[Asterisk-Users] iconnect quality?

Lubomir Christov voip at minitelecom.org
Wed Mar 12 10:20:34 MST 2003 

Hi Gregg,

I'm using iconnect with LineJACK/PhoneCARD and G723.1 codec from about 1 
mount without any problems. The quality is perfect and everything is OK 
(only some little problems sometime).
But today morning, with the NEW CVS version update of asterisk I found 
that SIP(G723/ulaw) and iconnect aren't working anymore .... ???????
When I try to connect trough iconnect I receive this error message:

     -- Got SIP response 488 "Not Acceptable Media" back from 213.137.73.178

you can try asterisk from yesterday:
   cvs -z9 co -D "Mar 11 2003" asterisk

and test it: everything will be OK :)
Here is my working configuration:

sip.conf

[general]
port = 5060			
;bindaddr = 0.0.0.0	
context = incomming		
disallow=all
allow=g723.1
;allow=ulaw
tos=lowdelay
tos=184

[iconnect]
type=friend
username=12345678
password=1234
host=213.137.73.178
callerid=1234567890



phone.conf

format=slinear
echocancel=low
silencesupression=no


extension.conf

exten => _00.,1,Dial(Sip/${EXTEN:2}@iconnect,,C)

Lubo

P.S. for successfully using G723 codec and phonejack you will need 
g723.1 and g723.1b placed in your codecs directory. You can got it like 
this:

cvs -d :pserver:anoncvs at cvs.digium.com:/usr/cvsroot co g723.1
cvs -d :pserver:anoncvs at cvs.digium.com:/usr/cvsroot co g723.1b

and uncomment this line in Makefile in codecs directory
MODG723=codec_g723_1.so codec_g723_1b.so

I hope that the todays problem with asterisk and SIP/G723 will be fixed 
very soon.

