Re: [asterisk-users] zaptel and oslec
this is the output of dmesg [EMAIL PROTECTED] zaptel-1.4.7.1]# dmesg |tail wct1xxp: disagrees about version of symbol zt_transmit wct1xxp: Unknown symbol zt_transmit wct1xxp: disagrees about version of symbol zt_rbsbits wct1xxp: Unknown symbol zt_rbsbits wct1xxp: disagrees about version of symbol zt_unregister wct1xxp: Unknown symbol zt_unregister wct1xxp: disagrees about version of symbol zt_register wct1xxp: Unknown symbol zt_register wct1xxp: disagrees about version of symbol zt_alarm_notify wct1xxp: Unknown symbol zt_alarm_notify [EMAIL PROTECTED] zaptel-1.4.7.1]# Tzafrir Cohen wrote: > On Mon, Jan 28, 2008 at 04:37:25PM -0200, Pablo Allietti wrote: >> hi all i have a te110p installed in my system with a lot of Echo.. >> i decide to install the oslec echo supressor but when y try to add the >> module i have this problem. >> >> >> [EMAIL PROTECTED] zaptel-1.4.7.1]# insmod wct1xxp.ko >> insmod: error inserting 'wct1xxp.ko': -1 Unknown symbol in module >> [EMAIL PROTECTED] zaptel-1.4.7.1]# > > That's because there's a module missing. > > Why not just use modprobe? > > If you actually want to figure out what module it was: > > dmesg| tail > > (and it is probably either oslec or zaptel) > > Or, if you used insmod because you want modules from your working > directory, then please help me improve > http://svn.digium.com/svn/zaptel/branches/1.4/zap_auto > -- .- Pablo Allietti E-mail: [EMAIL PROTECTED] | LACNIC Phone : +598 2 604 | http://LACNIC.NET ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel and oslec
this is the output of dmesg [EMAIL PROTECTED] zaptel-1.4.7.1]# dmesg |tail wct1xxp: disagrees about version of symbol zt_transmit wct1xxp: Unknown symbol zt_transmit wct1xxp: disagrees about version of symbol zt_rbsbits wct1xxp: Unknown symbol zt_rbsbits wct1xxp: disagrees about version of symbol zt_unregister wct1xxp: Unknown symbol zt_unregister wct1xxp: disagrees about version of symbol zt_register wct1xxp: Unknown symbol zt_register wct1xxp: disagrees about version of symbol zt_alarm_notify wct1xxp: Unknown symbol zt_alarm_notify [EMAIL PROTECTED] zaptel-1.4.7.1]# Tzafrir Cohen wrote: > On Mon, Jan 28, 2008 at 04:37:25PM -0200, Pablo Allietti wrote: >> hi all i have a te110p installed in my system with a lot of Echo.. >> i decide to install the oslec echo supressor but when y try to add the >> module i have this problem. >> >> >> [EMAIL PROTECTED] zaptel-1.4.7.1]# insmod wct1xxp.ko >> insmod: error inserting 'wct1xxp.ko': -1 Unknown symbol in module >> [EMAIL PROTECTED] zaptel-1.4.7.1]# > > That's because there's a module missing. > > Why not just use modprobe? > > If you actually want to figure out what module it was: > > dmesg| tail > > (and it is probably either oslec or zaptel) > > Or, if you used insmod because you want modules from your working > directory, then please help me improve > http://svn.digium.com/svn/zaptel/branches/1.4/zap_auto > -- .- Pablo Allietti E-mail: [EMAIL PROTECTED] | LACNIC Phone : +598 2 604 | http://LACNIC.NET ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] zaptel and oslec
hi all i have a te110p installed in my system with a lot of Echo.. i decide to install the oslec echo supressor but when y try to add the module i have this problem. [EMAIL PROTECTED] zaptel-1.4.7.1]# insmod wct1xxp.ko insmod: error inserting 'wct1xxp.ko': -1 Unknown symbol in module [EMAIL PROTECTED] zaptel-1.4.7.1]# some advice? thanks. -- .- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Everyone is busy/congested: IP Trunk
On Fri, Oct 26, 2007 at 06:55:12AM -0700, bilal ghayyad wrote:
> Hi List;
Ip address to destination?
Unable to create channel of type SIP (cause 3 - No
route to destination)
i think you have the wrong ip information
>
> I established an SIP IP Trunk between Asterisk and
> another softswitch (asterisk registered on the
> softswitch successfully) and I saw this on the
> softswitch.
>
> >From firefly softphone, I was need to do a call to be
> via this softswitch (ofcourse, the softphone will send
> for asterisk and asterisk should route to the
> softswitch based on the extensions.conf
> configurations.
>
> But, always I receive this message (and the call does
> not even reach to the softswitch, it is not sended
> from Asterisk to the softswitch):
>
> Executing [EMAIL PROTECTED]:1]
> Dial("SIP/EgyptOeratorSIP-09f9bed0",
> "SIP/[EMAIL PROTECTED]") is new stack
>
> Unable to create channel of type SIP (cause 3 - No
> route to destination)
>
> Everyone is busy/congested at this time (1:0/0/1)
>
> Anyone faced that?
>
> Is it related to a paramater that control number of
> allowed channels per IP trunk? Maybe I have such
> parameters is 0 ? I do not know even if there is such
> parameter.
>
> At the softswitch, I do not see even any attempt
> (nothing related to the dialed number), so why
> Asterisk does not send the called number to the
> softswitch and why asterisk assume there is not
> available channel?
>
> The softphone codec is g729a and the softswitch
> support such codec. Also, if it is a codec matter,
> then call should be send to the softswitch, and the
> softswitch will gives an error related to the codec
> missmatch.
>
> Any help?
