:ISA bridge: Intel Corp. 82371SB PIIX3
> ISA [Natoma/Triton II] (rev 1).
> Bus  0, device   7, function  1:IDE interface: Intel Corp. 82371SB
> PIIX3 IDE [Natoma/Triton II] (rev 0).
> Bus  0, device   9, function  0:VGA compatible controller: IRQ 10.
> Bus  0, device  10, function  0:Tiger Jet Network Inc. Model 300 128k
> (rev 0). IRQ 12.
> Bus  0, device  12, function  0:Tiger Jet Network Inc. Model 300 128k
> (#2) (rev 0). IRQ 9.
> Bus  0, device  11, function  0:Realtek Semiconductor Co., Ltd.
> RTL-8029(AS) (rev 0). IRQ 11.
>
> Commands sent :
> => modprobe zaptel
> => modprob wcfxs
> => modprobe wcfxo
>
> No pb !
>
> THEN :
>
> wait about 2 minutes (no more command passed)
> => PCI Master abort
>
> OR
>
> => asterisk -cvvv
> => (zt_open): Unable to specify channel 1
> => PCI Master abort
> ___
>
> Does any body work with the same HW configuration (1 FXO + 1 FXS) ?
> (Wildcard S400P 4 modules / 1 installed   +   Wildcard X101P)
>
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Re: [Asterisk-Users] LD accontability

[Martin Pycko]

"show application authenticate" ?

Martin

On Fri, 4 Jul 2003, Kim C. Callis wrote:

> As I was working on my extensions.conf file, I started to segment
> calling privileges. For the everyday workers, I don't free reign to LD
> access unless it is business related. So I was wondering if there was a
> way to implement some type of accounting code to be entered before
> accessing LD, which of course would be noted in the CDR (however it is
> implemented, either comma delimited or MySQL).
>
> For example:
>
> I have a customer who is customer 2022. One of my tech support people
> has to make a call to that client for business related reasons. So when
> that tech person dials 1-212-555-1212, they have to dial in 2022 so that
> it is notated that this call was to that particular client. Assuming
> that is a valid account code, the LD call proceeds, if not, a message is
> played for the tech person to the effect of something like, "That
> account code is not valid" and notates that in the cdr as well.
>
> This is my view of how it would work, but if anyone has another way to
> provide accountability that would work as well.
>
> Kim C. Callis
>
>

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Re: [Asterisk-Users] Accounting info for SIP Calls

[Martin Pycko]

CDR is used for that.

Martin

On Fri, 4 Jul 2003, destan wrote:

> Hi all,
>
> Does Asterisk report or keep a database of the duration of SIP Calls? Is CDR
> used for this? If there are people using this facility, how accurate is it?
>
> Thanks in advance,
> Umut
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Re: [Asterisk-Users] dst number

[Martin Pycko]

Lets say that you're going to receive 20 diffrent DIDs

1000 - 1019

[incoming]
exten => _X.,1,SetVar,SAVE_DID=${EXTEN}
exten => _X.,2,Goto,${EXTEN}|s|1

[1000]
exten => s,1,blabla



[1019]
exten => s,1,..

Martin

On Fri, 4 Jul 2003, Steven Kawuma wrote:

> Hi all,
>
> I'm trying to write a set of contexts that will be executed for different 
> destination numbers. But for each of these numbers, different audio files are played 
> as specified in a postgres db depending on the destination number. Is this possible? 
> How can I tell which number has been called at the time when the call is answered?
>
> Steven
>

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Re: [Asterisk-Users] res parking patch

[Martin Pycko]

But in this case you should have said:
"What exten for picking *the oldest* or *the first* call that got there

Martin

On Fri, 4 Jul 2003, Matteo Brancaleoni wrote:

> It's an offence ;) ?
>
> yes I know that.
>
> But in that case I needed to be able to pick the older
> call in the parking lot, w/o knowing the parked number.
> Useful when multiple calls get in, get parked and
> I wanna to manage them in a FIFO way, dialling
> just a single exten.
>
> the queue app wasn't useful, since it rings
> the members (or I must be an agent), and I (read:
> the customer) didn't want that.
>
> Matteo.
>
> Scrive Steven Critchfield <[EMAIL PROTECTED]>:
>
> > Did you not know you could dial the parked number and pick it up
> > directly?
> >
> > On Thu, 2003-07-03 at 16:36, Brancaleoni Matteo wrote:
> > > Ok, a little patch that adds a little functionality to call parking.
> > > With that, you can pickup the older parked call, if many are in the
> > > parking lot. The default exten to do that is 750, but can be changed
> > > by setting "parkpick => exten" on parking.conf , like
> > >
> > > [general]
> > > parkext => 800   ; What ext. to dial to park
> > > parkpos => 801-820   ; What extensions to park calls on
> > > parkpick => 821  ; What exten for picking older call on park
> > > context => parkedcalls   ; Which context parked calls are in
> > > parkingtime => 60
> > >
> > > I've done that since a customer asked me a such function.
> > > Feel free to try it.
> > > Disclaimer : I haven't tested it heavily... just seems to work ;)
> > >
> > > Matteo.
> > --
> > Steven Critchfield <[EMAIL PROTECTED]>
> >
> > ___
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> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
> --
>
> Matteo Brancaleoni
> Espia System Administrator
> http://www.espia.it
>
> -
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Re: [Asterisk-Users] client reinvitation problem

[Martin Pycko]

That means that asterisk is sending SIP messages but gets no response from
the device.

Martin

On Thu, 3 Jul 2003 [EMAIL PROTECTED] wrote:

> Hello All!
>
> There is description of my problem with Asteriks below.
> Asteriks CLI says:
> "File chan_sip.c, Line 415 (retrans_pkt): Maximum retries exceeded on call"
>
> Sip debug on the server gives the next:
>
> Retransmitting #5 (no NAT):
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.0.26:5060
> From: ;tag=106403508
> To: ;tag=as0771c6f9
> Call-ID: [EMAIL PROTECTED]
> CSeq: 1 INVITE
> User-Agent: Asterisk PBX
> Contact: 
> Content-Type: application/sdp
> Content-Length: 209
>
> 8523 is Cisco ATA-186
>
> The sip.conf content:
>  - - - - -
> [cisco8523]
> type=friend
> username=8523
> secret=test
> nat=no
> host=dynamic
> canreinvite=no
> qualify=300
> defaultip=192.168.0.26
>  - - - - -
>
> Why I place a call to Asteriks. I hear some invitation but connection brokes
> when retransmit exceed.
>
> Could anyone give some advice or solution.
> Thanks in advance
>
> --
> Best regards
> Vlad
>
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RE: [Asterisk-Users] BIG problem with multiple rings before pickup

[Martin Pycko]

You "Answer" on analog channels and then you need to have
a "fax" extension in the current context.

regards
Martin

On Wed, 2 Jul 2003, Joe Antkowiak wrote:

> How do you tell asterisk to detect for fax tones?
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Steven
> Critchfield
> Sent: Wednesday, July 02, 2003 2:55 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] BIG problem with multiple rings before pickup
>
> On Wed, 2003-07-02 at 13:34, Jim Archer wrote:
> > Hi All...
> >
> > I have a maddening problem...
> >
> > I have Asterisk configured to pick up a line after 4 rings.  I do this to
> > allow my fax machine to pick up a particular distinctive ring pattern, so
> I
> > don't have to pay for a dedicated fax line.
> >
> > If someone calls the line, lets it ring 3 times and then hangs up,
> Asterisk
> > answers the line, and holds it off hook forever, constantly playing the
> > prompts.
> >
> > My hardware is 2 X100P cards.
> >
> > Any ideas?
>
> Get a TDM10B, cancel your distinctive ring, and let asterisk answer
> immediately and detect fax tones and forward it to your fax machine.
> --
> Steven Critchfield  <[EMAIL PROTECTED]>
>
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>
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Re: [Asterisk-Users] *8 pickup then transfer drops call

[Martin Pycko]

What configuration of hardware/software are you running. I just checked
picking up with *8/transfer on zaptel/SIP and it works on our Digium PBX.

I placed a call from SIP to Zap, picked it up with Zap (*8) and transfered
to Zap

and also
place a call from Zap to Zap, picked it up with SIP-Snom200 (*8) and
transfered with blind transfer to Zap.

It worked fine.

regards
Martin

On Tue, 1 Jul 2003, Chad Sawyer wrote:

> I have a small problem,
>
> Whenever we pickup a call using *8 then try to transfer it via flash or # transfer 
> the call is dropped.  Any ideas?  Whe have all called exten's in extension.conf 
> ending in |t som that # transfers work.  What am I doing wrong?
>
>
> Chad Sawyer, Manager, Network Administrator
>

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Re: [Asterisk-Users] How do i make best use of Macro?

[Martin Pycko]

The meetmecount app is supposed to tell you the number of participants in
a certain conf number. However it does not create the var variable.
The error about "wrong use of LEN(" was do to the fact that your var
variable does not exist and it was a bug. It's fixed now.

