[Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks
Hi there, I have very strange behaviors for my SIP endpoints with FS SVN trunk 15905. I tested with both Polycom IP650 and Bria 2.5.4, compared against port audio and googletalk endpoints in the same network. all SIP end points (Polycom and Bria) behind NAT but in the same subnet 192.168.0, I tried to change the settings below: param name=local-network-acl value=localnet.auto/ param name=apply-nat-acl value=nat.auto/ in /conf/sip_profiles/internal.xml using different combinations of either enabling or disabling them. the results are all the same, the audios on sip endpoints always got cut about 31 seconds, no issues at all with either port audio or gtalk, Could anyone point me to the right direction for the sofia_sip profile setup? Your helps are greatly appreciated Thanks, Chris ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks
On Fri, Dec 11, Chris Chen wrote: Hi there, I have very strange behaviors for my SIP endpoints with FS SVN trunk 15905. Is this a change in behavior or is this the first time you've run freeswitch? If this is your first time welcome aboard! Also if this is your first time you've probably have some IPs aliased on your interface and you still have stun enabled. This was the behavior I saw the first time I ran it on a box with aliases on an interface. The stun server tells freeswitch after some time that the IP is different then the one you've assigned. This is just one possibility. If this isn't the case then we will need to see sip traces on all of your profiles. --FC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks
Thanks Frank for sharing your experience. This is the behavior change just starting within three days, maybe because of some code changes in mod_sofia which I should change the settings accordingly I noticed that the acl automatically having 192.168.0.0 set as deny, that's why I tried to changed the settings regarding nat acl and localnet acl. Chris On Fri, Dec 11, 2009 at 11:25 AM, Frank Carmickle fr...@carmickle.comwrote: On Fri, Dec 11, Chris Chen wrote: Hi there, I have very strange behaviors for my SIP endpoints with FS SVN trunk 15905. Is this a change in behavior or is this the first time you've run freeswitch? If this is your first time welcome aboard! Also if this is your first time you've probably have some IPs aliased on your interface and you still have stun enabled. This was the behavior I saw the first time I ran it on a box with aliases on an interface. The stun server tells freeswitch after some time that the IP is different then the one you've assigned. This is just one possibility. If this isn't the case then we will need to see sip traces on all of your profiles. --FC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks
As i said multiple times on irc last night, we need to see debug logs with sip trace to see what is going on. Mike On Dec 11, 2009, at 11:44 AM, Chris Chen wrote: Thanks Frank for sharing your experience. This is the behavior change just starting within three days, maybe because of some code changes in mod_sofia which I should change the settings accordingly I noticed that the acl automatically having 192.168.0.0 set as deny, that's why I tried to changed the settings regarding nat acl and localnet acl. Chris On Fri, Dec 11, 2009 at 11:25 AM, Frank Carmickle fr...@carmickle.com wrote: On Fri, Dec 11, Chris Chen wrote: Hi there, I have very strange behaviors for my SIP endpoints with FS SVN trunk 15905. Is this a change in behavior or is this the first time you've run freeswitch? If this is your first time welcome aboard! Also if this is your first time you've probably have some IPs aliased on your interface and you still have stun enabled. This was the behavior I saw the first time I ran it on a box with aliases on an interface. The stun server tells freeswitch after some time that the IP is different then the one you've assigned. This is just one possibility. If this isn't the case then we will need to see sip traces on all of your profiles. --FC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks
Hi Mike, the fs console log with sip trace on the internal profile is attached in the pastebin below, http://pastebin.freeswitch.org/11483 could you please take a look at it? Thanks, Chris On Fri, Dec 11, 2009 at 12:51 PM, Michael Jerris m...@jerris.com wrote: As i said multiple times on irc last night, we need to see debug logs with sip trace to see what is going on. Mike On Dec 11, 2009, at 11:44 AM, Chris Chen wrote: Thanks Frank for sharing your experience. This is the behavior change just starting within three days, maybe because of some code changes in mod_sofia which I should change the settings accordingly I noticed that the acl automatically having 192.168.0.0 set as deny, that's why I tried to changed the settings regarding nat acl and localnet acl. Chris On Fri, Dec 11, 2009 at 11:25 AM, Frank Carmickle fr...@carmickle.comwrote: On Fri, Dec 11, Chris Chen wrote: Hi there, I have very strange behaviors for my SIP endpoints with FS SVN trunk 15905. Is this a change in behavior or is this the first time you've run freeswitch? If this is your first time welcome aboard! Also if this is your first time you've probably have some IPs aliased on your interface and you still have stun enabled. This was the behavior I saw the first time I ran it on a box with aliases on an interface. The stun server tells freeswitch after some time that the IP is different then the one you've assigned. This is just one possibility. If this isn't the case then we will need to see sip traces on all of your profiles. --FC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks
