[Asterisk-Users] RE: Asterisk-Users Digest, Vol 8, Issue 88
Jeff Glassman
jrglass at columbus.rr.com
Sat Mar 12 16:17:23 MST 2005
These allow and disallow work with NuFone for me
disallow=all
allow=ulaw
allow=alaw
allow=gsm
Jeff
Message: 11
Date: Fri, 11 Mar 2005 11:15:51 +0100
From: "Edward Banfa" <edward at radform.com>
Subject: [Asterisk-Users] NuFone Configuration [problem]
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
<asterisk-users at lists.digium.com>
Message-ID: <200503111016.j2BAFw1s014610 at wrench.thebook.com>
Content-Type: text/plain; charset="us-ascii"
Hello,
I am trying to configure the my asterisk box here with the following
**iax.conf***
[NuFone]
type=peer
host=switch-1.nufone.net
secret=xxxxxx
***extensions.conf:***
exten => _1NXXNXXXXXX,1,Dial,IAX2/xxxxxxx at NuFone/${EXTEN}
exten => _011N.,1,Dial,IAX2/xxxxxx at NuFone/${EXTEN}
I have a couple of Xlite softphones and 2 analogue phones connected to a
mediatrix 1102 connected to our lan. The mediatrix talks sip to the
asterisk
box on the lan. We are running asterisk on FC3 .
SOFTPHONES[XLITE] ---SIP--> ASTERISK----IAX--->NUFONE[ASTERISK]
ANALOGPHONES---MEDIATRIX_1102---SIP--->ASTERISK---IAX--->NUFONE[ASTERISK
]
Well the problem goes something like this.
1) I can dial a number form the softphones and when the call is answered
I
can hear the user on the other end but the user can't hear me
2) I can dial a number from the analog phones (via mediatrix tru to
asterisk)(the mediatrix is properly registered with our asterisk box)
and
when the call is answered both ends can't hear a word, its just silent.
I think I am having a codec problem here. What am I doing wrong. We
would
sincerely appreciate any help/pointers.
Thank you all
Edward Banfa
******EXTENSION.CONF*******
[general]
static=yes
[from-sip]
exten => 100,1,Dial(SIP/edward,20)
exten => 100,2,Hangup
exten => 101,1,Dial(SIP/phone1,20)
exten => 101,2,Hangup
exten => 102,1,Dial(SIP/phone2,20)
exten => 102,2,Hangup
exten => _1NXXNXXXXXX,1,Dial,IAX2/xxxxx at NuFone/${EXTEN}
exten => _011N.,1,Dial,IAX2/xxxxxx at NuFone/${EXTEN}
*****IAX.CONF*****
[general]
port=5036
bind=0.0.0.0
bandwidth=low
disallow=lpc10
[NuFone]
type=peer
host=switch-1.nufone.net
secret=xxxxxx
disallow=all
allow=ilbc
allow=gsm
allow=ulaw
disallow=all
allow=ulaw
allow=alaw
allow=gsm
******SIP.CONF*****
[general]
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
[edward] ;My Xlite softphone
type=friend
host=dynamic
secret=pass-da-word
context=from-sip
callerid="edward" <100>
mailbox=100
disallow=all
allow=gsm
allow=ulaw
allow=alaw
allow=ilbc
allow=g726
[phone1] ;First analog phone connected to mediatrix
type=friend
host=dynamic
secret=pass-da-word
context=from-sip
callerid="phone1" <101>
mailbox=101
disallow=all
allow=gsm
allow=ulaw
allow=alaw
allow=ilbc
allow=g726
[phone2] ;Second analog phone connected to mediatrix
type=friend
host=dynamic
secret=pass-da-word
context=from-sip
callerid="phone2" <102>
mailbox=102
disallow=all
allow=gsm
allow=ulaw
allow=alaw
allow=ilbc
allow=g726
------------------------------
Message: 12
Date: Fri, 11 Mar 2005 15:57:38 +0530
From: Jagan Mohan <jaganmk at gmail.com>
Subject: [Asterisk-Users] Load Balancing b/w 2 asterisk servers using
SIP load balancer
To: Asterisk <asterisk-users at lists.digium.com>
Message-ID: <52a9bccc05031102273d89f61d at mail.gmail.com>
Content-Type: text/plain; charset=US-ASCII
Hi,
I'm trying to do load balancing between 2 asterisk servers using SIP
load balancer, provided by http://www.vovida.org
I used the following options on lbproxy, but I get the below message
continuously.
