[Asterisk-Users] help
Shiraz Khalid
shiraz at tstoneinc.com
Thu Jan 19 11:35:31 MST 2006
-----Original Message-----
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Sent: Thursday, January 19, 2006 12:00 PM
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Subject: Asterisk-Users Digest, Vol 18, Issue 121
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Today's Topics:
1. Re: DTMF # ? (Mojo with Horan & Company, LLC)
2. Re: Brief silences during calls (Mojo with Horan & Company, LLC)
3. Problem with rxfax - Dropping incompatible voice frame?
(Micha?l Gaudette)
4. transfer and zap (Marcel Pennewi?)
5. Sound issue with Asterisk (Kevin)
6. Re: Brief silences during calls (Rob Lith)
7. Re: MeetMe Listen Only flag (|m) (Tony Mountifield)
8. Re: SAN Devices (Jared Watkins)
9. Disabling zap echo cancellor from dialplan (Massimo De Nadal)
----------------------------------------------------------------------
Message: 1
Date: Thu, 19 Jan 2006 08:22:22 -0900
From: "Mojo with Horan & Company, LLC" <mojo at horanappraisals.com>
Subject: Re: [Asterisk-Users] DTMF # ?
To: chris.songer at getblaze.com, Asterisk Users Mailing List -
Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <43CFCACE.1090800 at horanappraisals.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
It's mapped to blind transfer in features.conf -- If you want to use the
blind transfer feature, which I find easier than my phones' transfer
features, remap it to ## in features.conf. That way if you hit # it
dtmfs through to the target IVR, but you can hit ## real quick to get
the transfer function.
Or maybe I could refer you to the notes in the wiki: ;)
http://www.voip-info.org/wiki-Asterisk+config+features.conf
Using the blindxfer in [featuremap] section you can redefine the
transfer key. For example, if the blindxfer is set to "##", transfer
only happens when you press the "#" key twice very quickly. This solves
a problem using Asterisk phones to call IVR systems such as those used
by banks and credit card companies - "Enter you account number followed
by the # key".
Moj
chris songer wrote:
> Can the # be used as a valid key press for a user in a dial plan?
> if so how can the asterisk recognize it as a valid key press?
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
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--
Mojo <mojo at horanappraisals.com>
Office Manger, Horan & Company, LLC
(907) 747-6666 x112
------------------------------
Message: 2
Date: Thu, 19 Jan 2006 08:31:11 -0900
From: "Mojo with Horan & Company, LLC" <mojo at horanappraisals.com>
Subject: Re: [Asterisk-Users] Brief silences during calls
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <43CFCCDF.5090205 at horanappraisals.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
check irq supply on the * server -- When you run zttest do you maintain
over 98%? or like Steve suggested, the network may have congestion or
other errors ethereal may help you figure out.
I had a polycom 500 that was doing this to my user, 301s and 501s
wouldn't do it. Not sure if that was network issues or something with
the polycom itself.
Moj
Mimmus wrote:
> Where can I investigate the origin of brief silences during calls
from/to my
> SIP phone?
> Only rare pauses of 0.5-1.0 sec when I'm not able to hear anything.
>
> Thanks
> Mimmus
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Mojo <mojo at horanappraisals.com>
Office Manger, Horan & Company, LLC
(907) 747-6666 x112
------------------------------
Message: 3
Date: Thu, 19 Jan 2006 12:40:26 -0500
From: Micha?l Gaudette <michael.gaudette at virtutel.ca>
Subject: [Asterisk-Users] Problem with rxfax - Dropping incompatible
voice frame?
To: asterisk-users at lists.digium.com
Message-ID: <003201c61d1f$6ea88f50$0a01a8c0 at mike>
Content-Type: text/plain; charset=iso-8859-1
Hi,
I'm having problems with the rxFax app. One of the messages that appear
in
my console is:
Executing Set("SIP/something",
"FAXFILE=/var/spool/asterisk-fax/1137692307.5.tif") in new stack
-- Executing RxFAX("SIP/something",
"/var/spool/asterisk-fax/1137692307.5.tif") in new stack
Jan 19 12:38:30 NOTICE[12008]: channel.c:1906 ast_read: Dropping
incompatible voice frame on SIP/something of format slin since our
native
format has changed to ulaw
"Dropping incompatible voice frame on SIP/something of format slin since
our
native format has changed to ulaw"
This seemed particularly important, but I can't really say why....Could
this
be why my faxes are often interrupting during transmission and giving me
errors on my PSTN fax machine that is used for sending the fax?
