Displaying 10 results from an estimated 10 matches for "sip_provid".
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sip_provider
2006 May 26
0
SIP call problem
Hello,
I have problem to make SIP calls, i have asterisk +
PC InterP4 + Digium TDM400P
here is the content of the sip.conf:
[SIP_PROVIDER]
type=peer
fromuser=testcomclient
username=testcomclient
secret=testr
host=IP_SIP_PROVIDER
;dtmfmode=rfc2833
context=interne
canreinvite=no
;allerid=Beer
disallow=all
allow=ulaw
allow=gsm
allow=g723.1 ; Asterisk only
supports g723.1 pass-thru!
allow=g729
Th...
2005 Jul 01
2
Sip.conf problems
Hi,
I have been trying to configure my Asterisk to use a Sip provider for
out and incoming calls.
I only have one user and password for connect to my sip provider.
My sip.conf is:
[general]
;disallow=gsm
;allow=ulaw
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = default ; Default for incoming calls
callerid=No
2011 Jan 26
0
SIP channel status - Why is it different when calling an internal extension rather than an outside line over SIP?
...ck line right away. What is the
fault behind this and what is workaround?
This works:
*originate sip/101 extension s at dial_wait*
[dial_wait]
exten => s,1,Answer
exten => s,n,Playback(Please_wait_as_dial_the_second_party)
exten => s,n,NoOp(Calling second party)
exten => s,n,Dial(SIP/sip_provider/12145556666)
This doesn't wait for channel to come up and jumps to Playback (s,2) without
even the first party yet picking up:
*originate SIP/sip_provider/12148889999 extension s at dial_wait*
*
*
*Thanks,*
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2008 Dec 15
3
Variables for dial plan
...p.conf
[client1]
dialplan=NZ
..........
In extensions.conf (Logic expressed using PHP style)
if ($dialplan == NZ) {
$NAT = 0;
$INT = 00;
};
and in the [outgoing] section
; Australia
exten => _${INT}61[278]NXXXXXX.,1,Set(CDR(UserField)=AUSTRALIA)
exten => _${INT}61[278]NXXXXXX.,n,Dial(SIP/SIP_PROVIDER/0${EXTEN:4:9})
How can I implement this in Asterisk style?
Thanks,
Michael
2006 Apr 27
1
asterisk spandsp and txfax
...2email on my asterisk box.
The rxfax works fine in my setup.
The problem is with the txfax.
I have tryed all snadsp version (0.0.2x and 0.0.3x) but I get this
errors. Because I can't find anything on Internet I'm hoping u can give
me a hand.
here are my logs:
-- Attempting call on SIP/sip_provider/1234 for application txfax(/va
r/spool/asterisk/fax/215690048.1145968036.383.tif|caller|debug) (Retry 1)
> Channel SIP/mc3810-a20f was answered.
> Launching
txfax(/var/spool/asterisk/fax/215690048.1145968036.383.tif|ca
ller|debug) on SIP/mc3810-a20f
FLOW Slow carrier up
FLOW S...
2006 Mar 07
3
Re: [asterisk-dev] Is there a way to define an outbound proxy in sip.conf ?
...back to the sip proxy.
Is it possible ?
ASTERISK
| |
| |
==SIP proxy===sip agents
Thanks for help
Harry
--- Wilmar Campos <wilmar.campos@gmail.com> a ?crit :
> Yes:
>
>
> ;
> ; Provider or Remote PEER
> ;
> register => 800:345698:12345678@sip_provider/800
>
> [sip_provider]
> type=peer
> context=default
> ;secret=345698
> fromuser=800
> host=sip.provider.com
> ;language=es
> dtmfmode=rfc2833
> disallow=all
> allow=g729
> canreinvite=no
> I hope this is what you are looking for.
>
> Regards,
>...
2004 Jul 12
0
GnuGK + Asterisk + SIP Provider
...|
H.323 SIP
And I wanna configure a setup that the SIP terminal
talk with the H323 terminal. For this I would like use
the asterisk.
My h323.conf file is like:
[general]
gatekeeper=10.11.2.80
AllowGKRouted=yes
[H323Asterisk]
type=h323
context=sip_provider
prefix=113151,116462
[default]
type=h323
context=default
My sip.conf file is like:
[default]
context=default
My extension file is like:
exten=_1131517400,1,Dial(h323/1131517400@10.11.2.80,10)
exten=_1131517401,1,Dial(h323/OPENH323GK@10.11.2.80,10)
exten=_2001,1,Dial(h323/2001@10.11.2.80,10)
ex...
2010 Apr 20
0
SIP one-way audio
Hi,
This problem has been tackled over and over, I know.
I'm trying to understand why I'm having trouble with my "simple setup".
My setup is like this:
<SIP_PROVIDER>---<DSL1>---<LINUX_GATEWAY>---<ASTERISK_VIA_DSL1>
I've unloaded the nf_*_sip kernel modules from the LINUX_GATEWAY just in case they could interfere.
The DSL1 modem/router is a THOMSON SPEEDTOUCH and I've disabled "SIP ALG".
If I place a SIP call then I i...
2007 May 05
1
SIP registration problem
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2007 Mar 30
4
Speed Dial Application in *
Hi all,
Is there a "speed dial" type application in *? The NEC PBX we
currently use has a feature which allows any phone to access a
system-wide speed dail database simply by keying the speed-dial number
and pressing the 'redial' key from any extension. Even using a vinella
phone on an sli the user can dial 77+speedial# and access this
directory.
Does * have a similar