Displaying 20 results from an estimated 300 matches similar to: "SIP<->H323 "Failed to create smoother""
2005 Aug 13
2
forward incoming analog call to SIP?
I'm trying to setup a demo where my Asterisk box with a TDM01B (FXO)
answers an incoming call and forwards that call to a SIP softphone (X-lite.)
Seems all is built/installed okay:
# ztcfg -vv
Zaptel Configuration
======================
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
1 channels configured.
I'm pretty new at this and the extensions.conf file is eating my
2005 Oct 13
1
Noob help with IAX
Ok so I've just built and installed a CVS (HEAD) version of asterisk
on RHFC2 running a 2.6.13.3 kernel.org kernel. I installed the samples
via "make samples". Everything seems to work except one thing. I'm
trying to do the connect to the Digium IAX demo server portion of the
demo (dial 500) and I just get the following messages. I am behind a
NAT server and did NOT change
2005 Sep 02
2
chan_capi hfcpci mISDN linux 2.6.12 not working
Hello,
These are error messages I get when I try to call a number over CAPI channel.
-- Executing SetCallerID("SIP/xlite1-3b80", "0") in new stack
-- Executing Dial("SIP/xlite1-3b80", "CAPI/hfcpci/b17") in new stack
> data = hfcpci/b17
> capi request for interface 'hfcpci'
== hfcpci: Call CAPI/hfcpci/b17-1 (pres=0x00,
2005 Sep 05
0
ooh323c h323_convertAsteriskCapToH323Cap Don't know how to deal with mode 0x40 (slin)
Hello,
I have the following setup:
(*)<--->IP<--->Micronet 5012 H.323 box <---> POTS <---> PBX (Alcatel OmniPCX)
Grand idea is to use the micronet's POTS interfaces to connect SIP
phones to the PBX and to the PSTN. I think i even managed my way in
the arcane and cryptic management interface of that appliance, but I
am stuck against theese messages:
-- Executing
2005 Oct 17
1
Call transfer - atxfer
Hi,
I try to set up attended transfer in my Asterisk Box . My
features.conf look like this:
[general]
parkext => 100
parkpos => 1-5
context => parkedcalls
parkingtime => 100
transferdigittimeout => 3l
courtesytone = beep
xfersound = beep
xferfailsound = invalid
featuredigittimeout = 500
;adsipark = yes
pickupexten = *8
[featuremap]
atxfer => *2
blindxfer => #
disconnect
2003 Oct 21
1
Hangup
Hi,
Some calls I make trough my PSTN asterisk gateway just hangup
after some minutes. Even if I'm using sip or iax. I have callprogress=no
busydetect=no in my zapata.conf.
Anyone help? Or tell me what to look at /var/log/asterisk/debug. I
didn't find anything wrong.
[endpoint]---iax or sip----[asterisk]----E&M----PSTN.
As endpoint I had tested another asterisk box (with a FXS),
2004 Sep 04
1
Oh323, Please Help Newbie ;(
Hi,
I just installed OH323 Plugin and im now tryin to make
simple Configuration to connect Openphone and Xlite to
my Asterisk-Server.
All works fine, i just wanna know if there's a
better way to do it? Is there anything wrong with my
Config?
OH323.conf
[general]
listenAddress=0.0.0.0
listenPort=1720
connectPort=1720
tcpStart=10000
tcpEnd=20000
udpStart=8000
udpEnd=8005
fastStart=no
2006 Mar 07
1
Setting Vaaibles
Helo List,
First I would like to apologize for my bad spelling as
well as that I did not search the wiki first. I only
have email access at the moment.
I am having trouble setting both variables and global
variables thru an extension.
I am using Asterisk 1.2.4 with Ztdummy on CentOS 3.4
with an Xlite softphone. I have two xlite phones on
diffent computers. One logs in as xlite1 and the other
as
2008 Oct 13
1
Need help for debuging
I am running asterisk 1.2.27 and it dead today. The following is the backtrace of core file. Can anybody help me to identify what is the possible cause of crash?
It seems the mysql connection causing problem in Thread 2. But I can not tell what exactly happened.
This asterisk is using as ACD for over hundred agents.
#> thread apply all bt
........
........