>
> Regards
> Bilal Ghayad
>
> __
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.-
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E-mail: [EMAIL PROTECTED] | LACNIC
Phone : +598 2 604 | http://LACNIC.NET
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[asterisk-users] Congested/busy
hi all i have a TE110P connected to my PBX when i try to call a
extension number in other location 3525 the asterisk give me a error
-- User entered '3525'
-- Executing [EMAIL PROTECTED]:4] GotoIf("Zap/31-1", "0?6:5") in new
stack
-- Goto (lacnicuy,450,5)
-- Executing [EMAIL PROTECTED]:5] Dial("Zap/31-1",
"IAX2/lacnic:[EMAIL PROTECTED]/3525|30|r") in new stack
-- Called lacnic:[EMAIL PROTECTED]/3525
-- IAX2/nicbr-1 is circuit-busy
[Oct 11 10:08:02] NOTICE[2763]: chan_iax2.c:2925 __auto_congest:
Auto-congesting call due to slow response
-- Hungup 'IAX2/nicbr-1'
[Oct 11 10:08:02] NOTICE[2781]: cdr.c:434 ast_cdr_free: CDR on channel
'IAX2/nicbr-1' not posted
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [EMAIL PROTECTED]:6] Hangup("Zap/31-1", "") in new stack
== Spawn extension (lacnicuy, 450, 6) exited non-zero on 'Zap/31-1'
-- Hungup 'Zap/31-1'
anybody can help me with this? thanks
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[asterisk-users] show logged clients
hi all, in console mode how i can display the logged users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Echo cancell
hi all i have a newbrand phone Linksys spa941 and i realize that my asterisk have ECHO. :( in the zaptel file i have this parameters echocancel=128 echocancel=yes echocancelwhenbridged=yes echotraining=yes i need to add any other parameter to cancel the echo? thanks. any tips? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: g729 or another
On Fri, Jun 09, 2006 at 04:45:51PM -0400, William Piper wrote: > >GSM and what is the size in KB that gsm spent? > > > >bp > > > On 6/9/06, Pablo Allietti <[EMAIL PROTECTED]> wrote: > > hi all, i saw in digium that the codec g729 is not free. exist > another > codec with low bandwith to use in asterisk for free? > -- > .- > Pablo Allietti > E-mail: [EMAIL PROTECTED] | LACNIC > Phone : +598 2 604 | [3]http://LACNIC.NET > VoIp: [EMAIL PROTECTED] > ___ > --Bandwidth and Colocation provided by [5]Easynews.com -- > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >[6]http://lists.digium.com/mailman/listinfo/asterisk-users > > References > >1. mailto:[EMAIL PROTECTED] >2. mailto:[EMAIL PROTECTED] >3. http://LACNIC.NET/ >4. mailto:[EMAIL PROTECTED] >5. http://Easynews.com/ >6. http://lists.digium.com/mailman/listinfo/asterisk-users > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ---end quoted text--- -- .- Pablo Allietti E-mail: [EMAIL PROTECTED] | LACNIC Phone : +598 2 604 | http://LACNIC.NET ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Linksys SPA-941 NAT?
On Tue, Jun 13, 2006 at 09:53:36AM +0200, Filip Dr?gowski wrote: > Very nice phones. There is no problem when conected to Asterisk (for > about 6 months now) > >any body know this phone? support NAT? and standart codecs of asterisk ? thank you all!! > > > > -FD > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ---end quoted text--- -- .- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Linksys SPA-941 NAT?
any body know this phone? support NAT? and standart codecs of asterisk ? -- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] g729 or another
hi all, i saw in digium that the codec g729 is not free. exist another codec with low bandwith to use in asterisk for free? -- .- Pablo Allietti E-mail: [EMAIL PROTECTED] | LACNIC Phone : +598 2 604 | http://LACNIC.NET VoIp: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: meetme public
On Thu, Jun 08, 2006 at 12:03:32AM +0100, Marco Mouta wrote:
>
Marco. i solve this creating adding the meetme extension in the default
context. this extension now is valid for any user.
>Hi,
>Please check you [general] section in sip.conf
>; If you need to answer unauthenticated calls, you should change this
>; next line to 'from-trunk', rather than 'from-sip-external'.
>; You'll know this is happening if when you call in you get a message
>; saying "The number you have dialed is not in service. Please check
>the
>; number and try again."
>context = from-sip-external ; Send unknown SIP callers to this context
>callerid = Unknown
>It could be happening that your public sip call is arriving @
>asterisk, and seems unknow, so it is sent to from-sip-external
>context.
>In your extensions.conf look for section called [from-sip-external],
>there you need to paste your code to route the call to your meetme
>room.
>Hope it helps,
>Best regards,
>Marco Mouta
>Ps. Please give me some feeback if it solved.
>
>On 6/7/06, Pablo Allietti <[EMAIL PROTECTED]> wrote:
>
> hi all i have an asterisk working and i need to add a mettme public
> service.
> for example i need to download a soft (sjphone) and without any
> configuration call to [EMAIL PROTECTED] (meetme) and
> join a conference but when i do that i
> received an error saying nomber do not exist. but if i call a
> extension
> is work propperly.
> in the extensions.conf have
> exten => 411,1,Answer
> exten => 411,2,Wait(1)
> exten =>
> 411,3,SetVar(CALLFILENAME=/var/spool/asterisk/monitor/${TIMESTAMP})
> exten => 411,4,Monitor(wav,${TIMESTAMP},m)
> exten => 411,5,Meetme(4001,qM)
> exten => 411,6,Hangup
> 4001 is the room number
> in the mmetme conf have
> conf => 4001
> any comments?
> --
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>
>--
>Com os melhores cumprimentos,
>Marco Mouta
>
> References
>
>1. mailto:[EMAIL PROTECTED]
>2. mailto:[EMAIL PROTECTED]
>3. http://Easynews.com/
>4. http://lists.digium.com/mailman/listinfo/asterisk-users
> _______
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Phone : +598 2 604 | http://LACNIC.NET
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[Asterisk-Users] meetme public
hi all i have an asterisk working and i need to add a mettme public
service.
for example i need to download a soft (sjphone) and without any
configuration call to [EMAIL PROTECTED] (meetme) and join a conference but when
i do that i
received an error saying nomber do not exist. but if i call a extension
is work propperly.
in the extensions.conf have
exten => 411,1,Answer
exten => 411,2,Wait(1)
exten =>
411,3,SetVar(CALLFILENAME=/var/spool/asterisk/monitor/${TIMESTAMP})
exten => 411,4,Monitor(wav,${TIMESTAMP},m)
exten => 411,5,Meetme(4001,qM)
exten => 411,6,Hangup
4001 is the room number
in the mmetme conf have
conf => 4001
any comments?
