[Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #3387 - 9 msgs
Jain, Sonal
Sonal.Jain at Sterlingbancorp.com
Mon Apr 12 09:04:59 MST 2004
How do you setup the timing in Meetme conference? I have a x100p and tdm4x card.
When I dialing to my conference I get a request to schedule in the past error message.
thanks
-----Original Message-----
From: asterisk-users-admin at lists.digium.com [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of asterisk-users-request at lists.digium.com
Sent: Saturday, April 10, 2004 10:48 AM
To: asterisk-users at lists.digium.com
Subject: Asterisk-Users digest, Vol 1 #3387 - 9 msgs
Send Asterisk-Users mailing list submissions to
asterisk-users at lists.digium.com
To subscribe or unsubscribe via the World Wide Web, visit
http://lists.digium.com/mailman/listinfo/asterisk-users
or, via email, send a message with subject or body 'help' to
asterisk-users-request at lists.digium.com
You can reach the person managing the list at
asterisk-users-admin at lists.digium.com
When replying, please edit your Subject line so it is more specific
than "Re: Contents of Asterisk-Users digest..."
Today's Topics:
1. Re: Re: Analogue telephone cards for the UK (Iain Stevenson)
2. Re: Re: Analogue telephone cards for the UK (WipeOut)
3. Re: Re: Analogue telephone cards for the UK (Paul Tyreman)
4. No ringing tone with IAXY (and other bits and bobs) (Chris Orme)
5. Nothing to do? Go bounty-hunting! (Olle E. Johansson)
6. RE: No ringing tone with IAXY (and other bits and bobs) (Brian Cuthie)
7. RE: No ringing tone with IAXY (and other bits and bobs) (Rich Adamson)
8. Extensions and Include (Kevin )
9. RE: No ringing tone with IAXY (and other bits and bobs) (Brian Cuthie)
--__--__--
Message: 1
Date: Sat, 10 Apr 2004 10:53:15 +0100
From: Iain Stevenson <iain at iainstevenson.com>
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Re: Analogue telephone cards for the UK
Reply-To: asterisk-users at lists.digium.com
--On Saturday, April 10, 2004 10:42:26 +0100 Paul Tyreman
<paul at tyreman.org.uk> wrote:
>
> Thanks for all the replies.
>
> Can someone tell me if it is possible to put two of these X100P cards
> into the same machine to order to gain access to two BT landlines ?
I believe so although problems have been reported with certain motherboards
- best to search this list before buying.
> Would it also be possible for someone to outline in a bit more detail the
> procdue for limiting which phones have access via the card as I am new to
> Asterisk.
You need to define a context for outgoing calls which will include dial
commands for the X100P. You then define additional contexts for local
phones. Only those local contexts that "include" the outgoing context
will be able to make outgoing calls. Start with a "bare bones"
extensions.conf or you'll find * very hard going.
> What happens when someone calls the number of the line the card is on -
> Do all phones ring or what happens ?
You define that in extensions.conf. Incoming calls will land in the
context you specify in /etc/asterisk/zapata.conf
> Is that auto attendant thing a real
> possiblity. What I would idealy like is this...
> Welcome. If you know the extention you wish to call, press * now and
> then dial it. Otherwise, press 1 for Family A, 2 for Family B and 3 for
> Family C. If the user Presses 1, Press 1 for Person A, Press 2 for
> Person B. etc ?
>
> Is that possible ?
... I dunno - sorry
Iain
--__--__--
Message: 2
Date: Sat, 10 Apr 2004 11:15:30 +0100
From: WipeOut <wipe_out at users.sourceforge.net>
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Re: Analogue telephone cards for the UK
Reply-To: asterisk-users at lists.digium.com
Paul Tyreman wrote:
> Thanks for all the replies.
>
> Can someone tell me if it is possible to put two of these X100P cards
> into the same machine to order to gain access to two BT landlines ?
