[Asterisk-Users] sip phone problem

Tue May 25 04:55:33 MST 2004

Hi,
First you need to upgrade to the latest CVS and then insert a second /
third priority line with hangup in the dialplan.

Regds
Vivian Alan

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
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Sent: Tuesday, May 25, 2004 7:48 AM
To: asterisk-users at lists.digium.com
Subject: Asterisk-Users digest, Vol 1 #3893 - 15 msgs

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Today's Topics:

   1. Re: ZAPTEL not loading on FC2 (Fran Boon)
   2. Re: Meetme Options (new one) (Fran Boon)
   3. Sip/IAX Clients for Linux (reacend at gmx.net)
   4. Re: 11 instead of Star (Peter Corlett)
   5. sip phone problem (=?iso-8859-1?q?Antonio=20Diego?=)
   6. Re: Sip/IAX Clients for Linux (andrewg at felinemenace.org)
   7. RE: Sip/IAX Clients for Linux (Karl Dyson)
   8. Troubles with Kphone (enano)
   9. RE: 11 instead of Star (Paul Crick)
  10. RE: 11 instead of Star (Paul Crick)
  11. TerraCall Setting (cary at cary.net)
  12. Sound card problem (cary at cary.net)
  13. Answer App hanging in I4L (S.Murali Krishna)
  14. [Fwd: Answer App hanging in I4L] (Murali Krishnan)
  15. Re: Troubles with Kphone] (Murali Krishnan)

--__--__--

Message: 1
Date: Tue, 25 May 2004 09:52:48 +0100
From: Fran Boon <flavour at partyvibe.com>
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] ZAPTEL not loading on FC2
Reply-To: asterisk-users at lists.digium.com

Jorge Verastegui wrote:
> I have successfully compiled the last cvs zaptel drives in FC2 box and
> then load wcfxs module, but Kernel Freezes with zttool 

http://bugs.digium.com/bug_view_page.php?bug_id=0001704

F

--__--__--

Message: 2
Date: Tue, 25 May 2004 09:56:47 +0100
From: Fran Boon <flavour at partyvibe.com>
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Meetme Options (new one)
Reply-To: asterisk-users at lists.digium.com

Ben Merrills wrote:
> Seems like it would be a simple modification?
> Where would I post a feature request like this? :-)

bugs.digium.com
Ensure summary starts with [request]

F

--__--__--

Message: 3
Date: Tue, 25 May 2004 11:07:12 +0200
To: asterisk-users at lists.digium.com
From: reacend at gmx.net
Subject: [Asterisk-Users] Sip/IAX Clients for Linux
Reply-To: asterisk-users at lists.digium.com

Hi There,
i think all VOIP clients for Linux are unusable!

i got testet:

Linphone + Linphonec all in version 12.2
Kphone
gophone
and other...

the only programm that is usable is gnomemeeting...

does anybody knew some other tools?

Best Regards,
Mark

--__--__--

Message: 4
Date: Tue, 25 May 2004 10:09:19 +0100
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] 11 instead of Star
From: Peter Corlett <abuse at chapati.cabal.org.uk>
Reply-To: asterisk-users at lists.digium.com

On Mon, May 24, 2004 at 07:58:26PM -0700, Paul Crick wrote:
[...]
> *sighs* Yeah, that won't work.. which is a shame.. this goes back to
the
> whole debate of "should CLASS service codes be implemented in the dial
> plan or the channel driver?"

The most compelling reason to me to have them in the dial plan is that
CLASS
codes aren't universal. AFAICS, they're mainly limited to the USA and
telcos
that source cheap switches from the USA and don't bother to customise
them.

For my Asterisk setup, I'd rather my phone uses the same star codes as
BT
and GSM than some foreign standard that nobody here knows.

> From memory and reading the mailing list for a while now, I think
Mark's
> dead against having these features in the dial plan, but I can't
remember
> why.

Let me guess, is he American? ;)

[...]
> I think the way it was going to go was a flag which would allow you to
> disable all channel driver features like this and rely on the dial
plan to
> implement the features.

