Displaying 20 results from an estimated 4000 matches similar to: "Music on Hold for SIP"
2004 Sep 07
1
Asterisk + NetJet (ISDN4Linux)
Well, it has been a long time since I used my NetJet S cards with
asterisk (moved to the quad PRI card) but I am trying to get this
working again for home use.
Basically, what I am using is this:
Linux 2.6.8.1:
config lines:
# ISDN subsystem
CONFIG_ISDN=y
# Old ISDN4Linux
CONFIG_ISDN_I4L=y
# CONFIG_ISDN_PPP is not set
CONFIG_ISDN_AUDIO=y
# CONFIG_ISDN_TTY_FAX is not set
CONFIG_ISDN_DRV_HISAX=y
2004 Apr 07
0
Struggling with ISDN4Linux and Asterisk config
The card is an ASUSCOM ISDNLink PCI (passive) and the circuit is from
Qwest (in the US). I will be using this circuit only for voice (I'm
doing this because of the poor quality of my POTS lines).
I've compiled Hisax (as a module) into my 2.4.25 kernel, and with
'modprobe hisax type=35 protocol=4 id=hisax' I get the following:
Apr 7 10:34:24 dev kernel: HiSax: Linux Driver for
2004 Sep 22
1
(euro)ISDN: complete silence / can't hear a word.
Hello,
I just got my isdn-card working together with i4l and asterisk.
Everything seems to be working fine: I can accept calls coming from the
outside and I can dial out. Even setting the msn works like charm but my
problem is that I cannot hear a word. There's complete silence in both
directions.
Any idea what could be the cause?
Thanks for your help,
Gunther
Lspci:
0000:01:07.0 Network
2004 May 05
7
* & ISDN-BRI-PTP & DID & ISDN4Linux does not show incoming number
Hi,
I am using Asterisk on a DSS1 ISDN-BRI with ISDN4Linux (and a Fritz Card PnP).
The ISDN-BRI is in PTP-Mode (Point to Point "german: Anlagenanschluss")
which is enabled within I4L with "hisaxctrl fcpcipnp0 7 1".
Everything works fine except that I can not see the called number/MSN
of incoming calls within Asterisk and because of this I can not route
incoming calls
2004 Aug 03
0
OH323 not dial Modem[i4l]/g1
Hello everybody,
I have a strange comportment with oh323 and asterisk, I'start testing
asterisk but with this I can't understant plesae help me !
Thanks
Eltorio
----------------------------------------------------------
1/PB: I can't dial from a H323 extensions (registered on a GNU GK) to a
Modem[i4l] line
----------------------------------------------------------
Nothing happens
2004 Sep 21
2
ISDN problem: lacking dialtone
Hi all,
this is a rather "newbie-oriented" question, so please bear with me...
The system running Asterisk has been provided with an AVM FRITZ!Card
PnP. SuSE Linux 9.0 recognizes it right after booting the system and it
seems to be configured (MSN) correctly...
The hwinfo looks like this:
---
pbx:/etc/asterisk # hwinfo --isapnp
11: ISA(PnP) 01.0: 10300 ISDN Adapter
[Created at
2005 Jul 25
1
asterisk + i4l problems
Hello!
Im pretty new of asterisk world: my goal would be using iaxcomm to call ppl over POTN.
Yesterday i configured asterisk to be able to hear a song (with Playback() command) with
iaxcomm, and it was wonderful.
Now, today i tried to configure asterisk to use my ISDN4Linux supported card (a plain HFC card)
with asterisk.
i configured modem.conf this way, after reading
2003 Sep 12
0
Newbie (unfortunately =)) q regarding BRI
Hi all
I've only been working with Asterisk for a matter of days but have
already grown into a big fan =) Much as I've managed to get internal
calling working fine, I have a configuration running on an old PII-233
on RH9 with a (although not badged as is a) Dynalink IS64PH/Winbond
W6692 PCI Card as /dev/ttyI0.
The card works fine in minitel and dials out without a problem.. However
try
2003 Jun 18
2
== Everyone is busy at this time problem
hi,
i installed asterisk and works very well, the only problem is that
when i try to call a direct number of a company that has a normal PBX
i got this error:
to 10.8.210.153:5060
== Accepting call on 'SIP/a.sampietro-f7be' (a.sampietro)
-- Executing Goto("SIP/a.sampietro-f7be", "doisdn|BYEXTENSION|1") in new stack
-- Goto (doisdn,00115601992,1)
--
2004 Jan 07
1
Unexpected ISDN hangup on outbound call
We have setup an asterisk box to let everybody call into the university
internal network, but I get unexpected hangups when doing an outbound call
from SIP to the ISDN interface, and it happens from 20 seconds to some minutes into the
call.