Martin

On Tue, 1 Jul 2003 [EMAIL PROTECTED] wrote:

> Steven,
>
> I tried the following:
>
> [Conferences]
> exten => ,1,MeetMeCount(|var)
> exten => ,2,SayNumber(${var})
> exten => ,3,Meetme()
>
> but I get the following error:
>
> NOTICE[48152]: File pbx.c, Line 900 (pbx_substitute_variables_temp): Wrong
> use of LEN(VARIABLE)
>
> It still let's me join the conference but it says "zero" no matter how many
> people are in the conference.
>
> What am I doing wrong?
>
> Michael
>
> > Since there was some interest in this, here is the diff against current
> > cvs. Someone that is better at C should look into my use of strsep
> > because there is a couple of warnings. Also there is a warning on my use
> > of pbx_builtin_setvar_helper, but I can't see whats wrong here.
> >
> > BTW, SayNumber doesn't seem to say '0'.
> >
> > Usage is like this.
> >
> > exten => 1234,1,MeetMeCount(1234|var)
> > exten => 1234,2,SayNumber(${var})
> > exten => 1234,3,MeetMe(1234)
>
> >-Original Message-
> >From: [EMAIL PROTECTED] [mailto:asterisk-users-
> >[EMAIL PROTECTED] On Behalf Of Steven Critchfield
> >Sent: Thursday, June 26, 2003 2:18 PM
> >To: [EMAIL PROTECTED]
> >Subject: Re: [Asterisk-Users] hunt group
> >
> >Granted I haven't read the handbook in a while, but I think this might
> >be a unique question worth answering.
> >
> >The idea should be that you create each user as a group. And then you
> >dial with Zap to each of the 4 groups. Asterisk will call out using the
> >lowest port in each of the groups. This accomplishes what you mentioned.
> >
> >for the zapata interface use something like
> >group = 1 ; 4 line phone 1
> >channel => 1-4
> >group = 2 ; 4 line phone 2
> >channel => 5-8
> >group = 3 ; 4 line phone 3
> >channel => 9-12
> >group = 4 ; last 4 line phone
> >channel => 13-16
> >
> >and your dial string will be like
> >exten => _X.,1,Dial(Zap/g1&Zap/g2&Zap/g3&Zap/g4)
> >
> >
> >On Thu, 2003-06-26 at 16:08, Jeremy McNamara wrote:
> >> See zapata.conf.sample group keyword and the Asterisk Handbook, this is
> >> covered there.
> >>
> >>
> >>
> >> Jeremy McNamara
> >>
> >>
> >> Joe Antkowiak wrote:
> >>
> >> >I was wondering if someone could point me in the right direction with
> >> >this...
> >> >
> >> >I have 3 4-line phones, all connected to a * box via a T100P and channel
> >> >bank.
> >> >
> >> >I essentially want to create a "hunt group" for each phone, and then as
> >> >calls come in, Dial to include the 3 different hunt groups, so that if
> >one
> >> >user is on line 1 of their phone, a new incoming call will go to line 2
> >of
> >> >their phone, but line 1 of the others.
> >> >
> >> >Can I do this with hints?
> >> >
> >> >Thanks.
> >> >
> >> >-Joe
> >> >
> >> >___
> >> >Asterisk-Users mailing list
> >> >[EMAIL PROTECTED]
> >> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >> >
> >> >
> >>
> >>
> >> ___
> >> Asterisk-Users mailing list
> >> [EMAIL PROTECTED]
> >> http://lists.digium.com/mailman/listinfo/asterisk-users
> >--
> >Steven Critchfield  <[EMAIL PROTECTED]>
> >
> >___
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>
>
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Re: [Asterisk-Users] Unable to get SetMusicOnHold working...

[Martin Pycko]

You have to call Dial with ||m option to have music-on-hold while
transfering

Read the "show application dial"

Martin

On 1 Jul 2003, Fabrice Tereszkiewicz wrote:

> Hello,
>
> I'm trying to do something really easy : transfer a PSTN call to a H323
> client. This works great. Now I'm trying to use the SetMusicOnHold
> function. I din't find any doc about it, I've just seen some mails in
> the list archive, but it still doesn't work.
>
> That's my extension.conf :
>
> [incoming]
> exten => s,1,SetMusicOnHold,default
> exten => s,2,Dial(OH323/192.168.1.215)
>
> really short...
> my musiconhold.conf :
>
> [classes]
> default => mp3:/var/lib/asterisk/mohmp3/
>
> and there are mp3's in my /var/lib/asterisk/mohmp3/ directory. I can
> hear music when I use :
>
> exten => s,4,MP3Player(/var/lib/asterisk/mohmp3/rem.mp3)
>
> So I don't know what to try, thanks for your help
> --
> Fabrice Tereszkiewicz <[EMAIL PROTECTED]>
>
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Re: [Asterisk-Users] picking up a ringing extension

[Martin Pycko]

To pick up a call that rings someone elses phone that is in the callgroup
as your pickupgroup.

Martin

On Tue, 1 Jul 2003, carlos del mayor wrote:

> Well, I suposse is a very basic question but,,,for
> what is used: callgroup=1 and pickupgroup=1 ?
> thanks!
> c.mayor
>
> --- Louis-David Mitterrand <[EMAIL PROTECTED]>
> escribió: >
> > Hello,
> >
> > We are using asterisk 0.4.0 on debian sid with Cisco
> > 7960 and ATA186
> > phones.
> >
> > All sip entries have:
> >
> > callgroup=1
> > pickupgroup=1
> >
> > However I am unable to remotely pickup a ringing
> > phone using *8#. I get
> > fast busy tone. Is there some flag to add in
> > extensions.conf ?
> >
> > Thanks in advance,
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> >
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> ___
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> Super Webcam, voz, caritas animadas, y más...
> http://messenger.yahoo.es
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Re: [Asterisk-Users] Conference calls

[Martin Pycko]

You need to look at "show application meetme" in the asterisk CLI
but for it to work you need to have some kind of zaptel hardware or
emulate it with zttdummy (but for that you need to have usb-uhci like USB
controller)

and then

exten => 1000,1,Meetme,1000

Martin

On Tue, 1 Jul 2003, Serge Mankovski wrote:

> Hi
> I want to set up * as a conference bridge. I would like to be able to
> conference is SIP calls (up to 12)
>
> I am looking through all available documentation for * to get info on how it
> is done. No luck so far.
>
> Can somebody direct me to the info in this subject?
>
> Thank you
> Serge
>
> _
> Protect your PC - get McAfee.com VirusScan Online
> http://clinic.mcafee.com/clinic/ibuy/campaign.asp?cid=3963
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Re: [Asterisk-Users] ISDN PRI E1-CLI and DNIS

[Martin Pycko]

ISDN PRI E1 is enough to receive DID and CallerID (ANI).

Martin

On Mon, 30 Jun 2003, Surajee Ratnayake wrote:

> hi everybody,
>
> my question is specific to ISDN signalling,
> in my set up, i want to get cli and dnis into my asterisk box, and i am going to use
> ISDN PRI E1s coming from telco.
> To get cli and dnis, do i need to apply for QSIG from the telecom, or is there
> some other type? and i got to know that still asterisk does not support QSIG???
>
> sorry for asking this kind of question in the asterisk mailing list,
>
> Thank you inadvance,
> Surajee
>

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Re: [Asterisk-Users] Asterisk CPU usage

[Martin Pycko]

Try to put
noload => chan_oss.so

in modules.conf

also do you use mpg123 with musiconhold ?

Martin

On Fri, 27 Jun 2003, Dave Alan Caruana wrote:

> hi there..
> I have an asterisk installation with a PRI-E1 card
> running EuroISDN, installed on a 1GHz Intel Celeron
> box with 256Mbytes RAM.
> CPU usage is stuck at 100% all the time, even with
> no calls going through. Is this the normal ?
> Running "top" reveals that the CPU allocation is
> 99.6% to Asterisk.
>
>  13:41:48  up 17:55,  3 users,  load average: 1.07, 1.02, 1.00
> 44 processes: 43 sleeping, 1 running, 0 zombie, 0 stopped
> CPU states:   0.0% user 100.0% system   0.0% nice   0.0% iowait   0.0% idle
> Mem:   247188k av,  239664k used,7524k free,   0k shrd,  126572k
> buff
> 173676k actv,   0k in_d,2932k in_c
> Swap:  522104k av,2680k used,  519424k free   67164k
> cached
>
>   PID USER PRI  NI  SIZE  RSS SHARE STAT %CPU %MEM   TIME CPU COMMAND
>  2569 root  22   0  4712 4712   456 S99.8  1.9  1069m   0 asterisk
> 7 root  15   0 00 0 SW0.2  0.0   0:24   0
> kscand/Normal
> 1 root  15   0   108   8856 S 0.0  0.0   0:03   0 init
> etc.
>
> Second question is this :
> my asterisk server is currently configured to receive calls, and immediately
> forward them to a SIP hosts (an ITSP server in USA) that requires input
> via an IVR. Sometimes this works fine, but many times the connection just
> drops while typing in codes. Once a connection is established (ie. got
> past the IVR stage) the connection never drops ..
>
> help welcome :)
>
> cheers
> Dave
>
>
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Re: [Asterisk-Users] Asterisk and Digium E400P in EuroISDN environment

[Martin Pycko]

It's either EuroISDN or E&M w/E1. And for incoming calls you'll get what
you need if the telco sends it on the channels.