./lbProxy -name seneca -reqPort 5060 -respPort 5061 -proxy A1 -proxy A2
"No proxies are up - can not send message to anyone"
Xlite is not able to register to the asterisk server.
Is there anything which needs to be tweaked on Asterisk side to get this
working? Please help.
Thanks,
Jagan
------------------------------
Message: 13
Date: Fri, 11 Mar 2005 11:31:29 +0100
From: "Vledder, Hans" <Hans.Vledder at nl.compuware.com>
Subject: RE: [Asterisk-Users] Asterisk and USB ISDN controllers ...
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID:
<D913221A882FD31198D90008C75D69090F140249 at cwnl-ams-pri01.nl.compuware.co
m>
Content-Type: text/plain; charset="iso-8859-1"
Hi Steve,
>Since you don't mention what USB ISDN adapter specifically you are
>thinking about, what do you think we will be able to tell you.
All I know about the adapter is what I've told you. It's a USB
Colognechip
based ISDN controller - probably HCF-USB based. It's supported by Linux,
but
there's no info on access to B and D channels.
Regards,
Hans
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of Steven
Critchfield
Sent: Thursday, March 10, 2005 6:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk and USB ISDN controllers ...
On Thu, 2005-03-10 at 18:13 +0100, Vledder, Hans wrote:
> Guys,
>
> I am planning on building a small SIP PBX with a single ISDN line.
Currently
> I am looking into the specs of a very tiny barebone system that has an
> option Colognechip base ISDN controller. The only thing is that the
ISDN
> module that comes with this barebone hooks up to the motherboard using
USB.
> My intention is to allow incoming and outgoing calls from SIP to ISDN.
Is
> this setup in any way supported by *?
Since you don't mention what USB ISDN adapter specifically you are
thinking about, what do you think we will be able to tell you.
The first step would really be to ask if your specific ISDN adapter can
be used under linux. After that, can that specific ISDN adapter give
access to voice channels. What method is used to get access to the audio
and the signaling.
It may well be usable if the drivers for it implements the same API as
the current ISDN cards in use support.
--
Steven Critchfield <critch at basesys.com>
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------------------------------
Message: 14
Date: Fri, 11 Mar 2005 11:33:33 +0100
From: pbx <pbx at itcee.be>
Subject: Re: [Asterisk-Users] TE110P experiance
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <423173FD.2000106 at itcee.be>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Mario.Spoljar at hypo-alpe-adria.com wrote:
>
>
>Hello to all,
>I would like to ask some Digium TE110P users if they can share
experiance
>about this card. I put in service card yesterday but I noticed
following
>(strange) behaviar:
>- if I have to reboot my computer my zaptel driver fail to start and
>produce this error:
> ZT_SPANCONFIG failed on span 1: No such device or address (6)
>- to solve this problem I have to power cycle my computer and in all
cases
>this brings up card!
>
>- does anybody have any info about this hardware, example there are two
LED
>- what is the meaning of these LEDs. I bought this card and got anly
card
>without any papers (just bill :-( )
>Regards,
>
>mario.spoljar at hypo-alpe-adria.com
>
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users at lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
>To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
This is a well known bug ( can't remeber the number) but
the Card Identification on the bus PCI change (for some reason) and
the driver is not able to find the card anymore,
try to add the line marked with a + in the wcte11cxp.c (in zaptel
source), and recompile your driver.....
