[asterisk-users] Asterisk as SMSC to GSM-Phones

2008-02-27 Thread Hans-Peter Straub
Hello all,

i today have searched on the internet about a solution to let asterisk act as 
a SMSC (Short MessageServiceCentre) to deliver SMSes directly to GSM Phones. 
I only have found some cases with use of an extern SMSC (i.e. by the Mobile 
Net Provider)

Is there a possibillity to do that, or ist asterisk only able to send SMS to 
analog phones?

Do you have any idea or hint for me?

Thanks a lot

Hans-Peter Straub

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[asterisk-users] Gateway doesn't ring

2007-12-10 Thread Hans-Peter Straub
Hello all,

i have a problem on incoming call's from SIP Provider that ist going through 
the Asterisk to a Grandstream HT502. The first ring is executed on the HT502 
propperly, but no more ring will follow. But the call can nevertheless be 
answered by a phone on the gateway.

If i call the same Gateway through a connected second Asterisk the ringing is 
done well. 

If a call is coming through the same SIP Provider to another Gateway 
i.e. Inalp Patton SmartNode SN4552 all is working fine.

Is this a problem on the configuration of the Gateway itselfs or on Asterisk 
for this gateway?

Thanks

Hans-Peter Straub


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[Asterisk-Users] How to transfer outgoing calls

2006-04-28 Thread Hans-Peter Straub
Hello all,

is it possible to make an outgoing call transferable for the dialing phones 
like the 't' or T option on the Dial-Command does this for incoming calls?

Does someone have any idea?

Thanks

Hans-Peter Straub
 

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Re: [Asterisk-Users] How to transfer outgoing calls

2006-04-28 Thread Hans-Peter Straub
Am Freitag 28 April 2006 15:33, Eric ManxPower Wieling schrieb:
 Hans-Peter Straub wrote:
  Hello all,
 
  is it possible to make an outgoing call transferable for the dialing
  phones like the 't' or T option on the Dial-Command does this for
  incoming calls?

 The t and T option works for ANY call using Dial.  Incoming or outgoing.

Ok works fine. That was my fault, Sorry 

Thanks

Hans-Peter Straub

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[Asterisk-Users] Promblem dialin from an internal E1

2005-11-15 Thread Hans-Peter Straub
Hello all,

i have a problem when dialing from an analog-line from a PBX behind the 
Asterisk server. With the ISDN-Lines on that PBX it is possible to dial, when 
dialing before connecting. With the analog-phones Asterisk will not wait to 
finish the number, and returns the following:

Extension '' in context 'from-intern-s2m' from '58' does not exist.  Rejecting 
call on channel 0/1, span 2

How can i tell * to wait for the number finishing?

Do someone have any idea?

Thanks

Hans-Peter Straub


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[Asterisk-Users] Problem on Data-Connections through Asterisk

2005-11-04 Thread Hans-Peter Straub
Hello,

i'm using Asterisk as an intermediary between another PBX (Teles) and the E1 
of our Carrier. We us a TE205P Dual Span Card from Digium. All the analog 
connections are running fine, but digital data calls from local ISDN (BRI @ 
Teles PBX) to a remote syncPPP-dialup via the E1 doesn't work well. It looks 
that the connection is answered ok, but no data is going over this 
connection, so the handshake for the PPP-Login doesn't go to the server. I've 
also tested a rawip connection over ISDN (isdnx Interfaces of Linux) and the 
effect is the same. The connection will be made but no data goes through it.

When i conect the Teles-PBX directly to the E1 the calls and all data is going 
through the line.

I've played with some options i've found in the Mailinglist in the Dial-Tag 
but without any success.

Do someone have any idea if this fault can be removed or is there no chance to 
get this working?

Thanks a lot

Hans-Peter Straub


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Re: [Asterisk-Users] Problem on Data-Connections through Asterisk

2005-11-04 Thread Hans-Peter Straub

Hello again,

I've found the problem. I wrote some entries to the extensions.conf like

--
exten = _5X,1,Dial(Zap/g2/${EXTEN},120,rt)
--

but it seems that Asterisk don't like the timeout and/or option entries (i.e. 
120,rt) on digital calls. When i remove this entries like

--
exten = _5X,1,Dial(Zap/g2/${EXTEN})
--

all calls going fine through Asterisk :-)

Thanks

Hans-Peter Straub



 i'm using Asterisk as an intermediary between another PBX (Teles) and the
 E1 of our Carrier. We us a TE205P Dual Span Card from Digium. All the
 analog connections are running fine, but digital data calls from local ISDN
 (BRI @ Teles PBX) to a remote syncPPP-dialup via the E1 doesn't work well.
 It looks that the connection is answered ok, but no data is going over this
 connection, so the handshake for the PPP-Login doesn't go to the server.
 I've also tested a rawip connection over ISDN (isdnx Interfaces of Linux)
 and the effect is the same. The connection will be made but no data goes
 through it.