Thread 6 (process 20135):
#0
2003 Apr 20
1
Macros not working as expected with extension "h" in some circumstances
I have a question on how to handle the "h" routines. I have noticed that if the call is hung up by the side that originated the call, the "h" routine is not extendable via a macro, or at least I have been unable to do it. My tests have included only SIP->SIP calls.
If the originating side hangs up first: The macro is called from "exten =>
2004 Oct 03
0
Call gets disconnected upon connect
Hi Everybody,
I am trying to use SIP (Sipura 2000) to connect to Asterisk which then
dials out a local number using the Digium E100P. We have purchased the
G729 codec licenses from Digium and loaded them into Asterisk
successfully. However, the call drops immediately after being answered
with the debug error message saying something like: "channel.c:2646
ast_channel_bridge: Didn't get a
2008 Jul 07
1
DTMF on iax channel is not interpreted by asterisk
Hi!
I'm using asterisk 1.4.17 with twinkle and a custom phone based on
iaxclient 2.0.2 and I'm struggling a bit with DTMF and features.conf.
While the twinkle client is able to initiate an attended transfer using
*2 (as configured in features.conf), the iax client is not. I can see
the DTMF messages showing up on the asterisk console, but asterisk does
not invoke the features
2004 Oct 05
1
Why I don't hear Call Progress
I'm using sipgate.de as my sip provider. When I'm using xlite as client
on sipgate.de, everything works fine: I call number, hear ringing (real
progress tone form called party, not one generated in xlite) and then
talking with called person.
But, when I'm using Asterisk as sip client on sipgate.de, I don't hear
progress tones: I hear only one (locally generated) ring tone, and
2005 May 26
1
Little Php question
> -----Original Message-----
> From: Ronald [mailto:asterisk107@gmail.com]
> Sent: 26 May 2005 10:47
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Little Php question
>
>
> Hi
> I'm trying to make a call from a local webpagee through my
> xlite softphone
> (xlite1)
> BTW when I'm trying to do it through
2005 May 26
0
SV: Little Php question
Hi
I think you should have a look at the end of line - you are missing " :-)
Br,
dmirty
-----Oprindelig meddelelse-----
Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af Ronald
Sendt: 26 May 2005 11:47
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: [Asterisk-Users] Little Php question
Hi
I'm trying to make
2005 Jan 12
1
linphone -> NAT -> * -> NAT -> firefly woes.
Hi folks
an issue I don't understand. I'm running * stable 1.0.3 on public
internet, with following iax.conf / sip.conf entries:
iax.conf
[100]
type=friend
username=Foo
context=default
auth=md5,plaintext,rsa
secret=secret
host=dynamic
callerid="Foo" <100>
qualify=no
sip.conf
[10]
type=friend
username=Bar
context=default
callerid=Bar <10>
2005 Apr 07
1
"404 User Not Found" when calling between two SIP UA's
The configuration for kphone in sip.conf:
[177204]
;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
type=friend
;regexten=1234 ; When they register, create extension 1234
;username=xlite1
;callerid="Jane Smith" <5678>
host=dynamic
;nat=yes ;
2009 Oct 02
0
Sending a DTMF remotely with PlayDTMF problem.
Hello,
I need to be able to send a DTMF to an existing channel remotely. So I made
a php script to do such with the Manager command PlayDTMF. I need it for
example to start a transfer.
isb177*CLI> features show
Builtin Feature Default Current
--------------- ------- -------
Pickup *8 *8
Blind Transfer # #8
Attended Transfer
2009 Oct 03
0
Problem sending a DTMF remotely. Please need help...
Hello,
I need to be able to send a DTMF to an existing channel remotely. So I made
a php script to do such with the Manager command PlayDTMF. I need it for
example to start a transfer.
isb177*CLI> features show
Builtin Feature Default Current
--------------- ------- -------
Pickup *8 *8
Blind Transfer # #8
Attended Transfer
2009 Oct 06
0
Problem sending a DTMF remotely. Please need help!!!
Hello, how are you?
I need to be able to send a DTMF to an existing channel remotely. So I made
a php script to do such with the Manager command PlayDTMF. I need it for
example to start a transfer.
isb177*CLI> features show
Builtin Feature Default Current
--------------- ------- -------
Pickup *8 *8
Blind Transfer # #8
Attended