[Freeswitch-users] default timeout
Stephen Thwaites
stephen.thwaites at callstera.com
Wed Apr 10 20:03:23 MSD 2013
Hello all,
I still haven't managed to solve this one. Any further tips would be
greatly appreciated.
I feel that I am experimenting with various parameters without a full
understanding of what the behavior should be. The A leg (caller) hits
the dialplan in public, the public dialplan finds a match and
transfers to the appropriate dialplan in corresponding context, this
extension then uses 'set' or 'export' on the call_timeout variable
and attempts a bridge. Does this 'set' set a timeout for the A leg or
B leg? If I use 'export' does it set the timeout for both legs? Last
of all does setting or exporting the call_timeout override the default
bridge timeout of 60s?
Many thanks.
Regards,
Steve.
On Fri, Mar 29, 2013 at 4:50 PM, Stephen Thwaites
<stephen.thwaites at callstera.com> wrote:
> Hello,
> I have setup this scenario now with a test case and have tried using
> the 'export' application instead of 'set' for the call_timeout=25.
> Sadly the same behavior. Similarly no change when using
> hangup_after_bridge=true and continue_on_fail=true before each bridge
> as suggested. Further I have tried using originate_timeout=25 and also
> experimented with leg_timeouts. I then wondered if the voip provider
> set a timeout on the incoming call and they do, but this is set to
> 120s. I all tests I get an ORIGINATOR_CANCEL after 60s.
>
> I then did a sip trace and log level 6 from the fs_cli. Here is the (I
> think) the relevant bit that shows that FS is sending a CANCEL to the
> telephones.
>
> Any further help would be appreciated!
>
> Regards,
> Steve.
>
> 2013-03-29 15:41:55.117929 [NOTICE] sofia.c:6379 Hangup
> sofia/external/0610884128 at 91.195.160.3 [CS_EXECUTE]
> [ORIGINATOR_CANCEL]
> 2013-03-29 15:41:55.117929 [NOTICE] switch_ivr_originate.c:3349 Hangup
> sofia/internal/sip:1000 at 192.168.1.113:45060 [CS_CONSUME_MEDIA]
> [ORIGINATOR_CANCEL]
> 2013-03-29 15:41:55.117929 [NOTICE] switch_ivr_originate.c:3349 Hangup
> sofia/internal/sip:1001 at 192.168.1.112:35160 [CS_CONSUME_MEDIA]
> [ORIGINATOR_CANCEL]
> 2013-03-29 15:41:55.117929 [NOTICE] switch_ivr_originate.c:3349 Hangup
> sofia/internal/sip:1002 at 192.168.1.10:49535 [CS_CONSUME_MEDIA]
> [ORIGINATOR_CANCEL]
> 2013-03-29 15:41:55.117929 [INFO] mod_dptools.c:3052 Originate Failed.
> Cause: ORIGINATOR_CANCEL
> 2013-03-29 15:41:55.117929 [NOTICE] switch_core_session.c:1505 Session
> 9911 (sofia/external/0610884128 at 91.195.160.3) Ended
> 2013-03-29 15:41:55.117929 [NOTICE] switch_core_session.c:1509 Close
> Channel sofia/external/0610884128 at 91.195.160.3 [CS_DESTROY]
> 2013-03-29 15:41:55.117929 [NOTICE] switch_core_session.c:1505 Session
> 9917 (sofia/internal/sip:1000 at 192.168.1.113:45060) Ended
> 2013-03-29 15:41:55.117929 [NOTICE] switch_core_session.c:1509 Close
> Channel sofia/internal/sip:1000 at 192.168.1.113:45060 [CS_DESTROY]
> send 429 bytes to udp/[80.101.42.120]:52292 at 14:43:27.409611:
> ------------------------------------------------------------------------
> CANCEL sip:1000 at 192.168.1.113:45060 SIP/2.0
> Via: SIP/2.0/UDP 94.100.113.237;rport;branch=z9hG4bK5H6X2DKX8Dj1N
> Route: <sip:1000 at 80.101.42.120:52292>
> Max-Forwards: 67
> From: "Thwaites, Stephen"
> <sip:0610884128 at voiptest.callstera.com>;tag=1rv4jNU086aaQ
> To: <sip:1000 at 192.168.1.113:45060>
> Call-ID: d6322816-1321-1231-80a7-fa163e468f82
> CSeq: 41948035 CANCEL
> Reason: SIP;cause=487;text="ORIGINATOR_CANCEL"
> Content-Length: 0
>
> ------------------------------------------------------------------------
> 2013-03-29 15:41:55.117929 [NOTICE] switch_core_session.c:1505 Session
> 9918 (sofia/internal/sip:1001 at 192.168.1.112:35160) Ended