Martin

On Thu, 26 Jun 2003, Scott Stingel wrote:

> Hello-
>
> I know this is a basic question, but before I start down the road of using
> Asterisk open source software, I thought I would ask if someone could please
> tell me definitively whether Asterisk and Digim will connect with EuroISDN
> using E&M protocol, and pass the caller's number (CLI) and dialed number
> (DDI) to the software so I can see them in variables?
>
> I need this for an application that I'm proposing.
>
> Thanks in advance,
> Scott Stingel
>
>
>
> Scott M. Stingel
> Emerging Voice Technology Inc.
> Palo Alto, California and London, England
>
> Email:  [EMAIL PROTECTED]
> URL:www.evtmedia.com
>
>
>
>
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RE: [Asterisk-Users] VoiceMail recording dialtone

[Martin Pycko]

Unless your telco signals hangup with a dialtone . it should help.
The thing is that most propably your X100P hangs up and then picks up the
line due to something ... that was my original idea.

Martin

On Wed, 25 Jun 2003, Sam Bingner wrote:

> I don't understand how that would affect the voicemail recording dialtone
> when the phone never rang?
>
> 1, User calls
> 2, Nobody answers in 20 seconds
> 3, greeting is played (user hangs up somewhere in here, close to end)
> 4, voicemail is called, and records a dialtone
>
> -- phone never rings again here --
>
> Sam
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Martin Pycko
> Sent: Thursday, June 19, 2003 11:57 AM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] VoiceMail recording dialtone
>
>
> Well experiment yourself with the code.
>
> in wcfxo.c
> /* Don't accept a ring for another 1000 ms */
> wc->ringdebounce = 1000;
>
> Try a diffrent value (e.g. 3000 for 3 sec)
> and in zaptel.h
>
> #defineRING_DEBOUNCE_TIME  500 /* 500 ms ring debounce
> time */
>
> try the same value as in wcfxo.c
>
> recompile/reload and test
>
> regards
> Martin
>
> On Thu, 19 Jun 2003, Sam Bingner wrote:
>
> > Zaptel was the version from about 4 days ago when I sent this message,
> > I updated again yesterday night
> >
> > Sam
> >
> > Quoting Martin Pycko <[EMAIL PROTECTED]>:
> >
> > > How old is your zaptel code ?
> > > Mark recently increased some timer for that.
> > >
> > > Martin
> > >
> > > On Wed, 18 Jun 2003, Sam Bingner wrote:
> > >
> > > > I have an extension setup with voicemail, for incoming calls on an
> > > > X100P card.  It quite often will record about 15 seconds of
> > > > dialtone... I'm guessing that it picks up the line after the
> > > > outgoing line has been disconnected.
> > > >
> > > > Has anybody else run into this problem?  Shouldn't chan_zap be
> > > > detecting the hangup and ending the connection?
> > > >
> > > > Sam
> > > >
> > >
> > > ___
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> > >
> >
> >
> >
> >
> > -
> > This mail sent through IMP: http://horde.org/imp/
> >
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> >
>
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Re: [Asterisk-Users] indication tones and callwaiting chirp too loud

[Martin Pycko]

Try to change something in zaptel/tonezone.c around these lines
/* Bring it down -8 dbm */
gain = pow(10.0, (LEVEL - 3.14) / 20.0) * 65536.0 / 2.0;

td->fac1 = 2.0 * cos(2.0 * M_PI * (freq1 / 8000.0)) *
32768.0;
td->init_v2_1 = sin(-4.0 * M_PI * (freq1 / 8000.0)) *
gain;
td->init_v3_1 = sin(-2.0 * M_PI * (freq1 / 8000.0)) *
gain;

td->fac2 = 2.0 * cos(2.0 * M_PI * (freq2 / 8000.0)) *
32768.0;
td->init_v2_2 = sin(-4.0 * M_PI * (freq2 / 8000.0)) *
gain;
td->init_v3_2 = sin(-2.0 * M_PI * (freq2 / 8000.0)) *
gain;

Martin

On Wed, 25 Jun 2003, Surfer Dude wrote:

>
> I am wondering if anyone could help me figure out how to turn down the volume on all 
> the dial tones, indications, etc.. and especially the call-waiting CHIRP!
>
> I don't want to change the txgain and rxgain because they are working at levels that 
> I would like.  However, when voice conversations and voicemail recordings are at 
> good levels then the dial tones, busy tones, etc are way too loud.  This is 
> especially true for the chirp for call-waiting which is excruciating for someone who 
> is having a conversation.
>
> There is also a very loud chirp and the very end of Voicemail recordings.
>
> Is it possible to turn down these levels?  How about a software limiter?  Or a 
> filter that cuts out huge spikes?
>
> Could these problems be due to my equipment?
>
> Let's see, my system:
>
> T400P 4 port  T1
> port one attached to CAC II w/ 12 FXO set to ground start 12 FXS set to 
> loopstart
> port two CAC II w/ 24/FXS loopstart
> port three CAC II w/ 24/FXS loopstart
> port four free
>
>
> Thanks,
> Jason
>


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Re: [Asterisk-Users] Pattern matching: least-to-most specific PITA

[Martin Pycko]

I think that if you put

exten => _X.,1,DIal,Zap

it'll improve the matching dramatically

Martin

On Wed, 25 Jun 2003, John Todd wrote:

>
> My synapses are rather fried after a long few days of debugging other
> problems, so perhaps I'm being lazy in sending this to the general
> list, but I can't think straight about it.  Forgive me if there is an
> overly obvious solution to this.
>
> I have a list of phone numbers that are SIP extensions.  I'd like to
> dial them via SIP if ${EXTEN} is equal to one of those numbers.  If
> ${EXTEN} is not equal to one of those numbers, I'd like to send the
> call out to a PRI group, regardless of dialed sequence length or
> pattern.
>
> It seems I cannot do this with *'s pattern matching, due to the order
> in which extensions are parsed, which seems to be least-specific to
> most-specific.  This causes all kinds of headaches when trying to use
> wildcards, since wildcards are super-least-specific.
>
> My desire would be to have the more specific matches done first, so
> that if ${EXTEN} would be matched in an order that makes sense.  I
> understand why matching goes from least-to-most specific for analog
> equipment, but it makes certain tasks impossible from a dialplan
> point of view when I have the full number and I'm not waiting on a
> user to finish typing the digits.
>
> If presented with 12123669751 I would expect the match to happen and
> the SIP extension to be dialed.  It doesn't.  It dials the Zap
> extension.
>
> [foo]
> ;
> exten => _1212366975X,1,Dial(SIP/${EXTEN})
> exten => _181772721[8-9]X,1,Dial(SIP/${EXTEN})
> exten => _191481287[4-7]X,1,Dial(SIP/${EXTEN})
> exten => _141550926[0-2]X,1,Dial(SIP/${EXTEN})
> ;
> exten => _.,1,Dial(Zap/g1/${EXTEN})
> ;
>
>
> How do I invert this match examination to make it go most- to
> least-specific execution?
>
> JT
>
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Re: [Asterisk-Users] parsing bug? (using PGSQL)

[Martin Pycko]

If you use brackets () then you need to call it like this
PGSQL(blabla(bla)bla)

That should work

regards
Martin

On Tue, 24 Jun 2003, Thomas Haeger wrote:

> Hi all again,
>
> if i make a query with
> ...
> exten => _X.,2,PGSQL,"Query resultid ${connid} SELECT getdest('${EXTEN}')";
> ...
>
> an error like
>
> WARNING[32785]: File pbx.c, Line 1126 (pbx_extension_helper): No application
> 'PGSQL,"Query resultid ${connid} 'SELECT getdest'
> for extension (tel1, 00905888, 2)
>
> occurs.
>
> This looks like an parsing bug. As if the brackets be cut.
>
> Can somebody help?
>
> Thanks,
>
> Thomas.
>
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Re: [Asterisk-Users] Setting up the E100P

[Martin Pycko]

THat's not it.
in zapata.conf you *also* need to have
signalling=pri_cpe or pri_net

Martin

On Mon, 23 Jun 2003, Michael Bielicki wrote:

> On Monday 23 June 2003 2:58 pm, Anton Yurchenko wrote:
> > Hello,
> >
> > I have an E100P, and in the zaptel.conf I have:
> >
> > span=1,1,0,ccs,hdb4,crc4,yellow
> > fxsks=1-10
> delete the fxsks line and put:
> bchan=1-15,17-31
> dchan=16
> >
> > the light on the card is green( BTW what do all those states of the card
> > that zttool reports YELLO, RED, BLUE ..., is there a doc for zttool?, or
> > for the card?)
> >
> > in the asterisks` zapata.conf I have:
> >
> > [channels]
> > context=default
> > switchtype=euroisdn
> > signalling=fxs_ks
> > usecallerid=yes
> > hidecallerid=no
> > callwaiting=yes
> > callwaitingcallerid=yes
> > threewaycalling=yes
> > transfer=yes
> > cancallforward=yes
> > callreturn=yes
> > echocancel=yes
> > echocancelwhenbridged=yes
> > rxgain=0.0
> > txgain=0.0
> > group=1
> > callgroup=1
> > pickupgroup=1
> >
> > immediate=no
> > callerid="line1"< 238-20-31>
> > channel => 1
> > callerid="line2"< 238-20-31>
> > channel => 2
> >
> >
> > but Asterisk on startup reports that:
> >
> > WARNING[13326]: File chan_zap.c, Line 4173 (ss_thread): CallerID returned
> > with rror on channel 'Zap/1-1'
> > WARNING[14351]: File chan_zap.c, Line 4173 (ss_thread): CallerID returned
> > with rror on channel 'Zap/2-1'
> >
> >
> > and when I call the number that is supposed to terminate there I get
> > busy signal.
> > also in the zttool  I see strange thing, in the bottom portion where
> > there are RxA, RxB  the 111 ( ones) sometimes change into
> > 00, they go from up to down and if I enable all the channels then
> > firs on the first column then on the second column, in the bottom half
> > of the screen.
> >
> > while if I plug the same E1 into Cisco AS5300, whit this config( just
> > exepts):
> >
> > isdn switch-type primary-net5
> >
> > controller e1 ...
> >  clock source line primary
> >  pri-group timeslots 1-31
> > ...
> >
> > interface Serial0:15
> >  isdn switch-type primary-net5
> >  isdn incoming-voice modem 64
> >
> > Framing is CRC4, Line Code is HDB3, Clock Source is Line ...
> >
> >
> > on Cisco the swithtype is primary-net5 ( my guess its euroisdn ? )
> >
> > anybody could guess what is the problem?
> >
> > The admin that runs the Cisco says that signalyng should be PRI , and there
> > is an option for pri signaling in zapata.conf, but the zaptel,conf doesnt
> > have it and so I`d get up with a mismatch, and zasterisk would not start.
>
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Re: [Asterisk-Users] Asterisk CPU power requirements

[Martin Pycko]

You need to find out which way your SIP gateway wants to receive the
DTMFs. There are three ways to do that. Read sip.conf.sample.

