Displaying 20 results from an estimated 100 matches similar to: "Probs with oh323 driver: audio only in 1 direction"
2004 May 06
4
asterisk-oh323, new version 0.6.1
Hello all,
This new version (0.6.1) of asterisk-oh323 fixes the "one-way audio"
problem of the previous release.
Download from the usual location:
http://www.inaccessnetworks.com/projects/asterisk-oh323
Regards,
Michael.
2005 Aug 22
1
Asterisk ISDN CallerID identification failure
Hello,
We have 4 'Onramp-2' Telstra ISDN BRI services operating on Asterisk
Server with Eicon 4BRI card. For most part the service is okay.
However, we are are having problems with passing callerID to internal
extensions.
This is the set of command executed.
exten => <pattern>,1,Answer ; Answer the line
exten => <pattern>,2,NoOp(${DNIS}) ; debug statements
exten =>
2017 May 12
3
pjsip: asterisk can't decide which codec to use
Hello!
I'm facing completely choppy sound. The wireshark trace shows, that
there are a lot of codec changes without any trigger (means no options
or reinvite or any other package).
Background:
The call is initiated by asterisk and is received by the same asterisk
conference room via
Phone extension -> asterisk -> provider A -> provider B -> asterisk.
Asterisk initially sends
2004 Jul 12
2
Indications missing on Cisco FXO -> ATA-186 (SIP)
Calling from a Cisco FXO port to an ATA-186 (SIP 3.1 image) via *
(either CVS-HEAD-06/28/04-11:43:41 or CVS-HEAD-07/12/04-15:49:58)
I didn't hear any ringing sound & get the following on the console:
-- Called 5503
-- SIP/5503-f6b5 is ringing
WARNING[-1323201616]: channel.c:1375 ast_indicate: Unable to handle
indication 3 for 'SIP/10.10.2.250-9903'
-- SIP/5503-f6b5 answered
2004 Apr 29
1
CAPI ptp does not work
Hallo all,
I am trying to get * with chan_capi and a ptp-ISDN with 4 lines on a AVM C4
card to work.
But weather inbound nor outbound is working :(
My capi.conf:
;
; CAPI config
;
;
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
mode=immediate
isdnmode=ptp
msn=8993
incomingmsn=*
mode=immediate
controller=1,2,3,4
softdtmf=1
;accountcode=
context=demo
2003 Sep 03
8
Asterisk Jitters
Hi,
Every time I dial into my asterisk box i hear nothing but asterisk
jittering.
The following is an example of what I get on the asterisk CLI
Thanks
*CLI> DEBUG[81926]: File chan_sip.c, Line 3826 (check_user): Setting NAT
on RTP
to 0
DEBUG[81926]: File chan_sip.c, Line 4807 (handle_request): Check for res
DEBUG[81926]: File chan_sip.c, Line 952 (find_user): Call from user
2006 May 21
1
no ringtone
Hi,
I have a queue that plays music when a call comes in. To be able to do
that I need to Answer() the call first. After a timeout in this scenario
the call should be transfered to an extension using a GoTo statement to
the extensions context. The problem is that as soon as asterisk Answers
the call it can not play a ringtone (or other tones) back to the
original caller when executing a Dial
2008 Dec 05
2
async agi question
Hi,
I am developing asterisk support for our application using the Async AGI
and Asterisk-Java.
One thing I haven't been able to implement is how to stop playing a
sound. Something similar to StopIO for Dialogic GlobalCall or
DivaStopSending for Eicon.
Is there any way to achieve this today which I have missed? Or could
someone give me hints on how I could implement this in the res_agi.c The
2010 Aug 04
1
Asterisk not working with Festival
Hello,
I am having a Mac 10.6.4 (Snow Leopard). I have compiled and built Asterisk
1.6.2.9 and Festival 2.0.95:beta on my machine. Asterisk is working fine
with SIP channels without Festival. I have written following context in
extension.conf:
[connect-to-me]
exten => s,1,Answer
Exten => s,n,SayDigits(?1?)