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[Asterisk-Users] hangup on silence?
is possible to define a parameter to, hangup the line on silent? or ping dead or something? because all line have busy after the pc hangup :( -- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] calls only for logging users
hi all i have a asterisk configured and working perfectly. but i have a problem. if i download a softphone for example sjphone and digit for example [EMAIL PROTECTED] i receive this call. is possible to block this? i only want to received calls for login users... -- .- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dumb question... block 00
hi all a dumb question.. how i do to block the 00 for certain sips extensions? for example i have the extensions 400 to 500 i need to extension higher than 429 can't digit 00 in my extensions.conf i have exten => 420,1,Dial(SIP/420,20) exten => 420,2,Hangup exten => 421,1,Dial(SIP/421,20) exten => 421,2,Hangup exten => 430,1,Dial(SIP/430,20) exten => 430,2,Hangup exten => 431,1,Dial(SIP/431,20) exten => 431,2,Hangup is that possible? thanks a lot. -- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Solved] who is online
On Thu, Dec 29, 2005 at 02:36:05AM +1100, Adrian Carter wrote: > >you need to set the extensions paramters to qualify=yes or >qualify= and then FOP (flash operator panel) will reflect the >status of the extensions. > Pablo Allietti wrote: yep. this solve my problem Thanks!! > > On Wed, Dec 28, 2005 at 09:15:15AM -0600, Kevin P. Fleming wrote: > > > Pablo Allietti wrote: > > > hi all i use asdterisk in my company with Flash Panel Operator to know > who is talking or ringing. But i dont know any web application to know > who is online or offline. any body know any webapp for that ? > > > Flash Operator Panel _is_ a web application. > > > sure. but dont have the online and offline applet? or maybe have and i > dont know how to configure it? > > > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > [1]http://lists.digium.com/mailman/listinfo/asterisk-users > > > ---end quoted text--- > > > > -- > Adrian Carter > Technical Manager > Leading Edge Internet > > Web [2]http://www.lei.net.au [3]http://support.lei.net.au > Direct+61 2 6163 6162 Support 1 300 662 415 > E-mail[EMAIL PROTECTED] > > -- > Adrian Carter > Technical Manager > Leading Edge Internet > > Web [5]http://www.lei.net.au [6]http://support.lei.net.au > Direct+61 2 6163 6162 Support 1 300 662 415 > E-mail[EMAIL PROTECTED] > > References > >1. http://lists.digium.com/mailman/listinfo/asterisk-users >2. http://www.lei.net.au/ >3. http://support.lei.net.au/ >4. mailto:[EMAIL PROTECTED] >5. http://www.lei.net.au/ >6. http://support.lei.net.au/ >7. mailto:[EMAIL PROTECTED] > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ---end quoted text--- -- .- Pablo Allietti E-mail: [EMAIL PROTECTED] | LACNIC Phone : +598 2 604 | http://LACNIC.NET pgpVCiTqSPhRv.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: who is online
On Wed, Dec 28, 2005 at 09:15:15AM -0600, Kevin P. Fleming wrote: > Pablo Allietti wrote: > >hi all i use asdterisk in my company with Flash Panel Operator to know > >who is talking or ringing. But i dont know any web application to know > >who is online or offline. any body know any webapp for that ? > > Flash Operator Panel _is_ a web application. sure. but dont have the online and offline applet? or maybe have and i dont know how to configure it? > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ---end quoted text--- -- .- Pablo Allietti E-mail: [EMAIL PROTECTED] | LACNIC Phone : +598 2 604 | http://LACNIC.NET ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] who is online
hi all i use asdterisk in my company with Flash Panel Operator to know who is talking or ringing. But i dont know any web application to know who is online or offline. any body know any webapp for that ? -- thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Codecs.
mail(b403)
> exten => 403,103,Hangup
>
> With the above, extension 402 can call 403 as well as 403 can call 402.
>
> Your entry
> exten => s,1,Dial(SIP/402,20)
> exten => s,2,Hangup
> does not apply to the configuration that you've shown us. The "s" extension
> is typically used for calls that arrive via Zap and Iax channels where
> "no dialed digits" are received. The "s" is not a match-all option.
>
> We don't have any idea what you mean by "the other side". If you are
> trying to dial from one sip phone to another on your system, then you
> need to define each phone in sip.conf as shown above, and configure
> each phone so that it properly registers with asterisk. To see what
> is registered, do a "sip show peers". If you sip phones don't show in
> that list, they aren't registered. Fix that first before moving on.
>
> Once the above configs have been fixed and asterisk restarted, then
> watch the asterisk CLI to "see" what happens when one phone calls the
> other. If you still have problems, paste the CLI output into a posting
> for us to see. Without that, we can only guess.
>
> Given what you have posted, I don't have a clue what you are trying to
> do with:
> exten => _XXX,1,Dial(${TRUNK}/${EXTEN})
> exten => _XXX,2,Voicemail(u${EXTEN})
> However, when sip extension 402 dials 403, it will match the above _XXX
> and send that call out Zap/g1 (whatever that happens to be).
>
> If you really are working with two asterisk systems tied together with
> Zap channels, then I'd suggest modifying the above to something like
> exten => _5XX,1,Dial(${TRUNK}/${EXTEN})
> exten => _5XX,2,Voicemail(u${EXTEN})
> when the 4XX extensions are on one system and the 5XX extensions on the
> second system.
>
>
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LACNIC
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[Asterisk-Users] Re: Codecs.
On Sat, Dec 17, 2005 at 07:44:29AM -0600, Rich Adamson wrote:
ok rick all of my conf...
asterisk 1.2.1
zaptel 1.2.1
i have a pbx simple with digital phones in one side. and the other side
are xten with SIP.
my extencion.conf
[general]
static=yes
writeprotect=no
autofallthrough=yes
[globals]
CONSOLE=Console/dsp ; Console interface for
demo
TRUNK=Zap/g1
[local]
; ignorepat => 9
include => default
[default]
;
; By default we include the demo. In a production system, you
; probably don't want to have the demo there.
exten => 402,1,Dial(SIP/402,20)
exten => 402,2,Hangup
[teste]
exten => s,1,Dial(SIP/402,20)
exten => s,2,Hangup
exten => 402,1,Dial(SIP/402,20)
exten => 402,2,Hangup
exten => _XXX,1,Dial(${TRUNK}/${EXTEN})
exten => _XXX,2,Voicemail(u${EXTEN})
the sip.conf is the default for asterisk i didnt touch anything in this
file only the extention number and i dont have nothing about codecs in
this file
[402]
type=friend
host=dynamic
username=Pablo
secret=teste
callerid="Pablo" <402>
canreinvite=no
;nat=yes
;amaflags=billing
context=teste
> > > > Hi all i have some problems with my pbx and asterisk codecs.
> > > >
> > > > if i use g711u or g711a codecs. the line never hangup. and the origin
> > > > and destination are connected until i restart my pbx or asterisk
> > > >
> > > > But if i use GSM all work fine.