> 2013-03-29 15:41:55.117929 [NOTICE] switch_core_session.c:1509 Close
> Channel sofia/internal/sip:1001 at 192.168.1.112:35160 [CS_DESTROY]
> send 429 bytes to udp/[80.101.42.120]:42328 at 14:43:27.410670:
> ------------------------------------------------------------------------
> CANCEL sip:1001 at 192.168.1.112:35160 SIP/2.0
> Via: SIP/2.0/UDP 94.100.113.237;rport;branch=z9hG4bK6tZp48305p8KH
> Route: <sip:1001 at 80.101.42.120:42328>
> Max-Forwards: 67
> From: "Thwaites, Stephen"
> <sip:0610884128 at voiptest.callstera.com>;tag=21NXmgc45F1vj
> To: <sip:1001 at 192.168.1.112:35160>
> Call-ID: d6335070-1321-1231-80a7-fa163e468f82
> CSeq: 41948035 CANCEL
> Reason: SIP;cause=487;text="ORIGINATOR_CANCEL"
> Content-Length: 0
>
> ------------------------------------------------------------------------
> 2013-03-29 15:41:55.127924 [NOTICE] switch_core_session.c:1505 Session
> 9919 (sofia/internal/sip:1002 at 192.168.1.10:49535) Ended
> 2013-03-29 15:41:55.127924 [NOTICE] switch_core_session.c:1509 Close
> Channel sofia/internal/sip:1002 at 192.168.1.10:49535 [CS_DESTROY]
> send 427 bytes to udp/[80.101.42.120]:50175 at 14:43:27.413100:
> ------------------------------------------------------------------------
> CANCEL sip:1002 at 192.168.1.10:49535 SIP/2.0
> Via: SIP/2.0/UDP 94.100.113.237;rport;branch=z9hG4bK73rF63m42Zy6c
> Route: <sip:1002 at 80.101.42.120:50175>
> Max-Forwards: 67
> From: "Thwaites, Stephen"
> <sip:0610884128 at voiptest.callstera.com>;tag=3aFppBX72rQFe
> To: <sip:1002 at 192.168.1.10:49535>
> Call-ID: d634a26a-1321-1231-80a7-fa163e468f82
> CSeq: 41948035 CANCEL
> Reason: SIP;cause=487;text="ORIGINATOR_CANCEL"
> Content-Length: 0
>
> ------------------------------------------------------------------------
> recv 531 bytes from udp/[80.101.42.120]:50175 at 14:43:27.440064:
> ------------------------------------------------------------------------
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 94.100.113.237;rport=5060;branch=z9hG4bK73rF63m42Zy6c
> From: "Thwaites, Stephen"
> <sip:0610884128 at voiptest.callstera.com>;tag=3aFppBX72rQFe
> To: <sip:1002 at 192.168.1.10:49535>;tag=942234751
> Call-ID: d634a26a-1321-1231-80a7-fa163e468f82
> CSeq: 41948035 CANCEL
> Contact: <sip:1002 at 192.168.1.10:49535>
> Supported: replaces, path, timer, eventlist
> User-Agent: Grandstream GXP2200 1.0.1.40
> Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO,
> REFER, UPDATE, MESSAGE
> Content-Length: 0
>
> ------------------------------------------------------------------------
> recv 507 bytes from udp/[80.101.42.120]:50175 at 14:43:27.445183:
> ------------------------------------------------------------------------
> SIP/2.0 487 Request Terminated
> Via: SIP/2.0/UDP 94.100.113.237;rport=5060;branch=z9hG4bK73rF63m42Zy6c
> From: "Thwaites, Stephen"
> <sip:0610884128 at voiptest.callstera.com>;tag=3aFppBX72rQFe
> To: <sip:1002 at 192.168.1.10:49535>;tag=942234751
> Call-ID: d634a26a-1321-1231-80a7-fa163e468f82
> CSeq: 41948035 INVITE
> Supported: replaces, path, timer, eventlist
> User-Agent: Grandstream GXP2200 1.0.1.40
> Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO,
> REFER, UPDATE, MESSAGE
> Content-Length: 0
>
> ------------------------------------------------------------------------
> send 387 bytes to udp/[80.101.42.120]:50175 at 14:43:27.445316:
> ------------------------------------------------------------------------
> ACK sip:1002 at 192.168.1.10:49535 SIP/2.0
> Via: SIP/2.0/UDP 94.100.113.237;rport;branch=z9hG4bK73rF63m42Zy6c
> Route: <sip:1002 at 80.101.42.120:50175>
> Max-Forwards: 67
> From: "Thwaites, Stephen"
> <sip:0610884128 at voiptest.callstera.com>;tag=3aFppBX72rQFe
> To: <sip:1002 at 192.168.1.10:49535>;tag=942234751
> Call-ID: d634a26a-1321-1231-80a7-fa163e468f82
> CSeq: 41948035 ACK
> Content-Length: 0
>
> etc...