exten => s,n,Festival(hello john)
exten => s,n,Hangup
I use call files to
2006 Jan 24
2
Re: Asterisk-Users Digest, Vol 18, Issue 134
O.K. thanks a lot, Felix and Peer Oliver. But somehow asterisk keeps
telling me while startup:
[chan_capi.so] => (Common ISDN API for Asterisk)
Jan 24 14:30:47 NOTICE[9796]: chan_capi.c:3271 load_module: Unused
contr1
Jan 24 14:30:47 NOTICE[9796]: chan_capi.c:3271 load_module: Unused
contr2
Jan 24 14:30:47 NOTICE[9796]: chan_capi.c:3271 load_module: Unused
contr3
Jan 24 14:30:47
2004 May 24
1
Chan_capi 0.3.1 , Asterisk , 3 x C4 active ISDN card Segmentation fault
Hi,
i use chan_capi 0.3.1 with asterisk (stable branch cvs) and 3 x c4
active ISDN card.
From Controller 1 - 7 there are no problems making calls between
asterisk and the pstn.
But when i make calls from controller 8 - 12 i get on every controller
(8 - 12) a segmentation fault in asterisk :(
I tried different linux distributions (gentoo 2004.1, redhat 9.0 , suse
9.1) but same error.
2003 May 27
1
chan_h323 + Ericsson Webswitch 100
I'm haveing trouble connecting an Ericsson Webswitch 100 to asterisk.
Has anyone gotten a Webswitch running? When I try to connect asterisk
thinks everything works fine, while the webswitch just rings. I belive
chan_h323 is picking the wrong port to talk at the webswitch on, however
I'm not sure, nor am I sure how to fix it. Any clues/hints? A tcpdump
is attached to show the session.
2006 Mar 14
3
Attended Transfer - transfer timeout, how to change?
Hi,
We are trying to use attended transfer with Asterisk 1.2.5, but when we
do the transfer and dial the new number, it times out after 3 rings and
then the callee is put back to the original agent.
Where can I adjust the timeout which applies to the number we are
transferring to? I have changed the extension for this number to timeout
at 60 seconds, but that seems to make no difference.
--
2004 Aug 12
1
AgentLogin issue
Hi
i have an issue getting agentLogin working
/etc/asterisk/queues.conf
member => Agent/1001
member => Agent/1002
extension.conf
exten => 110,1,Wait,1
exten => 110,2,AgentLogin()
now, i call 110 by a firefly client, trying to login in as 1001 agent:
Aug 12 16:31:36 DEBUG[1103408048]: chan_sip.c:4423 build_route: build_route: Contact hop: <sip:sip3@192.168.1.151:5060>
--
2005 Jan 19
1
How to change the packet size
Hi,
We observed the packet size used in asterisk is about
20 ms.
We would like to know if is possible to change this
value to 10 or 30 ms for example.
If so, how could I change it?
Thanks in advance and best regards
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2004 Aug 31
2
Asterisk codecs and packet size
Does anybody knows if it's posible or if there is some develoment in
course to be able to use longer transmit packet sizes (as long as I know
this is fixed in 20ms now) with the compressed voip codecs in asterisk
(g729, g726, gsm, etc).
I need to use asterisk to connect remote sip clients with 24kb bandwidth
lines and I'm using a licences g729 codec but because I can't increase
2003 Nov 17
1
ISDN debugging and SIP dial-in issue]
(I have some problems with my mailing-list alias, I hope this
doesn't get sent twice)
On Sat, Nov 15, 2003 at 04:35:20PM +0100, Philipp von Klitzing wrote:
Thank you for your comments Philipp:
> > - with a SIP phone configured as 192.168.1.190, and with its SIP
> > server being 192.168.1.190
>
> That doesn't look right. Do you have another "SIP
2003 Aug 26
1
More questions. Call Waiting and Threeway
I can't do threeway from my Grandstream phone. Looking through the server
config files, I figured out why - zapata.conf has Threeway turned off for
the channels I use.
I do my work on someone else's Asterisk box and don't want to modify
zapata.conf for several reasons, the biggest being that the guy who owns
the box has a couple clients using it and I am deathly afraid of breaking
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
2005 Aug 02
0
Hang up as soon as other party picks up call
Hello,
I have an Asterisk box with a TE410P connected to a PRI line and agents with
X-Lite installed on the same LAN as the Asterisk server. Sometimes, when I
make outbound calls it hangs up as soon as other party tries to picks up the
call. Does someone ever experienced this situation? On X-Lite, only
G711-ulaw is enabled and here is what i put in sip.conf:
[4001]
type=friend
username=4001