Re: [Freeswitch-users] call disconnects after 32 seconds issue - FreeSWITCH-users Digest, Vol 77, Issue 117
Hi All, Its kaleem, i reported the issue with call disconnecting after 32 seconds, i have managed to fix the issue. please find the detailed fix below; i had to get the external Interface (internet facing) to pass 5060 TCP/UDP & 5080 TCP/UDP to freeswitch server. if you are experiencing similar issues then please follow following steps. run fswitch console using command "fs_cli" once in run command "sofia status" this should show all gateways and profiles. check if external profile has your external IP to it. (to confirm go to www.whatsmyip.org) this will confirm freeswitch is using NAT. now turn loggin on using command "sofia loglevel all 9" and "sofia profile external siptrace on" where external is name of profile which has your external IP addr. please note this will give you a lot of information and you might have digestion issues. i used putty and changed putty settings to show 3000 lines (right click on putty title bar, CHANGE SETTINGS and click on WINDOWS and change SCROLLBACK lines from default 200 to 3000) also you can enable log in putty so it saves all data in putty to a file for you. now make the call and when it disconencts go to log windows and type "/quit" to quit freeswitch terminal and no more logs are shown. now either copy and paste the data or open the log file and search for "50" and try to see what ports are being used. now log on to your router and change all port fowarding (please refer to documentation for your router) forward port you can see in log file(in my case 5060 & 5080) to freeswitch server. i restarted my server just to be on safe side and everything went fine after that. also i had a bad experience with BT SIP, British telecoms business sip, they dont respond with SRV & DNS queries so FreeSwitch doesnt register with them and gateway status goes to fail_wait. i had to play around and manage to fix it, i will do a proper guide so no one else has to suffer like me where it took me 4 days and nights to sort that out. i dont want to post messy documentation so will clean up a bit and hopefully FreeSwitch admins will import to WIKI hope this helps. regards, K On Sat, Nov 17, 2012 at 3:21 PM, < freeswitch-users-request@lists.freeswitch.org> wrote:
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Today's Topics:
1. changing vm default announcement (andy) 2. retreiving voicemail dropping after 30 seconds (Jason Holden) 3. Re: call disconnects after 32 seconds (Anthony Minessale)
---------- Forwarded message ---------- From: andy <packetandy@gmail.com> To: freeswitch-users@lists.freeswitch.org Cc: Date: Fri, 16 Nov 2012 20:20:50 -0800 Subject: [Freeswitch-users] changing vm default announcement hi Nick, thanks for the reply.
Greetings are in $${base_dir}/storage/voicemail/default/$${domain}/<user>/greeting_1.wav as specified in the wiki.
trying to force freeswitch to use this custom greeting by using *<action application="set" data="voicemail_greeting_number=1"/> *does not work*. *Default greeting from Callie still plays*. *If one records a greeting and then overwrites the recorded greeting with another file in the same directory - that works.
Guys, is this a bug, or do I not understand what the 'voicemail_greeting_number variable is supposed to do?
drew
BTW - I am on FS 1.2.1
---------- Forwarded message ---------- From: "Jason Holden" <jason.holden@start.ca> To: <freeswitch-users@lists.freeswitch.org> Cc: Date: Sat, 17 Nov 2012 01:33:04 -0500 Subject: [Freeswitch-users] retreiving voicemail dropping after 30 seconds
Hi.****
When accessing voicemail to listen to messages I am finding that it is dropping at 30 seconds each time with a message of 100 sleep timer.****
Does anyone have any recommendations?****
I am using a Sipura 3000 connected to my freeswitch server.****
** **
---------- Forwarded message ---------- From: Anthony Minessale <anthony.minessale@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Date: Sat, 17 Nov 2012 09:21:22 -0600 Subject: Re: [Freeswitch-users] call disconnects after 32 seconds The only reason that there would be a failure to get the ACK would be if the 2 endpoints have lost communication. If you can't get an ACK to the host, you can't get a BYE to them either. Even if you hacked it to tolerate no ACK, the session timers would fail in another 30 seconds.