Mike
------------------------------
Message: 4
Date: Thu, 19 Jan 2006 18:40:19 +0100
From: Marcel Pennewi? <mape2k at gmail.com>
Subject: [Asterisk-Users] transfer and zap
To: asterisk-users at lists.digium.com
Message-ID: <1445813335.20060119184019 at gmail.com>
Content-Type: text/plain; charset=ISO-8859-15
Hello,
some problems with transfer and zap...
one hfc-card in NT mode and one fritz isdn-card in server.
there is one gigaset SX353 isdn phone on the hfc-card.
anybody calls from external via capi and the call is bridged
to the zap-device. if you want to transfer the call via R-button on
the isdn-phone the caller get the music-on-hold. you get a dialtone
and dial - if the called person gets on phone - i will hang up the
phone. but the call did'nt transfer - the moh to the first caller
will not stop. how can i transfer the call?
i want to transfer back the origin call to the asterisk-server in an
extension for faxtransfer, so if anyone call and a faxtone is there i
want transfer it back, so that asterisk answer the call.
any ideas?
sorry for my bad english ;-)
Marcel Pennewiss
------------------------------
Message: 5
Date: Thu, 19 Jan 2006 12:45:50 -0500
From: Kevin <kevin at locker70.com>
Subject: [Asterisk-Users] Sound issue with Asterisk
To: asterisk-users at lists.digium.com
Cc: steve at daviesfam.org
Message-ID: <43CFD04E.8080608 at locker70.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hey Steve and everyone,
I looked at the configuration, and unless I am missing something I don't
think they are configured
# ztcfg -vv
Zaptel Configuration
======================
Channel map:
0 channels configured.
In the zapata.conf file, it is the sample version, but I didn't notice
anything in there that related to what you said. Or is it in a
different file or location?
I am in the office now so I am able to provide some more information
about the issue that I am having.
Here is the kernel if this helps Fedora core 4 -- 2.6.11-1.1369_FC4smp
I know that ztdummy is at least loaded now. Also as stated before there
is nothing plugged into the T1 card. So I wasn't sure if that was
causing a problem or not which is why I enabled ztdummy but it was not
the first time I e-mailed you.
# lsmod | grep ztdummy
ztdummy 7748 0
zaptel 192516 6 ztdummy,wct4xxp
If I look at the connections from tcpdump, I see my phone call coming
in, but no traffic is being sent back to the phone. With an Echo() test,
I see the traffic going back and forth, but when I call into a menu,
then there is nothing.
