[Freeswitch-users] Call drops after 30 seconds

Wed Jun 22 19:11:13 MSD 2011

Hi Matthew, continue my last reply here

2) Is your FS server using private IP address?  you have to setup your FS
external SIP/RTP IP address to the proper public IP address, by either using
UPNP enabled router, STUN, or hardcoded public IP address in sofia profiles.

Please check that.

Thanks,
Chris

On Wed, Jun 22, 2011 at 10:49 AM, Matthew Ralston
<freeswitch at mralston.com>wrote:

> Hi Chris,
>
> Thanks for the quick reply!
>
> The FreeSWITCH box is in the DMZ of a Cisco Linksys WAG160N. Best I can
> tell, the DMZ is doing its job and allowing all ports through, inbound and
> outbound, TCP & UDP.
>
> The external SIP phones are a Cisco SPA504G and Bria on the iPhone. These
> are behind behind a Cisco ASA5505, which has policy inspection for SIP
> switched on, i.e. ALG. I have also tested with Bria going over 3G (so it's
> not behind the Cisco ASA) and had the same problem.
>
> The other scenario we have is some Yealink T20P SIP phones on the same LAN
> segment as the FreeSWITCH box. These can make A-leg only calls into
> FreeSWITCH (like calling voicemail) and also calls to other internal SIP
> phones fine. However when they make an outbound call the problem happens
> again. In this case the b-leg of the calls are sent to an external SIP
> provider and get cut after 30 seconds. Incidentally, we also use the same
> SIP provider from an Asterisk box in our data centre and that doesn't have a
> problem, so I believe the SIP provider is fine.
>
> Kind regards,
>
>
> Matthew Ralston
> Web Developer & IT Consultant
>
> matt at mralston.co.uk
> www.mralston.com
>
> On 22 Jun 2011, at 14:21, Chris Chen wrote:
>
> Hi Matthew, this is typical behavior for the setup of SIP behind NAT.
>
>     1) Please provide the exact setup of remote SIP phones, what's the
> router model, does it have SIP ALG enabled, what kind  of SIP phones
>     2)
>
>
> On Wed, Jun 22, 2011 at 8:38 AM, Matthew Ralston <freeswitch at mralston.com>wrote:
>
>> Hi,
>>
>> I'm having a problem at the moment with calls being successfully set up,
>> with two-way audio, being terminated by FreeSWITCH after 30 seconds.
>>
>> Internal calls (i.e. between SIP phones on the same LAN segment as the
>> FreeSWITCH box) work flawlessly.
>>
>> The problem arises when at least one of the handsets is located elsewhere
>> on the Internet. This behaviour is exhibited under the following
>> circumstances:
>>
>> - A-leg only call, e.g. to voicemail when the handset is at another
>> location on the Internet
>> - A-leg-B-leg call if one or both of the handsets are at another location
>> on the Internet
>> - Inbound calls from our external SIP provider
>> - Outbound calls to our external SIP provider
>>
>> So it is obvious that the problem is related to the SIP going via the
>> Internet, but I'm having trouble understanding why.
>>
>> Whilst debugging this problem I have placed the FreeSWITCH box is in the
>> DMZ on our router, so there should not be any ports blocked. The FreeSWITCH
>> box itself is not running a software firewall.
>>
>> The calls themselves are absolutely fine for the first 30 seconds - each
>> party can hear the other talking fine.
>>
>> The fact that the call is consistently dropped after 30 seconds (give or
>> take a second or two for PDD) suggests that some timeout is being triggered.
>>
>> When FreeSWITCH terminates the call, the following is logged to the
>> console:
>>
>> 2011-06-22 13:33:50.514941 [DEBUG] sofia.c:4787 Channel
>> sofia/internal/1006 at public.ip.removed entering state [terminating][0]
>> 2011-06-22 13:33:50.514941 [DEBUG] switch_channel.c:2641 (
>> sofia/internal/1006 at public.ip.removed) Callstate Change ACTIVE -> HANGUP
>> 2011-06-22 13:33:50.514941 [NOTICE] sofia.c:5508 Hangup
>> sofia/internal/1006 at public.ip.removed [CS_EXECUTE] [NORMAL_UNSPECIFIED]
>> 2011-06-22 13:33:50.514941 [DEBUG] switch_channel.c:2657 Send signal
>> sofia/internal/1006 at public.ip.removed [KILL]
>> 2011-06-22 13:33:50.514941 [DEBUG] switch_core_session.c:1118 Send signal
>> sofia/internal/1006 at public.ip.removed [BREAK]
>> 2011-06-22 13:33:50.534966 [DEBUG] switch_ivr_play_say.c:1649 done playing
>> file
>> 2011-06-22 13:33:50.625988 [DEBUG] switch_ivr_play_say.c:244 Handle
>> play-file:[voicemail/vm-press.wav] (en:en)