This is very much my preferred solution. If there is still some bizarre
obligation to support alien phone standards in the channel drivers, we
should have the option of disabling this undesired behaviour.

--__--__--

Message: 5
Date: Tue, 25 May 2004 04:21:53 -0500 (CDT)
From: =?iso-8859-1?q?Antonio=20Diego?= <almodovarcebrian at yahoo.com>
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] sip phone problem
Reply-To: asterisk-users at lists.digium.com

Hi all.
I have 2 ip phones (Grandstream Budgetone):
   -budgetone1
   -budgetone2

All two are connected to an Asterisk server.
When I make a call from budgetone1 to budgetone2, I
can speak with budgetone2 whith no problem. But when
budgetone2 hangs up, budgetone1 does not play any tone
(like busy tone). Budgetone1 seems to be still in
conversation, but what conversation!

Has anyone had a problem connecting budgetones to
Asterisk?

Please help me.

Thanks in advance.

_________________________________________________________
Do You Yahoo!?
Información de Estados Unidos y América Latina, en Yahoo! Noticias.
Visítanos en http://noticias.espanol.yahoo.com

--__--__--

Message: 6
Date: Tue, 25 May 2004 02:24:13 -0700
From: andrewg at felinemenace.org
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Sip/IAX Clients for Linux
Reply-To: asterisk-users at lists.digium.com

iaxclient.sourceforge.net perhaps?

On Tue, May 25, 2004 at 11:07:12AM +0200, reacend at gmx.net wrote:
> 
> Hi There,
> i think all VOIP clients for Linux are unusable!
> 
> 
> i got testet:
> 
> Linphone + Linphonec all in version 12.2
> Kphone
> gophone
> and other...
> 
> 
> the only programm that is usable is gnomemeeting...
> 
> does anybody knew some other tools?
> 
> 
> Best Regards,
> Mark
> _______________________________________________
> 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
Subject: RE: [Asterisk-Users] Sip/IAX Clients for Linux
Date: Tue, 25 May 2004 10:31:35 +0100
From: "Karl Dyson" <kd at junesta.com>
To: <asterisk-users at lists.digium.com>
Reply-To: asterisk-users at lists.digium.com

I=20hear=20sjphone=20from=20sjlabs.com=20is=20usable.......

-----Original=20Message-----
From:=20asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]=20On=20Behalf=20Of
reacend at gmx.net
Sent:=2025=20May=202004=2010:07
To:=20asterisk-users at lists.digium.com
Subject:=20[Asterisk-Users]=20Sip/IAX=20Clients=20for=20Linux

Hi=20There,
i=20think=20all=20VOIP=20clients=20for=20Linux=20are=20unusable!

i=20got=20testet:

Linphone=20+=20Linphonec=20all=20in=20version=2012.2=20Kphone=20gophone=
20=
and=20other...

the=20only=20programm=20that=20is=20usable=20is=20gnomemeeting...

does=20anybody=20knew=20some=20other=20tools?

Best=20Regards,
Mark
_______________________________________________
Asterisk-Users=20mailing=20list
Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To=20UNSUBSCRIBE=20or=20update=20options=20visit:
=20=20=20http://lists.digium.com/mailman/listinfo/asterisk-users

________________________________________________________________________
This=20e-mail=20has=20been=20scanned=20for=20all=20viruses=20by=20Star=2
0I=
nternet.=20The
service=20is=20powered=20by=20MessageLabs.=20For=20more=20information=20
on=
=20a=20proactive
anti-virus=20service=20working=20around=20the=20clock,=20around=20the=20
gl=
obe,=20visit:
http://www.star.net.uk
________________________________________________________________________

________________________________________________________________________
This=20e-mail=20has=20been=20scanned=20for=20all=20viruses=20by=20Star=2
0I=
nternet.=20The
service=20is=20powered=20by=20MessageLabs.=20For=20more=20information=20
on=
=20a=20proactive
anti-virus=20service=20working=20around=20the=20clock,=20around=20the=20
gl=
obe,=20visit:
http://www.star.net.uk
________________________________________________________________________