Thanks,
Kevin
I ran a sip debug as well but I felt it was better at the end of the
e-mail:
<-- SIP read from 64.7.189.14:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.7.161.26:5060;branch=z9hG4bK2b87d1ab;rport
From: "asterisk" <sip:asterisk at 64.7.161.26>;tag=as4a36d77b
To: <sip:budgeTone-PubIP at 64.7.189.14>;tag=a0efbf44ecab5900
Call-ID: 286e70e60596cc1b34a1fcac4e4c5337 at 64.7.161.26
CSeq: 102 OPTIONS
User-Agent: Grandstream BT100 1.0.6.7
Contact: <sip:budgeTone-PubIP at 64.7.189.14>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Supported: replaces
Content-Length: 0
--- (11 headers 0 lines)---
Destroying call '286e70e60596cc1b34a1fcac4e4c5337 at 64.7.161.26'
<-- SIP read from 64.7.189.14:5060:
INVITE sip:500 at 64.7.161.26 SIP/2.0
Via: SIP/2.0/UDP 64.7.189.14;branch=z9hG4bKa4b619218b6ad43c
From: "Budge Tone"
<sip:budgeTone-PubIP at 64.7.161.26>;tag=b72941c93fe74588
To: <sip:500 at 64.7.161.26>
Contact: <sip:budgeTone-PubIP at 64.7.189.14>
Supported: replaces
Call-ID: 2143389df4877360 at 64.7.189.14
CSeq: 19606 INVITE
User-Agent: Grandstream BT100 1.0.6.7
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 337
v=0
o=budgeTone-PubIP 8000 8000 IN IP4 64.7.189.14
s=SIP Call
c=IN IP4 64.7.189.14
t=0 0
m=audio 5004 RTP/AVP 2 8 4 18 15 97 9
a=sendrecv
a=rtpmap:2 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:15 G728/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:9 G722/16000
a=ptime:20
--- (13 headers 16 lines)---
Using INVITE request as basis request - 2143389df4877360 at 64.7.189.14
Sending to 64.7.189.14 : 5060 (non-NAT)
Found peer 'budgeTone-PubIP'
Reliably Transmitting (no NAT) to 64.7.189.14:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
64.7.189.14;branch=z9hG4bKa4b619218b6ad43c;received=64.7.189.14
From: "Budge Tone"
<sip:budgeTone-PubIP at 64.7.161.26>;tag=b72941c93fe74588
To: <sip:500 at 64.7.161.26>;tag=as6f00184d
Call-ID: 2143389df4877360 at 64.7.189.14
CSeq: 19606 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:500 at 64.7.161.26>
Proxy-Authenticate: Digest realm="asterisk", nonce="351ca5f6"
Content-Length: 0
---
Scheduling destruction of call '2143389df4877360 at 64.7.189.14' in 15000
ms
<-- SIP read from 64.7.189.14:5060:
ACK sip:500 at 64.7.161.26 SIP/2.0
Via: SIP/2.0/UDP 64.7.189.14;branch=z9hG4bKa4b619218b6ad43c
From: "Budge Tone"
<sip:budgeTone-PubIP at 64.7.161.26>;tag=b72941c93fe74588
To: <sip:500 at 64.7.161.26>;tag=as6f00184d
Contact: <sip:budgeTone-PubIP at 64.7.189.14>
Call-ID: 2143389df4877360 at 64.7.189.14
CSeq: 19606 ACK
User-Agent: Grandstream BT100 1.0.6.7
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
--- (11 headers 0 lines)---
<-- SIP read from 64.7.189.14:5060:
INVITE sip:500 at 64.7.161.26 SIP/2.0
Via: SIP/2.0/UDP 64.7.189.14;branch=z9hG4bK1b2220ace977c3a7
From: "Budge Tone"
<sip:budgeTone-PubIP at 64.7.161.26>;tag=b72941c93fe74588
To: <sip:500 at 64.7.161.26>
Contact: <sip:budgeTone-PubIP at 64.7.189.14>
Supported: replaces
Proxy-Authorization: Digest username="budgeTone-PubIP",
realm="asterisk", algorithm=MD5, uri="sip:500 at 64.7.161.26",
nonce="351ca5f6", response="3748b6120c7f4ecc4873cbdaf178d507"
Call-ID: 2143389df4877360 at 64.7.189.14
CSeq: 19607 INVITE
User-Agent: Grandstream BT100 1.0.6.7
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 337
v=0
o=budgeTone-PubIP 8000 8001 IN IP4 64.7.189.14
s=SIP Call
c=IN IP4 64.7.189.14
t=0 0
m=audio 5004 RTP/AVP 2 8 4 18 15 97 9
a=sendrecv
a=rtpmap:2 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:15 G728/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:9 G722/16000
a=ptime:20
--- (14 headers 16 lines)---
Using INVITE request as basis request - 2143389df4877360 at 64.7.189.14
Sending to 64.7.189.14 : 5060 (non-NAT)
Found peer 'budgeTone-PubIP'
Found RTP audio format 2
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 15
Found RTP audio format 97
Found RTP audio format 9
Peer audio RTP is at port 64.7.189.14:5004
Found description format G726-32
Found description format PCMA
Found description format G723
Found description format G729
Found description format G728
Found description format iLBC
Found description format G722
Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x519