Martin

On Mon, 23 Jun 2003, Dave Alan Caruana wrote:

> hi there,
> I have an installed & working Asterisk server,
> which I am using to connect to a SIP service
> abroad. Although I can hear the IVR from the
> ITSP, I cannot seem to send them digits from
> my phone.
>
> I have also noticed that the CPU usage on my
> machine is up to 100% constantly and 99.9%
> of that is going to Asterisk, even when asterisk
> is just idle and doing nothing at all ..
>
> The machine is a Celeron 800 with 256Mb of RAM,
> and there is a Digium single span E1 card
> going into it.
>
> Is something wrong? or do I just need more
> CPU power?
>
> cheers
> Dave
>
>
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Re: AW: [Asterisk-Users] help with pri configuration..

[Martin Pycko]

Well how did you solve your previous problem then ?

Martin

On Mon, 23 Jun 2003, Thomas Haeger wrote:

> The problem before is solved. But now gives another problem ...
>
>
>
>   == Registered channel type 'Zap' (Zapata Telephony Driver w/PRI)
>   == Registered channel type 'Tor' (Zapata Telephony Driver w/PRI)
>   == Starting D-Channel on span 1
> ERROR[1024]: File chan_zap.c, Line 5947 (start_pri): Unable to open
> D-channel 47 (Device or resource busy)
> ERROR[1024]: File chan_zap.c, Line 6682 (load_module): Unable to start
> D-channel on span 2
> WARNING[1024]: File loader.c, Line 299 (ast_load_resource): chan_zap.so:
> load_module failed, returning -1
> WARNING[1024]: File loader.c, Line 394 (load_modules): Loading module
> chan_zap.so failed!
>
> All channels are registered successfully before. But then this error occur.
> I tried to deactivate following ports (spans)
>
>   1. second one
>
> but then the same message occure with the next dchannel
>
> and so on.
>
> Only the first one works.
>
> What's wrong now?
>
>
> Thanks,
>
> Thomas.
>
>
> -Ursprüngliche Nachricht-
> Von: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Auftrag von Thomas
> Haeger
> Gesendet: Montag, 23. Juni 2003 11:34
> An: Asterisk User
> Betreff: [Asterisk-Users] help with pri configuration..
>
>
> Hi all,
>
> can somebody help me with pri configuration?
>
> Here my zapata.conf:
>
>
> ; Zapata telephony interface
> ;
> ; Configuration file
>
> [channels]
>
> switchtype=euroisdn
> signalling=pri_cpe
>
>
> ;group=1
> channel => 1-15,17-31
>
> ;group=2
> channel =>32-46,48-62
>
> ;group=3
> channel => 63-77,79-93
>
> ;group=4
> channel => 94-108,110-124
>
>
>
> And here my zaptel.conf:
>
> zaptel.conf []  0 L:[  1+ 0   1/ 18] *(0   / 227b)= .  10 0x0A
>
> span=1,0,0,ccs,hdb3 #,crc4
> span=2,0,0,ccs,hdb3 #,crc4
> span=3,0,0,ccs,hdb3 #,crc4
> span=4,0,0,ccs,hdb3 #,crc4
>
>
>
>
> bchan=1-15,,32-46,63-77,94-108
> dchan=16,47,78,109
> bchan=17-31,48-62,79-93,110-124
>
> And here the messages after starting astersik:
>
> loadzone = fr
> defaultzone=us
>
>  [chan_zap.so] => (Zapata Telephony w/PRI)
>   == Parsing '/etc/asterisk/zapata.conf': Found
> -- Registered channel 1, PRI Signalling signalling
> -- Registered channel 2, PRI Signalling signalling
> -- Registered channel 3, PRI Signalling signalling
> -- Registered channel 4, PRI Signalling signalling
> -- Registered channel 5, PRI Signalling signalling
> -- Registered channel 6, PRI Signalling signalling
> -- Registered channel 7, PRI Signalling signalling
> -- Registered channel 8, PRI Signalling signalling
> -- Registered channel 9, PRI Signalling signalling
> -- Registered channel 10, PRI Signalling signalling
> -- Registered channel 11, PRI Signalling signalling
> -- Registered channel 12, PRI Signalling signalling
> -- Registered channel 13, PRI Signalling signalling
> -- Registered channel 14, PRI Signalling signalling
> -- Registered channel 15, PRI Signalling signalling
> -- Registered channel 17, PRI Signalling signalling
> -- Registered channel 18, PRI Signalling signalling
> -- Registered channel 19, PRI Signalling signalling
> -- Registered channel 20, PRI Signalling signalling
> -- Registered channel 21, PRI Signalling signalling
> -- Registered channel 22, PRI Signalling signalling
> -- Registered channel 23, PRI Signalling signalling
> -- Registered channel 24, PRI Signalling signalling
> -- Registered channel 25, PRI Signalling signalling
> -- Registered channel 26, PRI Signalling signalling
> -- Registered channel 27, PRI Signalling signalling
> -- Registered channel 28, PRI Signalling signalling
> -- Registered channel 29, PRI Signalling signalling
> -- Registered channel 30, PRI Signalling signalling
> ERROR[1024]: File chan_zap.c, Line 4757 (mkintf): Signalling requested is
> PRI Signalling but line is in Unkn
> own signalling 896 signalling
> ERROR[1024]: File chan_zap.c, Line 6403 (load_module): Unable to register
> channel '1-15'
> WARNING[1024]: File loader.c, Line 299 (ast_load_resource): chan_zap.so:
> load_module failed, returning -1
> WARNING[1024]: File loader.c, Line 394 (load_modules): Loading module
> chan_zap.so failed!
>
>
>
> Whats wrong ?
>
>
>
> Thanks for help,
>
> Thomas.
>
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Re: [Asterisk-Users] How can I log SIP debug messages to a file?

[Martin Pycko]

asterisk -vvvcn | tee /tmp/log
CLI> sip debug
CLI> stop now
or

script
asterisk -vvvcn
CLI> sip debug
CLI> stop now
shell> exit

Martin

On Sun, 22 Jun 2003, destan wrote:

> Hi everybody,
> I want to read to debug messages and try to interpret them but they happen
> too fast, how can I log these guys to a file, or is there a file like this
> already?
> I checked the /var/log/asterisk but there isn't much interesting there yet?
> How can i turn on logging for SIP,IAX and other things?
> Thanks,
> Umut
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Re: [Asterisk-Users] Need help with inbound/outbound PRI calls

[Martin Pycko]

Look carefully at your debug. The "cause" is written there. And 69 stands
for the DISCONNECT message number.

In your case asterisk sends it so it can be that your SIP phone is either
not responding to asterisk's sending him some SIP messages or it's sending
BYE or something else. It would help if you would send 'sip debug' along
with 'pri debug ...'

