[Asterisk-Users] iconnect quality?

Mark Spencer markster at digium.com
Wed Mar 12 11:14:47 MST 2003 

Please check latest CVS.  This issue has been fixed and was related to the
dynamic payload merger.

Mark

On Wed, 12 Mar 2003, Lubomir Christov wrote:

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

More information about the asterisk-users mailing list