You are just focused on your business needs and just disregarding logic. You ask if you are missing something and several individuals are telling you YES. This is the 4th time I am telling you that you should focus your enthusiasm on learning rather than affirming a m00t point because we are not going to change the behavior.
I have not seen a seen a single case of missing ACK that did not point out a real problem that can be simply fixed by either using some NAT related configuration and proper formed Contact hosts.
If this argument continues it will be ended with moderation.....
On Fri, Nov 16, 2012 at 6:07 PM, Yiftach Golan <yiftah@choochee.com>wrote:
Here is the reason from the RFC3261 : "...The reason for this separation is rooted in the importance of delivering all 200 (OK) responses to an INVITE to the UAC. To deliver them all to the UAC, the UAS alone takes responsibility for retransmitting them (see Section 13.3.1.4), and the UAC alone takes responsibility for acknowledging them with ACK (see Section 13.2.2.4). Since this ACK is retransmitted only by the UAC, it is effectively considered its own transaction..."
On Fri, Nov 16, 2012 at 3:50 PM, Yiftach Golan <yiftah@choochee.com>wrote:
Yes I you are right if the RTP is not tied with SIP this theory is not really valid But this is again true about BYE that does not reach to the destination so the risk of not closing the dialog still exist In most of the voice implementation that I saw there are three options : 1. Softswitch (FreeSWITCH, Asterisk, etc) 2. MGW (Avaya, Cisco, etc) 3. Media release but only between two phones You can tie your RTP to the SIP with option 1 and option 2 Option 3 is the problematic one but you will never release your media for billing purpose so usually your risk will be extension to extension open dialog which is less riskier Again as I said it is pretty philosophical debate but from my long experience in SIP I do not think that there is a practical use for it but maybe I am missing something
On Fri, Nov 16, 2012 at 2:31 PM, Sergey Okhapkin < sos@sokhapkin.dyndns.org> wrote:
Don't mix signaling (SIP) and media (RTP). Signaling and media could run different ways. Why do you think FS will always be in media path?
On Friday 16 November 2012 14:56:58 Yiftach Golan wrote:
This is where I am getting a bit confused, if the 200OK arrived to the other side and we checked that the RTP exists (with mod_sofia option) we will not get to hours of calls (unless the other side did not hanged up) In any case there is a good chance that the BYE is getting lost, so the danger exist even without the ACK
I am guessing that the designers of the SIP protocol came up with the ACK because there is a potential for open dialog that is not bound with time and they therefore wanted to know that the other side actually ACKs the request, but since ACK is tied to INVITE only (AFAIK) and INVITE always tied with RTP (at least in most normal SIP implementations) I'm not sure that this ACK is that needed but again as I said it is more of philosophical debate, maybe a potential request in the new RFC
On Fri, Nov 16, 2012 at 12:58 PM, Ken Rice <krice@freeswitch.org> wrote:
In this case masking the issue can lead to massive bills... Imaging
paying by the minute... All of a sudden you are now leaving 2 minute calls up for hours on end... And continuing to get billed for them... Or you are not continuing to bill a customer for them... And now you have unexpected HUGE bills coming in... Masking it is far worse then just fixing it...
If we mask this one issue, we might as well mask memory leaks, or passwords that don’t work, etc... Sure sometimes we might have to mask an issue for production to work in the short term, but that is never the correct answer fix a problem
On 11/16/12 2:19 PM, "Yiftach Golan" <yiftah@choochee.com> wrote:
While I agree on the details I disagree on the solution Sometimes masking the problems can be a good solution but I guess it is a philosophical debate
On Fri, Nov 16, 2012 at 10:27 AM, Ken Rice <krice@freeswitch.org> wrote:
That leaves to big a risk of open sessions and only masks the true issue which is a problem with FS getting the ACK back...