Martin

On Sat, 21 Jun 2003, Daryl Jones wrote:

> I'm running a pretty successful Asterisk system and recently moved our
> PRI to a T100P board.  The PRI was previously connected to a Cisco 2600
> that was serving as a voice gateway. We are having a frequent problem with
> inbound and outbound calls being disconnected shortly after they are
> answered since moving the PRI directly to the Asterisk box. Most calls work
> fine, but approx 3 out 10 are prematurely disconnected.
>
> Debug info from an outbound call is included below.  Note that the disconnect
> cause is 69, which I believe means "Requested facility not implemented".  Am
> I interpreting the debug info correctly?  What would cause this?
>
> We had a similar problem with the Cisco gateway, but worked around it by
> configuring the Cisco to force the numbering plan for outgoing calls from
> 'unknown' to 'national'.  Is there a way to do this in Asterisk?
>
>
> -- debugging info from dropped outbound call ---
>
> > Calling Number (len=14) [ Ext: 0  TON: National Number (2)  NPI: ISDN/Telephony 
> > Numbering Plan (E.164/E.163) (1)
> >   Presentation: Unknown (1) '6505901801' ]
> > Called Number (len=14) [ Ext: 1  TON: National Number (2)  NPI: ISDN/Telephony 
> > Numbering Plan (E.164/E.163) (1) '15592982000' ]
> -- Called g1/15592982000
> < Protocol Discriminator: Q.931 (8)  len=10
> < Call Ref: len= 2 (reference 32802/0x8022) (Terminator)
> < Message type: CALL PROCEEDING (2)
> < Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0
>  < Ext: 1  Coding: 0   Number Specified   Channel Type: 3
> < Ext: 1  Channel: 1 ]
> -- Processing IE 24 (Channel Identification)
> < Protocol Discriminator: Q.931 (8)  len=9
> < Call Ref: len= 2 (reference 32802/0x8022) (Terminator)
> < Message type: ALERTING (1)
> < Progress Indicator (len= 2) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   
> Location: Private network serving the local user
> (1)
> < Ext: 1  Progress Description: Called equipment is non-ISDN. (2) ]
> -- Processing IE 30 (Progress Indicator)
> -- Zap/1-1 is ringing
> -- Registered SIP '99051' at 209.234.100.2 port 1175 expires 120
> < Protocol Discriminator: Q.931 (8)  len=5
> < Call Ref: len= 2 (reference 32802/0x8022) (Terminator)
> < Message type: CONNECT (7)
> > Protocol Discriminator: Q.931 (8)  len=5
> > Call Ref: len= 2 (reference 34/0x22) (Originator)
> > Message type: CONNECT ACKNOWLEDGE (15)
> -- Zap/1-1 answered SIP/99050-a3fe
> > Protocol Discriminator: Q.931 (8)  len=9
> > Call Ref: len= 2 (reference 34/0x22) (Originator)
> > Message type: DISCONNECT (69)
> > Cause (len= 2) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   Location: Private 
> > network serving the local user (1)
> >Ext: 1  Cause: Normal Clearing (16), class = Normal Event (1) ]
> -- Hungup 'Zap/1-1'
> == Spawn extension (trusted, 15592982000, 1) exited non-zero on 
> 'SIP/99050-a3fe'
> < Protocol Discriminator: Q.931 (8)  len=5
> < Call Ref: len= 2 (reference 32802/0x8022) (Terminator)
> < Message type: RELEASE (77)
> > Protocol Discriminator: Q.931 (8)  len=5
> > Call Ref: len= 2 (reference 34/0x22) (Originator)
> > Message type: RELEASE COMPLETE (90)
>
> 
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Re: [Asterisk-Users] Asterisk hogging CPU resources

[Martin Pycko]

Well if you have lots of /dev/timer opened than you have to edit your
asterisk/Makefile and comment out ZAPTEL_OPTIMIZATIONS or something like
that.

Martin

On Fri, 20 Jun 2003, Derek Beaumont wrote:

> What is the recommended version of mpg123?
> I am running 0.59r
>
> 
> I just had this problem and Marin found it to be the fact that I was
> not running the recommended version of mpg123
>
> Try starting there
>
> John
>
> On Friday, June 20, 2003, at 12:03  PM, Steven Critchfield wrote:
>
> > This blows my main idea of not having a timing source to keep asterisk
> > from entering a busy loop.
> >
> > Are you running the most current CVS? I know there had been a bug some
> > time back that caused every asterisk thread to open handles on
> > /dev/zap/timer repeatedly and at some point my system had run out of
> > file handles to give out and performance started sucking. A CVS
> upgrade
> > fixed that. Oddly enough too was that it only happened on 1 of my 3
> > asterisk machines.
> >
> > On Fri, 2003-06-20 at 10:06, Derek Beaumont wrote:
> >> The interfaces I'm using are 2 X100Ps and a TDM400P
> >>
> >>
> >>
> >> 
> >> What kind of interfaces are you using?
> >>
> >> I'm using zap and IAX on my main asterisk server that deals in about
> >> 300-400 calls a day without the cpu load you are seeing.
> > --
> > Steven Critchfield  <[EMAIL PROTECTED]>
> >
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Re: [Asterisk-Users] databases for billing

[Martin Pycko]

cdr_mysql.conf

On Fri, 20 Jun 2003, carlos del mayor wrote:

> hi
> I want to do a database to save the cdr with a billing finality. I've created the 
> database in mysql (thanks for the table and all that!) but I'm not sure of how to 
> 'connect' asterisk to that database in order to save there the cdr. Is the 
> cdr_mysql.conf what I have to config? Or must I do a script, with the 'database' AGI 
> commands?
> Any help would be so apreciated!
> Thanks a lot
> carlos
>
>
> -
> Yahoo! Sorteos
> Juega a la Lotería Primitiva sin salir de casa



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Re: [Asterisk-Users] Billsec on CDR

[Martin Pycko]

You need to change the FREQs for the events. I don't know exactly how the
code works. There was someone on the list that claimed to have the UK
callprogress working.

regards
Martin

On Fri, 20 Jun 2003, Tan Aks wrote:

> Isn't there any way to make callprogress work for people in Europe? What is
> it that is needed to make it work?
>
> T
>
>
>
> - Original Message -
> From: "Martin Pycko" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Thursday, June 19, 2003 11:36 PM
> Subject: Re: [Asterisk-Users] Billsec on CDR
>
>
> It has to do with the fact that with analog channels like FXO
> we don't have a way to tell whether the call has been answered or not.
> So after the interfaces sends the called number we assume that the
> call got answered. This happens unless you have callprogress=yes
> in zapata.conf. But it's designed to be working only in US.
>
> Martin
>
> On Thu, 19 Jun 2003, Dan Fernandez wrote:
>
> > I have an X100P and when I place calls to the PSTN which are not answered,
> the Billsec field of the CDR still logs the seconds that the phone rang.
> >
> > Can someone please confirm that this has to do with the ringcadance of the
> indications.conf file? Is there anything else I need to check ?
> >
> > Thanks in advance
> >
>
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Re: [Asterisk-Users] Billsec on CDR

[Martin Pycko]

It has to do with the fact that with analog channels like FXO
we don't have a way to tell whether the call has been answered or not.
So after the interfaces sends the called number we assume that the
call got answered. This happens unless you have callprogress=yes
in zapata.conf. But it's designed to be working only in US.

Martin

On Thu, 19 Jun 2003, Dan Fernandez wrote:

> I have an X100P and when I place calls to the PSTN which are not answered, the 
> Billsec field of the CDR still logs the seconds that the phone rang.
>
> Can someone please confirm that this has to do with the ringcadance of the 
> indications.conf file? Is there anything else I need to check ?
>
> Thanks in advance
>

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Re: [Asterisk-Users] VoiceMail recording dialtone

[Martin Pycko]

Well experiment yourself with the code.

in wcfxo.c
/* Don't accept a ring for another 1000 ms */
wc->ringdebounce = 1000;

Try a diffrent value (e.g. 3000 for 3 sec)
and in zaptel.h

#defineRING_DEBOUNCE_TIME  500 /* 500 ms ring debounce
time */

try the same value as in wcfxo.c

recompile/reload and test

regards
Martin

On Thu, 19 Jun 2003, Sam Bingner wrote:

> Zaptel was the version from about 4 days ago when I sent this message, I
> updated again yesterday night
>
> Sam
>
> Quoting Martin Pycko <[EMAIL PROTECTED]>:
>
> > How old is your zaptel code ?
> > Mark recently increased some timer for that.
> >
> > Martin
> >
> > On Wed, 18 Jun 2003, Sam Bingner wrote:
> >
> > > I have an extension setup with voicemail, for incoming calls on an X100P
> > > card.  It quite often will record about 15 seconds of dialtone... I'm
> > > guessing that it picks up the line after the outgoing line has been
> > > disconnected.
> > >
> > > Has anybody else run into this problem?  Shouldn't chan_zap be detecting
> > > the hangup and ending the connection?
> > >
> > > Sam
> > >
> >
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>
>
>
>
> -
> This mail sent through IMP: http://horde.org/imp/
>
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Re: [Asterisk-Users] VoiceMail recording dialtone

[Martin Pycko]

How old is your zaptel code ?
Mark recently increased some timer for that.

Martin

On Wed, 18 Jun 2003, Sam Bingner wrote:

> I have an extension setup with voicemail, for incoming calls on an X100P
> card.  It quite often will record about 15 seconds of dialtone... I'm
> guessing that it picks up the line after the outgoing line has been
> disconnected.
>
> Has anybody else run into this problem?  Shouldn't chan_zap be detecting
> the hangup and ending the connection?
>
> Sam
>

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Re: [Asterisk-Users] asterisk -rx under cron?

[Martin Pycko]

This works for me.
Martin

#!/usr/bin/perl -w
use Socket;
use IO::Handle;

socket(SOCK, AF_INET, SOCK_STREAM, getprotobyname('tcp'))
or die "Cannot create a socket: $!\n";
connect(SOCK, sockaddr_in(5038, inet_aton('localhost')))
or die "Cannot connect to the manager port\n";
SOCK->autoflush(1);
$text = "Action: Login\r\n";
$text .= "Username: mark\r\n";
$text .= "Secret: pass\r\n\r\n";
$text .= "Action: Command\r\nCommand: show channels\r\n\r\n";
print SOCK $text;
while () {
print if not /Message:/ and not /Response:/ and not /END COMMAND/;
exit 0 if /END COMMAND/
}

exit 0;


On Thu, 19 Jun 2003, Christopher Arnold wrote:

>
>
> On Wed, 18 Jun 2003, Jeremy McNamara wrote:
>
> > Why not write a [insert favorite scripting language here] script to use
> > the Asterisk manager interface?
> >
> > I've wrote quite a few little perl scripts each doing their own specific
> > function.  Works quite well.
> >
> Could you give us a pointer to info about the Asterisk manager interface?
> Perhaps even an example script in your favorite scripting language ?
>
>   /Chris
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Re: [Asterisk-Users] SNOM 200 and MWI??