--__--__--

Message: 8
From: enano <klky3 at fibertel.com.ar>
To: asterisk-users at lists.digium.com
Date: Tue, 25 May 2004 06:52:46 -0300
Subject: [Asterisk-Users] Troubles with Kphone
Reply-To: asterisk-users at lists.digium.com

Hi , 

I'm triying to use kphone 4.02, but when i'm make a call the programs 
doesn't respond any command, so i can't hear any sound .. 

in sip.conf that's my codec config:

disallow=all                    
allow=gsm
allow=ulaw                      
allow=ilbc

and the kphone give the follow : 
SipClient: Sending: 06:46:28.116
--------------------------------
ACK sip:500 at 192.168.0.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2;rport
CSeq: 6121 ACK
To: <sip:500 at 192.168.0.3>;tag=as12aab0bf
From: "ivan2" <sip:ivan2 at 192.168.0.3>;tag=7F6911ED
Call-ID: 155660827 at 192.168.0.2
Content-Length: 0
User-Agent: kphone/4.0.2
Contact: "ivan2" <sip:ivan2 at 192.168.0.2;transport=udp>

res_search: NO result !
res_search: NO result !
SipClient: Sending to '192.168.0.3:5060'
SipCallMember: localStatusUpdated: 200
CallAudio: Using GSM for output
CallAudio: Sending to remote site 192.168.0.3:19696
UDPMessageSocket::SetTOS: Operation not permitted
CallAudio: OSS device already open (readwrite)

anyone can help me ??

thanks 

Ivan 

--__--__--

Message: 9
From: "Paul Crick" <web-asterisk-users at ivrl.com>
To: <asterisk-users at lists.digium.com>
Subject: RE: [Asterisk-Users] 11 instead of Star
Date: Tue, 25 May 2004 03:02:59 -0700
Reply-To: asterisk-users at lists.digium.com

> The most compelling reason to me to have them in the dial
> plan is that CLASS codes aren't universal.
Yup, I'm with you there.. they're mostly universal within US telcos but
even
then there are some odd ones.. I use *98 for voicemail from home, but
other
carriers don't. UK = 1571 for BT Message Minder?

> For my Asterisk setup, I'd rather my phone uses the same
> star codes as BT and GSM than some foreign standard that
> nobody here knows.
Exactly.. and maybe supplement as necessary - I like the US option of
*70
for single call deactivation of call waiting, but it'd be nice to have
the
toggle on and off-able ability of european switches with *43# and #43#..
and
of course, those nice diversion codes that are standard for landlines
and
mobiles - the whole of Europe can't be wrong, surely?

> Let me guess, is he American? ;)
Meow - let's not go there. Seriously - I think Mark's a cool bloke and
I've
never even dealt with him, but look at what we have today - sure, there
are
other contributors and stuff, but a lot of it comes down to him, the
start
he made, the continuing work and coordination that he does..

Convincing Americans that some of the European stuff actually isn't bad
takes a while.. Sure, CDMA is better than GSM given the
circumstances/situation etc blah blah yadda but don't you see the
benefits
of the way GSM goes together nicely? roaming? access codes? Whoah there
Paul, it's late, let's not start a holy war..

> This is very much my preferred solution. If there is still
> some bizarre obligation to support alien phone standards
> in the channel drivers, we should have the option of
> disabling this undesired behaviour.
Maybe someone a bit higher up the development chain than me can comment
on
the current state of play with this? The bug tracker has this bug:
http://bugs.digium.com/bug_view_page.php?bug_id=0001071 but there's been
no
recent activity.

