hi list.
i have the following problem.
if i dial an ip endpoint from my ip phone and the endpoint is busy, in
my cdr i see (answered). I think there must be busy.
why is that? any hints?
thx,
thomas
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Hi List,
we are usind a Teles PSTN Gateway for termination calls to PSTN.
the Swich want´s to register at asterisk without using user/passwort.
how can i implement this in sip.conf?
asterisk forces thw switch to authenticate with username/passwort, can
i disable this for one sip peers?
thx for
ist there any cdr dokumentaion about the cdr format?
thx!
thomas
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hi list,
how can i disable that one sip clients can login twice?
thx for help,
thomas
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hi,
i read a lot of papers about answering tone whenn i call outside and
inbound with sip phones.
in my case, there is no dial tone wenn i do a call outside or if
someone calls the sip phone.
how can i configure/play a ring/dial tone till the endpoint accepts the
call?
thx!
thomas
x-tad-biggerdueeseldorf ist nur ein vertriebsbüro:
http://firstwave.ch/kontakt/impressum.html
Markus Wingen/x-tad-biggerx-tad-bigger
/x-tad-biggerx-tad-bigger Business Development/x-tad-biggerx-tad-bigger
/x-tad-biggerx-tad-bigger[EMAIL PROTECTED]/x-tad-biggerx-tad-bigger
hi list,
no call into pstn and from pstn to asterisk endpoints displays the
calling party number. I see in my sip phones number unavailable. in the
asterisk cdrs i see the number which is calling but at the endpoints
its unavailable.
any hints how i can activate to display the calling from
use ssh instead of telnet. telnet is a bad idea.
Am 20.08.2004 um 11:39 schrieb neil:
x-tad-biggerSorry if this is posted to the wrong forum but as it is related to a problem I have with Asterisk it may just scrape through!!/x-tad-bigger
x-tad-bigger /x-tad-bigger
x-tad-biggerI am running
(SIP/${EXTEN:[EMAIL PROTECTED],60,tT)
can anyone tell me what the tT attribute behind 60 stands for?
thx
--
Thomas Küpper
01063 Telecom GmbH Co. KG
Mottmannstr. 2
53842 Troisdorf
Telefon: 02241-9434-506
Telefax: 02241-9434-846
E-Mail: [EMAIL PROTECTED]
E-Mail: [EMAIL PROTECTED]
Homepage:
hi,
kann a sip user login two times from different clients? if he can, how
does asterisk handle the call in this case?
--
Thomas Küpper
01063 Telecom GmbH Co. KG
Mottmannstr. 2
53842 Troisdorf
Telefon: 02241-9434-506
Telefax: 02241-9434-846
E-Mail: [EMAIL PROTECTED]
E-Mail: [EMAIL PROTECTED]
hi,
i want to use mystun because off nat problems by more than one device
behind one nat gw. i think it is the only solution to solve the nat
problem.
what i do not understand is why needs the stun server two ip addresses?
thx for any hints.
--
Thomas Küpper
01063 Telecom GmbH Co. KG
hi,
wenn i do a call from gsm to sip endpoint i dont see the gsm telephone
nummber. i can only see UNAVAILABLE.
how can i change this?
--
Thomas Küpper
01063 Telecom GmbH Co. KG
Mottmannstr. 2
53842 Troisdorf
Telefon: 02241-9434-506
Telefax: 02241-9434-846
E-Mail: [EMAIL PROTECTED]
E-Mail:
Hi,
for incomming calls, i have set an gatekkeper in h323.conf.
outgoing calls wich are no sip endpoints should be sent to a h323
gateway, not the gatekeeper. If i disable the Gatekeeper, all non sip
calls are routed to the Gateway. If i enable the Gatekeeper, the calls
are send to the
with a h323 client over my gatekepper a call comes over asrerisk to my
sip endpoint:
== Spawn extension (sip-phones, 01634255122, 1) exited non-zero on
'SIP/0699073201-528d'
-- Executing Dial(H323/ip$10.0.0.124:49638/18690,
SIP/0699073201) in new stack
-- Called 0699073201
--
hi agian,
i am pondering why no one ist answering to thiis problem. i found 3
list-useres who have all the same problems, but ei can not find any
solution for that.
wenn ich do a call to 1234 with a h323 softohone to a sip endpoint all
works fine. If i make a call from PSTN to the same sip
hi,
we are using a voip h323 switch. the switch sends all caals to our
Gatekeeper (gnugk).
gnugk musst send all calls to asterisk and asterisk must do his choice
(sip endpoint or out to PSTN)
Making calls to our h323 switch works fine over asterisk. what must i
configure to get inboung h323
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Thomas
Kuepper
Envoyé : jeudi 5 août 2004 13:31
À : [EMAIL PROTECTED]
Objet : [Asterisk-Users] h323 gnugk to h323 asterisk and then to
endpoint
hi,
we are using a voip h323 switch. the switch sends all caals to our
Gatekeeper (gnugk
) and 5XXX ISDN link
to an
ISDN PBX
Everyone can reach each others
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Thomas
Kuepper
Envoyé : jeudi 5 août 2004 14:47
À : [EMAIL PROTECTED]
Objet : Re: [Asterisk-Users] h323 gnugk to h323 asterisk
hi list,
i want to convert all none SIP calls to h323 and send them to our GnuGK
Gatekeeper.
with my setup (attached) i called the number 5678 and got the following
error msg:
Error msg:
Jul 29 10:19:45 WARNING[114696]: chan_sip.c:457 __sip_xmit: sip_xmit of
0x81210dc (len 635) to 0.0.22.46
Hi List,
we are using openh323 gatekeeper for voip telefony. We also have a voip
over ss7 TELES Switch for voip into POSTN Network. Know we want to use
Asterisk for converting SIP to h323.
Now my question. Is Asterisk an full h323 gatekeeper like openh323? Do
we need openh323 GK for astrerisk,
Am 27.07.2004 um 15:37 schrieb Philipp von Klitzing:
Hi!
Now my question. Is Asterisk an full h323 gatekeeper like openh323?
No, Asterisk has no gatekeeper functionality, at least not yet.
And how can i tell asterisk to sent all none SIP-ip calls to the
gatekeeper over h323?
One (standard) way to
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