> > > >
> > > > is possible to solve this problem? or use only gsm codec?
> > >
> >
> > > Yes, its possible to solve the problem.
> >
> > can you explain how?
>
> Not without you providing at least "something" to give us a clue what it
> is that you've programmed into your system.
>
> How about if you give us some clue as to which version of * you're
> using, what type of phones are associated with "origin" and "destination",
> if these are sip phones what do your sip.conf definitions look like,
> what does the appropriate sections of extensions.conf look like, and
> any other configuration pieces that might pertain to whatever it is
> that you've implemented. Your posting implies there might be more than
> one * system involved and possibly even iax trunking, etc.
>
>
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[Asterisk-Users] Re: Codecs.
On Fri, Dec 16, 2005 at 05:08:07PM -0600, Rich Adamson wrote: > > > Hi all i have some problems with my pbx and asterisk codecs. > > > > if i use g711u or g711a codecs. the line never hangup. and the origin > > and destination are connected until i restart my pbx or asterisk > > > > But if i use GSM all work fine. > > > > is possible to solve this problem? or use only gsm codec? > can you explain how? > Yes, its possible to solve the problem. > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ---end quoted text--- -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Codecs.
Hi all i have some problems with my pbx and asterisk codecs. if i use g711u or g711a codecs. the line never hangup. and the origin and destination are connected until i restart my pbx or asterisk But if i use GSM all work fine. is possible to solve this problem? or use only gsm codec? -- .- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Firewall Ports forward
hi all. i have my asterisk with a 192.168.0.1 address which ports i need to forward in my firewall to connect remote xten clients and make calls? thsnk -- .- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: pbx or asterisk?
On Wed, Nov 30, 2005 at 01:17:33PM -0500, Sean Cook wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Looks like your zap channels are droping into the default context...
> better to set up a from-pstn context and start there.
hi sean you have a example please?
>
>
> Pablo Allietti wrote:
> > hi all i have a pbx siemens connect via E1 to my asterisk box.
> >
> > the asterisk box can call without problems to pbx extensions. but when y
> > press the numbers form example 402 in the pbx phones asterisk give me
> > this
> >
> >-- Saved useragent "X-Lite release 1103m" for peer 402
> > -- Going to extension s|1 because of Complete received
> > -- Executing Playback("Zap/31-1", "vm-goodbye") in new stack
> > -- Accepting call from '' to 's' on channel 0/31, span 1
> > == Spawn extension (default, s, 1) exited non-zero on 'Zap/31-1'
> > -- Hungup 'Zap/31-1'
> >
> >
> > -- Accepting call from '' to 's' on channel 0/31, span 1 <<<< did not
> > receive any number or i have miss configure somenthing in asterisk box?
>
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.4.2 (MingW32)
> Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
>
> iD8DBQFDjey8y9wPyZpnL2URAiVCAJ4hQCz+eb1/MaABy2gxUMOcMw1AMwCfYEJI
> VTt9lDiRDMLZhJ2aOL4Qpnw=
> =KqmL
> -END PGP SIGNATURE-
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[Asterisk-Users] pbx or asterisk?
hi all i have a pbx siemens connect via E1 to my asterisk box.
the asterisk box can call without problems to pbx extensions. but when y
press the numbers form example 402 in the pbx phones asterisk give me
this
-- Saved useragent "X-Lite release 1103m" for peer 402
-- Going to extension s|1 because of Complete received
-- Executing Playback("Zap/31-1", "vm-goodbye") in new stack
-- Accepting call from '' to 's' on channel 0/31, span 1
== Spawn extension (default, s, 1) exited non-zero on 'Zap/31-1'
-- Hungup 'Zap/31-1'
-- Accepting call from '' to 's' on channel 0/31, span 1 <<<< did not
receive any number or i have miss configure somenthing in asterisk box?
--
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LACNIC
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[Asterisk-Users] Bad quality
hi all, i have asterisk configured and working but the quality is very poor. i ear noise and braks in the voice when the people talk to me, and the people that eared me have the same problem any recommendation? any files you need to post? -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: sipphone for freebsd
On Thu, Nov 10, 2005 at 12:57:45PM +0800, Dinesh Nair wrote: > > > On 11/10/05 08:52 Pablo Allietti said the following: > >yes but both of them have problem with voice. some skype too anybody can > >have this problems in freebsd? i hear cutted conversations`: > > perhaps there's contention for your sound/mic devices. what does the > hw.snd.pcm0.vchans say, also what's the output of cat /dev/sndstat ? yesss i solve the problem with that. and you know in linux how to setup for 1 channel only? > > with multiple virtual sound channels, you can have different apps sharing > the sound devices cleanly. > > -- > Regards, /\_/\ "All dogs go to heaven." > [EMAIL PROTECTED](0 0)http://www.alphaque.com/ > +==oOO--(_)--OOo==+ > | for a in past present future; do| > | for b in clients employers associates relatives neighbours pets; do | > | echo "The opinions here in no way reflect the opinions of my $a $b." | > | done; done | > +=+ > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ---end quoted text--- -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: sipphone for freebsd
On Wed, Nov 09, 2005 at 01:20:47PM +0800, Dinesh Nair wrote: > > > On 11/09/05 07:17 Pablo Allietti said the following: > >Hi all > >anybody can tell me what sipphone are available for Freebsd? > > /usr/ports/net/kphone > /usr/ports/net/linphone yes but both of them have problem with voice. some skype too anybody can have this problems in freebsd? i hear cutted conversations`: > > -- > Regards, /\_/\ "All dogs go to heaven." > [EMAIL PROTECTED](0 0)http://www.alphaque.com/ > +==oOO--(_)--OOo==+ > | for a in past present future; do| > | for b in clients employers associates relatives neighbours pets; do | > | echo "The opinions here in no way reflect the opinions of my $a $b." | > | done; done | > +=+ > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ---end quoted text--- -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sipphone for freebsd
Hi all anybody can tell me what sipphone are available for Freebsd? i cant find anyone -- .- Pablo Allietti ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incomming calls
Hi all i have a question. is my first time using [EMAIL PROTECTED] and i
need your help
i configure all my asterisk to go outside and work perfect via te110p
but now i need to receive calls. but when in my PBX i digit the number
for example 202 the asterisk receive a "s" i suppouse. the error message
is that
-- Going to extension s|1 because of Complete received
-- Executing Playback("Zap/31-1", "vm-goodbye") in new stack
-- Accepting call from '' to 's' on channel 0/31, span 1
== Spawn extension (default, s, 1) exited non-zero on 'Zap/31-1'
-- Hungup 'Zap/31-1'
can anybody help me with the instructions ?