>
> Would it also be possible for someone to outline in a bit more detail
> the procdue for limiting which phones have access via the card as I am
> new to Asterisk.
>
> What happens when someone calls the number of the line the card is on
> - Do all phones ring or what happens ? Is that auto attendant thing a
> real possiblity. What I would idealy like is this...
>
> Welcome. If you know the extention you wish to call, press * now and
> then dial it.
> Otherwise, press 1 for Family A, 2 for Family B and 3 for Family C.
> If the user Presses 1, Press 1 for Person A, Press 2 for Person B.
> etc ?
>
> Is that possible ?
>
> Thanks, Paul.
>
It sounds like you are trying to share the PBX between multiple people..
I would suggest getting an ISDN BRI line and an AVM Fritz card (using
the chan_capi driver).. This will give you two lines onto which you can
get 8 MSN's (an MSN is another number coming in on the same BRI).. You
can setup Asterisk to route the calls to the correct phones or group of
phones based on the number that was called..
If you are in the UK there are plenty of Fritz cards around and this
method will also allow you to have CallerID if you want it where the
analog cards have issues with CallerID..
Later..
--__--__--
Message: 3
From: "Paul Tyreman" <paul at tyreman.org.uk>
To: <Asterisk-Users at lists.digium.com>
Subject: Re: [Asterisk-Users] Re: Analogue telephone cards for the UK
Date: Sat, 10 Apr 2004 11:55:26 +0100
Reply-To: asterisk-users at lists.digium.com
This is a multi-part message in MIME format.
------=_NextPart_000_0005_01C41EF2.B6A5E1E0
Content-Type: text/plain;
charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable
What I want to do is have the asterisk server sat in my house and used =
by my family to access the BT landline and to recieve calls made to that =
landline. If it is not possible to do the auto attendant thing then so =
be it, I will just have all phones in my house ring when a call is made =
on the BT line. That should be easy, right ?
In addition to running the server just for my house, I want to have =
other memebers of my extended family link up to the server via their =
broadband connections so we can make free calls to each other over the =
internet connections.
What I don't want is for other members of my family (who are not =
resident in my house) to be able to make calls on my BT landline, but I =
do want them to be able to make unlimited calls to other extentions on =
the asterisk server.
Since I already pay monthly for broadband, I am not very keen to start =
paying more for an IDSN line which will only be used for this project. =
I don't use / need caller ID on external calls, so thats not an issue.
Does that all make sence ?
Thanks, Paul.
-----Original Message-----
From: asterisk-users-admin at lists.digium.com =
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of WipeOut
Posted At: 10 April 2004 11:16
Posted To: Asterisk-Users
Conversation: [Asterisk-Users] Re: Analogue telephone cards for the UK
Subject: Re: [Asterisk-Users] Re: Analogue telephone cards for the UK
It sounds like you are trying to share the PBX between multiple people..
I would suggest getting an ISDN BRI line and an AVM Fritz card (using=20
the chan_capi driver).. This will give you two lines onto which you can=20
get 8 MSN's (an MSN is another number coming in on the same BRI).. You=20
can setup Asterisk to route the calls to the correct phones or group of=20
phones based on the number that was called..
If you are in the UK there are plenty of Fritz cards around and this=20
method will also allow you to have CallerID if you want it where the=20
analog cards have issues with CallerID..
Later..
