Call exits after 120 seconds with hangup cause
hi, we are using fs for different services, but we never used it, to connect sip-phones directly with fs. now we want to do so, but we encounter big problems. everything works fine, but after 120 seconds fs hangs up with the hangup cause RECOVERY_ON_TIMER_EXPIRE. it seems that this has something to do with nat problems (i read about it in nabble), but we can not figure out, how to fix the problem. we are not using stun, because all ip-addresses are static. we opened our firewall for our ip-addresses and opened the following ports: 5060 - 5091 TCP / UDP 10000 - 32767 UDP we are using a quite actual version of fs - the revision ist 13783. we set up a profile in the internal.xml and played a lot with different settings. we can talk to each other, make outbound calls and receive calls - but we can not figure out, how to avoid the hangup after 120 seconds. thanks a lot for your help dennis
You have a NAT issue. /b On Aug 21, 2009, at 7:46 AM, Dennis wrote:
hi,
we are using fs for different services, but we never used it, to connect sip-phones directly with fs.
now we want to do so, but we encounter big problems. everything works fine, but after 120 seconds fs hangs up with the hangup cause RECOVERY_ON_TIMER_EXPIRE.
it seems that this has something to do with nat problems (i read about it in nabble), but we can not figure out, how to fix the problem.
we are not using stun, because all ip-addresses are static. we opened our firewall for our ip-addresses and opened the following ports: 5060 - 5091 TCP / UDP 10000 - 32767 UDP
we are using a quite actual version of fs - the revision ist 13783.
we set up a profile in the internal.xml and played a lot with different settings. we can talk to each other, make outbound calls and receive calls - but we can not figure out, how to avoid the hangup after 120 seconds.
thanks a lot for your help dennis
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Do you have those lines in switch.conf file? <!--RTP port range --> <param name="rtp-start-port" value="10000"/> <param name="rtp-end-port" value="32767"/> BR
In your SIP profiles this could be set. I beleive 120 is the default setting. param name="session-timeout" value="120" On Fri, Aug 21, 2009 at 11:35 PM, bakko<asannucci@gmail.com> wrote:
Do you have those lines in switch.conf file?
<!--RTP port range --> <param name="rtp-start-port" value="10000"/> <param name="rtp-end-port" value="32767"/>
BR
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-- Jim Burke Director Evolutiontel. http://www.evolutiontel.net
You have a NAT issue.
as i wrote, we are quite sure, that this is a nat problem. but we have no idea how we could fix this. there must be a reason, why the hangup always comes after 120 seconds.
<!--RTP port range --> <param name="rtp-start-port" value="10000"/> <param name="rtp-end-port" value="32767"/>
we tested that, but this does not change anything (nothing becomes worse or better).
param name="session-timeout" value="120"
this line is commented in our profile and we believe, that this has nothing to do with our problem. to tell a little bit more about our problem: under http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios#Scenario_2 you can find the scenario, which is exactly like ours. we played with all the mentioned settings, but it does not work. to mention one thing: if we connect with xlite (softphone), we do not have any problems. but we have snom-voip-phones and with them we have the problems. the big question is: wo or what might cause/trigger the hangup? is it freeswitch or something else? we have a firewall (IPCop) - might there be a setting, which needs to be set, to avoid theses problems? kind regards dennis
if you are talking to a device that pretends to do session timers but really does not, the re-invite can cause a hangup. We have seen this with sonus many times, do you have sonus involved in your call per chance? On Wed, Aug 26, 2009 at 7:59 AM, Dennis <odermann@googlemail.com> wrote:
You have a NAT issue.
as i wrote, we are quite sure, that this is a nat problem. but we have no idea how we could fix this. there must be a reason, why the hangup always comes after 120 seconds.