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[Asterisk-Users] Re: Siemens HI-path to ASTERISK
On Tue, Oct 25, 2005 at 12:31:41PM -0200, [EMAIL PROTECTED] wrote:
> Hi Pablo!
ok. i do all the changes but now i have this error
-- Channel 0/1, span 1 got hangup
Oct 25 11:46:40 WARNING[3639]: app_dial.c:416 wait_for_answer: Unable to
forward voice
-- Hungup 'Zap/1-1'
== No one is available to answer at this time
-- Executing Playback("SIP/205-0014", "invalid") in new stack
-- Playing 'invalid' (language 'en')
== Spawn extension (from-internal, 9122, 2) exited non-zero on
'SIP/205-0014'
maybe is a extensions.conf ?? can you paste your extensions.conf here
please?
>
> I understood your problem. It is related to Siemens PBX.
>
> With this topology, Asterisk is acting as a PSTN Central Office (a Public
> Central). What you asking is something like this:
>
> Asterisk acting as Central Office -> HiPath -> Public Central Office
>
> That is: the SIP devices connected to the Asterisk are not HI-Path's
> extensions! They seem "external" terminal/lines.
>
> So...
>
> You will have to enable, at HiPath, something called "Transit" or "External
> traffic". In other words, it is a feature that you enable on HiPath allowing
> traffic between two trunks (the trunk connected to Asterisk and the trunk
> connected to the PSTN Central Office).
>
> Here we had to create a "trunk access code". So, if a Asterisk user wants to
> call the outside number -1234, he/she will dial:
> 9 + -1234
> Asterisk with then route this call to HiPath prefixing the trunk access
> code, for example, "88". So, asterisk will dial:
> 88 + -1234
>
> Hope this helps,
>
> --hg
> - Original Message -
> From: <[EMAIL PROTECTED]>
> To: "Pablo Allietti" <[EMAIL PROTECTED]>
> Sent: Tuesday, October 25, 2005 11:52 AM
> Subject: Re: Siemens HI-path to ASTERISK
>
>
> >Hi Pablo!
> >
> >I understood your problem. It is related to Siemens PBX.
> >
> >With this topology, Asterisk is acting as a PSTN Central Office (a Public
> >Central). What you asking is something like this:
> >
> >Asterisk acting as Central Office -> HiPath -> Public Central Office
> >
> >That is: the SIP devices connected to the Asterisk are not HI-Path's
> >extensions! They seem "external" terminal/lines.
> >
> >So...
> >
> >You will have to enable, at HiPath, something called "Transit" or
> >"External traffic". In other words, it is a feature that you enable on
> >HiPath allowing traffic between two trunks (the trunk connected to
> >Asterisk and the trunk connected to the PSTN Central Office).
> >
> >Here we had to create a "trunk access code". So, if a Asterisk user wants
> >to call the outside number -1234, he/she will dial:
> >9 + -1234
> >Asterisk with then route this call to HiPath prefixing the trunk access
> >code, for example, "88". So, asterisk will dial:
> >88 + -1234
> >
> >Hope this helps,
> >
> >Huelbe.
> >
> >- Original Message -
> >From: "Pablo Allietti" <[EMAIL PROTECTED]>
> >To: <[EMAIL PROTECTED]>
> >Sent: Tuesday, October 25, 2005 12:41 PM
> >Subject: Re: Siemens HI-path to ASTERISK
> >
> >
> >>On Mon, Oct 24, 2005 at 06:42:02PM -0200, [EMAIL PROTECTED]
> >>wrote:
> >>>Yes, for sure, it works. With TE110P from Digium using E1/ISDN/Pri
> >>>signalling.
> >>>
> >>>By heart, I remember the following:
> >>>
> >>>1. Configure Siemens E1 port as "station" and Asterisk as "Pri_Net" (or
> >>>Central Office).
> >>>
> >>>2. At Siemens, set the E1 port as "S2 Point-to-Point net line without
> >>>CRC4"
> >>>or something like this.
> >>
> >>
> >>yep done. i only have a problem i can call any extension in the pbx but
> >>i can't take outside line with the 9
> >>
> >>you can send to me the extensions.conf please please/
> >>
> >>>
> >>>3. At Asterisk, put these lines (/etc/zaptel.conf):
> >>>span=1,1,0,ccs,hdb3
> >>>bchan=1-15
> >>>dchan=16
> >>>bchan=17-31
> >>>
> >>>You have to study the rest of * conf file, but these ones are the
> >>>important
> >>>ones.
> >>>
> >>>Regards,
> >>>
> >>>--hg
> >>>
> >>&
[Asterisk-Users] Re: Siemens HI-path to ASTERISK
On Mon, Oct 24, 2005 at 06:42:02PM -0200, [EMAIL PROTECTED] wrote: > Yes, for sure, it works. With TE110P from Digium using E1/ISDN/Pri > signalling. > > By heart, I remember the following: > > 1. Configure Siemens E1 port as "station" and Asterisk as "Pri_Net" (or > Central Office). > > 2. At Siemens, set the E1 port as "S2 Point-to-Point net line without CRC4" > or something like this. yep done. i only have a problem i can call any extension in the pbx but i can't take outside line with the 9 you can send to me the extensions.conf please please/ > > 3. At Asterisk, put these lines (/etc/zaptel.conf): > span=1,1,0,ccs,hdb3 > bchan=1-15 > dchan=16 > bchan=17-31 > > You have to study the rest of * conf file, but these ones are the important > ones. > > Regards, > > --hg > > - Original Message - > From: "Pablo Allietti" <[EMAIL PROTECTED]> > To: > Sent: Monday, October 24, 2005 6:55 PM > Subject: [Asterisk-Users] Siemens HI-path to ASTERISK > > > >anybody can connect a Siemens HI-PATH to ASterisk via e1 ? > > > >i need your help please. > >___ > >--Bandwidth and Colocation sponsored by Easynews.com -- > > > >Asterisk-Users mailing list > >Asterisk-Users@lists.digium.com > >http://lists.digium.com/mailman/listinfo/asterisk-users > >To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ---end quoted text--- -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Siemens HI-path to ASTERISK
anybody can connect a Siemens HI-PATH to ASterisk via e1 ? i need your help please. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] is possible connect?
hi all i have this structure.