------=_NextPart_000_0005_01C41EF2.B6A5E1E0
Content-Type: text/html;
charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2800.1400" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>What I want to do is have the asterisk =
server sat=20
in my house and used by my family to access the BT landline and to =
recieve calls=20
made to that landline. If it is not possible to do the auto =
attendant=20
thing then so be it, I will just have all phones in my house ring when a =
call is=20
made on the BT line. That should be easy, right ?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>In addition to running the server just =
for my=20
house, I want to have other memebers of my =
extended family link up to=20
the server via their broadband connections so we can make free calls to =
each=20
other over the internet connections.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>What I don't want is for other members =
of my family=20
(who are not resident in my house) to be able to make calls on my BT =
landline,=20
but I do want them to be able to make unlimited calls to other =
extentions=20
on the asterisk server.</FONT></DIV><FONT face=3DArial size=3D2>
<DIV><BR>Since I already pay monthly for broadband, I am not very keen =
to start=20
paying more for an IDSN line which will only be used for this =
project. I=20
don't use / need caller ID on external calls, so thats not an =
issue.</DIV>
<DIV> </DIV>
<DIV>Does that all make sence ?</DIV>
<DIV> </DIV>
<DIV>Thanks, Paul.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>-----Original Message-----<BR>From: <A=20
href=3D"mailto:asterisk-users-admin at lists.digium.com">asterisk-users-admi=
n at lists.digium.com</A>=20
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of =
WipeOut<BR>Posted=20
At: 10 April 2004 11:16<BR>Posted To: Asterisk-Users<BR>Conversation:=20
[Asterisk-Users] Re: Analogue telephone cards for the UK<BR>Subject: Re: =
[Asterisk-Users] Re: Analogue telephone cards for the UK</FONT></DIV>
<DIV> </DIV><FONT face=3DArial size=3D2>
<DIV><BR>It sounds like you are trying to share the PBX between multiple =
people..</DIV>
<DIV> </DIV>
<DIV>I would suggest getting an ISDN BRI line and an AVM Fritz card =
(using=20
<BR>the chan_capi driver).. This will give you two lines onto which you =
can=20
<BR>get 8 MSN's (an MSN is another number coming in on the same BRI).. =
You=20
<BR>can setup Asterisk to route the calls to the correct phones or group =
of=20
<BR>phones based on the number that was called..</DIV>
<DIV> </DIV>
<DIV>If you are in the UK there are plenty of Fritz cards around and =
this=20
<BR>method will also allow you to have CallerID if you want it where the =
<BR>analog cards have issues with CallerID..</DIV>
<DIV> </DIV>
<DIV>Later..<BR></FONT></DIV></BODY></HTML>
------=_NextPart_000_0005_01C41EF2.B6A5E1E0--
--__--__--
Message: 4
Date: Sat, 10 Apr 2004 12:36:40 +0100 (BST)
From: Chris Orme <chris at talisa.net>
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] No ringing tone with IAXY (and other bits and bobs)
Reply-To: asterisk-users at lists.digium.com
Hi!
I'm really hope you can help me solve a little mystery, the mystery is
probably just my misunderstanding ! sorry...
I've got an iaxy talking to my * box which connects to two providers.
I'm running the stable release of the pbx.
The only thing is that when dialling from the iaxy the ringing tone isn't
heard while calling someone - you just hear silence then, they either
answer or they don't on the remote end.
>From my extensions.conf is the following - I tried putting the ,r in and
it doesn't help. Is there some other option I could try here ?
Also I'm getting quite a bit of echo noticed at the remote end as well as
the iaxy end. All lines are digital, I guess only the jitter buffer is
there to be tweaked to try and help ?
There is also this echo problem with the sipura, but not with an ATA186 or
snom. The lack of a ringing tone is only with the iaxy.
The Answer,Hangup lines were to solve 'busy' situations with SIP phones,
without this or even with 'Congestion' they just rang forever if a number
was busy. They seem to need the 'Answer' line.
If you know a nicer or more correct way for me to do this please let me
know as most times the SIP phone user will hear half a ring and then the
hangup noise generated by the SIP device when a number they call is busy.
Many thanks!!
Chris
PS please Cc: me a copy as well as to the list in case I miss it -
Thanks.
