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" <[EMAIL PROTECTED]> 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: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] 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 > > > > _______________________________________________ > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- Jeff Ramsey MIS Administrator Tubafor Mill, Inc. _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