[Martin Pycko]

You could always have

exten => asterisk,1,VoicemailMain

Martin

On Wed, 18 Jun 2003, Test wrote:

> Does anyone know if this was implemented? If not then where should I look to
> try and make the mod?
>
> Thanks
> Tan
>
>
>
> - Original Message -
> From: "WipeOut ." <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Monday, April 28, 2003 9:03 AM
> Subject: Re: [Asterisk-Users] SNOM 200 and MWI??
>
>
> Hi Mark,
>
> If you decied to impliment this please let me know when it is done..
>
> Thanks..
>
> > Sounds like the SNOM expects to use our "Contact" to get a hold of us.  It
> > should be simple to add something like "voicemail=" in the general
> > section for setting the voicemail extension to use in the contact area.
> >
> > Mark
> >
> > On Thu, 24 Apr 2003, WipeOut . wrote:
> >
> > > Here is the trace if anyone is interested..
> > >
> > > NOTIFY sip:[EMAIL PROTECTED] SIP/2.0
> > > Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK523b1b63
> > > From: "asterisk" ;tag=as3da6a846
> > > To: 
> > > Contact: 
> > > Call-ID: [EMAIL PROTECTED]
> > > CSeq: 102 NOTIFY
> > > User-Agent: Asterisk PBX
> > > Event: message-summary
> > > Content-Type: application/simple-message-summary
> > > Content-Length: 36
> > >
> > > Message-Waiting: yes
> > > Voicemail: 1/0
> > >
> > >
> > > > Hi,
> > > >
> > > > The MWI is working on the SNOM 200 but the problem is that when you
> press the MWI button it attempts to dial
> > > > "asterisk" 
> > > > where 192.168.1.200 is the IP address of my * box.
> > > >
> > > > How can I modify this so the return path is correct, which on my setup
> is extension 8500 for voicmailmain??
> > > >
> > > > Thanks
> > > > --
> > > > __
> > > > http://www.linuxmail.org/
> > > > Now with e-mail forwarding for only US$5.95/yr
> > > >
> > > > Powered by Outblaze
> > > > ___
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> > >
> > > --
> > > __
> > > http://www.linuxmail.org/
> > > Now with e-mail forwarding for only US$5.95/yr
> > >
> > > Powered by Outblaze
> > > ___
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> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> > ___
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>
> --
> __
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> Now with e-mail forwarding for only US$5.95/yr
>
> Powered by Outblaze
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Re: chan_agent MOH was (Re: [Asterisk-Users] CVS Error 2003-06-19)

[Martin Pycko]

You can call setmusiconhold app and as an argument call class silence,
off, or whatever non-existant class and it works now.

Martin

On Wed, 18 Jun 2003, TC wrote:

> Yea, I have faked that with a silent mp3,
> but to do it right it should also be a config flag in the agent.conf file
> for each agent, prolly add another arg to each agent definition
> for the MOH class, & the arg 'none' means don't play music for that agent
>
>
>
> -Original Message-
> From: James Golovich <[EMAIL PROTECTED]>
> To: [EMAIL PROTECTED] <[EMAIL PROTECTED]>
> Date: June 18, 2003 9:17 AM
> Subject: Re: [Asterisk-Users] CVS Error 2003-06-19
>
>
> >
> >
> >
> >On Wed, 18 Jun 2003, John Congdon wrote:
> >
> >> That is exactly what happened.  I commented out the music on hold,
> >> none of my Customer Service reps like it.  They would rather listen to
> >> silence.
> >
> >Perhaps this would be a good feature to add.  It should be very easy to
> >implement.  We could have a hardcoded class called 'off', 'silence', or
> >'none' that would just not start the moh generator.
> >
> >Anyone have any thoughts on this?
> >
> >James
> >
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Re: [Asterisk-Users] Temporized AGI Scripts.

[Martin Pycko]

You can use 'at' utility to copy a file that you prepare
one you execute AGI script. Look at asterisk/sample.call.

Martin

On Wed, 18 Jun 2003, Xisco Mateu wrote:

> Hi all,
>
> Now I'm working with a E400P, and I don't now if it's possible to do the following. 
> I want that and AGI script (Perl) recieve a call, and the user introduce the date, 
> the time and the destination phone number (where the temporized AGI must call). 
> Before an AGI script will call to that number in the date and time introduced by 
> user. That's possible, and it's how can I do it¿?¿?
>
> Thanks a lot.


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Re: [Asterisk-Users] Errors when compiling from CVS this morning

[Martin Pycko]

That's what you get when you modify your code and that modification is in
conflict with the CVS.

Martin

On Wed, 18 Jun 2003, John Congdon wrote:

> O -fPIC-c -o chan_agent.o chan_agent.c
> chan_agent.c: In function `login_exec':
> chan_agent.c:595: parse error before '<<' token
> chan_agent.c:602: parse error before '>>' token
> make[1]: *** [chan_agent.o] Error 1
> make[1]: Leaving directory `/usr/src/asterisk/channels'
>
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Re: [Asterisk-Users] New busydetect routines for analog channels(FXO mostly)

[Martin Pycko]

Well it was in

#error You can't
  ^

sorry about that.

Martin

On Wed, 18 Jun 2003, The Traveller wrote:

> Yo Martin,
>
> On Tue, Jun 17, 2003 at 17:03:15 -0500, Martin Pycko wrote:
>
> > Hello,
> >
> > I've commited the new busydetect routine to CVS.
> > You need to cvs update asterisk of course and then choose it
> > in asterisk/Makefile and recompile asterisk.
> [...]
>
> It fails to compile here (Redhat 9, gcc version 3.2.2 20030222 (Red Hat
> Linux 3.2.2-5)):
>
> dsp.c:1055:15: missing terminating ' character
> make: *** [dsp.o] Error 1
>
> Seems to be something simple, like the need to escape the "'".
>
>
> Grtz,
>
>   Oliver
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Re: [Asterisk-Users] X100P Dialing either Too Soon or Too Fast?

[Martin Pycko]

Did you try to use 'w' as a digit before dialing the number like this:

exten => _X.,1,Dial,Zap/1/w${NUMBER}

You could also try to put 'w' inbetween the digits.

regards
Martin

On Tue, 17 Jun 2003, John Laur wrote:

>
>
> Quite frequently, outgoing calls from the X100P cards here will not dial
> properly. Instead of hearing the ringing after the Zap interface picks
> up, I'll hear silence for a while then the 'If you'd like to make a call
> please hang up and try again' recording as if zaptel picked up the line,
> punched only a couple digits and then left it.
>
> I assume that either zaptel is sending the dtmf before the telco is
> ready to get it or is sending the digits too fast. I have tried setting
> toggling overlapdial in zaptel.conf, and I have tried fiddling with the
> timing parameters a little, but I don't know what each number really
> does (and think that bad values here may be bad anyway)
>
> Does anyone have any ideas to accomplish:
>
> 1) Delay zaptel sending dtmf by a few milliseconds
> 2) Have zaptel send dtmf digits slower
>
> Thanks,
> John
>
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[Asterisk-Users] New busydetect routines for analog channels (FXO mostly)

[Martin Pycko]

Hello,

I've commited the new busydetect routine to CVS.
You need to cvs update asterisk of course and then choose it
in asterisk/Makefile and recompile asterisk.
All you X100P users that had the problems
with false hangups or the card not being able to detect the busy tone
please check that.

In the asterisk/Makefile you need to find a line
BUSYDETECT =

and uncomment what you want/ comment out what you don't want.

Also in /etc/asterisk/zapata.conf you need to have
busydetect=yes
busycount=

where  should be betwen 5 and 15. The best value to try is
busycount=10

Also if you have some strange signal like 3 beeps (200 ms tone, 200 ms
silence) and then silence of e.g. 500 ms then you need to comment out:

BUSYDETECT+= #-DBUSYDETECT_TONEONLY

But then you can't use it with

BUSYDETECT+= #-DBUSYDETECT_COMPARE_TONE_AND_SILENCE

The last 'flag' should enforce detection (less false hangups)
although I tested the algorithm without
-DBUSYDETECT_TONEONLY nor -DBUSYDETECT_COMPARE_TONE_AND_SILENCE
with busycount = 10 and after 1 hour of conversation I didn't have any
false hangups.

regards
Martin

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Re: [Asterisk-Users] Parking causes crash

[Martin Pycko]

Describe that a little bit.
The call came on what interface and then you dialed what interface
and how did you park it ? You pressed a flash button or '#' key ?

Martin

On Tue, 17 Jun 2003, John Congdon wrote:

> Has this been solved?  When I park a call, the caller hears a second of
> music on hold
> and then the whole system crashes.
>
> I can restart with a simple (asterisk -cvvv), I don't have to reboot or
> anything
>
> John
>
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Re: [Asterisk-Users] Directory Application question

[Martin Pycko]

We have it done at Digium so it can be done.
Just record your name I guess with voicemail but I'm not entirely sure
about that you can record that in voicemail.

