asterisk users - Apr 2006 - Re: Asterisk-Users Digest, Vol 21, Issue 113

Hi List!!

Thanks for the colaboration, especially to Richard Cavanna who gave me the
necessary support.

I followed your indications and the comunication was better for the test
users. The warning indication is no jumping anymore and the voice is not
delayed. This is my sip.conf:

[general]
context=default
;allowguest=no
;realm=mydomain.tld
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
;domain=mydomain.tld
;domain=mydomain.tld,mydomain-incoming
;domain=1.2.3.4
;allowexternalinvites=no
;autodomain=yes
;pedantic=yes
;tos=184
;tos=lowdelay
;maxexpiry=3600
;defaultexpiry=120
;notifymimetype=text/plain
;checkmwi=10
;vmexten=voicemail
;videosupport=yes
;recordhistory=yes
disallow=all
allow=g729
allow=gsm
allow=ulaw
jitterbuffer=yes
maxjitterbuffer=1500
;allow=ilbc
;musicclass=default
;language=en
;relaxdtmf=yes
rtptimeout=60
;rtpholdtimeout=300
;trustrpid = no
;sendrpid = yes
;progressinband=never
;useragent=Asterisk PBX
;promiscredir = no
;usereqphone = no
dtmfmode = rfc2833
;compactheaders = yes
;sipdebug = yes
;subscribecontext = default
;notifyringing = yes


And these are the extensions:

[xxxx]
type=friend
 host=dynamic
 dtmfmode=rfc2833
 username=xxxx
 secret=xxxx

[xxxx2]
type=friend
 host=dynamic
 dtmfmode=rfc2833
 username=xxxx
 secret=xxxx

As you can see I put the jitterbuffer, maxjitterbuffer and rtptimeout
options. I think with this, the call has a huge improvement and I still
reading about it. This is the CLI output with different commands:

sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
usuario2/usuario2          10.xxx.xxx.xxx       D          5060
Unmonitored
usuario1/usuario1          10.xxx.xxx.xxx      D          5060
Unmonitored
2 sip peers [2 online , 0 offline]

sip show users
Username                   Secret           Accountcode      Def.Context
ACL  NAT
usuario2                   usuario2
default          No   RFC3581
usuario1                   usuario1
default          No   RFC3581

--- (8 headers 0 lines)---
Looking for 200.xxx.xxxx.xxx in default (domain )
Transmitting (no NAT) to 10.xxx.xxx.xxx:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.xxx.xxx.xxx
;rport;branch=z9hG4bK0a0101e20000001044479388000070d3000000d4;received10.xxx.xxx.xxx
From: <sip:usuario2@200.xxx.xxx.xxx>;tag=312051512495
To: <sip:200.xxx.xxx.xxx>;tag=as767ed6bb
Call-ID: DBBDE928-A279-4194-B78C-319FF0FCCDD9@10.xxx.xxx.xxx
CSeq: 150 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:200.xxx.xxx.xxx>
Accept: application/sdp
Content-Length: 0


But I have another question. Our users surf the Internet by cable modems and
we have a CMTS Motorola BSR 1000 with QoS options. I know I can configure it
to manage QoS but I don't know very well how to do it. If somebody knows any
tutorial or experiences administrating this device, please let me know

Thanks again

Carlos Bernat


> Message: 8
> Date: Wed, 19 Apr 2006 15:46:21 -0500
> From: "Cavanna, Richard" <RCavanna@sychip.com>
> Subject: [Asterisk-Users] RE: Delayed voice for 10 secs
> To: <asterisk-users@lists.digium.com>
> Message-ID:
>        
<AB220F8DE6CD4F489EAB48B0020C64A976352A@tx01mailbox1.sychip.com>
> Content-Type: text/plain;       charset="us-ascii"
>
> Please post pertinent config files and a CLI output so the list can help
> with the 10 sec delay
>
> You set codec selection in SIP.conf. This selects preferred codec from
> top to bottom as well as jitter buffer settings and the RTP timeout.
>
> Sip.conf
> disallow=all
> allow=g729
> allow=gsm
> allow=ulaw
> jitterbuffer=yes
> ;forcejitterbuffer=yes
> maxjitterbuffer=1500
> rtptimeout=60
>
>
> As for the DTMF issue try to use rfc2833
>
> in sip.conf define your extention
>
> [XXXX]
> username=XXXX
> type=friend
> secret=XXXXX
> qualify=no
> port=5060
> nat=yes
> mailbox=XXXX@device
> host=dynamic
> dtmfmode=rfc2833
> context=from-internal
> canreinvite=no
> callerid=device <XXXX>
>
> Rich
>
>
>
>
>
>
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