Cheers
Paul

--__--__--

Message: 10
From: "Paul Crick" <web-asterisk-users at ivrl.com>
To: <asterisk-users at lists.digium.com>
Subject: RE: [Asterisk-Users] 11 instead of Star
Date: Tue, 25 May 2004 03:05:42 -0700
Reply-To: asterisk-users at lists.digium.com

It's late, I'm tired, and missed this one too:
http://bugs.digium.com/bug_view_page.php?bug_id=0000071

--__--__--

Message: 11
Date: Tue, 25 May 2004 18:08:46 +0800
From: cary at cary.net
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] TerraCall Setting
Reply-To: asterisk-users at lists.digium.com

Dear All,

My Asterisk PBX sip.conf for TerraCall setting as the follow:

register => xxxxxxxxxx:xxxxxx at xchange.terracall.com/xxxxxxxxxx

[terracallcom]
type=friend
username=xxxxxxxxxx (10 numbers)
secret=xxxxxx
fromuser=xxxxxxxxxx
fromdomain=realm.terracall.com
host=xchange.terracall.com
nat=yes
canreinvite=no
context=private

I tried to dial out, the reply is follow:

May 25 17:59:55 NOTICE[1142135600]: Failed to authenticate on INVITE to
'"xxxxxxxxxx" <sip:xxxxxxxxxx at realm.terracall.com>;tag=as6f19e84d'

May 25 18:00:00 WARNING[1142135600]: Maximum retries exceeded on call
5365be3a5a10efa50fc825c83d54b36f at xxx.xxx.xxx.xxx for seqno 103 (Critical
Request)
May 25 18:00:00 WARNING[1142135600]: Maximum retries exceeded on call
5365be3a5a10efa50fc825c83d54b36f at xxx.xxx.xxx.xxx for seqno 104 (Critical
Request)

I tried to change the host to ip address, but same reply.

What's wrong?

Anyone can help me?

Thank You.

Cary LEUNG

--__--__--

Message: 12
Date: Tue, 25 May 2004 18:15:31 +0800
From: cary at cary.net
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Sound card problem
Reply-To: asterisk-users at lists.digium.com

Dear All,

In my pbx log show that follow for the sound device:

May 25 10:56:21 WARNING[1074431744]: Requested 8000 Hz, got 48000 Hz --
sound
may be choppy
May 25 10:56:21 WARNING[1184099120]: Read error on sound device:
Resource
temporarily unavailable

Anyone know how can solve it?

Thank You.

Cary LEUNG

--__--__--

Message: 13
Date: Tue, 25 May 2004 16:00:34 +0530
From: "S.Murali Krishna" <ismk at myrealbox.com>
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Answer App hanging in I4L
Reply-To: asterisk-users at lists.digium.com

Hi,

Anyone using ISDN4Linux (Eicon Diva Hisax ) card.

If yes, please help me out.

After configuring extension.conf and modem.conf
I could make outward calls correctly from gnophone  and kphone. Still 
the inward
call to the configured MSN is correctly reaching Asterisk and also to
the
configured Context. But the issue is, it was hanging on 'Answer' 
application and
throwing out 'Unable to Spawn Extension (vpk, s, 1) ..... ".

When I debug, found that Asterisk is issuing the following AT commands
while
answering the call.

ATA
(expecting VCON )
AT+VRX+VTX
(expecting CONNECT )

In the above sequence, I found that after giving ATA, without waiting
for
VCON it is giving the AT+VRX+VTX command and getting CONNECT
successfully, but according to voice communication CONNECT without
VCON would fail and hanging up the line.

Though all the above things are my Points  on debugging, my basic issue
is to successfully ANSWER an incoming call.

Please throw some lights.

Regards
Murali Krishnan.S
<ismk at myrealbox.com>

--__--__--

Message: 14
Date: Tue, 25 May 2004 16:13:35 +0530
From: Murali Krishnan <ismk at myrealbox.com>
Organization: bk SYSTEMS (P) LTD.,
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] [Fwd: Answer App hanging in I4L]
Reply-To: asterisk-users at lists.digium.com

-------- Original Message --------
Subject: Answer App hanging in I4L
Date: Tue, 25 May 2004 13:49:50 +0530
From: Murali Krishnan <murali at bksys.co.in>
Reply-To: ismk at myrealbox.com
Organization: bk SYSTEMS (P) LTD.,
To: asterisk-users at lists.digium.com

Hi,

Anyone using ISDN4Linux (Eicon Diva Hisax ) card.

If yes, please help me out.

