A follow up to this issue: I resolved this issue by setting the ext-sip-ip and ext-rtp-ip to 10.0.1.252. in the vars.xml as this freeswitch is going nowhere to the internet. So basically this one way audio issue is the NAT issue, the typical SIP issue. now the sip default profile looks like this: freeswitch@callcenter> sofia status profile default API CALL [sofia(status profile default)] output: ================================================================================================= Name default Domain Name 10.0.1.252 DBName sofia_reg_default Dialplan XML RTP-IP 10.0.1.252 Ext-RTP-IP 10.0.1.252 SIP-IP 10.0.1.252 Ext-SIP-IP 10.0.1.252 URL sip:mod_sofia@10.0.1.252:5060 BIND-URL sip:mod_sofia@10.0.1.252:5060;maddr=10.0.1.252 HOLD-MUSIC local_stream://moh CODECS G722,PCMU,PCMA,GSM,H263-1998 TEL-EVENT 101 CNG 13 SESSION-TO 0 MAX-DIALOG 0 Registrations: ================================================================================================= Call-ID Zjg0YTVjMmU0NzRhN2NhZjA1MmNiOTEyZWExN2M4YTY. User 1000@10.0.1.252 Contact "1000" <sip:1000@10.0.1.253:36784 ;rinstance=6f9ee3cff586843a> Agent Bria release 2.2 stamp 45414 Status Registered(unknown) EXP(2008-04-11 00:15:24) Call-ID 1858798975-5064-1@10.0.1.160 User 1006@10.0.1.252 Contact "user" <sip:1006@10.0.1.160:5064> Agent Grandstream GXW-4004 V1.1A 1.0.0.67 Status Registered(unknown) EXP(2008-04-11 00:20:37) Call-ID 1412946610-5066-1@10.0.1.160 User 1007@10.0.1.252 Contact "user" <sip:1007@10.0.1.160:5066> Agent Grandstream GXW-4004 V1.1A 1.0.0.67 Status Registered(unknown) EXP(2008-04-11 00:20:37) Call-ID 51e43082-5491ec6d@10.0.1.251 User 1010@10.0.1.252 Contact 1010 <sip:1010@10.0.1.251:5060> Agent Linksys/PAP2-3.1.12(LS) Status Registered(unknown) EXP(2008-04-11 00:21:11) Call-ID f37727a1-518ac7da@10.0.1.251 User 1011@10.0.1.252 Contact 1011 <sip:1011@10.0.1.251:5061> Agent Linksys/PAP2-3.1.12(LS) Status Registered(unknown) EXP(2008-04-11 00:21:11) Call-ID 445786933-5060-1@10.0.1.160 User 1004@10.0.1.252 Contact "user" <sip:1004@10.0.1.160:5060> Agent Grandstream GXW-4004 V1.1A 1.0.0.67 Status Registered(unknown) EXP(2008-04-11 00:21:21) Call-ID 1697459759-5062-1@10.0.1.160 User 1005@10.0.1.252 Contact "user" <sip:1005@10.0.1.160:5062> Agent Grandstream GXW-4004 V1.1A 1.0.0.67 Status Registered(unknown) EXP(2008-04-11 00:21:21) ================================================================================================= I really appreciate your valuable advice to me for the happy freeswitch journey. Lots of good stuffs to be explored further. Thanks, Chris On Thu, Apr 10, 2008 at 3:16 PM, Nicolas Brenner <nicolas@medularis.com> wrote:
Hi Chris,
Your problem seems a lot like what's happening to me, although I'm only using xlite softphones on extensions 1000 and 1001. The phones are able to register and make calls (well two calls at least, before FS stops responding), but the media is not getting through. I'm also using CentOS 5, but the server and softphone computers are on public IPs connected directly to the Internet, and there's no firewall running.
I made a test of registering with extension 1000 and calling the number 9998 2 times, the first I can hear 3 secs of the tetris sound, and hanging up on the softphone ends the call, but on the second call I get no audio at all, and when I hang up on the softphone, the call is not ended on FS, so it stays open for a while until it times out and FS starts deperately trying to send BYE messages to the softphone.
Here are my logs and sip trace (user/pass: freeswitch/mailing): - http://www.medularis.com/fs/freeswitch.log - http://www.medularis.com/fs/consolelog.txt - http://www.medularis.com/fs/siptrace.log
I guess I'm not being very useful to you in solving the problem, but I thought the issues could be related, so maybe we could be able to get the same help.
On 4/10/08, Chris Chen <chris.chen2004@gmail.com> wrote:
Firewall is disabled on the centos 5 as this is purely LAN environment to start with. I believe there is no signaling issue here as every SIP endpoint is able to be registered and with presence information. The issue is about the media can be heard but not able to be sent out. on the sip endpoints.
Thanks
Chris
On Thu, Apr 10, 2008 at 2:49 PM, Chris Danielson <chris@maxpowersoft.com
wrote:
Just tossing this out there. Are iptables running? as root run: `iptables -L`
Make sure that ports 5060-5080 (tcp and udp) are not being blocked.
Chris Danielson chris@maxpowersoft.com
Chris Chen wrote:
Without the change, the domain name and rtp-ip sip-ip would be
127.0.0.1 which is the loopback IP address of the freeswitch server. After those changes, at least I had all sip end-points registered and I heard dialtones when off the hook.
Thanks
On Thu, Apr 10, 2008 at 2:31 PM, Brian West <brian@freeswitch.org> wrote:
Can you elaborate on why you did this change?
/b
On Apr 10, 2008, at 1:25 PM, Chris Chen wrote:
I am using the default profile, with the SIP-IP and RTP-IP changed
to 10.0.1.252
Brian West sip:brian@freeswitch.org <sip%3Abrian@freeswitch.org>
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