<< extensions.conf >>
exten => _00.,1,AbsoluteTimeout(3600)
exten => _00.,2,Dial(${PROVIDER1}/${EXTEN:2},30,r)
exten => _00.,3,Answer
exten => _00.,4,Hangup
exten => _00.,103,Dial(${PROVIDER2}/011${EXTEN:2},30,r)
exten => _00.,104,Answer
exten => _00.,105,Hangup
<<iax.conf>>
[iaxy]
type=friend
accountcode=iaxy
disallow=all
;;allow=adpcm
allow=ulaw
username=iaxy
secret=xxx
auth=md5
nat=yes <- nat=1 ??
notransfer=yes <-this doesn't seem to work, perhaps in the wrong order?
host=dynamic
qualify=10000
Is the definitive order these should be in listed anywhere as I know it
really seems critical and lines can be ignored if they're not in spot on
the right order?
--__--__--
Message: 5
Date: Sat, 10 Apr 2004 14:44:19 +0200
From: "Olle E. Johansson" <oej at edvina.net>
Organization: Edvina AB
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Nothing to do? Go bounty-hunting!
Reply-To: asterisk-users at lists.digium.com
Being bored to death by these long weekends with nothing to do?
**** Why not go bounty-hunting? ****
There are some feature requests in the bug tracker with monetary bounties attached.
* Windows manager
* FreeBSD Zaptel drivers
http://bugs.digium.com/bug_view_page.php?bug_id=0000847
* IAX incoming/outgoing limit
* 2B channel transfer on PRI
* MGCP media gateway support
All of these have money attached, that you may earn over the weekend!
Get rich, start coding :-)
...or add to the bounty to make sure that others start coding!
I added URL to the FreeBSD Zaptel bounty, since it's closed as a bug report even
though the bounty is open for takers!
We're moving the bounty list to
http://www.voip-info.org/wiki-Asterisk+Bounty
since they're not really bugs or patches. As soon as we have patches, open a
[patch] report in the bug tracker and add them there, but not before we have
patches.
For those of you that started bounties, please help us move the request description
and the bounty value to the Wiki. For each bounty, I think we need one maintainer that
is in charge of handling the bounty - alone or with a group of advisors that can judge
the contributed code. The maintainer also needs to keep track of each contributor to
the bounty to help collecting the fee when it's time to pay.
I know the Wiki isn't the best tool for this, but it's an option we have today.
We are working on finding a long-term platform for feature requests with or without
bounties.
I would like to see for each bounty a clear request list
* Description of the function
* The combined value of the bounty
* Licensing for the code - I would suggest that all bounties should be disclaimed so
they may be candidates for the CVS
Happy easter from one of your friendly bug-marshals!
/O
--__--__--
Message: 6
From: "Brian Cuthie" <brian at systemix.com>
To: <asterisk-users at lists.digium.com>
Subject: RE: [Asterisk-Users] No ringing tone with IAXY (and other bits and bobs)
Date: Sat, 10 Apr 2004 08:50:52 -0400
Organization: Systemix Software
Reply-To: asterisk-users at lists.digium.com
What version of the Asterisk code are you running? 1_0 stable is definitely
broken wrt ringback, and the latest stuff seems really broken in all kinds
of ways. After seeing that others were having similar problems, and that
someone had solved many of them by rolling back to the CVS version from 3/5,
I tried the same and things are working marvelously (well, mostly).
-brian
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Chris Orme
> Sent: Saturday, April 10, 2004 6:37 AM
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] No ringing tone with IAXY (and
> other bits and bobs)
>
> Hi!
>
> I'm really hope you can help me solve a little mystery, the
> mystery is probably just my misunderstanding ! sorry...
>
> I've got an iaxy talking to my * box which connects to two providers.
> I'm running the stable release of the pbx.
>
> The only thing is that when dialling from the iaxy the
> ringing tone isn't heard while calling someone - you just
> hear silence then, they either answer or they don't on the remote end.
>
> >From my extensions.conf is the following - I tried putting the ,r in
> >and
> it doesn't help. Is there some other option I could try here ?