<!--RTP port range --> <param name="rtp-start-port" value="10000"/> <param name="rtp-end-port" value="32767"/>
we tested that, but this does not change anything (nothing becomes worse or better).
param name="session-timeout" value="120"
this line is commented in our profile and we believe, that this has nothing to do with our problem.
to tell a little bit more about our problem:
under http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios#Scenario_2 you can find the scenario, which is exactly like ours. we played with all the mentioned settings, but it does not work.
to mention one thing: if we connect with xlite (softphone), we do not have any problems. but we have snom-voip-phones and with them we have the problems.
the big question is: wo or what might cause/trigger the hangup? is it freeswitch or something else? we have a firewall (IPCop) - might there be a setting, which needs to be set, to avoid theses problems?
kind regards dennis
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if you are talking to a device that pretends to do session timers but really does not, the re-invite can cause a hangup. We have seen this with sonus many times, do you have sonus involved in your call per chance?
hi anthony, this is what we found on the snom website about the session timer settings: "If SIP Session Timer Support is enabled, this option specifies the SIP session timer in seconds. For instance, a Re-INVITE will be sent after 50% of its value has elapsed." our setting is 3600. is session timer and re-invite helpful or can we disable it? behaps disabling works? is it possible, that snom does not support a REAL session timer? sonus is not involved. kind regards dennis
Is the snom firmware up to the latest? I believe session timers should work properly with snom? you can try disabling it but make sure you have your firmware to latest. On Wed, Aug 26, 2009 at 11:23 AM, Dennis <odermann@googlemail.com> wrote:
if you are talking to a device that pretends to do session timers but really does not, the re-invite can cause a hangup. We have seen this with sonus many times, do you have sonus involved in your call per chance?
hi anthony,
this is what we found on the snom website about the session timer settings: "If SIP Session Timer Support is enabled, this option specifies the SIP session timer in seconds. For instance, a Re-INVITE will be sent after 50% of its value has elapsed." our setting is 3600.
is session timer and re-invite helpful or can we disable it? behaps disabling works?
is it possible, that snom does not support a REAL session timer?
sonus is not involved.
kind regards dennis
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Is the snom firmware up to the latest? yes, the firmware is the latest.
I believe session timers should work properly with snom? i don't know, but i think so. we tested with session timers "off" and with setting the session timer to 0. both does not change anything.
we played with using a stun-server and had a small success. if we are connected with a stund-server, the hangup will not come after 120 seconds. we can also call the 5900 and listen to the moh. this works fine. here we have a totally different problem. if another voip phone is connected to the server using stun (for example the 1000), both sides can not talk to each other. one won't hear, if the other side is talking. very strange (for us), but we have no idea, what we can do about that.
Are the phones behind the same nat as FS? /b On Aug 27, 2009, at 10:39 AM, Dennis wrote:
we played with using a stun-server and had a small success. if we are connected with a stund-server, the hangup will not come after 120 seconds. we can also call the 5900 and listen to the moh. this works fine. here we have a totally different problem. if another voip phone is connected to the server using stun (for example the 1000), both sides can not talk to each other. one won't hear, if the other side is talking.
Are the phones behind the same nat as FS?
no, both phones are behind the same nat, fs is behind another nat (and the internet is inbetween). both phones are in our office and we are sitting behind a nat. we connect through our nat over the internet into the other nat, where fs is behind.
Can you paste in the line that actually hangs up your call it's BLUE and says something like 2009-08-27 11:39:04.903202 [NOTICE] sofia.c:327 Hangup sofia/internal/ 1004@dev.bkw.org [CS_EXECUTE] [NORMAL_CLEARING] find the line like this for each leg of you call and put them in your reply On Thu, Aug 27, 2009 at 11:01 AM, Dennis <odermann@googlemail.com> wrote:
Are the phones behind the same nat as FS?
no, both phones are behind the same nat, fs is behind another nat (and the internet is inbetween).
both phones are in our office and we are sitting behind a nat. we connect through our nat over the internet into the other nat, where fs is behind.