Box.(te110p)Pbx(e1)4 analogic lines to outside
is poosible connect asterisk to get outside lines? because i can call
any extension in my pbx with xten but i cant get outside lines. the
asterisk tellme all circuits are busy when i send the number 9 to get
the line. i remove all in my extensions.conf and have this
[EMAIL PROTECTED] asterisk]# cat extensions.conf
[GLobals]
; RECEPTIONIST=Zap/1
JOHN=SIP/203
MARY=SIP/202
LOCALTRUNK=Zap/1
[incoming]
exten => s,1,Answer()
exten => s,2,Background(current-movies)
exten => s,3,Hangup()
exten => 1,1,Playback(movie1)
exten => 1,2,Goto(incoming,s,1)
exten => 2,1,Playback(movie2)
exten => 2,2,Goto(incoming,s,1)
exten => 0,1,Dial(${RECEPTIONIST})
[internal]
; ignorepat => 9
exten => _1XX,1,Dial(${LOCALTRUNK}/${EXTEN})
exten => _1XX,2,Voicemail(u${EXTEN})
exten => _,1,Dial(${LOCALTRUNK}/${EXTEN})
exten => _,2,Playback(invalid)
exten => _,3,Hangup
my extensions in my pbx start with the number 1 and i need for example
to call 9,856 is a service number in my country.
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[Asterisk-Users] Siemens Hi-Path help
hi all, anybody have a siemens hipath 3500 with a sm2/pri card? because i need to connect to my box TE110P (e1) and i dont know how is the mode in the pbx to change it. thanks -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: (no subject)
On Wed, Sep 14, 2005 at 07:52:26PM +0200, Sander wrote:
> This is not a siemens pbx problem you set the
> pridialplan = to national and that adds a number to the outgoing call or
> something just use
>
>
> Pridialplan = local
> prilocaldialplan = local
>
> and it should work
no uuuaaa the same problem.. ring in the extension 100.
>
> I tried to open the file kds again and now it showed me your configuration
> :) don't know why it did not show me before
>
> Sander
>
> -Oorspronkelijk bericht-
> Van: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Namens Pablo Allietti
> Verzonden: woensdag 14 september 2005 17:31
> Aan: Asterisk Users Mailing List - Non-Commercial Discussion
> Onderwerp: [Asterisk-Users] Re: (no subject)
>
> On Wed, Sep 14, 2005 at 10:22:10AM -0400, Matt Ryanczak wrote:
>
>
> ok. didnt work :( i thinks is a pbx problem. because E1 is incomming in the
> pbx. and all incomming calls go to 100. thats the problem i will try to
> solve this.
>
>
>
> > It could potentially be both. I would look at your extensions.conf
> > first though. What does the extension entry for that context look like.
> >
> > For instance I have an entry in my extensions.conf for dialing outside
> > lines (outside being from asterisk to my PBX and then onto the outside
> > world from there). The entry looks like this:
> >
> > [to-analog]
> > exten => _9XXX.,1,Dial(ZAP/G1/${EXTEN}) exten => _9XXX.,2,Congestion
> > exten => _9XXX.,103,Hangup
> >
> >
> > To dial a PBX extension the entry would look almost the same:
> >
> > [to-pbx-extension]
> > exten => _9XXX.,1,Dial(ZAP/G1/${EXTEN:1}) exten => _9XXX.,2,Congestion
> > exten => _9XXX.,103,Hangup
> >
> > Hope this helps,
> >
> > -Matt
> >
> > On Wed, 2005-09-14 at 11:46 -0300, Pablo Allietti wrote:
> > > hi all, i have a box with a te110p and a pbx siemens... connect both
> > > with a e1.
> > > with a xten soft i can call extensions numbers in my office example
> > > 100
> > > 102 etc. but when i truy to go outside with the 9 before the call
> > > rings in the first extensions (100). this is a asterisk problem? or
> > > a pbx problem?
> >
> > ___
> > --Bandwidth and Colocation sponsored by Easynews.com --
> >
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> ---end quoted text---
>
> --
>
> .-
>
> Pablo Allietti
> LACNIC
>
> ___
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>
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> http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users] Re: TE110P - [EMAIL PROTECTED] Install Problems
On Wed, Sep 14, 2005 at 10:45:36AM -0500, Robert Wagner wrote:
hi te110p is a et1 card. your sigfnalling is wrong i think
i have the same card and is work with this conf
/etc/zaptel.conf
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
loadzone= us
defaultzone = us
/etc/asterisk/zapata.conf
;
; Zapata telephony interface
;
; Configuration file
[trunkgroups]
[channels]
context=from-pstn
rxwink=400 ; Atlas seems to use long (250ms) winks
relaxdtmf=yes
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
;group=1
;callgroup=1
;pickupgroup=1
signalling=pri_net ; pri_cpe = PRI slave ; pri_net = PRI master
; national: National ISDN 2 (default)
; dms100: Nortel DMS100
; 4ess: AT&T 4ESS
; 5ess: Lucent 5ESS
; euroisdn: EuroISDN
; ni1:Old National ISDN 1
switchtype=national
echocancelwhenbridged=yes
echotraining=400 ; Asterisk trains to the beginning of the call, number
is in milliseconds
callerid=asreceived
group=1
;context=default ; Points to the default context of your extensions.conf
channel => 1-15,17-31 ; Set this to 1-15,17-31 for E1
>
>I am having problems sending and receiving calls over the T1. They
>never seem to connect - outbound keeps ringing, inbound gets busy.
>The T1 looks ok - no errors on the line. Any ideas on what is wrong?
>I have tried a variety of fxsks and fxoks configurations without
>avail. This is a single [EMAIL PROTECTED] system with a single T1 card.
>Robbed Bit T1 ami, d4.