>
> Also I'm getting quite a bit of echo noticed at the remote
> end as well as the iaxy end. All lines are digital, I guess
> only the jitter buffer is there to be tweaked to try and help ?
>
> There is also this echo problem with the sipura, but not with
> an ATA186 or snom. The lack of a ringing tone is only with the iaxy.
>
> The Answer,Hangup lines were to solve 'busy' situations with
> SIP phones, without this or even with 'Congestion' they just
> rang forever if a number was busy. They seem to need the
> 'Answer' line.
>
> If you know a nicer or more correct way for me to do this
> please let me know as most times the SIP phone user will hear
> half a ring and then the hangup noise generated by the SIP
> device when a number they call is busy.
>
> Many thanks!!
>
> Chris
>
> PS please Cc: me a copy as well as to the list in case I miss
> it - Thanks.
> << extensions.conf >>
>
> exten => _00.,1,AbsoluteTimeout(3600)
> exten => _00.,2,Dial(${PROVIDER1}/${EXTEN:2},30,r)
> exten => _00.,3,Answer
> exten => _00.,4,Hangup
> exten => _00.,103,Dial(${PROVIDER2}/011${EXTEN:2},30,r)
> exten => _00.,104,Answer
> exten => _00.,105,Hangup
>
> <<iax.conf>>
>
> [iaxy]
> type=friend
> accountcode=iaxy
> disallow=all
> ;;allow=adpcm
> allow=ulaw
> username=iaxy
> secret=xxx
> auth=md5
> nat=yes <- nat=1 ??
> notransfer=yes <-this doesn't seem to work, perhaps in the
> wrong order?
> host=dynamic
> qualify=10000
>
> Is the definitive order these should be in listed anywhere as
> I know it really seems critical and lines can be ignored if
> they're not in spot on the right order?
>
> _______________________________________________
> 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
>
--__--__--
Message: 7
Date: Sat, 10 Apr 2004 08:13:08 -0600
From: Rich Adamson <radamson at routers.com>
Subject: RE: [Asterisk-Users] No ringing tone with IAXY (and other bits and bobs)
To: asterisk-users at lists.digium.com
Reply-To: asterisk-users at lists.digium.com
Brian,
I need to roll back to an earlier version to identify a different problem,
but I dont have the cvs checkout command string that includes a date. Can
you post how to do that please?
Rich
------------------------
> What version of the Asterisk code are you running? 1_0 stable is definitely
> broken wrt ringback, and the latest stuff seems really broken in all kinds
> of ways. After seeing that others were having similar problems, and that
> someone had solved many of them by rolling back to the CVS version from 3/5,
> I tried the same and things are working marvelously (well, mostly).
>
> -brian
>
> > -----Original Message-----
> > From: asterisk-users-admin at lists.digium.com
> > [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Chris Orme
> > Sent: Saturday, April 10, 2004 6:37 AM
> > To: asterisk-users at lists.digium.com
> > Subject: [Asterisk-Users] No ringing tone with IAXY (and
> > other bits and bobs)
> >
> > Hi!
> >
> > I'm really hope you can help me solve a little mystery, the
> > mystery is probably just my misunderstanding ! sorry...
> >
> > I've got an iaxy talking to my * box which connects to two providers.
> > I'm running the stable release of the pbx.
> >
> > The only thing is that when dialling from the iaxy the
> > ringing tone isn't heard while calling someone - you just
> > hear silence then, they either answer or they don't on the remote end.
> >
> > >From my extensions.conf is the following - I tried putting the ,r in
> > >and
> > it doesn't help. Is there some other option I could try here ?
> >
> > Also I'm getting quite a bit of echo noticed at the remote
> > end as well as the iaxy end. All lines are digital, I guess
> > only the jitter buffer is there to be tweaked to try and help ?
> >
> > There is also this echo problem with the sipura, but not with
> > an ATA186 or snom. The lack of a ringing tone is only with the iaxy.