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this is the line (without stun - so we only have one leg) and we called the 5900 to moh: 2009-08-27 19:18:02.348232 [NOTICE] sofia.c:3863 Hangup sofia/internal/1004@212.18.215.102 [CS_EXECUTE] [RECOVERY_ON_TIMER_EXPIRE] we called the 5900 and waited 2 minutes... or did you mean something different?
do the following: 1) restart FS 2) type sofia profile internal siptrace on 3) type console loglevel debug 4) reproduce the issue 5) upload the resulting output to http://pastebin.freeswitch.org (*hint* the pass is in the dialog box) 6) let me know the pastebin number On Thu, Aug 27, 2009 at 12:25 PM, Dennis <odermann@googlemail.com> wrote:
this is the line (without stun - so we only have one leg) and we called the 5900 to moh:
2009-08-27 19:18:02.348232 [NOTICE] sofia.c:3863 Hangup sofia/internal/1004@212.18.215.102 [CS_EXECUTE] [RECOVERY_ON_TIMER_EXPIRE]
we called the 5900 and waited 2 minutes...
or did you mean something different?
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the pastebin number is 10129
I think the problem is the session-timeout is too long and your nat mapping is being deleted. try setting it to a smaller value like 20 or 40 sec or try setting those phones to register more frequently (every 10 or 20 sec) On Thu, Aug 27, 2009 at 1:11 PM, Dennis <odermann@googlemail.com> wrote:
the pastebin number is 10129
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I think the problem is the session-timeout is too long and your nat mapping is being deleted.
we changed the param session-timeout (in internal) to 20 and 40, without success. we changed the minimum-session-expires to 20, although we knew, the allowed minimum is 90 and 90 was shown in the console - no success.
try setting it to a smaller value like 20 or 40 sec or try setting those phones to register more frequently (every 10 or 20 sec)
we changed all possible options to 10 and 20, again without success :(
If you look at your trace the call is sending a re-invite over and over and over again with no reply you need to examine your network topology and find out why the packets FS is sending to your phone never make it. also try disabling session-timers on the snom On Thu, Aug 27, 2009 at 2:09 PM, Dennis <odermann@googlemail.com> wrote:
I think the problem is the session-timeout is too long and your nat mapping is being deleted.
we changed the param session-timeout (in internal) to 20 and 40, without success.
we changed the minimum-session-expires to 20, although we knew, the allowed minimum is 90 and 90 was shown in the console - no success.
try setting it to a smaller value like 20 or 40 sec or try setting those phones to register more frequently (every 10 or 20 sec)
we changed all possible options to 10 and 20, again without success :(
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If you look at your trace the call is sending a re-invite over and over and over again with no reply
you need to examine your network topology and find out why the packets FS is sending to your phone never make it.
also try disabling session-timers on the snom
are you talking about our problem, that we get a hangup after 120 seconds? i just ask to avoid, that we are talking about different things. if we connect without stun and call the 5900, we can hear the music, so i assume, that we are receiving packets from fs!? but after about 120 seconds, we receive the hangup. before everything was fine and as expected. we also tried disabling session-timers in the snom - same problem. thant's the problem, we tried soooooo many things, but simply nothing changes the problem or makes something different (like sending the hangup later or earlier).
yes i am talking about that same problem. I am not talking about audio I am talking about sip messages. look at the trace you made from my earlier request You can see the same invite being sent 10 times and never got any answer. so you need to investigate it. On Thu, Aug 27, 2009 at 2:42 PM, Dennis <odermann@googlemail.com> wrote:
If you look at your trace the call is sending a re-invite over and over and over again with no reply
you need to examine your network topology and find out why the packets FS is sending to your phone never make it.
also try disabling session-timers on the snom
are you talking about our problem, that we get a hangup after 120 seconds? i just ask to avoid, that we are talking about different things.
if we connect without stun and call the 5900, we can hear the music, so i assume, that we are receiving packets from fs!? but after about 120 seconds, we receive the hangup. before everything was fine and as expected.
we also tried disabling session-timers in the snom - same problem.
thant's the problem, we tried soooooo many things, but simply nothing changes the problem or makes something different (like sending the hangup later or earlier).
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participants (5)
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Anthony Minessale
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bakko
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Brian West
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Dennis
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Jim Burke