If you like tio verify the bug
start your your system from power down
do a lspci -v locate "Tiger Jet Network "
look the id
load zaptel driver
do lspci -v
ans see the difference
Hope this help
static struct pci_device_id t1xxp_pci_tbl[] = {
{ 0xe159, 0x0001, 0x71fe, PCI_ANY_ID, 0, 0, (unsigned long)
"Digium Wildcard TE110P T1/E1 Board" },
{ 0xe159, 0x0001, 0x79fe, PCI_ANY_ID, 0, 0, (unsigned long)
"Digium Wildcard TE110P T1/E1 Board" },
{ 0xe159, 0x0001, 0x795e, PCI_ANY_ID, 0, 0, (unsigned long)
"Digium Wildcard TE110P T1/E1 Board" },
{ 0xe159, 0x0001, 0x79de, PCI_ANY_ID, 0, 0, (unsigned long)
"Digium Wildcard TE110P T1/E1 Board" },
+ { 0xe159, 0x0001, 0x797e, PCI_ANY_ID, 0, 0, (unsigned long)
"Digium Wildcard TE110P T1/E1 Board" },
{ 0 }
};
------------------------------
Message: 15
Date: Fri, 11 Mar 2005 11:47:03 +0100
From: "Dennie Verstrepen" <Dennie.Verstrepen at secuteam.com>
Subject: [Asterisk-Users] FW: IAX Settings
To: <asterisk-users at lists.digium.com>
Message-ID:
<7A222250F3B4344F9E01786F77D4BEB13D298B at SCTSERVER.secuteam.local>
Content-Type: text/plain; charset="iso-8859-1"
> Hello,
>
> Has anyone a complete overview of all the settings you can use in the
iax.conf file and also where those settings can belong (e.g. in the
general section, in a context of type=peer or type=user)?
>
> Thank you in advance
>
> Dennie
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Message: 16
Date: Fri, 11 Mar 2005 10:50:37 +0000 (UTC)
From: tony at softins.clara.co.uk (Tony Mountifield)
Subject: [Asterisk-Users] Intermittent volume deterioration in
conferences
To: asterisk-users at lists.digium.com
Message-ID: <d0rt5t$8u3$1 at softins.clara.co.uk>
I wonder if anyone can suggest ways to diagnose an infuriating problem
being experienced by customers of a company I did a large Asterisk
project for.
First some background:
The system is a conferencing system using a modified MeetMe. There are
seven Asterisk boxes (we call them bridges) each with four T1 PRIs into
a
TE405P. No VoIP is involved. A conference is always local to a single
bridge. The conference leader has a control screen and may dial into
the
bridge, or may instruct the bridge to dial him/her. Once the leader is
in
the conference, they instruct the bridge to dial each other participant.
Each conference is recorded locally in the Asterisk system. The bridges
are in Oklahoma and all the leaders and most of the participants are all
over Texas.
The problem:
For the first three or four months of operation everything went very
well, but from early February the customer started reporting problems
with the volume of audio. Initially the reports seemed to be localized
to a particular area of Texas, and to be small in number. Over time,
they have increased in frequency and been reported from different areas.
Sometimes one participant can't be heard very well by the others, and is
also faint on the recording. Other times a participant has trouble
hearing the others, but the others are ok on the recording. There does
not seem to be any significant distortion, just faint volume.
It sounds to me like a phone network issue, but proving that is turning
out to be a nightmare. The fact that it is not confined to one bridge
but is randomly spread across them would seem to suggest it is not a
bridge hardware problem, because it is unlikely to happen in them all.
No changes were made to the hardware, Zaptel drivers or Asterisk on the
bridges since installation.
A day or so ago we disabled echo cancellation on the zap channels, to
see
if that would make a difference, but it doesn't seem to have. It still
wouldn't explain why the problem did not previously exist, and started
happening spontaneously.
Sometimes if it's really difficult for people to hear, the leader closes
the conference and reverts to their older conferencing system (that our
system replaced), and reports that the volume is then fine. I don't know
where the older system is located, but I believe it is more local. This
is
obviously a worrying scenario.
If anyone can suggest any ideas of ways to tackle the problem, and to
determine whether it really is the Asterisk bridges or the phone
systems,
I would be very, very grateful, as it is turning into a nightmare!
Cheers
Tony
--
Tony Mountifield
Work: tony at softins.co.uk - http://www.softins.co.uk
Play: tony at mountifield.org - http://tony.mountifield.org
------------------------------
Message: 17
Date: Fri, 11 Mar 2005 12:02:21 +0100
From: Giovanni Miano <giomiano at gmail.com>
Subject: [Asterisk-Users] Asterisk + Call hangup
To: Asterisk-Users at lists.digium.com
Message-ID: <d75be1ca0503110302165f7e7c at mail.gmail.com>
Content-Type: text/plain; charset=US-ASCII
Scenario
PSTN <-> ZAP CHANNEL <-> ASTERISK <-> SIP
When i recive call i fwd it to SIP Phone
-> SIP PHONE ringing
If From External Line PSTN hungup call SIP Phone Ringing too, why ?
------------------------------
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