Its not sending to the right Contact: header in the 200 OK packet. This was fixed in r15870, you have to update. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 11-Dec-09, at 2:03 PM, Chris Chen wrote: Hi Mike, the fs console log with sip trace on the internal profile is attached in the pastebin below, http://pastebin.freeswitch.org/11483 could you please take a look at it? Thanks, Chris On Fri, Dec 11, 2009 at 12:51 PM, Michael Jerris m...@jerris.com wrote: As i said multiple times on irc last night, we need to see debug logs with sip trace to see what is going on. Mike On Dec 11, 2009, at 11:44 AM, Chris Chen wrote: Thanks Frank for sharing your experience. This is the behavior change just starting within three days, maybe because of some code changes in mod_sofia which I should change the settings accordingly I noticed that the acl automatically having 192.168.0.0 set as deny, that's why I tried to changed the settings regarding nat acl and localnet acl. Chris On Fri, Dec 11, 2009 at 11:25 AM, Frank Carmickle fr...@carmickle.com wrote: On Fri, Dec 11, Chris Chen wrote: Hi there, I have very strange behaviors for my SIP endpoints with FS SVN trunk 15905. Is this a change in behavior or is this the first time you've run freeswitch? If this is your first time welcome aboard! Also if this is your first time you've probably have some IPs aliased on your interface and you still have stun enabled. This was the behavior I saw the first time I ran it on a box with aliases on an interface. The stun server tells freeswitch after some time that the IP is different then the one you've assigned. This is just one possibility. If this isn't the case then we will need to see sip traces on all of your profiles. --FC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks
Thanks Mathieu, but I am on SVN r15912 now. Chris On Fri, Dec 11, 2009 at 2:09 PM, Mathieu Rene mrene_li...@avgs.ca wrote: Its not sending to the right Contact: header in the 200 OK packet. This was fixed in r15870, you have to update. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 11-Dec-09, at 2:03 PM, Chris Chen wrote: Hi Mike, the fs console log with sip trace on the internal profile is attached in the pastebin below, http://pastebin.freeswitch.org/11483 could you please take a look at it? Thanks, Chris On Fri, Dec 11, 2009 at 12:51 PM, Michael Jerris m...@jerris.com wrote: As i said multiple times on irc last night, we need to see debug logs with sip trace to see what is going on. Mike On Dec 11, 2009, at 11:44 AM, Chris Chen wrote: Thanks Frank for sharing your experience. This is the behavior change just starting within three days, maybe because of some code changes in mod_sofia which I should change the settings accordingly I noticed that the acl automatically having 192.168.0.0 set as deny, that's why I tried to changed the settings regarding nat acl and localnet acl. Chris On Fri, Dec 11, 2009 at 11:25 AM, Frank Carmickle fr...@carmickle.comwrote: On Fri, Dec 11, Chris Chen wrote: Hi there, I have very strange behaviors for my SIP endpoints with FS SVN trunk 15905. Is this a change in behavior or is this the first time you've run freeswitch? If this is your first time welcome aboard! Also if this is your first time you've probably have some IPs aliased on your interface and you still have stun enabled. This was the behavior I saw the first time I ran it on a box with aliases on an interface. The stun server tells freeswitch after some time that the IP is different then the one you've assigned. This is just one possibility. If this isn't the case then we will need to see sip traces on all of your profiles. --FC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks
You set the extrtp ip to an IP exactly.. this is the issue we are fixing soon.. if you have natpmp or upnp set it to auto-nat and let it figure it out. The issue is we have restored the behavior in 1.0.4 that lies about the IP all the time... I'm going to commit a patch shortly that'll fix this. /b On Dec 11, 2009, at 1:34 PM, Chris Chen wrote: Thanks Mathieu, but I am on SVN r15912 now. Chris ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks
Thanks Brian for your explanation, could we still keep the option to set the extrip ip, as my DLINK DIR-655 UPNP is not working reliably, and I believe many other routers have similar issue. Chris On Fri, Dec 11, 2009 at 2:45 PM, Brian West br...@freeswitch.org wrote: You set the extrtp ip to an IP exactly.. this is the issue we are fixing soon.. if you have natpmp or upnp set it to auto-nat and let it figure it out. The issue is we have restored the behavior in 1.0.4 that lies about the IP all the time... I'm going to commit a patch shortly that'll fix this. /b On Dec 11, 2009, at 1:34 PM, Chris Chen wrote: Thanks Mathieu, but I am on SVN r15912 now. Chris ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks
Please test www.bkw.org/sofia_autonat_static_ip.diff /b On Dec 11, 2009, at 1:34 PM, Chris Chen wrote: Thanks Mathieu, but I am on SVN r15912 now. Chris ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