After configuring extension.conf and modem.conf
I could make outward calls correctly from gnophone  and kphone. Still
the inward
call to the configured MSN is correctly reaching Asterisk and also to
the
configured Context. But the issue is, it was hanging on 'Answer'
application and
throwing out 'Unable to Spawn Extension (vpk, s, 1) ..... ".

When I debug, found that Asterisk is issuing the following AT commands
while
answering the call.

ATA
(expecting VCON )
AT+VRX+VTX
(expecting CONNECT )

In the above sequence, I found that after giving ATA, without waiting
for
VCON it is giving the AT+VRX+VTX command and getting CONNECT
successfully, but according to voice communication CONNECT without
VCON would fail and hanging up the line.

Though all the above things are my Points  on debugging, my basic issue
is to successfully ANSWER an incoming call.

Please throw some lights.

Regards
Murali Krishnan.S
<ismk at myrealbox.com>

--__--__--

Message: 15
Date: Tue, 25 May 2004 16:14:11 +0530
From: Murali Krishnan <ismk at myrealbox.com>
Organization: bk SYSTEMS (P) LTD.,
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Troubles with Kphone]
Reply-To: asterisk-users at lists.digium.com

-------- Original Message --------
Subject: Re: [Asterisk-Users] Troubles with Kphone
Date: Tue, 25 May 2004 15:44:15 +0530
From: Murali Krishnan <murali at bksys.co.in>
Reply-To: ismk at myrealbox.com
Organization: bk SYSTEMS (P) LTD.,
To: asterisk-users at lists.digium.com
References: <200405250652.46370.klky3 at fibertel.com.ar>

enano wrote:

>Hi , 
>
>
>
>I'm triying to use kphone 4.02, but when i'm make a call the programs 
>doesn't respond any command, so i can't hear any sound .. 
>
>
>in sip.conf that's my codec config:
>
>disallow=all                    
>allow=gsm
>allow=ulaw                      
>allow=ilbc
>
>and the kphone give the follow : 
>SipClient: Sending: 06:46:28.116
>--------------------------------
>ACK sip:500 at 192.168.0.3 SIP/2.0
>Via: SIP/2.0/UDP 192.168.0.2;rport
>CSeq: 6121 ACK
>To: <sip:500 at 192.168.0.3>;tag=as12aab0bf
>From: "ivan2" <sip:ivan2 at 192.168.0.3>;tag=7F6911ED
>Call-ID: 155660827 at 192.168.0.2
>Content-Length: 0
>User-Agent: kphone/4.0.2
>Contact: "ivan2" <sip:ivan2 at 192.168.0.2;transport=udp>
>
>
>res_search: NO result !
>res_search: NO result !
>SipClient: Sending to '192.168.0.3:5060'
>SipCallMember: localStatusUpdated: 200
>CallAudio: Using GSM for output
>CallAudio: Sending to remote site 192.168.0.3:19696
>UDPMessageSocket::SetTOS: Operation not permitted
>CallAudio: OSS device already open (readwrite)
>
>
>anyone can help me ??
>
>
>thanks 
>
>
>Ivan 
>
>
>
>
>_______________________________________________
>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
>  
>
Check the following things.

1. Make sure your sound card is configured properly for record/playback
    - if not, do it with either kmix and test with gnome-sound-recorder
2. Make sure your identity is configured in sip.conf and extension.conf
correctly
3. Make sure kphone is registered with Asterisk
   File->Identity  - see whether 'Unregister' is there, (means you are
registered )
4. Watch for Asterisk Messages for any clue. ( asterisk -vvvvvc )
5. Make sure the destination extension you are dialing from kphone has
proper dialplan sequence in extension.conf
6. If you have  OSS sound configuration, immediately switch to ALSA.
  - visit alsa-project.org and search docs for your card type. Compile
and
    install the packages. ( this OSS would be the major headache if you
are not
getting sound ).

If you are registered with Asterisk and your sound card is proper, and
you
configured your destination extension routing properly in extension.conf
everything should work fine.

Get back with success.

Regards
Murali Krishnan.

--__--__--

_______________________________________________
Asterisk-Users mailing list
Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users

End of Asterisk-Users Digest