> >
> > The Answer,Hangup lines were to solve 'busy' situations with
> > SIP phones, without this or even with 'Congestion' they just
> > rang forever if a number was busy. They seem to need the
> > 'Answer' line.
> >
> > If you know a nicer or more correct way for me to do this
> > please let me know as most times the SIP phone user will hear
> > half a ring and then the hangup noise generated by the SIP
> > device when a number they call is busy.
> >
> > Many thanks!!
> >
> > Chris
> >
> > PS please Cc: me a copy as well as to the list in case I miss
> > it - Thanks.
> > << extensions.conf >>
> >
> > exten => _00.,1,AbsoluteTimeout(3600)
> > exten => _00.,2,Dial(${PROVIDER1}/${EXTEN:2},30,r)
> > exten => _00.,3,Answer
> > exten => _00.,4,Hangup
> > exten => _00.,103,Dial(${PROVIDER2}/011${EXTEN:2},30,r)
> > exten => _00.,104,Answer
> > exten => _00.,105,Hangup
> >
> > <<iax.conf>>
> >
> > [iaxy]
> > type=friend
> > accountcode=iaxy
> > disallow=all
> > ;;allow=adpcm
> > allow=ulaw
> > username=iaxy
> > secret=xxx
> > auth=md5
> > nat=yes <- nat=1 ??
> > notransfer=yes <-this doesn't seem to work, perhaps in the
> > wrong order?
> > host=dynamic
> > qualify=10000
> >
> > Is the definitive order these should be in listed anywhere as
> > I know it really seems critical and lines can be ignored if
> > they're not in spot on the right order?
> >
> > _______________________________________________
> > 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
> >
>
> _______________________________________________
> 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
---------------End of Original Message-----------------
--__--__--
Message: 8
From: "Kevin " <Asterisk at gtcus.com>
To: <asterisk-users at lists.digium.com>
Date: Sat, 10 Apr 2004 09:36:29 -0400
Subject: [Asterisk-Users] Extensions and Include
Reply-To: asterisk-users at lists.digium.com
This perhaps is a newbie question or I have been up too late working on
this. Shouldn't I be able to dial internal extensions via the
"inboundanalog1" menu? When I dial an extension from an external call
to the inboundanalog1 menu, I get a busy and a hangup?
Any suggestions?
[extensions]
exten => 0,1,Dial,${P6601}
exten => 0,2,Hangup
exten => 6601,1,Dial(${P6601},20,t)
exten => 6601,2,Voicemail(u6601)
exten => 6601,3,Hangup
[inboundanalog1]
include => extensions
exten => s,1,AGI,calleridnamelookup.agi
exten => s,2,SetMusicOnHold,default
exten => s,3,Dial(${P6601},18,r)
exten => s,4,Answer
exten => s,5,Wait(1)
exten =>
s,6,Background(/var/spool/asterisk/voicemail/default/6601/unavail)
exten => s,7,ResponseTimeout(10)
exten => s,8,Voicemail2(6601)
exten => s,9,Hangup
--__--__--
Message: 9
From: "Brian Cuthie" <brian at systemix.com>
To: <asterisk-users at lists.digium.com>
Subject: RE: [Asterisk-Users] No ringing tone with IAXY (and other bits and bobs)
Date: Sat, 10 Apr 2004 09:39:24 -0400
Organization: Systemix Software
Reply-To: asterisk-users at lists.digium.com
Sure. I used this to get the 3/5 version:
cvs co -D 20040305 zaptel asterisk
-brian
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of
> Rich Adamson
> Sent: Saturday, April 10, 2004 9:13 AM
> To: asterisk-users at lists.digium.com
> Subject: RE: [Asterisk-Users] No ringing tone with IAXY (and
> other bits and bobs)
>
> Brian,
>
> I need to roll back to an earlier version to identify a
> different problem, but I dont have the cvs checkout command
> string that includes a date. Can you post how to do that please?