>
>--inbound call
> -- Starting simple switch on 'Zap/7-1'
>-- Starting simple switch on 'Zap/14-1'
>-- Executing Playback("Zap/7-1", "vm-goodbye") in new stack
>-- Playing 'vm-goodbye' (language 'en')
>-- Executing Macro("Zap/7-1", "hangupcall") in new stack
>-- Executing ResetCDR("Zap/7-1", "w") in new stack
>-- Executing NoCDR("Zap/7-1", "") in new stack
>-- Executing Wait("Zap/7-1", "5") in new stack
>-- Executing Playback("Zap/14-1", "vm-goodbye") in new stack
>-- Playing 'vm-goodbye' (language 'en')
>-- Executing Hangup("Zap/7-1", "") in new stack
> == Spawn extension (macro-hangupcall, s, 4) exited non-zero on
>'Zap/7-1' in macro 'hangupcall'
> == Spawn extension (default, s, 2) exited non-zero on 'Zap/7-1'
>-- Hungup 'Zap/7-1'
>-- Executing Macro("Zap/14-1", "hangupcall") in new stack
>-- Executing ResetCDR("Zap/14-1", "w") in new stack
>-- Executing NoCDR("Zap/14-1", "") in new stack
>-- Executing Wait("Zap/14-1", "5") in new stack
>-- Executing Hangup("Zap/14-1", "") in new stack
> == Spawn extension (macro-hangupcall, s, 4) exited non-zero on
>'Zap/14-1' in macro 'hangupcall'
> == Spawn extension (default, s, 2) exited non-zero on 'Zap/14-1'
>-- Hungup 'Zap/14-1'
>
>
>--outbound call
>-- Executing SetVar("SIP/4901-cd04", "OUTNUM=mynum") in new stack
>-- Executing Cut("SIP/4901-cd04", "custom=OUT_1|:|1") in new stack
>-- Executing GotoIf("SIP/4901-cd04", "0?19") in new stack
>-- Executing Dial("SIP/4901-cd04", "ZAP/g0/mynum") in new stack
>-- Called g0/mynum
>-- Zap/1-1 answered SIP/4901-cd04
>-- Hungup 'Zap/1-1'
>---
>
>[zaptel.conf]
>span=1,1,0,d4,ami # have tried with 1,0,0 - same problem
>fxsks=1-24
>loadzone = us
>defaultzone=us
>
>[zapata.conf]
>[channels]
>signalling=fxs_ks
>group=0
>;context=incoming
>channel=>1-24
>echocancelwhenbridged=yes
>echotraining=400
>context=default
>faxdetect=incoming
>
>;Include genzaptelconf configs
>#include zapata-auto.conf
>
>;Include AMP configs
>#include zapata_additional.conf
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--
.-
Pablo Allietti
LACNIC
___
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[Asterisk-Users] Re: (no subject)
On Wed, Sep 14, 2005 at 10:22:10AM -0400, Matt Ryanczak wrote:
ok. didnt work :( i thinks is a pbx problem. because E1 is incomming in
the pbx. and all incomming calls go to 100. thats the problem i will
try to solve this.
> It could potentially be both. I would look at your extensions.conf first
> though. What does the extension entry for that context look like.
>
> For instance I have an entry in my extensions.conf for dialing outside
> lines (outside being from asterisk to my PBX and then onto the outside
> world from there). The entry looks like this:
>
> [to-analog]
> exten => _9XXX.,1,Dial(ZAP/G1/${EXTEN})
> exten => _9XXX.,2,Congestion
> exten => _9XXX.,103,Hangup
>
>
> To dial a PBX extension the entry would look almost the same:
>
> [to-pbx-extension]
> exten => _9XXX.,1,Dial(ZAP/G1/${EXTEN:1})
> exten => _9XXX.,2,Congestion
> exten => _9XXX.,103,Hangup
>
> Hope this helps,
>
> -Matt
>
> On Wed, 2005-09-14 at 11:46 -0300, Pablo Allietti wrote:
> > hi all, i have a box with a te110p and a pbx siemens... connect both
> > with a e1.
> > with a xten soft i can call extensions numbers in my office example 100
> > 102 etc. but when i truy to go outside with the 9 before the call rings
> > in the first extensions (100). this is a asterisk problem? or a pbx
> > problem?
>
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--
.-
Pablo Allietti
LACNIC
___
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[Asterisk-Users] (no subject)
hi all, i have a box with a te110p and a pbx siemens... connect both with a e1. with a xten soft i can call extensions numbers in my office example 100 102 etc. but when i truy to go outside with the 9 before the call rings in the first extensions (100). this is a asterisk problem? or a pbx problem? -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: siemens pbx what i ask techinician?
On Fri, Sep 09, 2005 at 09:41:58PM +0200, Sander wrote: > sure my mail is [EMAIL PROTECTED] i configure that but the system when i dial a 9 form get outside line give me the error "all circuits are busy now" and the cat /proc/zaptel/1 [EMAIL PROTECTED] asterisk]# cat /proc/zaptel/1 Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" HDB3/CCS YELLOW 1 WCT1/0/1 Clear (In use) 2 WCT1/0/2 Clear (In use) 3 WCT1/0/3 Clear (In use) 4 WCT1/0/4 Clear (In use) 5 WCT1/0/5 Clear (In use) 6 WCT1/0/6 Clear (In use) 7 WCT1/0/7 Clear (In use) 8 WCT1/0/8 Clear (In use) 9 WCT1/0/9 Clear (In use) 10 WCT1/0/10 Clear (In use) 11 WCT1/0/11 Clear (In use) 12 WCT1/0/12 Clear (In use) 13 WCT1/0/13 Clear (In use) 14 WCT1/0/14 Clear (In use) 15 WCT1/0/15 Clear (In use) 16 WCT1/0/16 HDLCFCS (In use) 17 WCT1/0/17 Clear (In use) 18 WCT1/0/18 Clear (In use) 19 WCT1/0/19 Clear (In use) 20 WCT1/0/20 Clear (In use) 21 WCT1/0/21 Clear (In use) 22 WCT1/0/22 Clear (In use) 23 WCT1/0/23 Clear (In use) 24 WCT1/0/24 Clear (In use) 25 WCT1/0/25 Clear (In use) 26 WCT1/0/26 Clear (In use) 27 WCT1/0/27 Clear (In use) 28 WCT1/0/28 Clear (In use) 29 WCT1/0/29 Clear (In use) 30 WCT1/0/30 Clear (In use) 31 WCT1/0/31 Clear (In use) show me that... is that correct? > > Ok i'm looking will try to make a small manual for you, please make sure you > have set the jumers of the pri card in asterik at the right position? Voor > TE mode > Here is my config of the e1 card the only thing that does not work on my pbx > is to do a reboot with de cable plugged in the asterisk pbx don't know why > but it just hangs. so boot after power down > > > Zaptel: > loadzone=us > defaultzone=us > > > # E1 definition: crc4 check achter hdb3 ,crc4 > span=1,1,0,ccs,hdb3 > > #E1: > bchan=1-15,17-31 > dchan=16 > > zapata > > [channels] > signalling=pri_cpe > switchtype=national > echocancel=yes > echocancelwhenbridged=no > callerid=asreceived > group=1 > immediate = no > context= incomming ; Points to the default context of your extensions.conf > channel => 1-15,17-31 > > > And on the siemens pbx turn of crc4 check on the e1 card configuration maybe > you can give me your mail adres so i can make screenshots of the manager e > configuration tool i can't mail pictures to the user list > > > > -Oorspronkelijk bericht- > Van: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Namens