----------the dial and the problem-----------
-- Executing Dial("SIP/57966-a19d", "Modem/g1:96121||rt|") in new
stack
--
2003 Nov 01
1
NetJet Cards
Hello,
I am trying to use 2 netjet cards under asterisk and isdn4linux. I am
having a hard time trying to get them to work in terms of dial out. Does
anyone have a working config I could look at for even one card (tried
that, not much luck either).
When i dial out:
-- Accepting AUTHENTICATED call from 172.16.11.2, requested format =
2, actual format = 2
-- Executing
2005 Mar 18
0
I4l + HiSax
I need HELP pls!
BRISTUFF: Bad Sound quality
CAPI : PTP Mode dont supported
mISDN : kernel is 2.4.x and not 2.6.x
HISAX : PTMP ok, PTP incoming ok but in outgoing asterisk dont
compose number(i listen dial tone and than i can compose number via
dtmf)
Asterisk CLI (g3 is group of Modem[i4l]/ttyI0 and ttyI1):
Called g3: 3453444444
2005 Oct 14
4
[ISDN] Problem: Device '/dev/ttyI1' lacking dialtone
All,
Currently I've got my Asterisk machine running smoothly on IP bases. Meaning I can reach all phones or softphones within my LAN or remote LAN's via VPN. The next step for me is connecting it to the PSTN network.
After some tweaking with the modem.conf I got the i4l driver running correctly, and it appears that my Fritz! ISDN v2 card is working correctly.
I have added the
2006 Feb 01
1
ISDN busy line
I'm trying using Asterisk over an isdn line. I configured my Eicon Diva
card by the hisax module and it works with the Asterisk's chan_modem
(and chan_modem_i4l) driver. The problem is that I can receive calls (on
/dev/ttyI0) but when I make a call I always have the line busy (on
ttyI0 but even if I use an other device like ttyI1). This is a part of
my consol output:
--
2004 Apr 13
0
Dialout from SIP to PSTN
Hi,
i install the Asterisk PBX on a linux machine with i4l to connect to PSTN
(EuroISDN). And i configure a very simple dial plan in extension.conf.
After this, i connect with a SIP program to asterisk and would call my
cellular phone, but got this error:
-- Executing Ringing("SIP/ACzerniak-0904", "") in new stack
-- Executing Dial("SIP/ACzerniak-0904",
2004 Jun 07
1
isdn4linux, NETjet, chan_modem help needed
I'm trying to get a basic Asterisk configuration together for ISDN incoming
/ outgoing calls. I have two Cisco 7905g phones working (at least talking to
each other) and have purchased a NETjet-S PCI ISDN card for routing calls to
/ from ISDN.
The state I've managed to get it to is:-
-- Executing Ringing("SIP/PHONE2-d557", "") in new stack
-- Executing
2004 Jun 22
0
Accessing ISDN with avm bluetooth hardware
Hi *,
This is my first message to this list, so I hope I am not breaking any rule with my post.
My question is the following: can I establish a voice connection over an ISDN interface by using a
bluetooth dongle to connect to the ISDN access point?
Here follow the details:
I have compiled and configured asterisk. Everything seems fine except that when I configure
asterisk to use the ISDN
2004 Apr 21
1
one-way audio and isdn4linux
Hi,
Apologies in advance for the lengthy email.
I'm new to asterisk and have trouble with isdn4linux.
The setup is very basic like this:
winxp ------- asterisk -------- winxp
x-lite | x-lite
|
pstn
The hardware involved is:
Compaq EVO with RH9/kernel 2.4.20-30.9.
Fritz!Card PCI v2
Asterisk CVS-04/17/04-21:36:18
Basically
2003 Feb 21
0
I4l outgoing dtmf problem.
Hi.
I'm working with i4l with asterisk CVS-02/21/03-13:59:12,
plus i4l (chan modem i4l *dsp patched* and kernel 2.4.19
patched to disable dtmf).
All seems ok (apart some echo issues that seems gone
with mec2 aggressive suppressor), but outgoing dtmf
doesn't work . or at least I hear the very first part
of the dtmf, but then it seems suppressed.
here's my modem.conf
[interfaces]
2003 Apr 17
1
timeout music on hold or ring tone
Is any way to limit music on hold (or ringtones) to specified time ? I
need it to play it ~ for 7 seconds .
How to do this ?
in dial plan i have:
exten => _021XXXXXX,4,Dial,Zap/1/BYEXTENSION||r
when go to this extension it rings once!
and then asterisk say :
-- Zap/1-1 answered Modem[i4l]/ttyI0
and it stop ringing ;) becouse mean that other end is ringing :) ..
BUT when the other