>
> Rich
>
> ------------------------
> > What version of the Asterisk code are you running? 1_0 stable is
> > definitely broken wrt ringback, and the latest stuff seems really
> > broken in all kinds of ways. After seeing that others were having
> > similar problems, and that someone had solved many of them
> by rolling
> > back to the CVS version from 3/5, I tried the same and
> things are working marvelously (well, mostly).
> >
> > -brian
> >
> > > -----Original Message-----
> > > From: asterisk-users-admin at lists.digium.com
> > > [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Chris
> > > Orme
> > > Sent: Saturday, April 10, 2004 6:37 AM
> > > To: asterisk-users at lists.digium.com
> > > Subject: [Asterisk-Users] No ringing tone with IAXY (and
> other bits
> > > and bobs)
> > >
> > > Hi!
> > >
> > > I'm really hope you can help me solve a little mystery,
> the mystery
> > > is probably just my misunderstanding ! sorry...
> > >
> > > I've got an iaxy talking to my * box which connects to
> two providers.
> > > I'm running the stable release of the pbx.
> > >
> > > The only thing is that when dialling from the iaxy the
> ringing tone
> > > isn't heard while calling someone - you just hear silence
> then, they
> > > either answer or they don't on the remote end.
> > >
> > > >From my extensions.conf is the following - I tried
> putting the ,r
> > > >in and
> > > it doesn't help. Is there some other option I could try here ?
> > >
> > > Also I'm getting quite a bit of echo noticed at the remote end as
> > > well as the iaxy end. All lines are digital, I guess only the
> > > jitter buffer is there to be tweaked to try and help ?
> > >
> > > There is also this echo problem with the sipura, but not with an
> > > ATA186 or snom. The lack of a ringing tone is only with the iaxy.
> > >
> > > The Answer,Hangup lines were to solve 'busy' situations with SIP
> > > phones, without this or even with 'Congestion' they just rang
> > > forever if a number was busy. They seem to need the
> 'Answer' line.
> > >
> > > If you know a nicer or more correct way for me to do this
> please let
> > > me know as most times the SIP phone user will hear half a
> ring and
> > > then the hangup noise generated by the SIP device when a
> number they
> > > call is busy.
> > >
> > > Many thanks!!
> > >
> > > Chris
> > >
> > > PS please Cc: me a copy as well as to the list in case I
> miss it -
> > > Thanks.
> > > << extensions.conf >>
> > >
> > > exten => _00.,1,AbsoluteTimeout(3600) exten =>
> > > _00.,2,Dial(${PROVIDER1}/${EXTEN:2},30,r)
> > > exten => _00.,3,Answer
> > > exten => _00.,4,Hangup
> > > exten => _00.,103,Dial(${PROVIDER2}/011${EXTEN:2},30,r)
> > > exten => _00.,104,Answer
> > > exten => _00.,105,Hangup
> > >
> > > <<iax.conf>>
> > >
> > > [iaxy]
> > > type=friend
> > > accountcode=iaxy
> > > disallow=all
> > > ;;allow=adpcm
> > > allow=ulaw
> > > username=iaxy
> > > secret=xxx
> > > auth=md5
> > > nat=yes <- nat=1 ??
> > > notransfer=yes <-this doesn't seem to work, perhaps in the wrong
> > > order?
> > > host=dynamic
> > > qualify=10000
> > >
> > > Is the definitive order these should be in listed
> anywhere as I know
> > > it really seems critical and lines can be ignored if
> they're not in
> > > spot on the right order?
> > >
> > > _______________________________________________
> > > 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
> > >
> >
> > _______________________________________________
> > 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
>
> ---------------End of Original Message-----------------
>
>
> _______________________________________________
> 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
>
--__--__--
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
End of Asterisk-Users Digest
More information about the asterisk-users
mailing list