 When i conect the Teles-PBX directly to the E1 the calls and all data is
 going through the line.

 I've played with some options i've found in the Mailinglist in the Dial-Tag
 but without any success.

 Do someone have any idea if this fault can be removed or is there no chance
 to get this working?

 Thanks a lot

 Hans-Peter Straub

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[Asterisk-Users] Problem loading misdn driver

2005-10-18 Thread Hans-Peter Straub
Hallo all,

i have a problem on loading chan_misdn. The misdn is running and all 
cards (TDM40B+AVMFritz) is initialized. When im going to start asterisk 
with the chan_misdn.so module i get the following error in the log (on 
console) and asterisk ist hanging.

i use the current CVS-HEAD of asterisk (7 Days old),
chan_misdn-0.2.1-rc2 and mISDN+mISDNuser from the automated
installation.

Is here anybody who have any idea why asterisk hangs. I searched all
the mailinglists, and doesn't get any information on what's wrong.


cp from the console:

 [chan_local.so] = (Local Proxy Channel)
  == Registered channel type 'Local' (Local Proxy Channel Driver)
 [chan_skinny.so] = (Skinny Client Control Protocol (Skinny))
  == Parsing '/etc/asterisk/skinny.conf': Found
  == Skinny listening on 0.0.0.0:2000
  == Registered channel type 'Skinny' (Skinny Client Control Protocol (Skinny))
 [chan_features.so] = (Feature Proxy Channel)
  == Registered channel type 'Feature' (Feature Proxy Channel Driver)
 [skipping chan_oss.so]
 [chan_modem_i4l.so] = (ISDN4Linux Emulated Modem Driver)
 [chan_phone.so] = (Linux Telephony API Support)
  == Parsing '/etc/asterisk/phone.conf': Found
  == Registered channel type 'Phone' (Standard Linux Telephony API Driver)
 [chan_oh323.so] = (InAccess Networks OpenH323 Channel Driver)
  == Parsing '/etc/asterisk/rtp.conf': Found
  == Parsing '/etc/asterisk/oh323.conf': Found
  == Registered channel type 'OH323' (InAccess Networks OpenH323 Channel Driver)
  == OpenH323 Channel Ready (v0.7.3)
 [chan_misdn.so] = (Channel driver for mISDN Support (Bri/Pri))
cb_log called with out-of-range port number!


Yours

Hans-Peter Straub


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[Asterisk-Users] How to rewrite a CALLERID on outgoing calls

2005-10-14 Thread Hans-Peter Straub
Hello all,

is here anybody who have any idea how i can insert a script or program to 
rewrite a callerid with special rules. This ist necessary because of many 
moving mobile offices who changes the telefonenumbers in short time 
distances. I've found the SetCIDName feature. But this doesn't work in my 
relation.

The phones from wich the calls are, are connected via OH323 Module and 
registered on a gnugk Gatekeeper.

Thanks

Hans-Peter Straub



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Re: [Asterisk-Users] How to rewrite a CALLERID on outgoing calls

2005-10-14 Thread Hans-Peter Straub
Am Freitag 14 Oktober 2005 21:22, Hans-Peter Straub schrieb:
 Hello all,

 is here anybody who have any idea how i can insert a script or program to
 rewrite a callerid with special rules. This ist necessary because of many
 moving mobile offices who changes the telefonenumbers in short time
 distances. I've found the SetCIDName feature. But this doesn't work in my
 relation.

 The phones from wich the calls are, are connected via OH323 Module and
 registered on a gnugk Gatekeeper.

Hello again,

sorry i made a fault in my tests. SetCIDName does now work for me fine.

Thanks

Hans-Peter Straub



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Re: [Asterisk-Users] OOH323C

2005-09-29 Thread Hans-Peter Straub
Am Donnerstag 29 September 2005 16:28, Kanishka Somaratne schrieb:
 hi
 has any one used OOH323C i tried this it is installed but do not know how
 to configure has any one used this, what is the best h323 addon to use with
 asterisk

Hello,

for me it seems that the OOH323 Development is not finished yet. I use the 
OH323 that you can reach at

http://www.inaccessnetworks.com/projects/asterisk-oh323

Yours

Hans-Peter Straub



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[Asterisk-Users] Keytouch without effect

2005-09-23 Thread Hans-Peter Straub
Hello anybody,

i have a problem on connecting an innovaphone ip202 to theAsterisk-PBX. When i 
dial in the PBX with the standard (make samples) configuration with the ip202 
the connection is fine, but to push any Key on the keypad dosn't take any 
effect. Is for H323-Phones a special DTMF config necessary?

Thanks

Hans-Peter Straub


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