>
> On Tue, Mar 26, 2013 at 12:22 AM, Stephen Thwaites
> <stephen.thwaites at callstera.com> wrote:
>> Nick, Michael,
>> Thanks for the advise, will give these a try and feed back to the list.
>>
>> Regards,
>> Steve.
>>
>> On Mon, Mar 25, 2013 at 11:21 PM, Michael Collins <msc at freeswitch.org> wrote:
>>> Try "export" instead of "set" on your call_timeout=25 lines.
>>> -MC
>>>
>>> On Fri, Mar 22, 2013 at 8:32 AM, Stephen Thwaites
>>> <stephen.thwaites at callstera.com> wrote:
>>>>
>>>> Hi All,
>>>> Apologies for the simple question but I can't find the answer anywhere
>>>> in the books, wiki, or our friend google.
>>>>
>>>> If somebody calls us, FS does an ORIGINATE_CANCEL after 60s but the
>>>> follow-me scheme we have configured is100s. How can I increase the
>>>> default call_timeout of 60s to 100s? Or maybe I am just doing
>>>> something wrong!
>>>>
>>>> I have tried leg_timeouts, originate_timeouts both on the transfer and
>>>> the bridge as well to no avail?
>>>>
>>>> Would be very grateful for any help or advise.
>>>>
>>>> Regards,
>>>> Steve
>>>>
>>>> Some Details:
>>>> - External call comes in on external profile from our voip provider.
>>>> - Dialplan in the public context does a transfer to a follow-me
>>>> extension 7777 in context creche-babys <action application="transfer"
>>>> data="7777 XML creche-babys "/>
>>>> - The 7777 extension is as follows and the call hangs up part way
>>>> through the third step if nobody picks up (after 60s total).
>>>>
>>>> <extension name="creche-fm">
>>>> <condition field="destination_number" expression="^7777$">
>>>> <action application="set" data="hangup_after_bridge=true"/>
>>>> <action application="set" data="continue_on_fail=true"/>
>>>> <action application="set" data="ringback=${nl-ring}"/>
>>>>
>>>> <action application="set" data="call_timeout=25"/>
>>>> <action application="bridge"
>>>>
>>>> data="{ignore_early_media=true}user/21@${domain_name},user/20@${domain_name}"/>
>>>>
>>>> <action application="set" data="call_timeout=25"/>
>>>> <action application="bridge"
>>>> data="{ignore_early_media=true}user/22@${domain_name}"/>
>>>>
>>>> <action application="set" data="call_timeout=25"/>
>>>> <action application="bridge"
>>>>
>>>> data={ignore_early_media=true}user/23@${domain_name},user/24@${domain_name},user/26@${domain_name},user/27@${domain_name},user/28@${domain_name}"/>
>>>>
>>>> <action application="set" data="ringback=${nl-ring}"/>
>>>> <action application="set" data="call_timeout=25"/>
>>>> <action application="bridge"
>>>> data="{ignore_early_media=true}sofia/gateway/3120
>>>> ...voipprovidergateway/06 ...mobile number"/>
>>>>
>>>> </condition>
>>>> </extension>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>>
>>>>
>>>>
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>>>
>>>
>>>
>>>
>>> --
>>> Michael S Collins
>>> Twitter: @mercutioviz
>>> http://www.FreeSWITCH.org
>>> http://www.ClueCon.com
>>> http://www.OSTAG.org
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
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>>> http://www.freeswitchsolutions.com
>>>
>>>
>>>
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