Martin

On Tue, 17 Jun 2003, Derek Beaumont wrote:

> I'm wondering if I can do the following:
>
>   Caller activates the Directory application
>   Caller enters the first 3 digits of a person's last name
>   =
>   Normally here, Asterisk will say the extension number of a
> person found.
>   Is there a way to get Asterisk to say the name as well? (perhaps
> using the same sound file that is used
>   for their name in the voicemail application)
>
> Can this be done?
>
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Re: [Asterisk-Users] play music in background, while wait in a queue

[Martin Pycko]

Do you have '-z' option with the definition of random in musiconhold.conf
? actually I just did see the options of mpg123 and it has to be an
uppercase Z:

-Z

Martin

On Tue, 17 Jun 2003, Rafael Gonzalez Lomeña wrote:

> Hello to all,
>
>   I want to put incoming calls in a queue and that user hear a "beauty song"
> :-)
>
>   But, although I think that parameters in the file "queues.conf" are
> corrected; is not possible to listen any melody ... and nevertheless,
> the message of " Started music on hold, class 'random', on H323:"
> appears on the CLI.
>
>   Are know some idea?
>
>   Thank's in advance
>
>
>   raglom
>
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Re: [Asterisk-Users] CLID trouble

[Martin Pycko]

Just do the "pri debug span 1" and see for yourself that asterisk sends
that. You might however send it without one digit or something ... or
maybe your telco doesn't support it. Just give then a call.

Martin

On Tue, 17 Jun 2003, Tom De Wispelaere wrote:

> Hey Martin, thnx for your reply,
>
> I've tried it out as you said by setting it explicitly in my
> extensions.conf as follows :
>
> exten => number,1,Wait,1
> exten => number,2,Answer
> exten => number,3,SetCallerID("somename" <0>)
> exten => number,4,Dial,Zap/g2/${phonenumber}
>
> in the console it says
>
>  -- Executing Answer("Zap/6-1", "") in new stack
>  -- Executing SetCallerID("Zap/6-1", ""somename" <0>") in new stack
>  -- Executing Dial("Zap/6-1", "Zap/g2/X") in new stack
>
> I still dont get any CLID on the receiving phone however :(
>
> Maybe i should contact the telco about this ?
>
> Regards,
>
> Tom
>
>
>
> On Tue, 17 Jun 2003, Martin Pycko wrote:
>
> > Try to explicitly add this line
> > ,1,SetCallerid,("somename" <12345>)
> > ,2,Dial,Zap/g1/${phonenumber}
> >
> > regards
> > Martin
> > On Tue, 17 Jun 2003, Tom De Wispelaere wrote:
> >
> > > Hey all,
> > >
> > > I have a E1 setup with a E400P digium card. Everything works just great
> > > except for the callerid. When i make an outgoing call via the E1 to a
> > > hardphone somewhere, all i get is "private number". In my zapata.conf
> > > however , i have defined the following:
> > >
> > > context=localE1
> > > group = 1
> > > channel=1-15
> > > group = 2
> > > callerid="somename" 
> > > channel=17-31
> > >
> > > and I explicitly dial out with something alla
> > > "Dial,Zap/g2/${phonenumber}"
> > >
> > > Am i forgetting something ? Thnx for any advice.
> > >
> > > Ciao
> > >
> > > Tom
> > >
> > >
> > > ___
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> > >
> >
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Re: [Asterisk-Users] New Module app_perl

[Martin Pycko]

Of course you can use Gotoif with expressions.

Gotoif,$[${VAR} > 12]?1|4:1|5

Martin

On Tue, 17 Jun 2003, Anthony Minessale wrote:

>
> I just made my first 2 modules for asterisk (The 1st one is depriciated already).
>
> I was annoyed that i couldn't get GotoIf to take any expressions besides a boolean
> then i made a module to mimic gotoif and parse a few expressions like (${var} > 12)
> exten => 1,1,gotoif_expr,${var} > 12:1|4:1|5
>
> Then I immediatly obseleted it with this new embedded perl module that lets
> you implement stuff at will from perl instead of needing to make a module
>
> Say you want to make gotoif that can parse expressions
>
> exten => 1,1,Perl,gotoif:${var} > 12:1|4:1|5
>
> say ${var} = 4
>
> then gotoif perl sub is launched as
>
> gotoif(4,"1|4","1|5");
>
> and gotoif() looks like:
>
>
> sub gotoif(@) {
>   my($expr,$goto1,$goto2) = @_;
>   my $test;
>   eval "\$test = ($expr)";
>   my $ret = "goto:" . (($test) ? $goto1 : $goto2);
>   print "test: [$expr] = [$test] [$ret]\n";
>   $ret;
> }
>
> Funy thing is I was too obcessed with what I had already seen so far in asterisk that
> I didnt even notice AGI until yesterday =)
>
> here is my beta README
>
>
>
> app_perl 1.0
>
>
> This is app_perl the "mod_perl" of sorts for asterisk.  It is an actual live Perl 
> Interpreter
> embeded in a module so when it starts up the perl stays resident in memory the whole 
> life of the
> Asterisk process.
>
> FEATURES:
>
> It can call perl functions from extensions.conf in a special package called 
> Asterisk::Perl.
> that is loaded on startup from /etc/asterisk/Asterisk::Perl.pm.
>
> exten => 1,1,Perl,myfunc:arg_1:arg_2:arg_n..
>
> It then does it's business and returns one or more special directives that
> are asterisk related operations processed by the module.
> they are returned as a list (array) of scalar each containing a specially 
> formated
> command to feed back into asterisk via the C module.
>
> These are counterparts to the real directives found in extensions.conf only 
> probably
> there is less error checking and more direct control a.k.a. likelyhood to 
> crash.
>
>
> Valid commands so far...
>
> setvar::
> goto:|
> 
> add_ext:::
> runapp::
> include::
> ignorepat::
> switch:::
>
>
> There are also some special commands: (1 for now)
>
> thread:
> This will spin off a thread with the perl sub you specify running inside it.
> the sub will get 1 scalar arg being the number of times it was run (starting 
> with 1)
> you can return an array full of more commands listed above with 1 exception:
> if you put the keyword "loop" in the list, it will call the function again 
> and increment the counter arg.
> You can use this to say run 1 thread to listen on a tcp port and populate a 
> shared data object
> and use another to run in a loop and pass the altered data back to asterisk 
> and relaunch
> every 1 min or so.  And of core all this stuff can be used to horribly crash 
> the program.
>
> The function can return as many commands as it wants but of corse several 
> goto or other such nonsense
> may not be too smart.
>
>
> The functions startup() and shutdown() are called respectively on load and 
> unload of the module
> allowing you to create on the fly contexts and configuration by looking up 
> data from a database,more files etc.
> you can even make the dialing of a certian extension cause a perl func to 
> create more extensions or to
> rewrite the existing one on the fly.  Of course, you can't do any channel 
> related commands (runapp etc)
> from the startup function because there is no call in progress so they get 
> ignored.
>
>
> BUGS:
>
> If you stare at the source code from about 2 feet back for approx 30 seconds 
> you will start
> to see that the entire thing is in itself 1 large bug.  I am not a C 
> programmer but rather
> a Perl programmer so that was my motivation for this idea but it probably is 
> sucky C style etc...
>
>
>
>
>
>
>
> -
> Do you Yahoo!?
> SBC Yahoo! DSL - Now only $29.95 per month!

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Re: [Asterisk-Users] X100P creating a short-circuit on line

[Martin Pycko]

Did you cvs update zaptel and recompiled ?

Martin

On Tue, 17 Jun 2003, K. C. Li wrote:

> On Tue, 17 Jun 2003, Mark Spencer wrote:
>
> > I'm in Paris right now and can't test this change, but I've been
> > researching the DAA and there are a few international settings I can
> > change, so I've changed the driver in CVS so that you can specify
>
> That's encouraging news.
>
> > the operational mode.  Try "modprobe wcfxo opermode=1" if you're in most
> > of Europe and that should switch to CTR21 mode which slightly modifies a
> > few of the electrical characteristics of the DAA.
>
> "modprobe wcfxo opermode=1" was rejected with the following error:
>
> /lib/modules/2.4.20/misc/wcfxo.o: invalid parameter parm_opermode
> /lib/modules/2.4.20/misc/wcfxo.o: insmod /lib/modules/2.4.20/misc/wcfxo.o failed
> /lib/modules/2.4.20/misc/wcfxo.o: insmod wcfxo failed
>
> I couldn't find "opermode" mentioned anywhere in the source tree either.
>
> Regards,
>
> Kwong Li
> [EMAIL PROTECTED]
> Laser Business Systems Ltd.
> http://www.laser.com
>
>
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Re: [Asterisk-Users] (no subject)

[Martin Pycko]

Try to explicitly add this line
,1,SetCallerid,("somename" <12345>)
,2,Dial,Zap/g1/${phonenumber}

regards
Martin
On Tue, 17 Jun 2003, Tom De Wispelaere wrote:

> Hey all,
>
> I have a E1 setup with a E400P digium card. Everything works just great
> except for the callerid. When i make an outgoing call via the E1 to a
> hardphone somewhere, all i get is "private number". In my zapata.conf
> however , i have defined the following:
>
> context=localE1
> group = 1
> channel=1-15
> group = 2
> callerid="somename" 
> channel=17-31
>
> and I explicitly dial out with something alla
> "Dial,Zap/g2/${phonenumber}"
>
> Am i forgetting something ? Thnx for any advice.
>
> Ciao
>
> Tom
>
>
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Re: [Asterisk-Users] GASTMAN AUTH QUESTION

[Martin Pycko]

/etc/asterisk/manager.conf

Martin

On Sun, 15 Jun 2003, Alvaro Parres wrote:

> Hi,
>
>Any of you know where to define the user and password for gastman.???
>
>
> PLEAS HELP ME!
>
> Alvaro Parres
>
>
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Re: [Asterisk-Users] CallerID forward???