Pablo Allietti > Verzonden: vrijdag 9 september 2005 20:29 > Aan: Asterisk Users Mailing List - Non-Commercial Discussion > Onderwerp: [Asterisk-Users] Re: siemens pbx what i ask techinician? > > On Fri, Sep 09, 2005 at 07:20:58PM +0200, Sander wrote: > > > uuauuu that will great! > i cant undertand too much about internal connection because. i have a PC > with a te110p and my siemens is a hipath 3500 with a s2m/pri card is a > E1 card. but i dont know how to connect between them. i have always the red > alarm in the te110p. my conf files are > > both of this files i copy and paste from internet. > > /etc/zaptel.conf > span=1,0,0,ccs,hdb3,crc4,yellow > bchan=1-15,17-31 > dchan=16 > loadzone= us > defaultzone = us > > and the /etc/asterisk/zapata.conf > > [channels] > context=zap-in > ;switchtype=qsig > pridialplan=national > signalling=pri_cpe > usecallerid=yes > hidecallerid=no > callwaiting=yes > callwaitingcallerid=yes > threewaycalling=yes > transfer=yes > cancallforward=yes > echocancel=yes > rxgain=0.0 > txgain=0.0 > > group=1 > callgroup=1 > pickupgroup=1 > > immediate=no > callprogress=no > > callerid=asreceived > group=1 > signalling=pri_net > channel => 1-15,17-31 > > please help me!!! thanks a lot for your time > > > > > Do you want to connect the asterisk with pri or with internal isdn? > > And what model pbx do you have? then i can tell you how to configure? > > Maybe some screenshots with it > > > > -Oorspronkelijk bericht- > > Van: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] Namens Pablo Allietti > > Verzonden: vrijdag 9 september 2005 19:35 > > Aan: Asterisk Users Mailing List - Non-Commercial Discussion > > Onderwerp: [Asterisk-Users] Re: siemens pbx what i ask techinician? > > > > On Fri, Sep 09, 2005 at 05:39:22PM +0200, Sander wrote: > > >
[Asterisk-Users] Re: siemens pbx what i ask techinician?
On Fri, Sep 09, 2005 at 07:20:58PM +0200, Sander wrote: uuauuu that will great! i cant undertand too much about internal connection because. i have a PC with a te110p and my siemens is a hipath 3500 with a s2m/pri card is a E1 card. but i dont know how to connect between them. i have always the red alarm in the te110p. my conf files are both of this files i copy and paste from internet. /etc/zaptel.conf span=1,0,0,ccs,hdb3,crc4,yellow bchan=1-15,17-31 dchan=16 loadzone= us defaultzone = us and the /etc/asterisk/zapata.conf [channels] context=zap-in ;switchtype=qsig pridialplan=national signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no callprogress=no callerid=asreceived group=1 signalling=pri_net channel => 1-15,17-31 please help me!!! thanks a lot for your time > Do you want to connect the asterisk with pri or with internal isdn? And what > model pbx do you have? then i can tell you how to configure? Maybe some > screenshots with it > > -Oorspronkelijk bericht- > Van: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Namens Pablo Allietti > Verzonden: vrijdag 9 september 2005 19:35 > Aan: Asterisk Users Mailing List - Non-Commercial Discussion > Onderwerp: [Asterisk-Users] Re: siemens pbx what i ask techinician? > > On Fri, Sep 09, 2005 at 05:39:22PM +0200, Sander wrote: > > > > thanks Sander but i have the soft, and i can enter to the pbx conf and > modify all settings, but i dont know how settings i need to change. > > > It's not that easy then everytime you want to change someting for > > testing you have to ask them to change something i can give you the > > software for programming siemens pbx if you want > > > > > > > > > > -Oorspronkelijk bericht- > > Van: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] Namens Pablo Allietti > > Verzonden: vrijdag 9 september 2005 16:09 > > Aan: asterisk-users@lists.digium.com > > Onderwerp: [Asterisk-Users] siemens pbx what i ask techinician? > > > > im really newbie, and i have a siemens digital pbx work in my work. i > > have 4 outside lines and the pbx has a E1/PRI card. what i need to ask > > my siemens > > provider(techinicians) to do in the pbx? > > > > i only have in my pbx the 9 to get a line to go outside is very > > simple. but i dont know what i need to ask them to programming. please > help me. > > -- > > > > .- > > > > Pablo Allietti > > LACNIC > > > > ___ > > --Bandwidth and Colocation sponsored by Easynews.com -- > > > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > > --Bandwidth and Colocation sponsored by Easynews.com -- > > > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > ---end quoted text--- > > -- > > .- > > Pablo Allietti > LACNIC > > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ---end quoted text--- -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: siemens pbx what i ask techinician?
On Fri, Sep 09, 2005 at 05:39:22PM +0200, Sander wrote: > thanks Sander but i have the soft, and i can enter to the pbx conf and modify all settings, but i dont know how settings i need to change. > It's not that easy then everytime you want to change someting for testing > you have to ask them to change something i can give you the software for > programming siemens pbx if you want > > > > > -Oorspronkelijk bericht- > Van: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Namens Pablo Allietti > Verzonden: vrijdag 9 september 2005 16:09 > Aan: asterisk-users@lists.digium.com > Onderwerp: [Asterisk-Users] siemens pbx what i ask techinician? > > im really newbie, and i have a siemens digital pbx work in my work. i have 4 > outside lines and the pbx has a E1/PRI card. what i need to ask my siemens > provider(techinicians) to do in the pbx? > > i only have in my pbx the 9 to get a line to go outside is very simple. but > i dont know what i need to ask them to programming. please help me. > -- > > .- > > Pablo Allietti > LACNIC > > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ---end quoted text--- -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] siemens pbx what i ask techinician?
im really newbie, and i have a siemens digital pbx work in my work. i have 4 outside lines and the pbx has a E1/PRI card. what i need to ask my siemens provider(techinicians) to do in the pbx? i only have in my pbx the 9 to get a line to go outside is very simple. but i dont know what i need to ask them to programming. please help me. -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
