[Asterisk-Users] Asterisk SIP sound issue

[Jeff Ramsey]

There are no firewalls involved. I've got a simple 10/100 switch running and
all connections involved are 100bT FD. switch configuration, nothing
spectacular. Both phones I am testing and the asterisk server are on the
same subnet, there are no routers or firewalls between any of the devices
involved. I have tested and tried some different settings, which is why the
'6200' and '6600' entries have different settings as the '6201-6201' and
'6601-6602' do. The extra settings are just the things I've found on the net
to try.

Here is my sip.conf:


--- START SIP.CONF ---
; Sample sip.conf file
;
; Anything in [general] will be the default setting for all sections of
; sip.conf unless overridden by a setting in a specific [section].  Also SIP
; clients that do not match a [section] will be allowed with the settings in
; the [general] section.
;
;[general]
;port=5060
;bindaddr=0.0.0.0
;tos=lowdelay
;context=INVALID
;static=yes
;writeprotect=no
;                  
;
; User1
[6200]
type=user
secret=secret1
host=dynamic
mailbox=6200
context=toll-access
;
[6200]
type=peer
secret=secret1
host=dynamic
;defaultip=192.147.167.42
context=toll-access
callerid="User1" <6200>
mailbox=6200
;
[6201]
type=user
secret=secret1
mailbox=6200
host=dynamic
context=toll-access
dtmfmode=rfc2833
progressinband=no
;
[6201]
type=peer
secret=secret1
host=dynamic
;defaultip=192.147.167.42
context=toll-access
callerid="User1" <6201>
mailbox=6200
dtmfmode=rfc2833
progressinband=no
;
[6202]
type=user
secret=secret1
mailbox=6200
host=dynamic
context=toll-access
dtmfmode=rfc2833
progressinband=no
;
[6202]
type=peer
secret=secret1
host=dynamic
;defaultip=192.147.167.42
context=toll-access
callerid="User1" <6202>
mailbox=6200
dtmfmode=rfc2833
progressinband=no
;
; User 2
[6600]
type=user
secret=secret2
mailbox=6600
host=dynamic
context=toll-access
;
[6600]
type=peer
secret=secret2
host=dynamic
;defaultip=192.147.167.10
context=toll-access
callerid="User2" <6600>
mailbox=6600
;
[6601]
type=user
secret=secret2
mailbox=6600
host=dynamic
context=toll-access
dtmfmode=rfc2833
disallow=all
allow=ulaw
progressinband=no
;
[6601]
type=peer
secret=secret2
host=dynamic
;defaultip=192.147.167.42
context=toll-access
callerid="User2" <6601>
mailbox=6600
dtmfmode=rfc2833
disallow=all
allow=ulaw
progressinband=no
;
[6602]
type=user
secret=secret2
mailbox=6600
host=dynamic
context=toll-access
dtmfmode=rfc2833
disallow=all
allow=ulaw
progressinband=no
;
[6602]
type=peer
secret=secret2
host=dynamic
;defaultip=192.147.167.42
context=toll-access
callerid="User2" <6602>
mailbox=6600
dtmfmode=rfc2833
disallow=all
allow=ulaw
progressinband=no
;
--- END SIP.CONF ---

On 4/28/05 2:31 PM, "Wiley Siler" <wsiler at education2020.com> wrote:

> Jeff,
> 
> Can you detail your network setup for the group as well.
> 
> Any firewalls involved?
> 
> What does your sip.conf file look like?
> Scrub it of its secret and send it over.
> 
> W
> 
> 
> 
>  
> 
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jeff
> Ramsey
> Sent: Thursday, April 28, 2005 2:25 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Asterisk SIP sound issue
> 
> I have an asterisk box build from cvs stable that I am trying to use
> with 5 IP500 Polycom SIP phones. I can receive calls in through a digium
> wctdm line. I can call out from the SIP IP500 phones to a PSTN number
> through the same card. In other words, incoming and outgoing calls work
> just fine. It is extension to extension calls that I have issues with.
> 
> When I call one SIP IP500 from the other, the call is connected, it is
> using ulaw, but I cannot hear the other person, from either end, no
> matter who makes the call. The only way I have found to make it work, it
> to put the call on hold, (which makes the hold music come on, and I can
> hear that...) and then pick the call back up. After picking the call
> back up, I can use the call like normal. I can hear and be heard. I've
> checked with the asterisk server, and the codec is ulaw the entire time
> the call is connected.
> 
> I have an extension setup that plays back the date and time, and the
> sound is fine on that extension, so I am really lost as to why this is
> happening.
> 
> Please help. I've been stuck here for days now.
> 
> --
> Jeff Ramsey
> 
> 
> 
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-- 
Jeff Ramsey
MIS Administrator
Tubafor Mill, Inc.