[Martin Pycko]

Then I guess in zapata.conf before the definition of the

callerid=asreceived
channel => 1;FXO port

Martin

On Fri, 13 Jun 2003, Derek Beaumont wrote:

> I don't understand how or where I would use setcallerid.
> I have tried to do
> exten=>400,1,Setcallerid,asreceived
> but that doesn't seem to work
>
>
> What am I doing wrong?
>
> -Derek
>
>
> 
> ==
> Check "show application setcallerid"
>
> Martin
>
> On Fri, 13 Jun 2003, Derek Beaumont wrote:
>
> > I only want to do this internally, from the reception phone to another
> > phone attached to my asterisk box.
> > I am using X100P and TDM400P.
> >
> > -Derek
> >
> >
> >
> > >>Over what interfaces ? (voip, analog t1, pri ?)
> > >>In general when you want to send it over T1 to the telco and further
> > on to
> > >>PSTN than it might not be possible since you're allowed most of the
> > times
> > >>to send the callerid that is one of your assigned DID numbers.
> >
> > >>regards
> > >>Martin
> >
> > On Fri, 13 Jun 2003, Derek Beaumont wrote:
> >
> > > Here is the situation that I would like to create:
> > >   Call comes in
> > >   Receptionist sees that the caller ID is Jenny <8675309>
> > >   Receptionist picks up phone and transfers call to Batman
> > >   Batman looks at his phone and sees that the caller ID is Jenny
> > > <8675309>
> > >
> > > I can't seem to figure out how to forward the caller ID.  Is this
> > > possible with Asterisk?
> > >
> > > ___
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> > > [EMAIL PROTECTED]
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [Asterisk-Users] Asterisk switch => statement

[Martin Pycko]

I think it's per context.

Martin

On Fri, 13 Jun 2003, Andy Powell wrote:

>
> So is that one switch statement per installation or one per context.
> ie can i have multiple switch statements each one applicable to a
> different section in extensions.conf
>
> Andy
>
>
> On 13/06/2003 at 13:28 Martin Pycko wrote:
>
> >The idea of switch is for every box to know what it can reach locally. And
> >then to do the 'switch' to remote boxes if the called number can't be find
> >locally.
> >
> >Martin
>
>
>
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Re: [Asterisk-Users] CallerID forward???

[Martin Pycko]

Check "show application setcallerid"

Martin

On Fri, 13 Jun 2003, Derek Beaumont wrote:

> I only want to do this internally, from the reception phone to another
> phone attached to my asterisk box.
> I am using X100P and TDM400P.
>
> -Derek
>
>
>
> >>Over what interfaces ? (voip, analog t1, pri ?)
> >>In general when you want to send it over T1 to the telco and further
> on to
> >>PSTN than it might not be possible since you're allowed most of the
> times
> >>to send the callerid that is one of your assigned DID numbers.
>
> >>regards
> >>Martin
>
> On Fri, 13 Jun 2003, Derek Beaumont wrote:
>
> > Here is the situation that I would like to create:
> > Call comes in
> > Receptionist sees that the caller ID is Jenny <8675309>
> > Receptionist picks up phone and transfers call to Batman
> > Batman looks at his phone and sees that the caller ID is Jenny
> > <8675309>
> >
> > I can't seem to figure out how to forward the caller ID.  Is this
> > possible with Asterisk?
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
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Re: [Asterisk-Users] CallerID forward???

[Martin Pycko]

Over what interfaces ? (voip, analog t1, pri ?)
In general when you want to send it over T1 to the telco and further on to
PSTN than it might not be possible since you're allowed most of the times
to send the callerid that is one of your assigned DID numbers.

regards
Martin

On Fri, 13 Jun 2003, Derek Beaumont wrote:

> Here is the situation that I would like to create:
>   Call comes in
>   Receptionist sees that the caller ID is Jenny <8675309>
>   Receptionist picks up phone and transfers call to Batman
>   Batman looks at his phone and sees that the caller ID is Jenny
> <8675309>
>
> I can't seem to figure out how to forward the caller ID.  Is this
> possible with Asterisk?
>
> ___
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> [EMAIL PROTECTED]
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>

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Re: [Asterisk-Users] Asterisk switch => statement

[Martin Pycko]

The idea of switch is for every box to know what it can reach locally. And
then to do the 'switch' to remote boxes if the called number can't be find
locally.

Martin

On 13 Jun 2003, Eric Wieling wrote:

> Cool.
>
> Now if I am on ast-1 and want to call 2200, which is a Zap interface on
> the ast-1 then I have to define 2200 on ast-1 and can't put it in the
> master/central dial plan on ast-2?
>
> On Fri, 2003-06-13 at 12:34, Martin Pycko wrote:
> > You're missing that then the IAX call will be started between ast1 and
> > ast2 and you'll get connected to ast2 Zap/1
> >
> > Martin
> >
> > On 13 Jun 2003, Eric Wieling wrote:
> >
> > > As I understand it (and my understanding is obviously incorrect) the
> > > switch => statement sells the Asterisk box to resolve (aka lookup)
> > > extensions by querying the remote Asterisk server defined in the switch
> > > => statement.  The switch => statement is used to centralize dialplans.
> > >
> > > I've not used the switch => statement yet, I'm just trying to understand
> > > the ramifications of using it before I try it and blow up my dial plan.
> > >
> > > Here's an example of a sample setup
> > >
> > > Asterisk Server ast-1
> > >   switch => IAX/[EMAIL PROTECTED]
> > >
> > > Asterisk Server ast-2
> > >   exten => 12010,1,Dial(Zap/1)
> > >
> > > If I'm on a device connected to ast-1 and I dial 12010 I assume ast-1
> > > asks ast-2 to resolve the extension 12010, and I also assume that ast-2
> > > returns "exten => 12010,1,Dial(Zap/1)" then ast-1 tries to Dial(Zap/1)
> > > which is not an interface on ast-1 and the call fails.
> > >
> > > What information am I missing?
> > >
> > > --Eric
> > >
> > >
> > >
> > >
> > > --
> > > BTEL Consulting
> > > 850-484-4535 x2111 (Office)
> > > 504-595-3916 x2111 (Experimental)
> > > 877-552-0838 (Backup Phone)
> > >
> > > ___
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> > > [EMAIL PROTECTED]
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> > ___
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> --
> BTEL Consulting
> 850-484-4535 x2111 (Office)
> 504-595-3916 x2111 (Experimental)
> 877-552-0838 (Backup Phone)
>
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Re: [Asterisk-Users] Call queues for phone operator

[Martin Pycko]

But he didn't think about agent. Just a regular SIP phone.
It should be in general like the original author of this thread thinks.
Besides it's easy to test so why not to test it :)

Martin

On Fri, 13 Jun 2003, TC wrote:

> >Hi.
> >
> >I was wondering how can I make incoming calls to wait if the phone
> >operator is busy. I've 8 incoming lines, with 30 extensions.
> >What I need is if the operator is busy with call nr #1 , the new
> >incoming call waits until the op. is free.
> >Looking into app_queue seems the way to go.
> Thats correct
> >So I want to ask if I'm right or wrong:
> >and when a call arrives, dial the operator and if he's busy,
> >fire up app_queue .
> NO agents log into an agent q their phone is OFF-HOOK always
> thus if you Dial that agent ext it is always busy
> >So what I expect, when the operator hangs up, his phone
> >will automagically rings playing the announce "from-queue" and
> >bridge it with the call that's waiting.
> the agent will just hear beep beep & the optional announcement
> on the handset/speakerphone or headset
> then the inbound caller is bridged
>
>
>
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Re: [Asterisk-Users] Asterisk asterisk => statement

[Martin Pycko]

You're missing that then the IAX call will be started between ast1 and
ast2 and you'll get connected to ast2 Zap/1

Martin

On 13 Jun 2003, Eric Wieling wrote:

> As I understand it (and my understanding is obviously incorrect) the
> switch => statement sells the Asterisk box to resolve (aka lookup)
> extensions by querying the remote Asterisk server defined in the switch
> => statement.  The switch => statement is used to centralize dialplans.
>
> I've not used the switch => statement yet, I'm just trying to understand
> the ramifications of using it before I try it and blow up my dial plan.
>
> Here's an example of a sample setup
>
> Asterisk Server ast-1
>   switch => IAX/[EMAIL PROTECTED]
>
> Asterisk Server ast-2
>   exten => 12010,1,Dial(Zap/1)
>
> If I'm on a device connected to ast-1 and I dial 12010 I assume ast-1
> asks ast-2 to resolve the extension 12010, and I also assume that ast-2
> returns "exten => 12010,1,Dial(Zap/1)" then ast-1 tries to Dial(Zap/1)
> which is not an interface on ast-1 and the call fails.
>
> What information am I missing?
>
> --Eric
>
>
>
>
> --
> BTEL Consulting
> 850-484-4535 x2111 (Office)
> 504-595-3916 x2111 (Experimental)
> 877-552-0838 (Backup Phone)
>
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