L

Gregg Lebovitz wrote:
> Hi Lubo,
> 
> I appreciate your email to help with this issue, but I don't understand
> your message. I assume your comment about format=slinear is to use
> format=slinear in phone.conf instead of format=ulaw. If so, how does
> this get you g723 to iconnect? Using format=g723.1 doesn't seem to work.
> 
> Gregg
> 
> On Wed, 2003-03-12 at 00:50, Lubomir Christov wrote:
> 
>>Dan, why are you using phonejack with ulaw codec? g723 (format=slinear 
>>only) is working just perfect with phonejack and iconnect :)
>>
>>Lubo
>>
>>Dan Fernandez wrote:
>>
>>>I found similar problems.
>>>
>>>With my phonejack I can make a call with ulaw with decent quality (I have a
>>>64k line).
>>>
>>>However, with Messenger I hear a brief horrible noise and that­s it.
>>>
>>>----- Original Message -----
>>>From: "Jim Archer" <jim at archer.net>
>>>To: <asterisk-users at lists.digium.com>
>>>Sent: Tuesday, March 11, 2003 8:17 PM
>>>Subject: Re: [Asterisk-Users] iconnect quality?
>>>
>>>
>>>
>>>
>>>>Ok!  When I use the 7777 prefix and I allow gsm it does work!  And the
>>>>quality is fine.
>>>>
>>>>There are two problems we're having now.
>>>>
>>>>1 - From watching the udp fly by, it seems that iconnect does not know
>>>
>>>when
>>>
>>>
>>>>we hang up.  For example, if I call a voice mail and it starts giving me
>>>>its speal and I hang up, iconnect stays connected until the VM hangs up at
>>>>its end.
>>>>
>>>>Next, if we try to call out via iconnect from a sip client extension (like
>>>>a windows soft phone) all we hear is horrible noise.
>>>>
>>>>Has anyone else had these issues?
>>>>
>>>>Jim
>>>>
>>>>
>>>>--On Tuesday, March 11, 2003 3:34 PM -0500 Gregg Lebovitz
>>>><gregg at lebovitz.net> wrote:
>>>>
>>>>
>>>>
>>>>>I haven't play around enough to know whether or not the 7777 prefix is a
>>>>>toggle. I will do some experimenting and let you know. Right now I am
>>>>>prefixing all my calls with 7777.
>>>>>
>>>>>My experience is that when the carrier's format is G723.1, you can't
>>>>>hear the incoming voice. When it is in G711 you can. I have made several
>>>>>calls using G711 and they are acceptable quality. Note that if you
>>>>>disallow=gsm in the sip.conf file you will get the 488 media errors you
>>>>>reported earlier.
>>>>>
>>>>>Below are my config files for sip and the linejack cards:
>>>>>
>>>>>;
>>>>>; SIP Configuration for Asterisk
>>>>>;
>>>>>[general]
>>>>>port = 5060 ; Port to bind to
>>>>>bindaddr = 0.0.0.0 ; Address to bind to
>>>>>context=iconnect ; Default for incoming calls
>>>>>allow=gsm
>>>>>allow=ulaw
>>>>>allow=alaw
>>>>>
>>>>>;register=1813342XXXX:XXXXXX at sipauth.deltathree.com
>>>>>;register=1202454XXXX:XXXXXX at sipauth.deltathree.com
>>>>>
>>>>>[iconnecthere]
>>>>>type=friend
>>>>>username=XXXXXXXX
>>>>>secret=XXX
>>>>>host=sipauth.deltathree.com
>>>>>
>>>>>;
>>>>>; Linux Telephony Interface
>>>>>;
>>>>>; Configuration file
>>>>>;
>>>>>[interfaces]
>>>>>
>>>>>mode=dialtone
>>>>>format=ulaw
>>>>>echocancel=medium
>>>>>silencesupression=no
>>>>>
>>>>>context=local
>>>>>context=default
>>>>>
>>>>>txgain=100%
>>>>>rxgain=100%
>>>>>device => /dev/phone0
>>>>>
>>>>>
>>>>>
>>>>>On Tue, 2003-03-11 at 14:28, Jim Archer wrote:
>>>>>
>>>>>
>>>>>>Hi Greg and thanks very much...
>>>>>>
>>>>>>A few questions...
>>>>>>
>>>>>>First, regarding the 7777 prefix, it seemed that this acts as a toggle,
>>>>>>switching from the one codec to the other.  But how do I set which me
>>>>>>system uses by default?  Or does iconnect always use the high bandwidth
>>>>>>one  by default (such that the 7777 always switches to the low
>>>
>>>bandwidth
>>>
>>>
>>>>>>one)?
>>>>>>
>>>>>>Next, I am still struggling to understand the SIP options and what goes
>>>>>>where.  Could you please tell me where the format command goes?  Is
>>>
>>>this
>>>
>>>
>>>>>>an  option on the channel?  I thing the allow goes in sip.conf.
>>>>>>
>>>>>>Finally, does this have any impact on the problem where the person
>>>>>>called  can not be heard?
>>>>>>
>>>>>>Thanks!!!
>>>>>>
>>>>>>Jim
>>>>>>
>>>>>>--On Tuesday, March 11, 2003 1:35 PM -0500 Gregg Lebovitz
>>>>>><gregg at lebovitz.net> wrote:
>>>>>>
>>>>>>
>>>>>>
>>>>>>>Jim,
>>>>>>>
>>>>>>>I changed my extensions entry for iconnect to:
>>>>>>>
>>>>>>>exten => _1XXXXXXXXXX,1,Dial,SIP/7777${EXTEN}@iconnecthere
>>>>>>>
>>>>>>>and my calls work fine with ulaw. I am calling from a linejack card
>>>>>>>with format=ulaw and SIP with allow=ulaw.
>>>>>>>
>>>>>>>Gregg
>>>>>>>
>>>>>>>On Mon, 2003-03-10 at 23:01, Jim Archer wrote:
>>>>>>>
>>>>>>>
>>>>>>>>--On Monday, March 10, 2003 4:47 PM -0300 Dan Fernandez
>>>>>>>><danfernandez00 at hotmail.com> wrote:
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>>Iconnect uses codecs g723 and g711 that can be configured for each
>>>>>>>>>account (you can change them by the 7777 prefix)
>>>>>>>>
>>>>>>>>I tried adding the 7777 in front of a number and it reliably
>>>
>>>generates
>>>
>>>
>>>>>>>>error "488 invalid media."
>>>>>>>>
>>>>>>>>
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>>>>>>>>Asterisk-Users at lists.digium.com
>>>>>>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>>>
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>>>>>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>>
>>>>>>
>>>>>>_______________________________________________
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>>>>>>Asterisk-Users at lists.digium.com
>>>>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>
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>>>>
>>>>
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>>>
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>>
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