>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_session.c:2063
>> sofia/internal/1006 at public.ip.removed skip receive message
>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:371 (
>> sofia/internal/1006 at public.ip.removed) State EXECUTE going to sleep
>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:325 (
>> sofia/internal/1006 at public.ip.removed) Running State Change CS_HANGUP
>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:565 (
>> sofia/internal/1006 at public.ip.removed) State HANGUP
>> 2011-06-22 13:33:50.727027 [DEBUG] mod_sofia.c:458 Channel
>> sofia/internal/1006 at public.ip.removed hanging up, cause:
>> NORMAL_UNSPECIFIED
>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:46
>> sofia/internal/1006 at public.ip.removed Standard HANGUP, cause:
>> NORMAL_UNSPECIFIED
>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:565 (
>> sofia/internal/1006 at public.ip.removed) State HANGUP going to sleep
>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:356 (
>> sofia/internal/1006 at public.ip.removed) State Change CS_HANGUP ->
>> CS_REPORTING
>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_session.c:1118 Send signal
>> sofia/internal/1006 at public.ip.removed [BREAK]
>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:325 (
>> sofia/internal/1006 at public.ip.removed) Running State Change CS_REPORTING
>> 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:625 (
>> sofia/internal/1006 at public.ip.removed) State REPORTING
>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:53
>> sofia/internal/1006 at public.ip.removed Standard REPORTING, cause:
>> NORMAL_UNSPECIFIED
>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:625 (
>> sofia/internal/1006 at public.ip.removed) State REPORTING going to sleep
>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:350 (
>> sofia/internal/1006 at public.ip.removed) State Change CS_REPORTING ->
>> CS_DESTROY
>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_session.c:1118 Send signal
>> sofia/internal/1006 at public.ip.removed [BREAK]
>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_session.c:1290 Session 5 (
>> sofia/internal/1006 at public.ip.removed) Locked, Waiting on external
>> entities
>> 2011-06-22 13:33:50.740064 [NOTICE] switch_core_session.c:1308 Session 5 (
>> sofia/internal/1006 at public.ip.removed) Ended
>> 2011-06-22 13:33:50.740064 [NOTICE] switch_core_session.c:1310 Close
>> Channel sofia/internal/1006 at public.ip.removed [CS_DESTROY]
>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:454 (
>> sofia/internal/1006 at public.ip.removed) Callstate Change HANGUP -> DOWN
>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:457 (
>> sofia/internal/1006 at public.ip.removed) Running State Change CS_DESTROY
>> 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:467 (
>> sofia/internal/1006 at public.ip.removed) State DESTROY
>> 2011-06-22 13:33:50.740064 [DEBUG] mod_sofia.c:363
>> sofia/internal/1006 at public.ip.removed SOFIA DESTROY
>> 2011-06-22 13:33:50.780056 [DEBUG] switch_nat.c:570 unmapped public port
>> 31484 protocol UDP to localport 31484
>> 2011-06-22 13:33:50.840070 [DEBUG] switch_nat.c:570 unmapped public port
>> 31485 protocol UDP to localport 31485
>> 2011-06-22 13:33:50.840070 [DEBUG] switch_core_state_machine.c:60
>> sofia/internal/1006 at public.ip.removed Standard DESTROY
>> 2011-06-22 13:33:50.840070 [DEBUG] switch_core_state_machine.c:467 (
>> sofia/internal/1006 at public.ip.removed) State DESTROY going to sleep
>>
>> The above example was from an externally situated SIP phone ringing
>> voicemail (4000) on FreeSWITCH.
>>
>> I have experimented changing various timers and timeouts in the config of
>> FreeSWITCH (one at a time, being careful to put them back afterwards!) but
>> been unable to resolve the issue.
>>
>> Incidentally, we have no long term intention of running off-site SIP
>> phones with the PBX and I'm hoping not to have to leave it in the DMZ
>> either, it's just like that for debugging. What is a real issue is the calls
>> to our external SIP provider (i.e. outbound calls) being dropped.
>>
>> Any suggestions would be greatly appreciated.
>>
>> Thanks,
>>
>> Matthew Ralston
>> Web Developer & IT Consultant
>>
>> matt at mralston.co.uk
>> www.mralston.com
>>
>>
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