(g723|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x400 (ilbc)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0
(nothing), combined - 0x0 (nothing)
Looking for 500 in local (domain 64.7.161.26)
list_route: hop: <sip:budgeTone-PubIP at 64.7.189.14>
Transmitting (no NAT) to 64.7.189.14:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
64.7.189.14;branch=z9hG4bK1b2220ace977c3a7;received=64.7.189.14
From: "Budge Tone"
<sip:budgeTone-PubIP at 64.7.161.26>;tag=b72941c93fe74588
To: <sip:500 at 64.7.161.26>
Call-ID: 2143389df4877360 at 64.7.189.14
CSeq: 19607 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:500 at 64.7.161.26>
Content-Length: 0
---
-- Executing Playback("SIP/budgeTone-PubIP-7e44", "demo-abouttotry")
in new stack
We're at 64.7.161.26 port 13648
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x400 (ilbc) to SDP
Reliably Transmitting (no NAT) to 64.7.189.14:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
64.7.189.14;branch=z9hG4bK1b2220ace977c3a7;received=64.7.189.14
From: "Budge Tone"
<sip:budgeTone-PubIP at 64.7.161.26>;tag=b72941c93fe74588
To: <sip:500 at 64.7.161.26>;tag=as324cfd6f
Call-ID: 2143389df4877360 at 64.7.189.14
CSeq: 19607 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:500 at 64.7.161.26>
Content-Type: application/sdp
Content-Length: 182
v=0
o=root 3026 3026 IN IP4 64.7.161.26
s=session
c=IN IP4 64.7.161.26
t=0 0
m=audio 13648 RTP/AVP 0 97
a=rtpmap:0 PCMU/8000
a=rtpmap:97 iLBC/8000
a=silenceSupp:off - - - -
---
-- Playing 'demo-abouttotry' (language 'en')
<-- SIP read from 64.7.189.14:5060:
ACK sip:500 at 64.7.161.26 SIP/2.0
Via: SIP/2.0/UDP 64.7.189.14;branch=z9hG4bKb9dc124752ee52c9
From: "Budge Tone"
<sip:budgeTone-PubIP at 64.7.161.26>;tag=b72941c93fe74588
To: <sip:500 at 64.7.161.26>;tag=as324cfd6f
Contact: <sip:budgeTone-PubIP at 64.7.189.14>
Proxy-Authorization: Digest username="budgeTone-PubIP",
realm="asterisk", algorithm=MD5, uri="sip:500 at 64.7.161.26",
nonce="351ca5f6", response="eb1f3ad109804e62f1ba1edcc3706bf9"
Call-ID: 2143389df4877360 at 64.7.189.14
CSeq: 19607 ACK
User-Agent: Grandstream BT100 1.0.6.7
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
--- (12 headers 0 lines)---
<-- SIP read from 64.7.189.14:5060:
BYE sip:500 at 64.7.161.26 SIP/2.0
Via: SIP/2.0/UDP 64.7.189.14;branch=z9hG4bKd592db0d7856fb18
From: "Budge Tone"
<sip:budgeTone-PubIP at 64.7.161.26>;tag=b72941c93fe74588
To: <sip:500 at 64.7.161.26>;tag=as324cfd6f
Proxy-Authorization: Digest username="budgeTone-PubIP",
realm="asterisk", algorithm=MD5, uri="sip:500 at 64.7.161.26",
nonce="351ca5f6", response="677ef75492337c32f50f3a936f4c0919"
Call-ID: 2143389df4877360 at 64.7.189.14
CSeq: 19608 BYE
User-Agent: Grandstream BT100 1.0.6.7
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
--- (11 headers 0 lines)---
Sending to 64.7.189.14 : 5060 (non-NAT)
Transmitting (no NAT) to 64.7.189.14:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
64.7.189.14;branch=z9hG4bKd592db0d7856fb18;received=64.7.189.14
From: "Budge Tone"
<sip:budgeTone-PubIP at 64.7.161.26>;tag=b72941c93fe74588
To: <sip:500 at 64.7.161.26>;tag=as324cfd6f
Call-ID: 2143389df4877360 at 64.7.189.14
CSeq: 19608 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:500 at 64.7.161.26>
Content-Length: 0
---
== Spawn extension (local, 500, 1) exited non-zero on
'SIP/budgeTone-PubIP-7e44'
Destroying call '2143389df4877360 at 64.7.189.14'
------------------------------
Message: 6
Date: Thu, 19 Jan 2006 19:47:19 +0200
From: Rob Lith <rob at connection-telecom.com>
Subject: Re: [Asterisk-Users] Brief silences during calls
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID:
<32b7e7b80601190947v808f974la6225f06afc9ed6c at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Just look through the devices settings for suppress silence or transmit
silence and don't supress or prevent transmission... this is a common
problem inX-Lite
Rob
On 1/19/06, Mojo with Horan & Company, LLC <mojo at horanappraisals.com>
wrote:
>
> check irq supply on the * server -- When you run zttest do you
maintain
> over 98%? or like Steve suggested, the network may have congestion or
> other errors ethereal may help you figure out.
