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;received=10.xxx.xxx.xxx
From: <sip:[EMAIL PROTECTED] >;tag=312051512495
To: <sip:200.xxx.xxx.xxx>;tag=as767ed6bb
Call-ID: [EMAIL PROTECTED]
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" <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] RE: Delayed voice for 10 secs
To: <[email protected]>
Message-ID:
<[EMAIL PROTECTED] >
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
[EMAIL PROTECTED]
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=device <XXXX>
Rich
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