Theres a reason FS is not getting the ACK, and FS will make several attempts to get an ack by retransmitting the 200 OK several times before that timeout occurs.
The real fix here is to fix the underlying cause, not masking it....
On 11/16/12 11:45 AM, "Yiftach Golan" <yiftah@choochee.com < http://yiftah@choochee.com> > wrote:
I know that it is kind out of the what RFC3261 instructs, but did anyone think on giving the option in configuration not to hang up calls in case of an ACK does not arrive? I know that it has the risk of open sessions but there some other ways to handle those cases
On Thu, Nov 15, 2012 at 6:35 PM, Ken Rice <krice@freeswitch.org < http://krice@freeswitch.org> > wrote:
This is probably the same scenario as this is exactly what to expect... Call gets answered far end doesn’t ACK FS sending them a 200OK , fs hangsup the call....
Quite common on networks with NAT issues or broken endpoints
On 11/15/12 7:43 PM, "Vitalie Colosov" <vetali100@gmail.com < http://vetali100@gmail.com> <http://vetali100@gmail.com> > wrote:
I saw this happened earlier when the remote party does not send SIP ACK after receiving SIP OK, so the call is being disconnected after exactly 32 seconds. Not sure if this is exact same scenario here, but just something to consider...
Regards. Vitalie
2012/11/15 kaleem rehman <k4kaleem@gmail.com < http://k4kaleem@gmail.com>
Hi All,
my inbound calls are fine with no issues, my outbound calls get disconnected after 32 seconds and its on all calls. i tried 2 different suppliers and its same result. please find the attached log file with sofia in debug mode. - caller was extension 1234 and desination was 01908321682
your help will be greately appreciated.
regards, Kaleem
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On 17-11-2012 18:14, kaleem rehman wrote:
Hi All,
Its kaleem, i reported the issue with call disconnecting after 32 seconds, i have managed to fix the issue. please find the detailed fix below;
i had to get the external Interface (internet facing) to pass 5060 TCP/UDP & 5080 TCP/UDP to freeswitch server.
<snip>
now log on to your router and change all port fowarding (please refer to documentation for your router) forward port you can see in log file(in my case 5060 & 5080) to freeswitch server. Aren't those ports defined in your profile? Seems there's no need to go hunting through the log if you can fish them out of your fs config? See here: http://wiki.freeswitch.org/wiki/Firewall
<snip>
i dont want to post messy documentation so will clean up a bit and hopefully FreeSwitch admins will import to WIKI
Actually, you can (and are encouraged to) edit the wiki yourself. Having this info on the mailing list is nice but not as useful.
I just use stun to get around these issues, does half of what you've just described automatically. <param name="ext-rtp-ip" value="stun:stun.freeswitch.org"/> <param name="ext-sip-ip" value="stun:stun.freeswitch.org"/> in my sip profiles did the trick. On 17 November 2012 19:53, Jim <hakkie42@gmail.com> wrote:
On 17-11-2012 18:14, kaleem rehman wrote:
Hi All,
Its kaleem, i reported the issue with call disconnecting after 32 seconds, i have managed to fix the issue. please find the detailed fix below;
i had to get the external Interface (internet facing) to pass 5060 TCP/UDP & 5080 TCP/UDP to freeswitch server.
<snip>
now log on to your router and change all port fowarding (please refer to documentation for your router) forward port you can see in log file(in my case 5060 & 5080) to freeswitch server. Aren't those ports defined in your profile? Seems there's no need to go hunting through the log if you can fish them out of your fs config? See here: http://wiki.freeswitch.org/wiki/Firewall
<snip>
i dont want to post messy documentation so will clean up a bit and hopefully FreeSwitch admins will import to WIKI
Actually, you can (and are encouraged to) edit the wiki yourself. Having this info on the mailing list is nice but not as useful.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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participants (3)
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Andrew Cassidy
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Jim
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kaleem rehman