>
> I had a polycom 500 that was doing this to my user, 301s and 501s
> wouldn't do it. Not sure if that was network issues or something with
> the polycom itself.
>
> Moj
>
> Mimmus wrote:
> > Where can I investigate the origin of brief silences during calls
> from/to my
> > SIP phone?
> > Only rare pauses of 0.5-1.0 sec when I'm not able to hear anything.
> >
> > Thanks
> > Mimmus
> >
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
> --
> Mojo <mojo at horanappraisals.com>
> Office Manger, Horan & Company, LLC
> (907) 747-6666 x112
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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------------------------------
Message: 7
Date: Thu, 19 Jan 2006 17:47:29 +0000 (UTC)
From: tony at softins.clara.co.uk (Tony Mountifield)
Subject: [Asterisk-Users] Re: MeetMe Listen Only flag (|m)
To: asterisk-users at lists.digium.com
Message-ID: <dqojbh$i17$1 at softins.clara.co.uk>
In article
<B0CF4196F21DC0448367514774331AB7E831BF at scl-exch2k3.phoenix.com>,
Dan Austin <Dan_Austin at Phoenix.com> wrote:
> Tony wrote:
> > I should tidy it up and submit it, but haven't got round to it :-(
>
> Let us know if you can. I'm already maintaining a grocery list
> of patches to make MeetMe viable in my orginization, so one more
> won't kill me.
I should be able do so this weekend. That's the plan, anyway :-)
I'll post the Mantis bug# when I've submitted it.
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: 8
Date: Thu, 19 Jan 2006 12:52:36 -0500
From: Jared Watkins <jared at watkins.net>
Subject: Re: [Asterisk-Users] SAN Devices
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <43CFD1E4.3070302 at watkins.net>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Adam Robins wrote:
>Anyone out there using small-midsized (2-4 TB) SAN solution among
>multiple Asterisk systems? I don't have the budget for an EMC-caliber
>solution, and can't seem to find much else out there.
>
>
I designed a virtualized san and have been running it in production for
the last two years... Speaking from experience... stay away from EMC!
We have several storage systems in production.. from multiple vendors...
and I've had nothing but problems with the CX line of emc systems.
Performance problems... hardware/crashing problems.. (they run embedded
xp you know) and dead fibre port problems. If I didn't have two of
everything.. mirroring across cabinets with IpStor we would have had
serious problems.
Just my two cents on the issue of 'EMC-caliber' storage...
Jared
------------------------------
Message: 9
Date: Thu, 19 Jan 2006 18:52:35 +0100
From: Massimo De Nadal <maxx at digital-system.it>
Subject: [Asterisk-Users] Disabling zap echo cancellor from dialplan
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <43CFD1E3.6020108 at digital-system.it>
Content-Type: text/plain; charset=ISO-8859-15; format=flowed
Anybody knows if it's possible to disable zap echo cancellor from
dialplan only for certain outbound calls ??
I share the same phone lines for voice calls and faxes. Iaxmodem works
fine for me only turning off the echo cancellor, but I need it for
voice calls.
Any ideas ?
maxx
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